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


________________________________________________
Get your own "800" number
Voicemail, fax, email, and a lot more
http://www.ureach.com/reg/tag


---- 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
[EMAIL PROTECTED]




Message Posted at:
http://www.groupstudy.com/form/read.php?f=7&i=36590&t=36566
--------------------------------------------------
FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html
Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]

Reply via email to