* Mark Burton on Friday, April 14, 2017 at 16:57:06 +0100 > I appreciate this is a tricky area and there appear to be different ways that > some encoders create AAC streams with regards to the padding and remaining > samples etc. I won’t pretend to fully understand all the factors, but I would > like to ask a genuine question which purely comes from the point of view of > wanting to create a file for my working environment - an environment > dominated by Quicktime 7 and X playback / decoding tools. I’m no great fan of > Quicktime and appreciate its not well loved here also. In my industry it is > still very much the defecto playback engine though, so if I’m able to tailor > a file for this decoder, it would be an enormous help. > > Let me also say, I am not accusing ffmpeg of having an issue, I have read a > number of ‘bug’ reports surrounding ffmpeg and AAC priming and there being a > sync discrepancy in the resultant encode when played back in certain > decoders. I happen to see the exact same issue, but from some of the > developers replies, I accept their position is that they feel ffmpeg is doing > it the right way and its the decoders that are at fault. > > With that said, I’d like to approach this question purely from the point of > view of finding out whether there is a way to tweak a command in order to > change this way the aac stream is created to produce an mp4 or mov file using > the native aac encoder which decodes in Quicktime 7 or X, in sync. Currently > an encoded file plays back 1 frame out of sync (audio is early by approx. 1 > frame). In VLC its about 1/2 a frame out of sync. > > The source file is a .mov, DNx115 24p (true 24p, not 23.976), PCM 24bit 48khz > audio, which is in sync. This is film material where sync is crucial and > always expected. > > Here is the basic command to reproduce. I have attached the uncut loglevel 99 > console output for this command: > ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p -movflags faststart > -c:a aac -b:a 128k ffmpeg.mp4
Can you try: -filter:a aresample=first_pts=0 Also, when you run with -v verbose, you'll see a delay (depends on audio codec), for you case it's probably 1024. Maybe try: -filter:a aresample=first_pts=0,asetpts=PTS-STARTPTS+1024 Especially the latter could be exactly the wrong thing for your purpose, but it doesn't hurt trying. -- Was heißt hier Dogma, ich bin Underdogma! [ What the hell do you mean dogma, I am underdogma. ] free movies --->>> https://blacktrash.org/underdogma http://itunes.apple.com/podcast/underdogma-movies/id363423596 _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
