Mercurial > libavcodec.hg
diff audioconvert.h @ 7459:283eeda62184 libavcodec
Modify av_audio_convert() to use AVAudioConvert context struct; add av_audio_convert_alloc() and av_audio_convert_free() support functions.
author | pross |
---|---|
date | Fri, 01 Aug 2008 13:53:18 +0000 |
parents | d1d15f2dca4c |
children | c4a4495715dd |
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--- a/audioconvert.h Fri Aug 01 11:26:22 2008 +0000 +++ b/audioconvert.h Fri Aug 01 13:53:18 2008 +0000 @@ -54,4 +54,38 @@ */ 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 */