[vorbis] FREE at last

[Kenneth Arnold]

On Sun, Oct 08, 2000 at 08:54:42PM -0700, Ralph Giles wrote:
> On Sun, 8 Oct 2000, Kenneth Arnold wrote:
> 
> > Tarkin? Where is that anyway?
> 
> I'd just start my own version. The thread on the format starts here:
> 
> http://www.xiph.org/archives/vorbis-dev/1329.html
> 
> A wavelet and a vector quantization tutorial should be enough to get
> something working; Monty &c. can clean it up for us later. :)

Hmmm... good idea. Any data suggesting how good a compression this achieves
and how it compares to MPEG-whatever?

I saw a link earlier on vorbis-dev for a chirplet-based system -- I'll have
to dig that up and look at it some more. Can't remember who it was, but
it was probably Ralph because he has done a lot of (NS)MLP. :)

Finally, did it include motion comp. or not? Could this improve things?
fiasco (as pointed out earlier) has a motion comp engine that looks like it
was stolen from somewhere, and the overall result of the two combined looks
pretty darn decent at low-bitrate. The scary thing is that I might actually
be understanding how weighted finite automata work...

I have found other things, including a DCT in 3D, that might be interesting...
I think any Ogg video project should pull together, hopefully without too
much work, all the research the graduate students have spent years working
on. :) The wonders of the Internet :)

> > I've found some video codec stuff myself, and am seriously considering
> > porting them over to the Ogg framework to ease playing around. Having
> > not yet delved into code, I wonder about frame-sync issues -- how can
> > I get a frame of video to match up with a position in the audio stream?
> > Forgive me for asking if this is blatantly simple.
> 
> Well, it is and it isn't. Ogg provides minimal timestamping in the
> framing which you can use for rough seeking and prediction. On top of
> that, most codecs provide a time or sample index within that.
> 
> What's a little harder is deciding how to maintain sync between the
> audio and the video. Most consumer applications synchronize the video to
> the audio. Frame-to-display = current-audio-sample / audio-samples-rate *
> video-frame-rate. Current-audio-sample is determined by what just went
> into the audio output buffer. There are two reasons for this: audio
> glitches are perceptually much more damaging than video glitches, and the
> clocks in most sound cards are terrible. So, sync video to whatever rate
> the audio's playing at, and nobody will notice the few-precent difference.
> 
> However, there are applications where it's important to sync to either
> "realtime" or an external clock. Maybe you're mixing live sources, or
> driving multiple unslaved outputs, or just care on general principles.
> This is a little harder, since you have to synchronize both audio and
> video to an external clock. The video code can be the same, but I
> imagine you'd need to do some kind of smooth stretching on the audio
> to avoid glitches at the adjustments. I believe most professional setups
> just slave everything (including the ADC/DACs) to an external clock;
> not something we can really do in software.

Maybe rtlinux could do sync issues better. But anyway, there needs to be some
sort of marker, either in the metadata or built into the stream itself, to
synchronize all the elements of the stream in time. So far we have been able
to ignore all that, because it's just audio, but there can be video, lyrics,
closed-captioning, links, markers, etc. that all will need to be synced to
something, probably the audio. From what you describe, it looks like the
facilities to do that are pretty minimal.

> Hope that helps,

Sure does.

Kenneth

>  -ralph
> 
> --
> giles at ashlu.bc.ca
> 
> 
> 
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-- 
Kenneth Arnold <ken at arnoldnet.net>
Slashdot and K5: kcarnold
Proudly registered Linux user #180115! See http://counter.li.org/

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