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author | Maciej S. Szmigiero <mail@maciej.szmigiero.name> | 2015-05-23 18:32:29 +0200 |
---|---|---|
committer | Mark Brown <broonie@kernel.org> | 2015-05-25 13:00:32 +0100 |
commit | e712bfca1ac1f63f622f87c2f33b57608f2a4d19 (patch) | |
tree | ab34256289bfda62f4ff9bbdf3f9e9ce5ded0395 | |
parent | b787f68c36d49bb1d9236f403813641efa74a031 (diff) | |
download | op-kernel-dev-e712bfca1ac1f63f622f87c2f33b57608f2a4d19.zip op-kernel-dev-e712bfca1ac1f63f622f87c2f33b57608f2a4d19.tar.gz |
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>
-rw-r--r-- | sound/soc/codecs/88pm860x-codec.c | 16 | ||||
-rw-r--r-- | sound/soc/codecs/stac9766.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8900.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm9713.c | 4 |
4 files changed, 13 insertions, 13 deletions
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index a0f2653..f62da48 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/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, }, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 6464caf..7f939ae 100644 --- a/sound/soc/codecs/stac9766.c +++ b/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, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 2eb986c..fdb7656 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 7955295..1b20b8d 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, |