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author | Anton Khirnov <anton@khirnov.net> | 2012-08-05 11:11:04 +0200 |
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committer | Anton Khirnov <anton@khirnov.net> | 2012-08-07 16:00:24 +0200 |
commit | 36ef5369ee9b336febc2c270f8718cec4476cb85 (patch) | |
tree | d186adbb488e7f002aa894743b1ce0e8925520e6 /libavdevice/alsa-audio.h | |
parent | 104e10fb426f903ba9157fdbfe30292d0e4c3d72 (diff) | |
download | ffmpeg-36ef5369ee9b336febc2c270f8718cec4476cb85.tar.gz |
Replace all CODEC_ID_* with AV_CODEC_ID_*
Diffstat (limited to 'libavdevice/alsa-audio.h')
-rw-r--r-- | libavdevice/alsa-audio.h | 10 |
1 files changed, 5 insertions, 5 deletions
diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h index ee43463696..a0e81d9e3e 100644 --- a/libavdevice/alsa-audio.h +++ b/libavdevice/alsa-audio.h @@ -38,7 +38,7 @@ /* XXX: we make the assumption that the soundcard accepts this format */ /* XXX: find better solution with "preinit" method, needed also in other formats */ -#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE) +#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE) #define ALSA_BUFFER_SIZE_MAX 32768 @@ -62,15 +62,15 @@ typedef struct { * @param sample_rate in: requested sample rate; * out: actually selected sample rate * @param channels number of channels - * @param codec_id in: requested CodecID or CODEC_ID_NONE; - * out: actually selected CodecID, changed only if - * CODEC_ID_NONE was requested + * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE; + * out: actually selected AVCodecID, changed only if + * AV_CODEC_ID_NONE was requested * * @return 0 if OK, AVERROR_xxx on error */ int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode, unsigned int *sample_rate, - int channels, enum CodecID *codec_id); + int channels, enum AVCodecID *codec_id); /** * Close the ALSA PCM. |