Merge tag 'asoc-v3.14-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: More updates for v3.14

A few more updates for v3.14 since the last set, highlights include:

 - Lots of DMA updates from Lars-Peter
 - Improvements to the constraints handling code from Lars-Peter
 - A very helpful conversion of the TWL4030 driver to regmap from Peter
 - A new driver for the Freescale ESAI controller from Nicolin Chen
 - Conversion of some of the drivers to use params_width()

63 files changed, 2338 insertions, 982 deletions

diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 43f24cce3de..4560ca0e565 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -514,3 +514,42 @@ unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit)
 	return 0;
 }
 EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate);
+
+static unsigned int snd_pcm_rate_mask_sanitize(unsigned int rates)
+{
+	if (rates & SNDRV_PCM_RATE_CONTINUOUS)
+		return SNDRV_PCM_RATE_CONTINUOUS;
+	else if (rates & SNDRV_PCM_RATE_KNOT)
+		return SNDRV_PCM_RATE_KNOT;
+	return rates;
+}
+
+/**
+ * snd_pcm_rate_mask_intersect - computes the intersection between two rate masks
+ * @rates_a: The first rate mask
+ * @rates_b: The second rate mask
+ *
+ * This function computes the rates that are supported by both rate masks passed
+ * to the function. It will take care of the special handling of
+ * SNDRV_PCM_RATE_CONTINUOUS and SNDRV_PCM_RATE_KNOT.
+ *
+ * Return: A rate mask containing the rates that are supported by both rates_a
+ * and rates_b.
+ */
+unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
+	unsigned int rates_b)
+{
+	rates_a = snd_pcm_rate_mask_sanitize(rates_a);
+	rates_b = snd_pcm_rate_mask_sanitize(rates_b);
+
+	if (rates_a & SNDRV_PCM_RATE_CONTINUOUS)
+		return rates_b;
+	else if (rates_b & SNDRV_PCM_RATE_CONTINUOUS)
+		return rates_a;
+	else if (rates_a & SNDRV_PCM_RATE_KNOT)
+		return rates_b;
+	else if (rates_b & SNDRV_PCM_RATE_KNOT)
+		return rates_a;
+	return rates_a & rates_b;
+}
+EXPORT_SYMBOL_GPL(snd_pcm_rate_mask_intersect);
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
index 7f91a86dd73..6058c1fd507 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -236,8 +236,7 @@ static int axi_i2s_probe(struct platform_device *pdev)
 	if (ret)
 		goto err_clk_disable;
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
-			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret)
 		goto err_clk_disable;
 
diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c
index 8db7a992069..198e3a4640f 100644
--- a/sound/soc/adi/axi-spdif.c
+++ b/sound/soc/adi/axi-spdif.c
@@ -229,8 +229,7 @@ static int axi_spdif_probe(struct platform_device *pdev)
 	if (ret)
 		goto err_clk_disable;
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
-			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret)
 		goto err_clk_disable;
 
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index 06082e5e5dc..b79a2a86451 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -50,7 +50,6 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = {
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_RESUME |
 				  SNDRV_PCM_INFO_PAUSE,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.period_bytes_min	= 256,		/* lighting DMA overhead */
 	.period_bytes_max	= 2 * 0xffff,	/* if 2 bytes format */
 	.periods_min		= 8,
diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c
index 054ea4d9326..33ec592ecd7 100644
--- a/sound/soc/atmel/atmel-pcm-pdc.c
+++ b/sound/soc/atmel/atmel-pcm-pdc.c
@@ -58,7 +58,6 @@ static const struct snd_pcm_hardware atmel_pcm_hardware = {
 				  SNDRV_PCM_INFO_MMAP_VALID |
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_PAUSE,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192,
 	.periods_min		= 2,
diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig
index 3d82a29ce3a..6a834e109f1 100644
--- a/sound/soc/bcm/Kconfig
+++ b/sound/soc/bcm/Kconfig
@@ -1,7 +1,6 @@
 config SND_BCM2835_SOC_I2S
 	tristate "SoC Audio support for the Broadcom BCM2835 I2S module"
 	depends on ARCH_BCM2835 || COMPILE_TEST
-	select SND_SOC_DMAENGINE_PCM
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select REGMAP_MMIO
 	help
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index d7c983862cf..77f45986857 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -168,15 +168,15 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 	int word_len = 0;
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		word_len = AD1836_WORD_LEN_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		word_len = AD1836_WORD_LEN_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 24:
+	case 32:
 		word_len = AD1836_WORD_LEN_24;
 		break;
 	default:
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 12c27eb363d..5a42dca535b 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -249,15 +249,15 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 	struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		word_len = 3;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		word_len = 1;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 24:
+	case 32:
 		word_len = 0;
 		break;
 	}
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 59654b1e7f3..eb836ed5271 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1078,17 +1078,17 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream,
 		ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK,
 		(div << 2) | ADAU1373_BCLKDIV_64);
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		ctrl = ADAU1373_DAI_WLEN_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		ctrl = ADAU1373_DAI_WLEN_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		ctrl = ADAU1373_DAI_WLEN_24;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		ctrl = ADAU1373_DAI_WLEN_32;
 		break;
 	default:
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index ebff1128be5..d71c59cf7bd 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -71,7 +71,7 @@
 
 #define ADAU1701_SEROCTL_WORD_LEN_24	0x0000
 #define ADAU1701_SEROCTL_WORD_LEN_20	0x0001
-#define ADAU1701_SEROCTL_WORD_LEN_16	0x0010
+#define ADAU1701_SEROCTL_WORD_LEN_16	0x0002
 #define ADAU1701_SEROCTL_WORD_LEN_MASK	0x0003
 
 #define ADAU1701_AUXNPOW_VBPD		0x40
@@ -299,20 +299,20 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
 }
 
 static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
-		snd_pcm_format_t format)
+					   struct snd_pcm_hw_params *params)
 {
 	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
 	unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK;
 	unsigned int val;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAU1701_SEROCTL_WORD_LEN_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAU1701_SEROCTL_WORD_LEN_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAU1701_SEROCTL_WORD_LEN_24;
 		break;
 	default:
@@ -320,14 +320,14 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
 	}
 
 	if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) {
-		switch (format) {
-		case SNDRV_PCM_FORMAT_S16_LE:
+		switch (params_width(params)) {
+		case 16:
 			val |= ADAU1701_SEROCTL_MSB_DEALY16;
 			break;
-		case SNDRV_PCM_FORMAT_S20_3LE:
+		case 20:
 			val |= ADAU1701_SEROCTL_MSB_DEALY12;
 			break;
-		case SNDRV_PCM_FORMAT_S24_LE:
+		case 24:
 			val |= ADAU1701_SEROCTL_MSB_DEALY8;
 			break;
 		}
@@ -340,7 +340,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
 }
 
 static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
-			snd_pcm_format_t format)
+					    struct snd_pcm_hw_params *params)
 {
 	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
@@ -348,14 +348,14 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
 	if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
 		return 0;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAU1701_SERICTL_RIGHTJ_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAU1701_SERICTL_RIGHTJ_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAU1701_SERICTL_RIGHTJ_24;
 		break;
 	default:
@@ -374,7 +374,6 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = dai->codec;
 	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
 	unsigned int clkdiv = adau1701->sysclk / params_rate(params);
-	snd_pcm_format_t format;
 	unsigned int val;
 	int ret;
 
@@ -406,11 +405,10 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream,
 	regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL,
 		ADAU1701_DSPCTRL_SR_MASK, val);
 
-	format = params_format(params);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		return adau1701_set_playback_pcm_format(codec, format);
+		return adau1701_set_playback_pcm_format(codec, params);
 	else
-		return adau1701_set_capture_pcm_format(codec, format);
+		return adau1701_set_capture_pcm_format(codec, params);
 }
 
 static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai,
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index f7bf4555274..f78b27a7c46 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -453,22 +453,22 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
 }
 
 static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, snd_pcm_format_t format)
+		struct snd_soc_dai *dai, struct snd_pcm_hw_params *params)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAV80X_CAPTURE_WORD_LEN16;
 		break;
-	case SNDRV_PCM_FORMAT_S18_3LE:
+	case 18:
 		val = ADAV80X_CAPTRUE_WORD_LEN18;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAV80X_CAPTURE_WORD_LEN20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAV80X_CAPTURE_WORD_LEN24;
 		break;
 	default:
@@ -482,7 +482,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
 }
 
 static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, snd_pcm_format_t format)
+		struct snd_soc_dai *dai, struct snd_pcm_hw_params *params)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
@@ -490,17 +490,17 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
 	if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J)
 		return 0;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16;
 		break;
-	case SNDRV_PCM_FORMAT_S18_3LE:
+	case 18:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
 		break;
 	default:
@@ -524,12 +524,10 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		adav80x_set_playback_pcm_format(codec, dai,
-			params_format(params));
+		adav80x_set_playback_pcm_format(codec, dai, params);
 		adav80x_set_dac_clock(codec, rate);
 	} else {
-		adav80x_set_capture_pcm_format(codec, dai,
-			params_format(params));
+		adav80x_set_capture_pcm_format(codec, dai, params);
 		adav80x_set_adc_clock(codec, rate);
 	}
 	adav80x->rate = rate;
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 256c364193a..d3036283482 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -714,17 +714,17 @@ static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
 	iface &= ~ALC5623_DAI_I2S_DL_MASK;
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		iface |= ALC5623_DAI_I2S_DL_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		iface |= ALC5623_DAI_I2S_DL_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		iface |= ALC5623_DAI_I2S_DL_24;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		iface |= ALC5623_DAI_I2S_DL_32;
 		break;
 	default:
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 19e9f222d09..fb001c56cf8 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -869,14 +869,14 @@ static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream,
 	iface &= ~ALC5632_DAI_I2S_DL_MASK;
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		iface |= ALC5632_DAI_I2S_DL_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		iface |= ALC5632_DAI_I2S_DL_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		iface |= ALC5632_DAI_I2S_DL_24;
 		break;
 	default:
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 10b39847720..16df0f91335 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -166,20 +166,21 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 	ARIZONA_MIXER_INPUT_ROUTES(name " Input 4")
 
 #define ARIZONA_DSP_ROUTES(name) \
-	{ name, NULL, name " Aux 1" }, \
-	{ name, NULL, name " Aux 2" }, \
-	{ name, NULL, name " Aux 3" }, \
-	{ name, NULL, name " Aux 4" }, \
-	{ name, NULL, name " Aux 5" }, \
-	{ name, NULL, name " Aux 6" }, \
+	{ name, NULL, name " Preloader"}, \
+	{ name " Preloader", NULL, name " Aux 1" }, \
+	{ name " Preloader", NULL, name " Aux 2" }, \
+	{ name " Preloader", NULL, name " Aux 3" }, \
+	{ name " Preloader", NULL, name " Aux 4" }, \
+	{ name " Preloader", NULL, name " Aux 5" }, \
+	{ name " Preloader", NULL, name " Aux 6" }, \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 1"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 2"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 3"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 4"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 5"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 6"), \
-	ARIZONA_MIXER_ROUTES(name, name "L"), \
-	ARIZONA_MIXER_ROUTES(name, name "R")
+	ARIZONA_MIXER_ROUTES(name " Preloader", name "L"), \
+	ARIZONA_MIXER_ROUTES(name " Preloader", name "R")
 
 #define ARIZONA_RATE_ENUM_SIZE 4
 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 1e0fa3b5f79..6e9ea8379a9 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -423,21 +423,17 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream,
 		intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24);
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		switch (params_format(params)) {
-		case SNDRV_PCM_FORMAT_S16_LE:
-		case SNDRV_PCM_FORMAT_S16_BE:
+		switch (params_width(params)) {
+		case 16:
 			fmt = CS42L51_DAC_DIF_RJ16;
 			break;
-		case SNDRV_PCM_FORMAT_S18_3LE:
-		case SNDRV_PCM_FORMAT_S18_3BE:
+		case 18:
 			fmt = CS42L51_DAC_DIF_RJ18;
 			break;
-		case SNDRV_PCM_FORMAT_S20_3LE:
-		case SNDRV_PCM_FORMAT_S20_3BE:
+		case 20:
 			fmt = CS42L51_DAC_DIF_RJ20;
 			break;
-		case SNDRV_PCM_FORMAT_S24_LE:
-		case SNDRV_PCM_FORMAT_S24_BE:
+		case 24:
 			fmt = CS42L51_DAC_DIF_RJ24;
 			break;
 		default:
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 8166dcb2e4a..e62e294a803 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -778,17 +778,17 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 
 	dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1);
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		dai_cfg1 |= DA7210_DAI_WORD_S16_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		dai_cfg1 |= DA7210_DAI_WORD_S20_3LE;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		dai_cfg1 |= DA7210_DAI_WORD_S24_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		dai_cfg1 |= DA7210_DAI_WORD_S32_LE;
 		break;
 	default:
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 4a6f1daf911..0c77e7ad742 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1067,17 +1067,17 @@ static int da7213_hw_params(struct snd_pcm_substream *substream,
 	u8 fs;
 
 	/* Set DAI format */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		dai_ctrl |= DA7213_DAI_WORD_LENGTH_S16_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		dai_ctrl |= DA7213_DAI_WORD_LENGTH_S20_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE: