The Black Hole......

Here's how Tektronix specifies their 10X sample rate, relative to the usable bandwidth (pic shows 500MHz, 5GHz sample rate).
 

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Here's how Tektronix specifies their 10X sample rate, relative to the usable bandwidth (pic shows 500MHz, 5GHz sample rate).

We are not building single-shot data acquisition systems. I don't know how many times this needs to be said.

You also cherry picked one example, btw. Your rule of thumb does not even hold among any oscilloscope mfg's product line. In fact, from what I can tell, only one or two models satisfy your "10X rule". Here are some noteworthy production scopes that don't satisfy your rule.

Tektronix MSO64: 6 GHz, 25 GSPS
Keysight X4000 series: 1.5 GHz, 5 GSPS
Keysight Infiniium S series: 8 GHz, 20 GSPS
LeCroy WavePro HD: 8 GHz, 20 GSPS
 
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Everything that fits under the LPF can (ideally) be reproduced perfectly. It only takes two samples per cycle, and more is not better.

Regarding choice of cutoff frequency I have no comment. Very high mileage ears, so I can't hear the proposed differences.

All good fortune,
Chris
Agree on the high mileage ears.
Two samples per cycle is incorrect. As an example, that could sample every zero crossing. Need more..

Jn
 
We are not building single-shot data acquisition systems. I don't know how many times this needs to be said.

Well, maybe you should be. Its a better fit.

There is no cost nor technical reason any more to band limit to an min. BW and use min CW Nyq sample rate.

Higher sampling rates sound better. IMO But then, i have always owned very low distortion loudspeakers, too.

THx-RNMarsh
 
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10 would undo that little zero crossing coincidence problem.

And I concur with the ears, one singing with tinnitus. Signed up for a free hearing test, so someone can try to sell me 'aids. I'm only interested in how much wax is in there and what the FR of each is these days... Havent had that "professionally measured" in I dont know how long. Could probably DIY using my FG and a set of 'phones.
 
Agree on the high mileage ears.
Two samples per cycle is incorrect. As an example, that could sample every zero crossing. Need more..
Okay, but wouldn't that also happen at every integer sub-multiple of Fs, creating a big nasty comb-filter response? Why don't we see that?

(Please forgive me if this is an insufferably dumb question. I am trying to keep up here, but the cold medications aren't helping.)
 
And I concur with the ears, one singing with tinnitus. Signed up for a free hearing test, so someone can try to sell me 'aids. I'm only interested in how much wax is in there and what the FR of each is these days... Havent had that "professionally measured" in I dont know how long. Could probably DIY using my FG and a set of 'phones.

Don't do it unless you really want to know; it can be disheartening. For me, motorcycles (wind noise, not noisy exhaust) and firearms (the military doesn't use hearing protection), plus: I'm a geezer.

All good fortune,
Chris
 
From what I can see, Richard is saying that signal processing that assumes steady state is insufficient for non continuous signals.

That is exactly what I said. Nothing more. Nothing less.

That's all I need to say besides higher sampling rates sound more accurate.

The rest is going back to recycling old info. again and again.

So, I will leave it at that.


Richard
 
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Maybe think about the issue with this model: we're observing the musical signal to be reproduced, using an old-time spectrum analyzer made with a very narrow tunable filter followed by a voltmeter.

We have all the time needed to wait for readings and to repeat the signal as needed. We observe single "pure" tones to have very narrow spiky responses, and "modulated" tones to have broader responses. The curve humps don't end abruptly, but continue down into the noise floor.

So, we can say that even very low musical tones have frequency components above audibility, they're just low amplitude. All non-continuous signals have an infinitely wide spectrum.

Because we can (ideally) perfectly reproduce everything below Fs/2, the only question worth discussion is about the location of Fs. Scott W has given the correct method; it's up to listening tests with his method to decide.

All good fortune,
Chris
 
I gave a method also. but, it is based upon Real-Time sampling. Which is also valid but a better choice IMO. It yields a higher sampling rate. And, the higher sampling rates do sound better.

Enjoy.

-Richard
 
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Please forgive me, I try to follow as much as possible (at least so as not to be rude) but I missed that somehow. This current thread? As long as it can make Fs/2 the only variable, it'll do the job. I can't imagine a method other than Scott W's that could work, so I'm eager to see it.

Much thanks, as always,
Chris

ps: Congratulations on still being able to hear this stuff. You're living a charmed life!
 
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I gave a method also. but, it is based upon Real-Time sampling. Which is also valid but a better choice IMO. It yields a higher sampling rate. And, the higher sampling rates do sound better.
The sample rate you suggest may be advantageous but it seems the reasoning isn't quite right, shouldn't it be based upon the rate required to render temporal information as accurately as necessary?
 
Yes. The CD's BW is not the main issue. 40KHz would have been better. And it is not what makes the sound less accurate as the freq is increased. Nor is it the filter... though a small affect may be detectable if not well designed. It is the sampling rate is far too low.

It is almost a smoke and mirrors trick to use Continuous Wave waveforms to help the sampling of 2X only. Since music has no CW to help construct the waveform with a simple 2X sample rate, it is the wrong way to choose sampling rate. As suggested, it may have been required rate for other compatibility reasons. You have to use Real-Time sampling which means a much higher sampling rate to capture the complex music waveform accurately.

That has been corrected in HD/Hi-Rez recordings.


THx-RNMarsh

You have to excuse me but this is what I think is referred to as Fourier denial. The CD will reproduce any sharp bend in the signal as long as that bend is not sharper than the corresponding HF limit.

It's really that simple.

Your derivation of how to calculate what you believe is a proper Fs, from , I presume, a hearing perspective seem "home-brew" and cant be taken serious. It's just your "feeling" convolved into tecno-speak? You have not put forth any hearing threshold limits on relevant parameters so how can you specify system requirements.

//
 
For continuity: Scott W. proposed that a musical signal proven to contain content above RedBook limits, in some unimpeachably high sample rate, be LPF'd to 22.05KHz but kept at its original sample rate. The original and the LPF'd could then be compared as apples-to-apples as possible.

It's hard to design an experiment as elegant as this one.

All good fortune,
Chris
 
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