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 ;-)
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