Digital Sampling

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VirtualLarry

No Lifer
Aug 25, 2001
56,452
10,120
126
Originally posted by: Bobthelost
No, not true, if you're sampling at 2x the frequency you might be able to reconstruct the signal precisely. If you're sampling a sine wave at the peaks and troughs then you can reproduce it exactly. But say you were to drop the frequency down by a third, that brings the ratio to 3x (and then play with the phase) so we're within the nyquist's theorem, you really want to see what godawful monstrosities i can make a wave into sampling around three times per cycle?

How many people know that, how many care? How many really think that sampling at 3x the signal's highest freq component means it's possible to reconstruct the signal properly? It doesn't work like that.

That is what i hate about the nyquist theorem and the way it's taught.

That's interesting. I understand Nyquist, sampling, etc.,.. and although I'm not completely up on the math behind Fourier decomposition, I think I grasp it at an abstract level. But you bring up an interesting question - what about the relative phase relationships between all of these superimposed (as per Fourier) sinusoidal waveforms, that ostensibly form the original signal? Is that not actually a *third* dimension to the signal, one that isn't being recorded by the sampling process? IOW, at any particular given point in time in the signal, the phase information is lost - shouldn't the reconstructed signal use some sort of error-diffusion/dithering technique, to rebuild the signal such that the overall phase-error of all of the waveforms approaches zero.. although that still would give a waveform that only approximates the original in phase-space, and doesn't re-create it.

It makes one wonder how much of a difference phasing matters to the human ear, or whether it only primarily reacts to amplitude and frequency.

Actually, hmm, there seems to be some sort of analogous similarity between sampling a waveform's frequency and phase information, which really can only exist relative to another reference waveform, and Heisenburg's uncertainty principle with "sampling" the position/velocity of particles.. or something along those lines. If they are really only just waveforms anyways, and our conscious observation causes them to be "sampled" into particles.. does that provide just one more piece of evidence that our conscious existance is actually digital? Would that imply that God, as an infinite being, is analog?
 

Navid

Diamond Member
Jul 26, 2004
5,053
0
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Fourier transform takes phase into account. It is a series of sine and cosine components. A sine wave is 90 degrees out of phase with respect to a cosine wave. So, the phase is not lost!

http://www.ph.tn.tudelft.nl/Courses/FIP/noframes/fip-Properti-2.html

Perfect reconstruction is theoretically possible. The only reason for degradation of signal is noise or if you do not satisfy the Nyquist criterion or if you do not consider anti-aliasing requirements.
 

Spinne

Member
Sep 24, 2003
57
0
0
You can do it if you like the Axiom of Choice.

Originally posted by: BigT383
I'm having an argument with this guy at work...

This is one of those "never possible in reality" things...

Given an analog waveform, and an infinite number of sampling points, can't you choose your sampling points so that there are no gaps between them- IE, produce a sampled function that, given any real number, returns sampled data without any interpolation?

My conjecture is that you can, and at that point your perfectly sampled function is the same as the original waveform. He believes that there would still be gaps between your sample points. I said that yes, you could have an infinite number of sample points and choose to still have gaps (only sample for rational numbers, for instance), but you didn't need to.

This all started from an argument we were having about whether it was possible for digital representations to ever be as good as analog. He said it was not possible, I said it was.

Correct me if I'm wrong about this: but in the real world there comes a point when, no matter how good your detector of the original waveform is there is still a margin of error (uncertainty principle), and that as soon as the distance between every two adjacent sample points becomes less than that margin of error, the digital representation becomes just as good as the analog.

 

ChaiBabbaChai

Golden Member
Dec 16, 2005
1,090
0
0
How did this turn into a discussion on Nyquists ideas? All it says is you need a sampling frequency of AT LEAST twice the audible frquency desired because if you want to capture a cycle of those tones, then it needs to be captured each time it has any power on EITHER (both) sides of the Zero point, and that is only possible if it snaps a sample faster than the wave changes phase. Right? Well, anyways, Let's think about the original question!

You can't tell me, without looking like a complete fool, that you can't hear the difference between a record and a CD IF you are listening to both through a digital amplifier, because both would be digitized. How many of you actually listen to music through purely analog system? How many of you arguing that you can't tell the difference actually OWN a purely analog system? If you don't then you are one blatant fool.

Without analog you would have no digital. Digital music is to Analog what Flourescent light is to the sunlight. It's like shredding up a fifty dollar bill and burning 2 of the shreds, then taping it back together.

It doesn't even matter about the highest frequency because A) most people can't hear as well as dogs B) It's not about high frequencies, and Analog recordings have high frequency limitations.
 

sdifox

No Lifer
Sep 30, 2005
96,210
15,787
126
Originally posted by: ChaiBabbaChai
How did this turn into a discussion on Nyquists ideas? All it says is you need a sampling frequency of AT LEAST twice the audible frquency desired because if you want to capture a cycle of those tones, then it needs to be captured each time it has any power on EITHER (both) sides of the Zero point, and that is only possible if it snaps a sample faster than the wave changes phase. Right? Well, anyways, Let's think about the original question!

You can't tell me, without looking like a complete fool, that you can't hear the difference between a record and a CD IF you are listening to both through a digital amplifier, because both would be digitized. How many of you actually listen to music through purely analog system? How many of you arguing that you can't tell the difference actually OWN a purely analog system? If you don't then you are one blatant fool.

Without analog you would have no digital. Digital music is to Analog what Flourescent light is to the sunlight. It's like shredding up a fifty dollar bill and burning 2 of the shreds, then taping it back together.

It doesn't even matter about the highest frequency because A) most people can't hear as well as dogs B) It's not about high frequencies, and Analog recordings have high frequency limitations.



I seriously doubt many people here use digital amp. Most are still on analogue, sure, it's on chip, but that just makes it more reliable/predictable than tube. It is still analogue, otherwise they wouldn't need a DAC.
 

ChaiBabbaChai

Golden Member
Dec 16, 2005
1,090
0
0
You just knit-picked one stupid typo. You get what I'm saying. The converters being in the chain AFTER the audio is the corruption of the experiment. Both being equal, you would hear the difference between analog and digital recordings, if you had decent headphones (at least $80 phones)
 

CSMR

Golden Member
Apr 24, 2004
1,376
2
81
The answers to the OP are really quite appaling.
Originally posted by: BigT383
I'm having an argument with this guy at work...

This is one of those "never possible in reality" things...

Given an analog waveform, and an infinite number of sampling points, can't you choose your sampling points so that there are no gaps between them- IE, produce a sampled function that, given any real number, returns sampled data without any interpolation?
Pedantic answer: take all the points
More sensible answer: yes, you can do it even with a countable infinity of sampling points, assuming the analog waveform is continuous
My conjecture is that you can, and at that point your perfectly sampled function is the same as the original waveform. He believes that there would still be gaps between your sample points. I said that yes, you could have an infinite number of sample points and choose to still have gaps (only sample for rational numbers, for instance), but you didn't need to.
You are right, except sampling rational numbers is an example of not having gaps.
This all started from an argument we were having about whether it was possible for digital representations to ever be as good as analog. He said it was not possible, I said it was.
Digital representations (in normal usage) contain a finite amount of information not an infinite amount. A countable infinite digital representation contains the same amount of information as a continuous "analog" function.
Correct me if I'm wrong about this: but in the real world there comes a point when, no matter how good your detector of the original waveform is there is still a margin of error (uncertainty principle), and that as soon as the distance between every two adjacent sample points becomes less than that margin of error, the digital representation becomes just as good as the analog.
You should be able to approach the error with more and more accurate (finite) digital representations. Subjectively you may get something just as good eventually. Mathematically probably not depending on how you define things.
 

sdifox

No Lifer
Sep 30, 2005
96,210
15,787
126
Originally posted by: ChaiBabbaChai
You just knit-picked one stupid typo. You get what I'm saying. The converters being in the chain AFTER the audio is the corruption of the experiment. Both being equal, you would hear the difference between analog and digital recordings, if you had decent headphones (at least $80 phones)

The discussion is about whether it is possible to reconstruct the original wave form through sampling. My answer was yes, sampling is not the issue, A to D process is the issue because of aliasing. That was all I said.

A perfect analogue system would be superior to digital, alas, our technology is not sufficiently advanced for that so we are stuck with digital facsimile of the original signal. Have you tried DVD-A/SACD? They are pretty good.
 

kobymu

Senior member
Mar 21, 2005
576
0
0
From a pure mathematical prospective, NO!
Infinity != infinity for infinity (by definition) is not an obsolete value (X=X, while , and only while "X" is defined as an absolute value).
Infinity, again from a pure mathematical prospective, "breaks" the classical mathematical operands ("=", "+", "-", "*", "/", "^", root, etc.)
Example:
5 + Y != 5 ; for any "Y" different from zero
infinity + Y = infinity for ; every Y.
4 * M != 4 ; for any "M" different from 1
infinity * M = infinity ; for every M (excluding 0)
0 * A = 0 ; always (for every A)
infinity * 0 =...? ; is not defined !!!

so... NO!



 
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