[FFmpeg-devel] AAC decoder handles start of audio stream differently (2.6.2 vs. current git)

[Hendrik Leppkes]

On Mon, May 9, 2016 at 11:20 PM, Gregory J Wolfe
<gregory.wolfe at 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