Re: Zaptel Pseudo TDM Bus

From: Daniel Phillips (phillips@arcor.de)
Date: Sun Jul 21 2002 - 15:04:06 EST


On Sunday 21 July 2002 21:01, Mark Spencer wrote:
> > A random question: is there any reason why Ogg isn't among the codecs?
>
> It wasn't ready when I started. Ogg, like mp3, is generally a very poor
> choice of codec for telephony, and even for the storage of files, unless
> its performance has improved greatly.

I don't know about performance (except for quality: it's said to require
about half the bitrate for the same quality, compared to mp3) however, it
has one killer advantage over mp3: it's patent-free, and hence, royalty-free.
I'd think that would be important for your project.

> On a 900 Mhz Athlon, you can get *hundreds* of simultaneous full-duplex
> GSM full-rate codecs running. Certainly that's an unrealistic expectation
> even for half-duplex ogg or mp3.

But that would be an argument against supporting mp3 as well.

> As for using ogg as an actual telephony
> protocol, its frame size is (or at least was at the time I contacted the
> author) much too long to be practical. Frame sizes for VoIP should be
> around 160 to 240 samples in general.

On a quick reading, this appears to indicate you can easily have what you
want:

   http://www.xiph.org/ogg/vorbis/doc/framing.html

Perhaps the last time you looked at Ogg the streaming format had not yet been
completed?

Also, doesn't part of telephony consist of having lots of pre-recorded audio
around, for voice mail etc? Granted, an encoder optimized for music is not
necessarily optimzed for voice. However, would that not be a matter of
tweaking the encoder? As I understand it, the vorbis compression format is
quite general, and in fact, all the recent work that improved the quality so
noticably involved only the encoder.

OK, this isn't a really kernel issue, so... I'll clamp my hams and post it
anyway ;-)

-- 
Daniel
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