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Modify av_audio_convert() to use AVAudioConvert context struct; add av_audio_convert_alloc() and av_audio_convert_free() support functions.

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Originally committed as revision 14496 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Peter Ross
2008-08-01 13:53:18 +00:00
parent 8a464e7580
commit 82cee279a5
2 changed files with 69 additions and 11 deletions
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34
libavcodec/audioconvert.h
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@ -54,4 +54,38 @@ const char *avcodec_get_sample_fmt_name(int sample_fmt);
*/
enum SampleFormat avcodec_get_sample_fmt(const char* name);
struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param out_channels Number of output channels
* @param in_fmt Input sample format
* @param in_channels Number of input channels
* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
* @param flags See FF_MM_xx
* @return NULL on error
*/
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
const float *matrix, int flags);
/**
* Free audio sample format converter context
*/
void av_audio_convert_free(AVAudioConvert *ctx);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] out_stride distance between consecutive input samples (measured in bytes)
* @param[in] in array of input buffers for each channel
* @param[in] in_stride distance between consecutive output samples (measured in bytes)
* @param len length of audio frame size (measured in samples)
*/
int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len);
#endif /* FFMPEG_AUDIOCONVERT_H */