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Re: sample rate

Adrian Bartholomew
8439 Lee Blvd
Leawood, KS 66206
(913) 660-6918

On Dec 20, 2005, at 1:12 PM, Brian Cass wrote:

oh man. i tried so hard not to get sucked into this debate. but here i am.

well u did:-)

the connect the dot explanation does help to explain how it's done, but the reason that does not quite work is just that the Max and Min points of the source are very unlikely to be the 2 points sampled. You have the same chance of sampling at the zero crossings. most likely the 2 samples per cycle are going to hit during it's ascending and descending, and get no information about max or min. 

completely correct. ur actually on my side. what i described was an ideal situation. so ur comment just proves MORE how "all the information of the original signal" was NOT true.

Can a 44.1k sampling rate accurately PRODUCE a 20k wave? 
Sure. The details of that wave are mostly lost, but it's frequency and amplitude are approximated pretty well.

i would say that for practical purposes, the details are lost.

Can a 44.1k sampling rate accurately RECORD a 20k wave? 
Not as easily. The waveshape is sure to be lost and the amplitude is subject to mis-representation.

i agree

Do kids today care? Do they know what they are missing?
Probably not. I am 28 and had a record player growing up. I love the sound of 2 inch tape.

subliminally i believe that they do. they wont know WHY they prefer one recording to another, all other things being equal, but they can pick out a professional recording over an ameteur one. its like Film and Video. video is actually cleaner and more true to form BUT, film just looks like cinema. when confronted with an A/B test most would say they prefer the good quality of digital broadcast video over the 35mm film. so why do the "behind the scenes" shots never look, to these SAME people, like the real film edits of the same scene?
i did this experiment myself.
its so subjective it isnt funny.

u take a real good, say, quincy jones production of say Michael J and transfer it to cassette. take ur best home recording to CD 44.1. the cassette will STILL sound better. even after a couple generations degration. most will still recognize the intrinsic quality of the QJ production.
well its back to the subliminal.
a recording originally done at 192k/24bit is like using a tube neumann with neve preamps in a good isolation booth. its the source of the recording that lasts through the subsequent mediums. 

even if u take a 70mm film project and bump it down to a cheap video format, itll still look better than one recorded originally on a professional video format bumped UP to film.

so to cut this short, the better the source, the better the final bump down to CD 44.1k

I love the new DSD technology. I hate mp3s. I am (we are) the minority on audio quality preference. DSD has the potential to take over the recording industry in the next 5 years. If so PCM will be like our old friend the VHS. Unless handheld mini disc recorders take over first, then we are all doomed.

Did I get this question/answer format from watching seinfeld last night?
Yes. Yes I did.


- b

On Dec 20, 2005, at 1:52 PM, Adrian Bartholomew wrote:

this is where SOME info is worse than NO info.
dude think about it.
u have a wave at 1/2 the sample frequency. think about it like connect-the-dots.
the only ones u can plot are the max(positive) and min(negative) points of the wave. NOW connect the dots and what do u have. thats right a sawtooth wave. even if the original was a sine.
but at least u have the frequency. forget about phase what about shape or tone?
even if u sampled at a frequency high enough to give u three or even 4 points to connect, its STILL approximate, very far from the shape of the original and certainly not "all the information of the original signal".
Adrian Bartholomew
8439 Lee Blvd
Leawood, KS 66206
(913) 660-6918

On Dec 20, 2005, at 1:53 AM, Bill Fox wrote:

a k butler wrote:

Bell Labs researcher Harry Nyquist develops Sampling Theory. It states provides that if a signal is sampled at twice its nominal highest frequency, the samples will contain all of the information in the original signal.

Which is clearly not true :-)
There's no way to keep the phase information for a signal sampled
at only twice it's frequency.
Only the amplitude.
I guess that the Nyquist Theorum is misquoted somewhat here
(and generally).

Although you might be correct for a frequency of f when the sampling frequency is 2f, the theorem correctly stated says that it will be good for frquencies UP TO f Hz, i.e. not including f.  So while you're correct for one frequency, f, the theorem holds 100% true for all frequencies below f and no information is lost.  The mathematics bear out.  For shorthand, the bandwidth of a system is stated as f Hz, not (f - 1) Hz.

BTW, I dare anyone to tell me they can HEAR that 20kHz has a wrong phase relationship in a system sampled at 40kHz.  Plus, in the real world, where there are no ideal filters, a guard band is built in.  That's why an audio system that is designed to have a 20kHz bandwidth uses a sampling frequency of 44.1kHz.  This also avoids the problem of 20kHz not having a proper phase relationship since it is less than half the sampling frequency, not exaclty half the sampling frequency.