[alsa-devel] [PATCH] ASoC: codecs: use SNDRV_PCM_FMTBIT_* for format bitmask

[Takashi Iwai]

At Sat, 23 May 2015 18:32:29 +0200,
Maciej S. Szmigiero wrote:
> 
> 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>

Wow, that made me wonder how these drivers could actually work.

BTW, how did you detect it?  Any static analyzer like sparse or
smatch?  sparse didn't detect it at the last time I tried, IIRC...


thanks,

Takashi

> 
> 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,
>