John Curl's Blowtorch preamplifier part III

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How many times have we heard this as Thee reason 44.1Khz sampling is fine:

"Just like Nyquist tells that the sample frequency Fs should be at least two times the highest signal frequency for correct reconstruction of the signal,"

Dont you know that only applies to a repetitive waveform? You cant construct a waveform from just 2 data points in one waveform cycle.

Anyone ever use a "digital" scope? Try to capture a single transient event with 2 samples. You need at least 5-10 data samples to even get a clue what the waveform shape and timing is. More data points and more detail is shown.

Nyquist works in both directions... more sampling needed for single cycle and fewer sampling points for many repeating cycles (2 being the minimum) With music it is of transient and never repeating waveform SO .... This is why (IMO) it sounds worse as the music content has higher frequencies... like cymbals. Not enough data points or too low sampling rate.

So, I answered the question of which is responsible for the improved sound... bits or rate. its mostly sampling rate.

Ok. sharpen your pencils... I am bracing myself for another wiz-bang EE101 thing. Remember though... keep in mind the central point -- Music is not repetitive waveforms ... and the phrase AT LEAST 2 points are needed really means with music waveforms you will need more than 2. More than 44.1Khz.

Someone told me at LLNL that if you run the sampling rate up high enough you will have the original analog signal. Seems he was right. But his problem with digitizing ultra fast transient signals was the sampling rate always had to be much higher than the signal being measured and they could barely keep up using wideband fast rise time analog techniques.

TEKtronix, when they first came out with digital scope techniques tried to use 2 times to make thier BW spec look better. They were called on it by people who really needed accurate waveform capture. Now they use 5 times. My TEK scope claims 300Mhz BW using 2 GHz sampling rate.

If you use the 5 times number, 44Khz barely gets you out of the midrange before falling apart. And that is exactly how it sounds, too.




THx-RNMarsh
 
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"Just like Nyquist tells that the sample frequency Fs should be at least two times the highest signal frequency for correct reconstruction of the signal,"

Dont you know that only applies to a repetitive waveform? You cant construct a waveform from just 2 data points in one waveform cycle.

Anyone ever use a "digital" scope? Try to capture a single transient event with 2 samples. You need at least 5-10 data samples to even get a clue what the waveform shape and timing is. More data points and more detail is shown.

Nyquist works in both directions... more sampling needed for single cycle and fewer sampling points for many repeating cycles (2 being the minimum) With music it is of transient and never repeating waveform SO .... This is why (IMO) it sounds worse as the music content has higher frequencies... like cymbals. Not enough data points or too low sampling rate.

As usual, displaying 0 (zero) knowledge of Nyquist, sampling, discrete Fourier transform, and other basics in signal processing (waaaaaay before you could get into dithering and noise shaping). And yes, I know your opinion, knowledge is for losers only.

As I said multiple times, in my book, ingnorance is not a sin, proudly displaying it is.
 
Dont you know that only applies to a repetitive waveform? You cant construct a waveform from just 2 data points in one waveform cycle.

Ok. sharpen your pencils... I am bracing myself for another wiz-bang EE101 thing. Remember though... keep in mind the central point -- Music is not repetitive waveforms ... and the phrase AT LEAST 2 points are needed really means with music waveforms you will need more than 2.


This is not really true, although you'll hear it a lot. Two samples per cycle can perfectly reconstruct any waveform that doesn't violate Nyquist. Examples of exceptions always violate Nyquist if examined more carefully.


All good fortune,
Chris
 
As usual, displaying 0 (zero) knowledge of Nyquist, sampling, discrete Fourier transform, and other basics in signal processing (waaaaaay before you could get into dithering and noise shaping). And yes, I know your opinion, knowledge is for losers only.

As I said multiple times, in my book, ingnorance is not a sin, proudly displaying it is.

Beat me to it, stunning in fact blatant denial of the basic theory. The hired and fired experts must have humored him. There are limits to not being an expert in anything.
 
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yeh yeh yeh I knew this was coming. here we go with the EE101 thing ... Sampling rate is too low and sounds like it too. Figure out why. I gave you some clues is all.

Address the point please. Nyquist works in both directions... more sampling needed for single cycle and fewer sampling points for many repeating cycles (2 being the minimum). Address a single fast rise time pulse recording with sampling theory.... what do you need for sampling rate for accurate reproduction?

JC gave clue also when he said the wide BW of mc analog... Tr was a factor in what was accurate sounding compared to CD.

The higher sampling rate sounds more real or more accurate.


THx-RNMarsh
 
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Sigh...

Sampling a sine at exactly 2x will give any value from full amplitude to zero, depending on phase.

Mr Marsh is close.

Jn

Now you're obfuscating talking about the Nyquist limit literally which has little to do with the prattle above. A 22.04kHz sine sampled at 44.1kHz is reconstructed exactly any frequency or amplitude modulation, etc. can violate Nyquist if not filtered out.
 
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yeh yeh yeh I knew this was coming. here we go with the EE101 thing ... Sampling rate is too low and sounds like it too. Figure out why. I gave you some clues is all.

Address the point please. Nyquist works in both directions... more sampling needed for single cycle and fewer sampling points for many repeating cycles (2 being the minimum). Address a single fast rise time pulse recording with sampling theory.... what do you need for sampling rate for accurate reproduction?

JC gave clue also when he said the wide BW of mc analog... Tr was a factor in what was more accurate sounding compared to CD.

The higher sampling rate sounds more real or more accurate.

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Sampling a sine at exactly 2x will give any value from full amplitude to zero, depending on phase.

Mr Marsh is close.

Jn

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Another clue from JN.

Now you experts go away and figure out why a higher sampling rate is needed for non-repetitive waveforms for accurate sound.


THx-RNMarsh
 
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This violates Nyquist. As do any of the other examples I've ever heard purporting to disprove sampling theory. The classic goes "music is complicated" or something equally fuzzy along those lines.


All good fortune,
Chris
No, it does not.
A 1khz sine sampled at 2k can be captured every zero crossing.

I said nothing of complicated...
Granted, I am fuzzy, but it's the weekend.. I'll shave Monday.

Jn
 
This violates Nyquist. As do any of the other examples I've ever heard purporting to disprove sampling theory. The classic goes "music is complicated" or something equally fuzzy along those lines.

No it doesn't it just places unnecessary focus on the limit. I'm with Hans.

I’m sure many won’t agree, but when playing at normal listening levels, I could not hear any difference at all.

Hans
 
Here's a test you can perform yourself, using only Audacity, to attack your question. From a very high sample rate source make three "downsampled" files. First bandlimit appropriate for a 96kHz sampling rate, then sample at 96kHz. Second bandlimit appropriate for Red Book, then sample at 44k1Hz and 96kHz.


This will tell you which part of the A/D/A process is bugging you.


All good fortune,
Chris
 
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