All right, John--

A couple of years ago (discreet cough), Cisco gave away copies of books as
promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw Hill, 2000).
GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate and voice
sampling:

"...Thus, if an analog voice signal reaching up to 3400Hz is to be sampled
at the Nyquist rate, the sampling frequency must be at least twice that, or
6800Hz, or samples per second.

"Sampling does not have to be done at the Nyquist rate. The Nyquist rate is
a minimal requirement to reproduce the input waveform, but sampling can be
done at rates higher or lower than the Nyquist rate. If sampling takes place
at rates lower than the Nyquist rate, the result is distortion of the
waveform known as (italics) aliasing. Aliasing just means that there is more
than one output waveform that fits the 'connect the dots' pattern of the
samples. There is no aliasing ast the Nyquist rate and above."

They go on to point out that, by sampling at a rate above the Nyquist rate,
you have more than the minimum required information to reliably reconstruct
the voice signal at the destination. This allows you to lose a few samples
in transit (not that such things would ever happen, of course) and still
have only one possible reconstruction. Sampling at 8000Hz means there is a
4000Hz voice bandwidth (overly generous but convenient because 4 is a power
of 2 and that makes it easier to code in a binary system).

And from the 8000 samples/sec, each of which sends 1 8-bit word, we have the
DS0 of 64000 bps (why only 56000 bps may be usable is a separate issue,
having to do with signaling on telephone links).

Annlee
""John Neiberger""  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
> This is OT, but the upper limit of human hearing is actually
> around 20KHz at best and usually drops to around 16KHz or so.
> If your upper limit starts to drop below that you'll start to
> notice that it's difficult to hear clearly.  (Sorry, in my
> other life I'm a sound engineer and musician.)
>
> I've heard that the 4KHz limit is because there is a low-pass
> filter used for voice.  I can't remember the exact reason, but
> that information plugged into the Nyquist theorem explains--as
> Priscilla mentions--why a DS0 is 64Kbps.
>
> Okay, time to do some serious studying once I'm through being
> lazy and drinking this coffee...
>
> John
>
> ---- On Tue, 26 Feb 2002, Priscilla Oppenheimer
> ([EMAIL PROTECTED]) wrote:
>
> > At 08:06 PM 2/26/02, Rafay wrote:
> > >How do you describe Sample Rate.?
> >
> > In what context? The term is sometimes used when describing
> the analog
> > to
> > digital process, for example when digitizing voice. Voice
> produces an
> > analog wave as your lungs and tongue press against the air.
> An analog
> > wave
> > has infinite possible values. Computers can't deal with
> infinity. They
> > work
> > with discreet numbers. The solution is to sample the analog
> voice many
> > times per second. Sampling means to take a snapshot.
> >
> > The sample rate is how often the analog wave is sampled.
> Nyquist showed
> > that you have to sample at twice the rate of the highest
> frequency that
> > may
> > occur in the original data. Most humans don't output (and
> can't hear)
> > anything about 4 KHz. So sample 8,000 times per second (8Khz)
> and the
> > result will be good enough. When using a sample rate of 8,000
> KHz, if
> > each
> > sample is saved in an 8-bit byte, the resulting data rate is
> 64 Kbps.
> > That's one DS0. Compression allows us to use a smaller data
> rate, with
> > some
> > loss in fidelity.
> >
> > Priscilla
> > ________________________
> >
> > Priscilla Oppenheimer
> > http://www.priscilla.com
> [EMAIL PROTECTED]
> >
> >
>
>
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