[Ffmpeg-devel] FAAC rate control broken?

[Luca Abeni]

Hi all,

I am trying to encode some AAC audio with ffmpeg (+ libfaac, version
1.24), but it seems that the "-ab" option does not work...

Here is a way to reproduce the problem:
1) capture some PCM audio:
./ffmpeg -ac 2 test.wav

2) encode it with aac at 60kbps:
./ffmpeg -i test.wav -ab 64 -acodec aac a.wav
ffmpeg version CVS, build 3342336, Copyright (c) 2000-2004 Fabrice Bellard
  configuration:  --cc=gcc-4.0 --enable-faac --disable-network --disable-zlib --disable-vhook --disable-ffplay --extra-cflags=-I /tmp/aac/include --extra-ldflags=-L /tmp/aac/lib 
  built on Dec 21 2005 00:30:58, gcc: 4.0.3 20051201 (prerelease) (Debian 4.0.2-5)
Input #0, wav, from 'test.wav':
  Duration: 00:00:35.5, start: 0.000000, bitrate: 1411 kb/s
  Stream #0.0: Audio: pcm_s16le, 44100 Hz, stereo, 1411 kb/s
File 'a.wav' already exists. Overwrite ? [y/N] y
Output #0, wav, to 'a.wav':
  Stream #0.0: Audio: aac, 44100 Hz, stereo, 64 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Press [q] to stop encoding
size=     356kB time=35.5 bitrate=  82.1kbits/s    
(so, the output bitrate is 82.1kbps)

3) Encode at 40kbps
 ./ffmpeg -i test.wav -ab 40 -acodec aac b.wav
ffmpeg version CVS, build 3342336, Copyright (c) 2000-2004 Fabrice Bellard
  configuration:  --cc=gcc-4.0 --enable-faac --disable-network --disable-zlib --disable-vhook --disable-ffplay --extra-cflags=-I /tmp/aac/include --extra-ldflags=-L /tmp/aac/lib 
  built on Dec 21 2005 00:30:58, gcc: 4.0.3 20051201 (prerelease) (Debian 4.0.2-5)
Input #0, wav, from 'test.wav':
  Duration: 00:00:35.5, start: 0.000000, bitrate: 1411 kb/s
  Stream #0.0: Audio: pcm_s16le, 44100 Hz, stereo, 1411 kb/s
Output #0, wav, to 'b.wav':
  Stream #0.0: Audio: aac, 44100 Hz, stereo, 40 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Press [q] to stop encoding
size=     356kB time=35.5 bitrate=  82.1kbits/s    
(still 82.1kbps...)

So, it seems that the bitrate control is not working very well... :(

I tried to have a look at libavcodec/faac.c, and it seems that the
problem is that Faac_encode_init() does not set faac_cfg->bandWidth...
As an experiment, I tried to set it to
avctx->bit_rate / avctx->channels / 2
and things look a little bit better:
./ffmpeg -i test.wav -ab 40 -acodec aac a.wav
ffmpeg version CVS, build 3342336, Copyright (c) 2000-2004 Fabrice Bellard
  configuration:  --cc=gcc-4.0 --enable-faac --disable-network --disable-zlib --disable-vhook --disable-ffplay --extra-cflags=-I /tmp/aac/include --extra-ldflags=-L /tmp/aac/lib 
  built on Dec 21 2005 00:30:58, gcc: 4.0.3 20051201 (prerelease) (Debian 4.0.2-5)
Input #0, wav, from 'test.wav':
  Duration: 00:00:35.5, start: 0.000000, bitrate: 1411 kb/s
  Stream #0.0: Audio: pcm_s16le, 44100 Hz, stereo, 1411 kb/s
File 'a.wav' already exists. Overwrite ? [y/N] y
Output #0, wav, to 'a.wav':
  Stream #0.0: Audio: aac, 44100 Hz, stereo, 40 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Press [q] to stop encoding
size=     275kB time=35.5 bitrate=  63.5kbits/s    
still the wrong bitrate (63.5 instead of 40), but it is less than with
-ab 64...

Does anyone have any idea about how to fix the problem?


			Thanks,
				Luca
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