I have two questions 1) How to detect garbage data in front of the audio frame ( for example if it's at the middle of a frame when i start to capture it from a streaming audio over internet. ) so i can remove first x bytes from the file automatically in my code.
2) ffmpeg does not multiply sample rate of lc aac sbr, so final file plays in slow motion (22050 instead of 44100). ( note: faad itself perfectly converts and multiplys it by just itself out of ffmpeg ). How to solve this problem for ffmpeg? Thanks. _______________________________________________ libav-user mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/libav-user
