Speaking of sample rates, I am playing with the idea of offering audio
training using MP3 files. I have prepared such a training on WAN
Troubleshooting. I'd love to get some feedback. This audio training will
help people studying for the Support test especially. It's 40 minutes. This
means
[mailto:[EMAIL PROTECTED]]
Sent: 01 March 2002 20:15
To: [EMAIL PROTECTED]
Subject: Re: Re: Sample Rate [7:36566]
Speaking of sample rates, I am playing with the idea of offering audio
training using MP3 files. I have prepared such a training on WAN
Troubleshooting. I'd love to get some feedback
I think this would be great. I would also think focusing on a certain
subject topics
would be valuable too. Reading does not always make sense and sometimes
hearing it from
someone else helps it make sense. Make sense. :-O
Ko
Priscilla Oppenheimer wrote:
Speaking of sample rates, I am
I know that I like this sort of product and I wish I'd had more of this
sort of thing available when I first started out in this field. For
certain types of learners, hearing someone discuss the topic allows it
to sink in better than reading alone. I'm definitely that type of
learner.
One
I just heard Priscilla's audio training on WAN Troubleshooting and I think
it's great. I spent the time listening to the audio file while organizing
my desk at work; very convenient!!. I'd definitely pay for audio training
if it was available.
Priscilla, do you have any other audio training
As you know with MP3, you won't really make much money because it's so
easily copy-able. You'd be on Napster, Morpheus, WinMX in no time. If you
don't mind the piracy, I think it would be a great idea!
--
RFC 1149 Compliant.
Audy Bautista wrote in message
news:[EMAIL PROTECTED];
I just
Two questions to prolongue the distraction:
1. If a 4KHz signal is sampled at 8Khz, it is sampled twice per cycle. Once
in the positive half and once in the negative. Doesn't this mean that the
value is dependant totally on the point at which it is sampled (and if you
sample it as it passes zero
The first question is pretty interesting. It seems that if you took a
1Hz signal and sampled it twice per second but you kept sampling the
points where the level was zero, it does at least seem possible.
However, that would only be with sine waves, not complex waves.
Still, it's an
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:
Well, if it comes from MANY years ago, before
John Neiberger wrote:
What I'm trying to find out is why the original 4KHz limit on
voice calls was put into place. It sounds like it was simply
an arbitrary decision. 4KHz is sufficient for a telephone call
and to provide clear calls that included higher frequencies
might have added some
Thanks, that's exactly the sort of thing I was hoping to find. It's
also interesting to note that the human ear is most sensitive to
midrange frequencies, say between 1000Hz and 4000Hz. So, even if you
filter out what's above 4KHz, you can make up for the lack of clarity
with a little
: Wednesday, February 27, 2002 9:58 AM
To: [EMAIL PROTECTED]
Subject: Re: Re: Sample Rate [7:36566]
John Neiberger wrote:
What I'm trying to find out is why the original 4KHz limit on
voice calls was put into place. It sounds like it was simply
an arbitrary decision. 4KHz is sufficient
You are right, John. With digital telephony, the analog speech signal is
filtered before sampling. High and low frequency components are removed. I
think it was just a tradeoff. We don't expect the human voice to sound that
great over the phone anyway, and by filtering we can reduce bandwidth
Female opera singers probably hate it when people ask them to sing over the
phone!?
OK, have we distracted you enough, John? ;-) Seriously, I think this was a
great discussion. Thanks to everyone who contributed.
Priscilla
At 10:58 AM 2/27/02, David L. Blair wrote:
John Neiberger wrote:
Usually thought about in analog to digital conversion, it is the number of
times per second a piece of data is captured. Accepted theory states that
for the sample to be an accurate representation of the data, the sample rate
should be twice the highest frequency you are trying to capture, so
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
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.)
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
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
It's the average frequency of human voice. if you look at a conversation on
a osciliscope, it averages out to 4k, so you double that and get the 8k
sample rate.
--
RFC 1149 Compliant.
John Neiberger wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
I understand all that, but
64 kbps comes about from sampling 8 bits at 8khz, 8x8000=64000
Bri
On Tue, 26 Feb 2002, John Neiberger wrote:
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
Exactly, that's what Priscilla and I both just said. :-)
What I'm trying to find out is why the original 4KHz limit on
voice calls was put into place. It sounds like it was simply
an arbitrary decision. 4KHz is sufficient for a telephone call
and to provide clear calls that included
The reason is for economics. The 300hz filter to 3300hz filter would carry
the voice fairly well so this is the least amount of bandwidth to do a
decent job of it. Not too much is lost because the voice is between
100 to 5000 hz. The hearing range is not considered as far as I know...
Message
Letterman
Cisco Systems
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of
John Neiberger
Sent: Tuesday, February 26, 2002 9:23 PM
To: [EMAIL PROTECTED]
Subject: Re: Re: Sample Rate [7:36566]
Exactly, that's what Priscilla and I both just said
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
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