[FFmpeg-devel] Exposing preroll and remainder info for aac and mp3.
michaelni at gmx.at
Wed Jul 4 23:21:04 CEST 2012
On Wed, Jul 04, 2012 at 04:24:51PM +0200, Michael Niedermayer wrote:
> On Wed, Jul 04, 2012 at 04:05:17PM +0200, Nicolas George wrote:
> > Le septidi 17 messidor, an CCXX, Michael Niedermayer a écrit :
> > > are there some reference files (mp3/aac + wav refernce of what should
> > > be the delay free output ?)
> > > These would be very usefull to implement this as well as for
> > > regression tests so it doesnt break in the future once it works.
> > Is it really necessary? We can see for ourselves with any audio file that
> > does not have too much silence on the beginning. For example:
> > ffmpeg -f lavfi -i aevalsrc='abs(sin(2*PI*t))*sin(2*PI*440*t)::d=10' \
> > test.mp3 test.wav
> > ffmpeg -i test.wav -i test.mp3 -filter_complex '
> > [0:0] showwaves=n=15:s=1470x64, pad=iw:ih*2 [a] ;
> > [1:0] showwaves=n=15:s=1470x64 [b] ;
> > [a] [b] overlay=0:64' -vframes 1 test.png
> > The output.png shows clearly a 74 pixels delay for the MP3 file, which makes
> > about 1110 samples.
> > I believe our purpose should be that a round trip encode+decode gives a file
> > that is in sync with the original. lame + lame --decode does that, for
> > example.
> we should aim toward getting our decoder round trip exact with any
> other encoder be that lame, faac, ffmpeg, itunes or other.
> for this we need sample files from all these encoders.
just posted a patchset implementing initial silence skiping for mp3
doing the same with mov/mp4 aac should be trivial, id need a reference
sample though to make sure the aac/mov/mp4 stuff is correct. Its easy
to get it wrong without a testcase ...
the mp3 code should be compatible with lames way of indicating the
amount of padding.
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