ASoC: codecs: use SNDRV_PCM_FMTBIT_* for format bitmask

snd_soc_pcm_stream.formats is a bitmask of SNDRV_PCM_FMTBIT_*,
not of SNDRV_PCM_FORMAT_* (which are sequential integers),
however some of ASoC CODEC drivers use these values instead.

Found out by sparse on 0-day kernel tester.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

sound/soc/codecs/88pm860x-codec.c

@@ -1187,16 +1187,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = {
 			.channels_min	= 2,
 			.channels_max	= 2,
 			.rates		= PM860X_RATES,
-			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
-					  SNDRV_PCM_FORMAT_S18_3LE,
+			.formats	= SNDRV_PCM_FMTBIT_S16_LE | \
+					  SNDRV_PCM_FMTBIT_S18_3LE,
 		},
 		.capture = {
 			.stream_name	= "PCM Capture",
 			.channels_min	= 2,
 			.channels_max	= 2,
 			.rates		= PM860X_RATES,
-			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
-					  SNDRV_PCM_FORMAT_S18_3LE,
+			.formats	= SNDRV_PCM_FMTBIT_S16_LE | \
+					  SNDRV_PCM_FMTBIT_S18_3LE,
 		},
 		.ops	= &pm860x_pcm_dai_ops,
 	}, {
@@ -1208,16 +1208,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = {
 			.channels_min	= 2,
 			.channels_max	= 2,
 			.rates		= SNDRV_PCM_RATE_8000_48000,
-			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
-					  SNDRV_PCM_FORMAT_S18_3LE,
+			.formats	= SNDRV_PCM_FMTBIT_S16_LE | \
+					  SNDRV_PCM_FMTBIT_S18_3LE,
 		},
 		.capture = {
 			.stream_name	= "I2S Capture",
 			.channels_min	= 2,
 			.channels_max	= 2,
 			.rates		= SNDRV_PCM_RATE_8000_48000,
-			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
-					  SNDRV_PCM_FORMAT_S18_3LE,
+			.formats	= SNDRV_PCM_FMTBIT_S16_LE | \
+					  SNDRV_PCM_FMTBIT_S18_3LE,
 		},
 		.ops	= &pm860x_i2s_dai_ops,
 	},

sound/soc/codecs/stac9766.c

@@ -321,7 +321,7 @@ static struct snd_soc_dai_driver stac9766_dai[] = {
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_32000 | \
 			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
+		.formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
 	},
 	/* alsa ops */
 	.ops = &stac9766_dai_ops_digital,

sound/soc/codecs/wm8900.c

@@ -998,8 +998,8 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 		      SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
 #define WM8900_PCM_FORMATS \
-	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
-	 SNDRV_PCM_FORMAT_S24_LE)
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	 SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct snd_soc_dai_ops wm8900_dai_ops = {
 	.hw_params	= wm8900_hw_params,

sound/soc/codecs/wm9713.c

@@ -1054,8 +1054,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 			  SNDRV_PCM_RATE_48000)
 
 #define WM9713_PCM_FORMATS \
-	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
-	 SNDRV_PCM_FORMAT_S24_LE)
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	 SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
 	.prepare	= ac97_hifi_prepare,