[alsa-devel] [PATCH] ASoC: codecs: use SNDRV_PCM_FMTBIT_* for format bitmask
Maciej S. Szmigiero
mail at maciej.szmigiero.name
Sat May 23 18:32:29 CEST 2015
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 at maciej.szmigiero.name>
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index ee31fa7..38b3dad 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1186,16 +1186,16 @@ 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,
}, {
@@ -1207,16 +1207,16 @@ 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,
},
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 2341e8e..ed4cca7 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -320,7 +320,7 @@ 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,
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index e7d2ecd..493faf5 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/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,
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 9d18a0e..89cd2d6 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/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,
More information about the Alsa-devel
mailing list