Input file original.wav
ffmpeg -i original.wav ... Duration: 00:03:36.93, bitrate: 1411 kb/s
Convert wav to m4a with fdk-aac encoder:
ffmpeg -i original.wav -c:a libfdk_aac -vbr 5 ffmpeg.m4a
Check duration of resulting m4a:
ffmpeg -i ffmpeg.m4a ... Duration: 00:03:36.98, start: 0.046440, bitrate: 240 kb/s
So, ffmpeg adds ~2000 zero samples at beginning of resulting file. How to disable this?
I try few different codecs and all work fine, problem only with fdk-aac.
- ffmpeg versions: 2.2.10 and 2.4.3
- fdk-aac version: 0.1.1 and 0.1.3
UPD1: I try fdkaac and vlc (it also use fdk-aac library). Both produce correct file (no zero samples at begin). So problem only with ffmpeg+fdk.