I can't SEE the difference?!

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The analogies are always poor, I know, I was just quoting from the other thread for the possible benefit or musings of others :)

I think where most people stumble up is when they can't understand how everything below the nyquist frequency can be faithfully reproduced (assuming everything above the nyquist frequency was filtered so no aliasing has occurred), and have a notion that having more samples (which they visualize as 'data points' on the waveforms) will mean a more accurate reproduction - the word 'smoother' also pops up from time to time. I think part of this problem and the way people visualise digital sampling as being 'jagged' is because you only really get to see the blocky dot-to-dot "waveforms" of most editors and not the smooth interpolations and reconstructions that actually occur during digital-analogue conversion. Intersample peaks are usually a good starting point for explaining this. I recall seeing some good images accompanied by some basic but insightful explanations of how the signal is reconstructed on a website somewhere and if I can find it then I'll post a link.
Hey mattr, yeah, I know you were just referencing someone else. I didn't mean to make it sound like I was correcting *you*. My apologies if it came out that way :o.

I think a big part of the misconception is the "connect-the-dots" or "stairstep" illustrations that most lay books use to illustrate the concept of sample rate. With such pictures it only makes sense that the more "dots" or "stairs" you have, the "smoother' the reproduction will be.

The problem is that at real sample rates, there's nowhere near as many "dots" as they imply in those diagrams. For example, a 20kHz sine wave samples at 44.1k will only have slightly more than two samples or dots per wave. there's no way to connect those dots based upon those dots alone to come even close to guarantying accurately redrawing that sine (the same would be true for a triangle, sawtooth or square wave, BTW). Even at 1/4th the signal frequency, at 5k - which I don't think anyone seriously doubts that 44.1k can handle just fine without breaking a sweat - we're only talking 8 or 9 samples per wave. A far cry what what they usually show in those illustrations, and still nowhere near enough to draw a very smooth or accurate wave by connecting the dots or building a staircase. That''s because that's not how digital reconstruction via Nyquist-Shannon actually works.

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Also, just a bit more about Harvey's square wave example - which was a good point to bring up. The thing to remember is that a square wave (also true to a degree of sawtooth and triangle waves) is so loaded with harmonic components in the form of higher-frequency sine waves that one cannot consider their fundamental frequency alone when looking at the issue of bandwidth limiting as required by Nyquist-Shannon. When you get a *true* square wave of high enough frequency, like Harvey illustrated, it contains components that are going to exist beyond the Nyquist frequency. For this reason, Nyquist-Shannon cannot apply to such waves with 100% accuracy *unless* the sampling frequency is raised to infinity, which is impractical.

Harvey did bring up a borderline example popular amongst many proponents of higher sampling rates of a square of sufficiently high frequency where one could and should see a significant difference between the square sampled at 44.1 and one sampled at, say 88.2 or 96. This is true, because of the complex and extended harmonic structure of the square wave takes it well beyond the fundamental frequency. This can on first blush appear to be a hole in Nyquist-Shannon and a real reason for extended sample rates.

It's not a hole in N-S theory, because of the theory's stipulation for bandwidth limiting. No rules are broken, no holes punched. Nyquist does not claim to be able to reproduce a square wave with a 7.5k fundamental exactly, because the fundamental is not a super quorum portion of the actual square wave's frequency content.

And as far a it being a "real reason" to justify higher sample rates, I reiterate that the likelihood of finding anything close to true square wave (or any other such significantly harmonically-complex waveform) at such high frequencies of any significant amplitude - even if running a synth direct* - is not only astronomically small, but does not address the OP question here, nor does it come close to explaining the much larger audible differences that proponents can hear between sampling rates.

*I grew up on analog synths; mostly Arp, but also Moog, PAIA, Oberheim Sequential Circuits and others, and regularly ran their outputs through an oscilloscope. I never did find one that could generate a square wave that any reasonable carpenter or EE could consider anything close to square. And as far as the VSTis, I've only looked at a couple, but they as often as not get what the original synth actually did wrong (the Arturia sawtooth for the ARP 2600 is a joke, for example), and what they do get right is usually no more 'pure" in shape than the original.

Now there many be some scientific tone generators and test gear that can do a better job, but I have yet to see one of those used as a high-frequency musical instrument.


G.
 
*I grew up on analog synths; mostly Arp, but also Moog, PAIA, Oberheim Sequential Circuits and others, and regularly ran their outputs through an oscilloscope. I never did find one that could generate a square wave that any reasonable carpenter or EE could consider anything close to square. And as far as the VSTis, I've only looked at a couple, but they as often as not get what the original synth actually did wrong (the Arturia sawtooth for the ARP 2600 is a joke, for example), and what they do get right is usually no more 'pure" in shape than the original.

Now there many be some scientific tone generators and test gear that can do a better job, but I have yet to see one of those used as a high-frequency musical instrument.


G.
Unless you're using modular analog synths, the output of most other subtractive synths (MiniMoog, Prophet 1, 5, etc) always pass through the filter stage, so you wouldn't necessarily see the direct output of the oscillators themselves, which in turn will color and mangle their waveform.

Another thing to consider, and to which you allude to above, is the fact that in musical synthesizers the oscillators are made to be reasonably accurate in the audio/playable range, which in most cases you're limited to what you have on a 61 key or less keyboard, which in most cases means, even if you use octave shifts (4'), your fundamental frequency is not going to push much higher than 4,186Hz (C8). And even then, not too many people play stuff that high up the keyboard, unless they're trying to annoy Glen :p

So, yeah, while there are test analog oscillators capable of generating oscillations at ultrasonic frequencies, they have pretty much zero bearing on music, and one might argue that due to their clinical and accurate nature, they don't even sound as good as the slightly skewed and non-accurate analog oscillators intended for musical synthesizers.
 
OK, I'm probably a victim of my own vocabulary; I should have said that the minimum sample rate *limit* is 2x. in order to be technically correct. The problem is that language almost always leads into a misunderstanding by those new to the idea that that means that the further above that rate one goes, the better, thus justifying the ultra-high sample rates.

I don't know for sure how close one can push that ">" towards ">=" - I believe it's pretty darn close to infinitely close - but it's an effect that is most certainly swamped by the requirements imposed by the filtering constraint that brings the sample rate way up to 44.1k. Which leads us to:I was just continuing the explanation of what I had said before that. The point is that A/D conversion in the real world does include a low pass filter stage on purpose in order to ensure bandwidth-limiting of the signal. Higher frequency noise and harmonics and transients (oh my!) and such can otherwise sneak through. Because of gear limitations and musicality and such it won't be a large amount, but it's out there, we just can't hear it. The problem is, if you let that stuff through to the converter, it gets aliased back into the (poetntially) audible range below the Nyquist target. Like George said, the half-sample rate becomes kind of a mirror that reflects higher frequencies back into lower frequency ranges.

This is why we want to - and do - low pass at ~20k and filter out above that. But because we can't build an "ideal" brick wall filter right at 20k, we have to let that filter slope past 20k a bit. The idea is that we should be able to build such a filter that reaches full attenuation by about 22k or so. Allowing for that and a little extra slack, we wind up with a sample rate of 44.1k to cover the full 22k of fully limited bandwidth.

G.

Yeah, I agree with all of that assessment. I think we're pretty much on the same page now.
 
Yeah, at my age, I have a hard time hearing a 5K simple sine wave, let alone the overtones of a 7,500 Hz square wave, but it is a theoretical problem - in theory anyway, if not really in practice. But the idea of it still bugs me just a little.
 
Yeah, at my age, I have a hard time hearing a 5K simple sine wave, let alone the overtones of a 7,500 Hz square wave, but it is a theoretical problem - in theory anyway, if not really in practice. But the idea of it still bugs me just a little.
Agreed. I just ticked over the half-century mark myself, and doubt I can hear much of anything above about 16k these days, to be honest.

And I completely understand your (and others) apprehension about that potential for missing overtones. But I tend to look at it this way; there are so many mitigating reality factors of greater magnitude, that to me worrying about sample rate (other than picking the speed my piece of gear likes best) is like worrying about whether my windows are cracked open or not when it's about to get hit by a tornado.

Between the limited response characteristics of the human ear - (even the good ones ;), the limited response characteristics of the recoding chain, the natural geometric decrease in energy as one climbs the frequency ladder, the natural lack of high-frequency complex waves like squares, sawtooths and such, and the other distortions natural to the music production process, sample rate seems so far down the list of factors as to be something I personally don't lose any sleep over.

If I worked for Telarc and recorded a lot of high-dynamic acoustic pieces, perhaps I might err on the side of caution, but working with mostly blues/rock./jazz stuff, it seems kind of pointless. And even if I were doing Sejji Ozawa conducting Yo Yo Ma, I'd probably have Top Shelf house clocking and conversion anyway, and not my li'l ol' MOTU 2408 ;).

G.
 
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And I completely understand your (and others) apprehension about that potential for missing overtones. But I tend to look at it this way; there are so many mitigating reality factors of greater magnitude, that to me worrying about sample rate (other than picking the speed my piece of gear likes best) is like worrying about whether my windows are cracked open or not when it's about to get hit by a tornado.

I agree with you...it can be silly focusing too much on one variable when you are faced with a plethora of variables in a typical recording environment!
Though still, my thought has always been to always try and squeeze as much as possible out of everything even if at times it seems like the effort is not worth it.
I guess it's the notion that the sum of a lot of small, almost insignificant choices might add up to give you maybe an extra 5%-10% improvement for the end product.
Sometimes...yeah, the decisions are almost made on faith that they will contribute positively even if it's hard to isolate that one variable and prove if it’s worth it or not.

I think much of that comes from the somewhat anal nature of recording, especially for people who've spent a lot of time with all the processes and minutia over the years, and it's never about looking for less work and shortcuts, as you become somewhat of a slave to your goal for the highest possible perfection as you see it.
Yeah...at times it does seem like a LOT of work for some “shadow of improvement”...but if you don't do it, it nags at you!!! :D
There is certainly an OCD quality to many recording processes. ;)
 
Glen and I don't disagree....occasionally we just don't agree. ;)
 
I haven't read through the rest of the post yet, so if there is any repetition, I apologize.

What I know: sample-rate means how often a sample is taken; because samples cannot be taken infinitely quickly, it seems to me that the more often a sample is taken, the more 'accurate' the representative waveform.
It depends on how you define 'accurate'. What the increased sample rate does for you is allow you to record higher frequencies. To the extent that what you are recording has frequency content above 20k, it will be present in the 96k sampled version and not the 44.1k version. If your audio doesn't have any content up there, there is nothing for the added samples to capture.

And the higher the bit-depth, the greater the dynamic range (and consequently, signal-to-noise ratio). I understand that each bit affords ~6dB in dynamic range.
Yes, but it adds dynamic range at the bottom. It pushes the theoretical noise floor down. If you wanted to see any changes, you would have to look close to the zero crossing point below -96db FS.
 
With those kinds of lines, you could run for a political office and get elected... I'll vote for you. :p :D
Awww, George, you're missing out on a deal. I'm from Chicago, you could vote for me twice! And so could your dead aunt! Your cat can only vote once though.... :D

BTW, everybody, George is so old, he started recording with a chisel on stone. He couldn't record too hot because they hadn't invented fire yet. He's so old, that on the seventh day, God downloaded George's MP3s. His first client wasn't T Rex, it was a *real* T Rex.

[Me] He's so old...

[Everybody at once] How old is he?

[Me] He's so old his first percussion track was the Big Bang!

I could keep going but it's 1am here, I just came home from a live gig, I'm tired and I have nasty headache. :( But I'm here all week, matinee on Sundays...


:D


G.
 
Awww, George, you're missing out on a deal. I'm from Chicago, you could vote for me twice! And so could your dead aunt! Your cat can only vote once though.... :D

BTW, everybody, George is so old, he started recording with a chisel on stone. He couldn't record too hot because they hadn't invented fire yet. He's so old, that on the seventh day, God downloaded George's MP3s. His first client wasn't T Rex, it was a *real* T Rex.

[Me] He's so old...

[Everybody at once] How old is he?

[Me] He's so old his first percussion track was the Big Bang!

I could keep going but it's 1am here, I just came home from a live gig, I'm tired and I have nasty headache. :( But I'm here all week, matinee on Sundays...


:D


G.
LMAO! Somebody help me! I... can't... stop... laughing! :D :D :D

My belly is actually aching. Showed my wife, she almost pissed her pants.

I guess the gig did go well :D
 
Glen and I don't disagree....occasionally we just don't agree. ;)
I disagree!

;) :D :D

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Hey, Moses...I mean George...take it easy on the laughing conniption; your ancient heart can't take it :).

G.
 
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