John Curl's Blowtorch preamplifier part IV

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This one might be somewhat related..
Inspired by Scott's suggestion that FFTs should be taken at exact number of bins the sample rate is, I tried to trick FFT analyzers that I have in my DAWs by generating the test files at odd (65536 kHz) frequencies with test tones being at f/2. Guess what - none of the generators that I have were able to produce test tones at exactly the fs/2! No problem with tones at fs/2 + 1Hz or fs/2 - 1Hz, though.
Anyway, to cut the long story short, this is how a sine tones offset by 1 Hz from fs/2 look in time domain - some strange AM! Is this aliasing? (I assume that there is some built-in filter in the generator).
Pictures show 3 sec snippet fully zoomed out and 22049 kHz signal fully zoomed in.
I'd love to hear what you think guys.
P.S Test files are zipped wavs at "regular" 44.1k, not some odd frequencies.
The output of a sampled sine at fs/2 (bin 32768 for 65536 block size) depends on the start phase. If the generator code works with the sin(x) function, you need a start phase that is not a integer multiple of pi (including zero) to get any output. At pi/2 you get your sine at full scale, with 3pi/2 the inverted sine.
And when you're of by one bin in either direction, you get beating (not AM!) with a period of one blocksize of, say 65536. Again, the difference between 32767 and 32769 bins is the polarity of the output. And the start position of the beating pattern is determined by the start phase. With phase=zero the beating waveform starts at zero, whereas with startphase=pi/2 it starts at the max amplitude.
 
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Don't know that this type of thing reads across to biological matters that easily other than anecdotally.

Hearing, in particular brain processing of sound before (some of) it is sent on to conscious awareness, is nothing but biological.

How we typically measure audio gear does not fully correlate with how such gear sounds to humans.

We tend to focus attention on measurements that are easy to do and easy to interpret. Doesn't mean those are all there is to it.

Again, there is a correspondence to doctors evaluating patients. They prefer to focus on diseases that are easy measure and diagnose using easy to interpret standard tests. They don't like treating, say, fibromyalgia, another medical problem that doctors used to believe was fake, or all in the patient's head. They still don't classify it as a disease, rather it is a syndrome. The reason it is not a disease is because there is no standard test to make it easy to diagnose.
 
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Good found it...


This seems to be based on an unmodulated groove. The pressures and temperatures of the groove accelerating the stylus effective mass are a completely different matter. I worked out an example of this once before on this very thread, for SY IIRC, and can do it again if needed, but it's simple enough to do for oneself.


Choose a large but trackable groove modulation; calculate acceleration. From typical stylus effective mass and typical contact area calculate pressure. I don't know how to calculate temperature from pressure.


All good fortune,
Chris
 
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Is there any other research that counters it, or proves it wrong ?

Maybe the mythology is what needs closer scrutiny instead of dismissal?

The analysis that quoted this computation included an extensive bibliography and there was no actual computation based on real physical material properties there, as they said only speculations essentially all speculation based on throwing large numbers around (my words).

Anyone can do the experiment (I have), these computations all have groove velocity as a variable and you can take a test LP and let a tone freewheel down to a very low velocity. There is essentially no change in the groove/vinyl geometry with groove velocity.

EDIT - Chris, missed your post. My measurements show no change in groove/vinyl contact geometry with an almost 10:1 change in RPM (the distortion remains unchanged). Obviously the acceleration the stylus experiences changes sign so would the pressure, that clearly seen in tracking torture tracks where the stylus leaves the surface. I would seem to me that a large change in pressure between peaks and valleys would dramatically increase seconds. On a left or right only track the other is blank.
 
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Well, does your dac sound time smeared?

If so, you have some work to do to 'clean it up,' or else just get a better dac.

well I wouldn't know what time smearing sounds like!

maybe it's what some find different between 16/44 and 24/96?

In a low pass, does a slow roll off filter sound different then a sharp roll off? If so is the slow roll off preferred subjectively?

I'd like to know more about that pre-ringing chirp though......is it something everyone hears?
 
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That's a big can of worms in there. I recall Dr. Hans van Maanen in Linear Audio (vol. 5 if memory serves) promoting the same "time smearing" stuff - and the the tough reprimands he got from Marcel van de Gevel and yours truly.

I even visited Hans v Maanen after his publication in attempt to convince him in person that he was wrong on several issues.
However that turned out to be a mission impossible and was the reason for me to write a response in one of the next Linear Audio volumes. 😀😀

Hans
 
Is there any other research that counters it, or proves it wrong ?

Maybe the mythology is what needs closer scrutiny instead of dismissal?

Bob, that article you've linked is correct in the technical details, but if the conclusion that _all_ upsampling filters were of the slow transition kind (transistion means from passband to stopband) at that time (it's original version is from 2000, and the "upsampling hype" started around 1996 afair), but of course it is a possible explanation for the reported sound differences.

But if it is correct the positive (if existent) effects should strongly depend on the type of music (which means on the signal level of the music from 10 kHz - to 20 kHz).

A long time ago a japanes company promoted in their cd players a technology called "legato link" that added some (?arbitrarily?) ultrasonic content with descending level over frequency and claimed in intense listening tests, participants strongly preferred this "improved" reproduction. Unfortunately I've never seen an analysis of this added signals and never got one of these player to look at it myself (and listen).
 
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I even visited Hans v Maanen after his publication in attempt to convince him in person that he was wrong on several issues.
However that turned out to be a mission impossible and was the reason for me to write a response in one of the next Linear Audio volumes. 😀😀

Hans

OTOH, Marcel wrote in a letter to the editor in the same publication that phono cartridges were reciprocal motors. BTW isn't Hans Joe's physics buddy, he does write some wrongs at times.
 
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Nobody cares to prove one Doug Rife wrong, life is too short for that. Short of some obvious (but not uncommon) confusions, like between the reconstruction and anti-aliasing filters, the article collides, in the name of the Audio Gods, with some well established facts in digital signal processing and introduces without justification some useless, audio context specific, concepts like "time smearing". These constitute in a body of extraordinary claims, for which extraordinary proof is not provided. There is only one way to separate the grain from the chaff: start reading a good book on digital signal processing (unfortunately not an easy/casual reading) and then the truth will come up clearly. BTW, this is what such authors are relying on: not many are willing to go through such an exercise, and those that already did usually couldn't care less about Doug Rife and his opinions.

You go ahead and scrutinize the mythology, not my cup of tea.

General rule, it helps if you (and that really means you) actually read a paper before commenting instead of responding to trigger words.

1.) No, he does not promote the "time smear theory" instead he does exactly the opposite
2.) No,he does not confuse anti-aliasing and anti-imaging filters
3.) Obviously you don't know who Doug Rife is
4.) You are often doing what seems to be projecting; in this case it's an author like you relying on the fact that others (presumably from the same cult section) do not read but instead post supporting comments
 
But if it is correct the positive (if existent) effects should strongly depend on the type of music (which means on the signal level of the music from 10 kHz - to 20 kHz).

A long time ago a japanes company promoted in their cd players a technology called "legato link" that added some (?arbitrarily?) ultrasonic content with descending level over frequency and claimed in intense listening tests, participants strongly preferred this "improved" reproduction. Unfortunately I've never seen an analysis of this added signals and never got one of these player to look at it myself (and listen).

Thx Jakob,

Pre-ringing is in a sense time smear, no?

I have a old pioneer ‘elite’ player with the hi-bit legato link,it does have a ease to it at higher volumes but falls short in excitement (kindly flat) at normal levels. I hardly have used it since I started hi-res streaming almost 2 years ago. The same recordings (streamed) sound much more dynamic.
 
OTOH, Marcel wrote in a letter to the editor in the same publication that phono cartridges were reciprocal motors. BTW isn't Hans Joe's physics buddy, he does write some wrongs at times.

Hans v Maanen developed a phono preamp in 1980, in an attempt to counteract the mechanical resonances of phono cartridges.
Don't know who Joe is, but Menno van der Veen was one of his buddies.
Steven van Raalte tried to make a more stable correction network, based on the previous work of Hans in one of the first LA's.
v Maanen is still a convinced time-smear adept as can be read on his website.

https://www.temporalcoherence.nl/cms/images/docs/Cross-Over-Time-Smear.pdf

Hans
 
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