Vorbis dev - Dec 2007 - Experimental release of Ghost/CELT 0.0.1

Hi everyone,

I've just made the first public release of some new *experimental* codec
work I've been doing (part of the vague Ghost project) with help from
Monty and Timothy. This is mainly intended with developers with DSP
knowledge, not for doing anything useful with it (but it does encode and
decode already). Also, the main idea is *not* to replace either Speex or
Vorbis, but to code audio with really low latency -- currently 8 ms.

This is still very experimental and everything is still likely to
change, including the exact goals. The algorithm is called (temporary
name) Code-Excited Lapped Transform (CELT) and the main ideas are:
- Using an MDCT on very short frames
- Dividing into 15 bands and transmitting the energy for each band
- Using a pitch predictor (good for speech, but helps for music as well).
- The rest is coded using a unit-pulse codebook.
At this point, I'm still trying to figure out how to fit psychoacoustics
into this.

CELT is based on a paper I submitted to ICASSP and which I'm hoping will
be accepted so I can make it available to everyone. The only difference
is that the ICASSP paper was based on the FFT (non critically sampled),
whereas this version is based on the MDCT. One part that is already
published though is Tim's explanation of the pulse codebook encoding
(http://people.xiph.org/~tterribe/cwrs.html).

The full source for CELT is available at:
http://downloads.us.xiph.org/releases/celt/celt-0.0.1.tar.gz
or through git at http://git.xiph.org/celt.git

I've put some music samples at 56 kbps CBR at
http://people.xiph.org/~jm/comp_celt58cbr.wav with the original at
http://people.xiph.org/~jm/comp44.wav . As you can hear, it definitely
doesn't suck as much as Speex on music, but there's still room for
improvement.

I'm open to interesting ideas, but don't bother complaining if it
doesn't work or if it explodes in your face :-) Oh, and don't expect a
final codec any time soon.

Have fun (for some definition of fun),

	Jean-Marc

>>>>> "Jean-Marc" == Jean-Marc Valin
<jean-marc.valin@usherbrooke.ca> writes:
> Hi everyone, I've just made the first public release of some new
> *experimental* codec work I've been doing (part of the vague Ghost
> project) with help from Monty and Timothy. This is mainly intended
> with developers with DSP knowledge, not for doing anything useful with
> it (but it does encode and decode already). Also, the main idea is
> *not* to replace either Speex or Vorbis, but to code audio with really
> low latency -- currently 8 ms.
I just wanted to complain about your codec using (patent-encumbered)
arithmetic encoding, but now I noticed you're actually using range
compression, all of this wrapped in a neat reusable library.

Not bad.  I somewhat had the impression that the non-adaptive huffman
coding was one thing holding patent free codecs (ie Vorbis, Theora)
back.

So any chance we're also going to see upgrades to Vorbis, Theora? :)

Personally I would be more interested in listening to a 32kbit sample.
This is a bitrate I never managed Vorbis to output any decent quality
at.  Compared to some 32kbit realaudio webcasts I've listened to which
were almost artifact free.  Hmm, your code compiles, but I don't seem to
find the bitrate knob.  I guess this is because it's not yet
implemented?

cheers,

David

PS: the URLs to papers in 'bitree.c' are broken and need to be
updated...
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