On Apr 26, 2013, at 1:45 PM, Bruce Wheaton <[email protected]> wrote:
> It may be that your source of dest buffers, if not from ffmpeg, are not > correctly aligned. To the best of my knowledge, the source data is aligned, exactly as needed. Reasons for saying so: 1. The meta-data on the QTSampleBuffer coming over indicates that this is so. 2. The same code / source data being used with the FLTP sample format / AAC encoder works perfect with a S16 sample format / ADPCM_SWF encoder. An alignment problem with the source data most likely would cause a problem in both scenarios. 3. I've tried both 0 and 1 for align parameters throughout my code, and it has absolutely zero effect on the output audio -- same distortion problem. 4. As an alternative approach to setting buffer/channel pointers, I've manually moved every value from the captured buffer to the source data array for resampling. Again, no change in output. I'll take other avenues. Thanks, Brad _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
