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MQA creator Bob Stuart answers questions.

Blumlein 88

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You might wish to check out Deltawave. A handy nulling software.

https://deltaw.org

I'll look at things later when I have a chance. It wasn't clear what all the differences were from. Though I did look at and listen to A, B, and C versions.
 

MRC01

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The idea was to determine what information was lost while capturing at 48/24 vs capturing at 192/24. Also, what was lost when resampling and dithering to 44.1/16 from the 48/24 and 192/24 captures. After that is determined, we can look at the lost information, try to hear it, and figure out whether it indicates a lost transfer of mechanical momentum, or otherwise just represents useless true-ultrasonic-sinusoids and noise.
...
Theory says that digital encoding & recreation perfectly reconstructs the original waveform so long as it is sampled at least twice the highest frequency you want to capture. That means if both sides (sampling & reconstruction) are done properly, there is no loss of information, we get perfect fidelity. This implies that ultrasonic frequencies cannot cause in-band changes to the signal of any kind, no matter how they appear (whether as transients or whatever). The notion of any such mechanism, "momentum" or whatever, violates this theory.

This implies that if there are any audible differences between these files (e.g. in-passband frequency content), it means either side (recording or playback) wasn't done properly. [or, the listener can hear beyond 44-16 resolution, which is unlikely] Perhaps the AA filter used during recording allowed some high frequency through and aliased into the encoding. Perhaps playback used an imperfect reconstruction formula/method.
 

Krunok

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Theory says that digital encoding & recreation perfectly reconstructs the original waveform so long as it is sampled at least twice the highest frequency you want to capture.

It's actually like this ".. so long as it is sampled at least twice the frequency of the highest harmonic of the original waveform you want to capture."
 

Krunok

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Of course. Harmonics are frequencies, so "the highest frequency you want to capture" includes all frequencies of any kind, whether harmonics or fundamentals.

Nope, only harmonics, because harmonics are sine waves. For example, with 44.1 kHz sampling you cannot catch any other 22kHz waveform except sine wave.
You can catch any waveform of lower frequency whose harmonics don't go over 22kHz. That means, for example, you cannot catch a triangle shaped wave of 15kHz as it will have harmonics higher than 22kHz.
 
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SIY

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Nope, no fundamentals, only harmonics, because harmonics are sine waves. For example, with 44.1 kHz sampling you cannot catch any other 22kHz waveform except sine wave.
You can catch any waveform of lower frequency whose harmonics don't go over 22kHz. That means, for example, you cannot catch a triangle shaped wave of 15kHz as it will have harmonics higher than 22kHz.

Well you capture it, but only the fundamental. Which is fine because there's no audible difference when you play a triangle wave at 15k versus a sine wave with the same magnitude as the fundamental. Of course, at my age, they sound identical because they're just past my HF limit.:cool:
 

Krunok

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Well you capture it, but only the fundamental. Which is fine because there's no audible difference when you play a triangle wave at 15k versus a sine wave with the same magnitude as the fundamental. Of course, at my age, they sound identical because they're just past my HF limit.:cool:

Sure, you capture only fundamental. I can't hear shit above 14kHz so 15kHz triangle wave is anyhow only a mathematical abstraction to me. :D
 
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Krunok

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You and me both, bro.

809.png
 
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March Audio

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Theory says that digital encoding & recreation perfectly reconstructs the original waveform so long as it is sampled at least twice the highest frequency you want to capture. That means if both sides (sampling & reconstruction) are done properly, there is no loss of information, we get perfect fidelity. This implies that ultrasonic frequencies cannot cause in-band changes to the signal of any kind, no matter how they appear (whether as transients or whatever). The notion of any such mechanism, "momentum" or whatever, violates this theory.

This implies that if there are any audible differences between these files (e.g. in-passband frequency content), it means either side (recording or playback) wasn't done properly. [or, the listener can hear beyond 44-16 resolution, which is unlikely] Perhaps the AA filter used during recording allowed some high frequency through and aliased into the encoding. Perhaps playback used an imperfect reconstruction formula/method.

Sergei is trying to disprove proven theory.

I demonstrated earlier that if you high pass filter (at 40kHz in that example) a wide band recording you hear nothing of the content that is in the pass band. It's above your hearing range.

Well no shit Sherlock :)

The test is so simple and conclusive, yet sergei persists.
 

MRC01

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Nope, only harmonics, because harmonics are sine waves. For example, with 44.1 kHz sampling you cannot catch any other 22kHz waveform except sine wave.
You can catch any waveform of lower frequency whose harmonics don't go over 22kHz. That means, for example, you cannot catch a triangle shaped wave of 15kHz as it will have harmonics higher than 22kHz.
I don't get the distinction you're making. Technically speaking, every sound we hear consists of a bunch of sin waves, each at a different frequency and amplitude, all superimposed together. Put differently: every real sound is a bundle of frequencies all piled together. These real sounds contain many frequencies we can hear, and some frequencies we can't hear. If you surgically remove the ones we can't hear, it sounds the same to us. By surgically, I mean, remove them without side effects that distort frequencies we can hear.

If you use a bandwidth wide enough to capture every frequency we can hear as a sin wave, you're done. It will transparently capture every real sound we can hear, regardless of fundamental vs. harmonic. No need to worry about or distinguish fundamentals from harmonics. SIY gave a good example: to me, a 14 kHz sin wave sounds the same as a 14 kHz square or triangle wave. I can only (barely) hear the fundamental, and all the harmonics that make them sound different are well above my hearing.
 

Krunok

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I don't get the distinction you're making. Technically speaking, every sound we hear consists of a bunch of sin waves, each at a different frequency and amplitude, all superimposed together. Put differently: every real sound is a bundle of frequencies all piled together. These real sounds contain many frequencies we can hear, and some frequencies we can't hear. If you surgically remove the ones we can't hear, it sounds the same to us. By surgically, I mean, remove them without side effects that distort frequencies we can hear.

If you use a bandwidth wide enough to capture every frequency we can hear as a sin wave, you're done. It will transparently capture every real sound we can hear, regardless of fundamental vs. harmonic. No need to worry about or distinguish fundamentals from harmonics. SIY gave a good example: to me, a 14 kHz sin wave sounds the same as a 14 kHz square or triangle wave. I can only (barely) hear the fundamental, and all the harmonics that make them sound different are well above my hearing.

Here is what you said:
"Theory says that digital encoding & recreation perfectly reconstructs the original waveform so long as it is sampled at least twice the highest frequency you want to capture. "

That is not true for any original waveform but only for waveforms which doesn't have harmonics over 1/2 of the sampling frequency. That means that 15kHz triangle wave won't be "perfectly reconstructed" with 44.1kHz sampling rate as it has harmonics over 22kHz - it will be reconstructed as a 15khz sine wave with the same amplitude. SIY is of course right when saying we won't hear the difference, but it will still not be perfectly reconstructed.
 

Krunok

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Sergei is trying to disprove proven theory.

I demonstrated earlier that if you high pass filter (at 40kHz in that example) a wide band recording you hear nothing of the content that is in the pass band. It's above your hearing range.

Well no shit Sherlock :)

The test is so simple and conclusive, yet sergei persists.

When Sergei was absent for one day you were missing him, and now when he's here you're bitching him. Eh, human nature.. :D
 

MRC01

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Here is what you said:
"Theory says that digital encoding & recreation perfectly reconstructs the original waveform so long as it is sampled at least twice the highest frequency you want to capture. "

That is not true for any original waveform but only for waveforms which doesn't have harmonics over 1/2 of the sampling frequency. That means that 15kHz triangle wave won't be "perfectly reconstructed" with 44.1kHz sampling rate as it has harmonics over 22kHz - it will be reconstructed as a 15khz sine wave with the same amplitude. SIY is of course right when saying we won't hear the difference, but it will still not be perfectly reconstructed.
At first I thought you were making a distinction between fundamentals & harmonics that I never stated or implied. But you may be making a different point: the term original wave is ambiguous.

Does it mean the actual sound wave that was recorded, or does it mean the lowpass filtered version of that wave that was digitally encoded? I meant the latter, and thought it was obvious since lowpass AA filtering is a necessary part of encoding, and I said, "as long as both sides (sampling & reconstruction) are done properly".
 

Sergei

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You might wish to check out Deltawave. A handy nulling software.

https://deltaw.org

I'll look at things later when I have a chance. It wasn't clear what all the differences were from. Though I did look at and listen to A, B, and C versions.

Thanks! I think I'm OK with the A/B/C after today's rechecking. However, now that I looked at D/E/F during a day, the A vs F difference seems to be too large for my taste in parts not involving cymbals. Either guitar(s) also threw in some ultrasonic parts as the spectrum of S192 suggests, or there is a systematic processing error. Please disregard the D/E/F/I/J for now.
 

Krunok

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At first I thought you were making a distinction between fundamentals & harmonics that I never stated or implied. But you may be making a different point: the term original wave is ambiguous.

Does it mean the actual sound wave that was recorded, or does it mean the lowpass filtered version of that wave that was digitally encoded? I meant the latter, and thought it was obvious since lowpass AA filtering is a necessary part of encoding, and I said, "as long as both sides (sampling & reconstruction) are done properly".

Here is how I see our discussion:

original signal --> ADC --> digital link/media --> DAC --> output signal

I thought we were comparing the original analog signal that enters input ADC connectors vs output analog signal on the output DAC connectors. In that terms original signal is what has been recorded. While I'm aware that AD process involves input lowpass filter, as well as DA process involves it on the output, I don't really see them as part of our discussion about Nyquist theorem.
 

Blumlein 88

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Again, what about this shows us some momentum transfer above 20 khz is going on?
1561628738618.png

What is with the high frequency noise above 60 khz running at about a 17 hz rate?
1561628866953.png
 
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