"Theoretically" no, but 99.99% of guitar pickups are wound with lots of fine (like AWG 40) wire (this "tradition" dates to using high-impedance vacuum tube circuitry, to give as much voltage output as practical from the pickup). The large inductance and self-capacitance gives a resonance in the 6 to 8 kHz range, and response drops off above that. This resonance is pretty much an inherent part of electric guitar sound. Likewise drivers used in guitar amplifiers have response falling off over 5kHz or so.
The only exception I've heard of is the Les Paul Recording Guitar, with low-impedance pickup and XLR output meant to connect to a low-impedance balanced mic preamp as commonly used in recording studios. It solved the common electric guitar problems of hum and noise, but was a commercial failure due to its flat (and thus boring) frequency response. But it probably has the ultrasonic frequency response you're implying with the guitar pick thing.
To the larger point, nothing "violates Nyquist" as long as the signal goes through an appropriate anti-aliasing filter before the sampling part.
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We all agree on that point. The discussion has been about what is being filtered out, and if it matters.
Jn
Edit: and while I've been using the phrase violate nyquist, what I refer to is signal content that exceeds CD Fs.
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Could the water be colored for visibility? Like natural gas is perfumed.
All good fortune,
Chris
Time for a picture. Just to summarize I'm trying to use the analytic signal to extract the instantaneous frequency and amplitude, this is a fairly old and standard technique. For the 16kHz and 24kHz sum of sines the Hilbert transform returns 20kHz as the instantaneous frequency, remove the 24kHz you get 16kHz and the 4kHz envelope modulation disappears. This seems obvious to me. Why not just sum the two sines turning one off and on? Lots of care is necessary to eliminate all possible confounders.
This is the envelope of a damped 15kHz sine wave before and after 22,050Hz filtering (192k sampled original). The transient affects the Hilbert transform as it must. The horizontal axis is 192k sample periods, nothing here excites me.
This is the envelope of a damped 15kHz sine wave before and after 22,050Hz filtering (192k sampled original). The transient affects the Hilbert transform as it must. The horizontal axis is 192k sample periods, nothing here excites me.
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Could this be an issue of two different meanings of the word "modulation" being conflated?
All good fortune,
Chris
Excellent. This is precisely what I stated very early on, when I provided the plots with the 20 kHz overlayed on the summed sines.
Nice, I have been kinda lonely, trying to explain all this.
How about just changing the ratio of the sines. As I also said, as one is lowered, the instantaneous frequency will drift towards the dominant one. I called this effect phase walking.
I'm not quite sure what you are showing here. But the peaks of both do not line up.
Jn
PS. Scott, thank you, your work as always, is awesome.
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The general public doesn’t really need to know how records or sausages are made. 😀
Ok, just a couple of links related to current talk.
If you have 3 hours to spare I highly recommend to watch this recording session. It is the only one of its kind put up on youtube.
That's real life, gentleman. (A "reality show", if you wish). Read the comments too.
Fly On The Wall Film Of A Large Orchestral Session At Air Studios
YouTube
From the same composer as above. See how many ("more that ever before") modern high bandwidth mics you can count
HOW TO MIC A STRING SECTION
YouTube
The Decca Sound: Secrets Of The Engineers
The Decca Sound: Secrets Of The Engineers – The Polymath Perspective
Decca Phase 4 - Stereo Concert Series: Recording and Mixing Demonstration
YouTube
B&K aka DPA mics get mentioned in these two threads:
M50 substitute
M50 substitute
The Myth of the Accurate Microphone
https://repforums.prosoundweb.com/index.php/topic,37170.0.html
Enjoy!
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The blue is the envelope of the original 192k damped sine. Unfortunately the analytic signal is computed using FFT's so the same issues of truncation of the time record will cause ripples. The green one shows the Gibbs ripple, the envelope is rectified hence the behavior at zero crossing.
JC and MR. Marsh should spend some time reading this thread. From Less Watts...
I'm done with the microphone discussion.
Here we go again.
The Gibbs ripples are added to the FFT ripples and also the content above 22.05 has been subtracted.
Nevertheless you expect the FFT ripple peaks after filtering to stay where they were before.
That would be a mathematical impossibility.
Hans
I've been researching "instantaneous frequency". It seems one(?) of the problems throughout this discussion has been a combination of different levels of understanding and inaccurate use of terminology. It seems to me if "instantaneous frequency" had been used instead of "frequency shifting" it would have saved a lot of confusion?
I read somewhere that the rise time on a bass drum, when struck hard, was 80us.
Surely if this discussion on a plucked string is to have merit we need to understand the rise time we are talking about. It would need to be less than 50us to have a problem with a 20 kHz BW digital system.
Separately, a wide band mic may be able to pick this stuff up but not your ears.
Surely if this discussion on a plucked string is to have merit we need to understand the rise time we are talking about. It would need to be less than 50us to have a problem with a 20 kHz BW digital system.
Separately, a wide band mic may be able to pick this stuff up but not your ears.
@elektroj - very nice article on the ‘Decca’ sound. My Decca Maazel recordings are amongst my favorites (1963-1967 with the VSO).
Every one has their opinion when it comes to Sound of somethings. proves zero.
Its another 'listening' evaluation and opinions. That which you used to object to using in arguments.
😉
My concern is which is more accurate.... not what a person Likes. I prefer music which is closer to what I hear in person/live. especially, acoustic. Does that cello sound like a real cello etc.
Why are you using 192K sampling? when all this fuss has been about 16/44 ?
THx-RNMarsh
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The discussion is about the effect of what is filtered out. The high sampling rate gives accuracy to the measurements
Yes. I would think it does same for recording/play and why CD isnt good enough?
THx-RNMarsh
Maybe. It also seems to show how easily measurements can be misunderstood, they don't lie, but they do require interpretation
If it's an "isolated" transient, in the case of a guitar string (even if it were fast enough to violate Nyquist), I'm presuming the timing won't matter, but if there were a very fast transient cluster (can I lay claim to inventing that term?😉) such as maybe exists in the cymbal strike it may be audible?
You used the same verbage to "explain" that the 20 kHz carrier didn't exist.
You used the same verbage to "explain" that the frequency shifted carrier in your plots didn't exist.
You used the incorrect modulation to "explain" how my assertions were "incorrect", despite the fact that it was visually evident.
Now that the bulk of what I have been saying has been supported analytically, you choose the words "here we go again", neglecting the fact that everything I have been saying has been borne out as a accurate.
It would be nice if you would spend less time going after me, and more time understanding and revisiting the fundamentals. And discussing.
Jn