Path: utzoo!attcan!utgpu!jarvis.csri.toronto.edu!cs.utexas.edu!wuarchive!decwrl!shelby!csli!poser
From: poser@csli.Stanford.EDU (Bill Poser)
Newsgroups: comp.unix.wizards
Subject: Re: FCC doing it again...
Message-ID: <11416@csli.Stanford.EDU>
Date: 12 Dec 89 21:09:51 GMT
References: <21728@adm.BRL.MIL>
Sender: poser@csli.Stanford.EDU (Bill Poser)
Reply-To: poser@csli.stanford.edu (Bill Poser)
Organization: Center for the Study of Language and Information, Stanford U.
Lines: 23

In article <21728@adm.BRL.MIL> Kemp@DOCKMASTER.NCSC.MIL writes:

>Bill's facts are correct but not his conclusions.
>Speech with a bandwidth of 3.6 KHz can be transmitted with very good
>quality at 4800 bps...
>The telephone companies presently use ADPCM coding at 32 Kbps for most
>of their trunks, but work is underway on a low-delay CELP at 16 Kbps.
>Speech coding at 4.8 - 8.0 Kbps will be used first on digital cellular
>circuits where bandwidth is extremely tight...
>Bill Poser's comments on bit rate and distortion apply only to straight
>PCM, not speech compression systems.

Actually, my conclusions are essentially correct for the technology
in use, which I believe is what were discussing. ADPCM produces a relatively
small compression over straight PCM. I am well aware of the existence
of a variety of speech compression techniques that produce lower bit rates.
Indeed, the theoretical lower limit is considerably lower than 4800bps.
I have heard demos of research coding techniques at as low as ~50 bps.
(The theoretically ideal technique is to recognize the speech
at the input, transmit codes for the recognized segments, and then resynthesize
it again. Since speech recognition is hard, this kind of compression is hard
too.) But the present telephone system does not use the more elaborate
compression techniques.