diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index 15c438f0f22d..abafadc0b534 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -655,23 +655,10 @@ static int cygnus_ssp_hw_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
 		value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE);
-		/* Configure channels as mono or stereo/TDM */
-		if (params_channels(params) == 1)
-			value |= BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
-		else
-			value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
+		value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
 		writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
 
 		switch (params_format(params)) {
-		case SNDRV_PCM_FORMAT_S8:
-			if (aio->port_type == PORT_SPDIF) {
-				dev_err(aio->cygaud->dev,
-				"SPDIF does not support 8bit format\n");
-				return -EINVAL;
-			}
-			bitres = 8;
-			break;
-
 		case SNDRV_PCM_FORMAT_S16_LE:
 			bitres = 16;
 			break;
@@ -842,6 +829,7 @@ int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai, int len)
 		return -EINVAL;
 	}
 }
+EXPORT_SYMBOL_GPL(cygnus_ssp_set_custom_fsync_width);
 
 static int cygnus_ssp_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 {
@@ -998,7 +986,7 @@ static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
 
 	active_slots = hweight32(tx_mask);
 
-	if ((active_slots < 0) || (active_slots > 16))
+	if (active_slots > 16)
 		return -EINVAL;
 
 	/* Slot value must be even */
@@ -1136,15 +1124,21 @@ static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = {
 	.set_tdm_slot	= cygnus_set_dai_tdm_slot,
 };
 
+static const struct snd_soc_dai_ops cygnus_spdif_dai_ops = {
+	.startup	= cygnus_ssp_startup,
+	.shutdown	= cygnus_ssp_shutdown,
+	.trigger	= cygnus_ssp_trigger,
+	.hw_params	= cygnus_ssp_hw_params,
+	.set_sysclk	= cygnus_ssp_set_sysclk,
+};
 
 #define INIT_CPU_DAI(num) { \
 	.name = "cygnus-ssp" #num, \
 	.playback = { \
-		.channels_min = 1, \
+		.channels_min = 2, \
 		.channels_max = 16, \
 		.rates = SNDRV_PCM_RATE_KNOT, \
-		.formats = SNDRV_PCM_FMTBIT_S8 | \
-				SNDRV_PCM_FMTBIT_S16_LE | \
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | \
 				SNDRV_PCM_FMTBIT_S32_LE, \
 	}, \
 	.capture = { \
@@ -1152,7 +1146,7 @@ static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = {
 		.channels_max = 16, \
 		.rates = SNDRV_PCM_RATE_KNOT, \
 		.formats =  SNDRV_PCM_FMTBIT_S16_LE | \
-					SNDRV_PCM_FMTBIT_S32_LE, \
+				SNDRV_PCM_FMTBIT_S32_LE, \
 	}, \
 	.ops = &cygnus_ssp_dai_ops, \
 	.suspend = cygnus_ssp_suspend, \
@@ -1174,7 +1168,7 @@ static const struct snd_soc_dai_driver cygnus_spdif_dai_info = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE |
 			SNDRV_PCM_FMTBIT_S32_LE,
 	},
-	.ops = &cygnus_ssp_dai_ops,
+	.ops = &cygnus_spdif_dai_ops,
 	.suspend = cygnus_ssp_suspend,
 	.resume = cygnus_ssp_resume,
 };
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index cc0b2d2eaf15..41d9b1da27c2 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1220,6 +1220,7 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
 	u8 dai_clk_mode = 0, dai_ctrl = 0;
+	u8 dai_offset = 0;
 
 	/* Set master/slave mode */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -1234,17 +1235,46 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	}
 
 	/* Set clock normal/inverted */
-	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_NB_NF:
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			dai_clk_mode |= DA7213_DAI_WCLK_POL_INV;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			dai_clk_mode |= DA7213_DAI_CLK_POL_INV;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			dai_clk_mode |= DA7213_DAI_WCLK_POL_INV |
+					DA7213_DAI_CLK_POL_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
 		break;
-	case SND_SOC_DAIFMT_NB_IF:
-		dai_clk_mode |= DA7213_DAI_WCLK_POL_INV;
-		break;
-	case SND_SOC_DAIFMT_IB_NF:
-		dai_clk_mode |= DA7213_DAI_CLK_POL_INV;
-		break;
-	case SND_SOC_DAIFMT_IB_IF:
-		dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | DA7213_DAI_CLK_POL_INV;
+	case SND_SOC_DAI_FORMAT_DSP_A:
+	case SND_SOC_DAI_FORMAT_DSP_B:
+		/* The bclk is inverted wrt ASoC conventions */
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			dai_clk_mode |= DA7213_DAI_CLK_POL_INV;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			dai_clk_mode |= DA7213_DAI_WCLK_POL_INV |
+					DA7213_DAI_CLK_POL_INV;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			dai_clk_mode |= DA7213_DAI_WCLK_POL_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
 		break;
 	default:
 		return -EINVAL;
@@ -1261,6 +1291,13 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_RIGHT_J:
 		dai_ctrl |= DA7213_DAI_FORMAT_RIGHT_J;
 		break;
+	case SND_SOC_DAI_FORMAT_DSP_A: /* L data MSB after FRM LRC */
+		dai_ctrl |= DA7213_DAI_FORMAT_DSP;
+		dai_offset = 1;
+		break;
+	case SND_SOC_DAI_FORMAT_DSP_B: /* L data MSB during FRM LRC */
+		dai_ctrl |= DA7213_DAI_FORMAT_DSP;
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -1271,6 +1308,7 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	snd_soc_write(codec, DA7213_DAI_CLK_MODE, dai_clk_mode);
 	snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK,
 			    dai_ctrl);
+	snd_soc_write(codec, DA7213_DAI_OFFSET, dai_offset);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h
index 16ef56f77cd4..5a78dba1dcb5 100644
--- a/sound/soc/codecs/da7213.h
+++ b/sound/soc/codecs/da7213.h
@@ -188,6 +188,7 @@
 #define DA7213_DAI_FORMAT_I2S_MODE				(0x0 << 0)
 #define DA7213_DAI_FORMAT_LEFT_J				(0x1 << 0)
 #define DA7213_DAI_FORMAT_RIGHT_J				(0x2 << 0)
+#define DA7213_DAI_FORMAT_DSP					(0x3 << 0)
 #define DA7213_DAI_FORMAT_MASK					(0x3 << 0)
 #define DA7213_DAI_WORD_LENGTH_S16_LE				(0x0 << 2)
 #define DA7213_DAI_WORD_LENGTH_S20_LE				(0x1 << 2)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 23b0da7df1f2..804c6f2bcf21 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1721,7 +1721,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
 				PTR_ERR(chan));
 		return PTR_ERR(chan);
 	}
-	BUG_ON(!chan->device || !chan->device->dev);
+	if (WARN_ON(!chan->device || !chan->device->dev))
+		return -EINVAL;
 
 	if (chan->device->dev->of_node)
 		ret = of_property_read_string(chan->device->dev->of_node,
@@ -1867,6 +1868,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (irq >= 0) {
 		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
 					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_common_irq_handler,
 						IRQF_ONESHOT | IRQF_SHARED,
@@ -1884,6 +1889,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (irq >= 0) {
 		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
 					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_rx_irq_handler,
 						IRQF_ONESHOT, irq_name, mcasp);
@@ -1899,6 +1908,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (irq >= 0) {
 		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
 					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_tx_irq_handler,
 						IRQF_ONESHOT, irq_name, mcasp);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 533c822ca6e6..c0edac80df34 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2800,7 +2800,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
 /**
  * snd_soc_dai_set_fmt - configure DAI hardware audio format.
  * @dai: DAI
- * @fmt: SND_SOC_DAIFMT_ format value.
+ * @fmt: SND_SOC_DAIFMT_* format value.
  *
  * Configures the DAI hardware format and clocking.
  */