[FFmpeg-devel] Flac encoder and MPEG4 ALS

[Kostya]

On Sun, Mar 30, 2008 at 03:35:22PM +0000, Jai Menon wrote:
> On Sunday 30 March 2008 08:48:52 Benjamin Zores wrote:
> > Jai Menon a ?crit :
> > > Hi,
> > >
> > > I started some basic work on the MPEG4 ALS encoder and I get the feeling
> > > that I could reuse a lot of code from the flac encoder. So
> > > (hypothetically ;-)) if I were to submit a patch, is it okay if I move
> > > some code out of flacenc.c to a shared source file?
> >
> > IIRC we don't even have a decoder for that atm.
> > Do you have a working ffmpeg decoder for that or are you using the
> > closed mp4alsRM19 binary ?
> 
> Very correct. It would be more sensible to write a decoder for  libavcodec 
> before pursuing this. But this "work" on the encoder is with reference to the 
> GSoC task. I'm using the RM20 reference decoder and it suits my current 
> purpose. Also i think that the encoder would be tested/verified against the 
> reference.

Just curious - is it possible now to distinguish ALS from ordinary AAC in mp4
container without extra hack (i.e. looking into extradata)?  

> I have submitted what i hope will be the final patch for my qual task. Once 
> that is done i'll have time to maybe get a basic als decoder and some code 
> refactoring done. Also there is that TAK reversing thing.......... ;-)

There's a bunch of closed-source, NIH-suffering lossless audio codecs
that are not worth looking at and TAK is one of them IMO (well, IIRC the author
promised to release specs somewhere in the distant future).

At least your (ALS) encoder quality should be better than mine (AAC).