[FFmpeg-devel] Exposing preroll and remainder info for aac and mp3.
michaelni at gmx.at
Wed Jul 4 16:24:51 CEST 2012
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
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.
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
Into a blind darkness they enter who follow after the Ignorance,
they as if into a greater darkness enter who devote themselves
to the Knowledge alone. -- Isha Upanishad
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