On date Friday 2012-11-30 21:51:18 +0200, Rafi Fisher encoded: > hello > > i try to trancode audio frames decoded from mp2 file into wav file. > > so from what i understand > > my input buffer should be the: *uint8_t *data *field in the*AVFrame > * structure > > my output buffer should be allocated using : >
> int* av_samples_alloc*(uint8_t **audio_data, int *linesize, int > nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align); > > what size should i put in* nb_samples* parameter ? It depends, I suppose it will be the number of decoded samples. > > then i understand i should call : > > int *swr_convert*(struct SwrContext *s, uint8_t **out, int > out_count, const uint8_t **in , int in_count); > > and pass* AVFrame data* filed as input buffer & the allocated > samples buffer as the output buffer. > > how do i calculate the* in_count & the out_count*. parameters. It should be big enough to contain the samples after conversion samples. Check this code: http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/155151 [...] _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
