Well, according to a voice manual I have, "Lower bandwidths need less
expensive equipment, which affects the sound quality... this was
determined to be the best cost-performance trade-off".
So I think it was reasonably arbitrary. My ancient but still useful
Tanenbaum (Computer Networks) doesn't seem to give any reason, which sort
of supports that. Horak's "Communications Systems and Networks" agrees
that it "is considered sufficient for voice communications and certainly
is more cost-effective than if full-fidelity voice were to be supported."
And by the way, most references I have refer to the range as being 200 Hz
to 3500 Hz, not 4000Hz - for noise I think - Nyquist assumes a noise-less
channel.
So... you're trying to study, I'm about to give up trying to work today
;-)
JMcL
----- Forwarded by Jenny Mcleod/NSO/CSDA on 27/02/2002 05:15 pm -----
"John Neiberger"
Sent by: [EMAIL PROTECTED]
27/02/2002 03:17 pm
Please respond to "John Neiberger"
To: [EMAIL PROTECTED]
cc:
Subject: Re: Re: Sample Rate [7:36566]--long reply [7:36566]
I understand all that, but what I don't remember is why there
is a 4KHz low-pass filter on voice lines. I know I've read the
reason before but I just can't recall what it was. Was it
simply arbitrary? A 4KHz upper limit is obviously sufficient
for voice quality. Did someone just pick that limit and filter
out everything above it, possibly to filter noise or something?
Hmm...this is bugging me now. :-)
But I can't be distracted right now, I'm trying to
study...which explains why I keep taking time out to check my
email and search the internet for MP3s of Clannad. :-) I did
just find a killer sampler of Celtic stuff. Very relaxing...
John
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---- On Tue, 26 Feb 2002, Annlee Hines ([EMAIL PROTECTED])
wrote:
> 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]
> > >
> > >
> >
> >
> > ________________________________________________
> > Get your own "800" number
> > Voicemail, fax, email, and a lot more
> > http://www.ureach.com/reg/tag
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