[vorbis] further development

[Robert Michel]

Salve Stan,

thank you for your mail and the link to speex.sf.net.
> Additionally, telephony applications usually are interested in encoding
> speech, which Vorbis is not optimized for.  Have you looked at Speex?
> (http://speex.sf.net)  It is a speech codec which will probably have
> much better compression on voice.  I don't know how the latency is on
> it.
The frame is 20ms - so the latency should be <30 ms :-)))

But saw vorbis ogg as powerfull to be replacement for ACC.
Imgagine a Journalist is giving an interview with low bitrate 48kBit/s from
a hotspot somewere in the world. The radiostation is streaming with
vorbis ogg in 128kBit/S 64/kBits and 32 KBit/s.
So why reencode the orginal stream with loosing qualitiy and blow it up
to 128kBit/s?
Why not bypass the original stream and send 48kBit/s to the 128kBit/s
listeners, too?

So bitrate is not everythink to have a good quality a strategie to avoid
reencoding is important, too.

I think speex is powerfull and good for telephonie via internet. To build up
a direct stream technogie for Internet-streamers under GNU / Free software
will be not important engough to motivate others to build a one solution
coodec Hight quality, fast and low delay if wanted. ;-)

But I still see a need to build a 48kBit/s,-90 KBit/s Mono
and 96kBit/s - 196Kbit/s Stereo codec with Low Delay and 
good quality for voice and music. So if someone share my 
points, please contact me.

Thanks again for your mails,
rob

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