What harmonic does to sound?

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Suppose you fed a step waveform into the the circuit. If there is only phae delay but no time delay the output will respond immediately, albeit an exponential-shaped rise. If there is a time-delay the output will not respond immediately.

It is misleading to think of a filter as a form of time delay, It isn't. It sort of looks like it is in the special case of a continuous sinewave when you look at it on a scope. But it isn't a delay.

Real time delays cause problems, especially in control systems. Imagine driving a car if the steering column had a 5 second delay. You'd soon be in a ditch. Delays tend to make feedback loops unstable and lead to oscillation. There's a special set of Z functions for designing control loops for systems involving time delays.

I imagine the 1.5us delay quoted earlier is not entirely a time delay. I'm not sure what the time delay might be for a power transistor operating in its linear region...pretty darned small though.
 
If there is only phae delay but no time delay the output will respond immediately, albeit an exponential-shaped rise.

No

It is misleading to think of a filter as a form of time delay, It isn't.

Yes it is !!! Never hear of group-delay (which can be quite frequency-independant for critical- and Bessel- lowpass-filters) ?

Real time delays cause problems, especially in control systems. Imagine driving a car if the steering column had a 5 second delay. You'd soon be in a ditch. Delays tend to make feedback loops unstable and lead to oscillation. There's a special set of Z functions for designing control loops for systems involving time delays.

Agreed, but phase-delay causes problems TOO !!!! Otherwise it would be fairly easy to use NFB of whatever amount around any circuit topology one can imagine. This way we would end up with the perfect "wire with gain" at almost no cost. Which is unfortunately not the case, at least to my knowledge.

Regards

Charles
 
hi
If there is a time-delay the output will not respond immediately.

Yes

How can I simulate it for instance with a cad
software?
I mean, does a module exist to introduce time
delay (with no shape change) in every component?
also in a cable? Where to put it?

How I have to think at the info propagation in a 3 leg
component?

I think (IMHO) this is very important to simulate
true feedback behavior

Bye

Federico
 
Suppose you had a simple nonlinear transfer function like below:
vo = a1*vi + a3*vi3

To all interested
I noted that with a transfer like the above I
obtain spectra with only the fundamental and the third harmonic, irrespective of the level.

If I use this:

vo = a1*vi + a2*vi2

I obtain only the second harm.

But if I use any other exponent I have all the harmonics,
even or odd, below that exponent.

If I try, for example
vo = a1*vi + a5*vi5

I obtain 1th,3rd and 5th harmonic. If, at that level, I add
(or subtract) the right amount of 3rd order in this way:

vo = a1*vi + a3*vi3 + a5*vi5

I succeed to cancel the 3rd harm, but only at that particular level. Changing it the 3rd order come back.

So 2nd and 3rd components behave differently from the others.

bye
federico
 
So, there is time delay and phase shift. Is this makes contribution to distortion in sound? I imagine the feedback is coming too late than the input signal, so there must be some confusion in the differential.

Nelson Pass said that if the transistor gain stage is too long, there is some kind of "Hall of Mirror" effect. I assume this is due to time delay too. This effect makes the differential not effectively working to make correction signal for the next stage, because what had happened in the output stages comes too late to the differential, it makes the differential makes a wrong error signal. That is why he likes a 2 stages amp, maybe because the control is much better than many stages power amp.

I dont know about the fact. Is this delay good or bad to sound? There are 9 stages power amp out there in the market.
 
lumanauw said:
So, there is time delay and phase shift. Is this makes contribution to distortion in sound? I imagine the feedback is coming too late than the input signal, so there must be some confusion in the differential.

Nelson Pass said that if the transistor gain stage is too long, there is some kind of "Hall of Mirror" effect. I assume this is due to time delay too. This effect makes the differential not effectively working to make correction signal for the next stage, because what had happened in the output stages comes too late to the differential, it makes the differential makes a wrong error signal. That is why he likes a 2 stages amp, maybe because the control is much better than many stages power amp.

I dont know about the fact. Is this delay good or bad to sound? There are 9 stages power amp out there in the market.

Yes, time delay and phase shift. I believe a typical bandwidth measurement showing gain magnitude and phase, has the delay time effects added to the RLC caused phase shift. (Delay time)/(sine period)*360 = phase shift due to delays, for sinewave. If the delay times are constant, then they produce more phase shift as frequency increases, I think.

To me this effect is like the tail wagging the dog instead of the dog wagging the tail. I think heavy global feedback with phase shift (delays + RLC phase shifts) causes more TIM, IM, and THD. I think an amp that does the job with the least active devices with adequate individual bandwidths, will make a better amp. I think faster parts can have less delays. I think delays come from different sources. Delay related to the speed of current flow and delay related to digital gate propagation being different. Part of propagation delay is related to slew rates in digital circuits, I think. There are probably several physical effects that contribute to an overall delay in an active device with others in addition for multiple stages and the PCB. The physics of each active device type should dictate the additive delays in the individual part. I think we see the net effects as phase shift, usually.

I'm sure I've got this all wrong, and it will be presented correctly by the experts. I'm now ready for my intellectual whipping.
 
Hi All
You might pick the input voltage to be a sinusoid so that the third harmonic was, say, 80 dB down from the fundamental. But in the weakly nonlinear region, if you reduced the input signal by 1 dB, the third harmonic would now be 82 dB down from the fundamental.

I have a question for someone
expert in musical instruments.

Does the spectrum of an instrument,e.g. a violin,
depend on the level?

I mean, it is like a distorted sinusoidal signal or the spectrum does not depend on level?

thank you

Federico
 
Re: There's Life Above 20 Kilohertz!

dimitri said:

I just found the same survey :cool:

Comments on James Boyk's Course
"Projects in Music & Science" (EE/Mu 107)
California Institute of Technology
"Exactly the kind of thing I came to Caltech for ... my most memorable, unique class. —Ben Brantley '00. (TA in 107.)

By far my favorite and most memorable course at Caltech... I would rank EE/Mu 107 among the top of all my Caltech courses, along with CS91 and the EE5x series taught by Glenn George. —David Barksdale '96 BS E&AS. (TA in 107. Projects: Class A headphone amp, Class AB high-bias-current minimal-global-feedback power amplifier.)

A perfect blend of engineering and science.... Best class I took at Caltech. —Bruce Miller '89 BSEE. (Project: Stereo miking demo CD.)

Taught me how to approach engineering like science.... filled in many of the gaps left in the more "mainstream" classes while still being a lot of fun. —Ken Walsh '96 BSEE, MS'97. (Project: Very high speed linear phase D/A.)

EE/Mu 107 provided a great opportunity to apply the theoretical knowledge I gained as a Caltech student to practical problems.... As a working engineer, I find that practical experience and good instincts are often as important as theoretical background. EE/Mu 107 is one the relatively few courses at Caltech that can provide that practical background.... One of the most enjoyable and interesting of all my courses. —Chris Ulmer '93 BS/EE 1993, MS/EE 1994. (Project: Software to implement a spectrum analyzer that could display spectral change of sounds.)

"More real laboratory science went on in that class than in any other I took in my Caltech career." —Bruce J. Sams III '83. (Ph.D. Astrophysics, Harvard; Max Planck Institute)

"The highest standards of scientific honesty and thoroughness were demanded.... Supremely valuable." —Denes Zsolnay '84.

"The theory of double-blind testing was not presented in any other course I took at Caltech."
—William Snyder '82.
(Later chief engineer, Krell Electronics,
makers of Class A audio amps.)

jb-04-30th-71.jpg


Curriculum Vitae — James Boyk
Current:
* Concert pianist and internationally-known recording artist
* Artist teacher of piano; coach of interpretation for all instruments
* Consultant on high-resolution audio (recording and playback)
* Consultant on choice and care of fine pianos
* Writer
About single, double & treble-blind testing
Hi-Fi News & Record Review, March, 1986.
Copyright © 1986, 1997 James Boyk

Dear Sir,

Professor Lipshitz complains ("Views" December 1985) that Martin Colloms's amplifier test was "zero-blind." The Editor's response claims it was "double-blind."
What the Editor described is actually a "single-blind" test; that is, one in which the subject does not know which unknown he is judging at any moment. "Double-blind" properly refers to tests in which neither the subject nor the operator know which is which. In a double-blind test, the experimenter (different from the operator) has made provision for identifying the unknowns, but the operator is not privy to these arrangements.
And if I may use this forum to introduce a new concept demanded by these trying times in audio, the triple-blind test is one in which neither the subject, the operator nor the experimenter knows which was which, nor what it all means. Of course, triple-blind tests lead to double-talk; and this is the explanation for so much of today's audio writing.
On the other hand, if we all keep our ears open, we may learn something. As long as we don't have the double-blind leading the double-deaf, there's hope.


Yours,
James Boyk

Links:
http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
http://www.its.caltech.edu/~musiclab/
http://www.cco.caltech.edu/~boyk/
http://www.performancerecordings.com/
 
There's Life Above 20 Kilohertz!
A Survey of Musical Instrument Spectra to 102.4 KHz James Boyk
California Institute of Technology
XI. What Next?

A natural next step would be to measure the ultrasonic content of orchestral sound
as heard from normal listening or recording distances.
This will automatically allow for the absorption of ultrasonics by the air.

The project will be expensive, because musicians' union rules require players to be paid at recording rates,
which are several times ordinary "scale," whenever a live microphone is present;
and I anticipate difficulty in having these rules waived for our research.

We solicit funding for this project!

fig1a.gif


http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
 
Dear Lineup , maybe at a live show there is life above 20khz
but in the living room it should be considered undesirable.
(We can't hear it, why should we waste amp power..)
An externally hosted image should be here but it was not working when we last tested it.

(0 - 50k log)
This FFT of vivadi-four seasons doesn't show much content
>20Khz. I took this off the output of a self amp at near full
output with a 47pf VAS miller cap.

from Lumanauw:
About miller cap in VAS, between C-B. I also experimented some values of it from 5pf to 470pf. The bigger this C, the sound seems to be less detailed/bright especially in high frequencies.

I've found that the miller cap value is very dependant on the
use of the amp. For a sub amp a large value seems to give
more tight bass and less high harmonics at clipping but for
full range a small 22-47pf value gives "crispness" to
various instruments.

The loudspeaker that is being driven is often ignored for
what it does to even and odd order distortions. I've
seen the amp above show different characteristics driving
different loudspeakers. Isn' t the speaker just another L-C-R
network that is a part of the amp's circuit ?

As far as the "tube" sound goes.. I'd love to build a 12ax7
preamp to hear real vacuum tube compression as opposed
to my DSP 32bit compression algorithms emulated by my pc
sound card. OS
 
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