> -----Original Message----- > From: ffmpeg-devel [mailto:ffmpeg-devel-boun...@ffmpeg.org] On Behalf > Of Hendrik Leppkes > Sent: Monday, May 09, 2016 5:27 PM > To: FFmpeg development discussions and patches <ffmpeg- > de...@ffmpeg.org> > Subject: Re: [FFmpeg-devel] AAC decoder handles start of audio stream > differently (2.6.2 vs. current git) > > On Mon, May 9, 2016 at 11:20 PM, Gregory J Wolfe > <gregory.wo...@kodakalaris.com> wrote: > > I am in the process of upgrading our FFmpeg from 2.6.2 to the latest > > git. One test I ran extracts audio from an AAC stream to a WAV file. > > When I examine the audio using Audacity, the stream extracted using > > the latest git is 1600 samples shorter, with the missing samples being > > from the beginning of the audio stream. Coincidentally, the first > > audio time stamp in the original audio stream is -1600 samples. So > > does 2.6.2 have a bug that is fixed in the latest git, or was a bug > > introduced into the latest git since 2.6.2? > > > > It is common for AAC to have padding at the beginning of the stream to > prime the decoder, those samples being dropped is the proper way to do > this. > And your timestamp seems to confirm this. > > So sounds like latest ffmpeg is doing it right to me. > > - Hendrik
OK, thanks, that makes sense. The codec info says that the audio sample delay is 1024 samples. So maybe that's the minimum amount, and in this case the author/original software that created the audio stream used 1600 samples because it works out to exactly 1/30 second. Greg Wolfe, Kodak Alaris _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
