Matti Otala - An Amplifier Milestone. Dead or Alive

Bob Cordell said:

Hi Rafael,

Good open-loop linearity is always a good thing with or without negative feedback. Those who have started with a really poor open loop performance and used NFB as a bandaid are the ones who have given NFB an undeserved bad rap.

High open-loop distortion will likely increase PIM, but such distotion will almost certainlt increase SMPTE IM (Amplitude Intermodulation Distortion, or AIM) just as much.

The conventional definition of PIM, as provided by Otala, is a test signal of 60 and 6000 Hz (sometimes 7000) in a 4:1 ratio of amplitude. SMPTE IM (AIM) measures the amplitude modulation that results on the 6 kHz carrier. PIM measures the phase modulation on the 6 kHz carrier. The fundamental process whereby NFB creates PIM depends on AIM in the open loop. There is thus an amplitude-to-phase conversion that results in PIM. However, this AM-PM conversion is not 100% efficient, so there will be both AIM and PIM in the output.

For this reason, if you have an amplifier with only 0.001% SMPTE IM, it is virtually impossible to have any substantial amount of feedback-generated PIM.

Some here like to refer to "FM distortion". This is just another way of saying PIM. Frequency is merely the rate-of-change of phase, so FM and PM are directly related.

Cheers,
Bob

Thanks Bob your answer was very informative.

High open-loop distortion will likely increase PIM, but such distotion will almost certainlt increase SMPTE IM (Amplitude Intermodulation Distortion, or AIM) just as much.

this would not be was the original proposal of Otalla, but maybe it was misinterpreted or he is confused to say that was the high negative feedback to cause distortion?

-You have interesting articles in your web page, unfortunately I can not translate the pdf :xeye:
But I eulogy, you share important information form of public and free, are few who do that.
 
Wavebourn said:
Do you call damping of resonances and minimizing of colorations caused by diffraction "tweaks"? If yes, how can they be performed by DSP? Also, what Scott meant, correction of non-linear distortions caused by speakers. It is impossible since they are level and frequency dependent.
Advanced signal processing techniques also can take care of non-linear, non-time-invariant systems. You just need to know more about the system (which is the hard part, of course -- finding a good model for the system you want to correct).

BTW, for a fixed position in space, cancelling of diffraction perfectly is doable, the common echo-cancellation stuff (each time you make a telephone call DSP-based echo-cancellation is at work, although it's only in 1D space, not 3D).

Damping true resonances is rather trivial, compared to that (notch out both the freqs where the resonance is and also those fractions of it which give rise to that ringing frequency, as their harmonics).

And, as mentioned, we all completely agree that it is futile to try to correct major design flaws (the "we'll fix in the mix, later" attitude known in recording business). Just like with feedback... it cannnot fix a mediocre system.

- Klaus
 
Let us try to better understand Matti Otala's philosophy, and not try to second guess him, especially those who have been criticized in print, in the past, for trying to second guess both him and the rest of us who did the research on TIM approximately 10 years before Bob Cordell entered the picture.
Matti Otala was associated with an amplifier that had been accidently miss-wired, back in the 1960's. It sounded better than a similar amplifier that was wired correctly, but it measured relatively poorly. Analysis of why showed that the miss-wired amp had much less feedback than was originally designed in, had a wider open-loop bandwidth, and maybe better slew rate, but that was probably not measured at the time. As he was pursuing his Ph'D degree, he also read the audio paper in the IEEE by Daughority (sp) and Greiner published in 1966. (I read the same paper at the same year in California, and puzzled over it). He wrote his first paper about 1969 on TIM.
Matti always felt, intuitively, that open loop bandwidth was important. First, he thought it was because of TIM, but by 1976, he knew that this was not PROVABLE by TIM alone, so he coined the term DIM at that time, of which TIM was a subset.
Once Matti, EERO, and I presented our paper on measuring TIM at the AES in the fall of 1976, a graph in our paper was noted by Mitch Cotter. He contacted Matti at some point and pointed out that the graph measuring the ua741 op amp had both TIM and PIM components. Matti (and I) had ignored these added components, because while they were real, there was no immediate obvious mathematical relationship to TIM that we could document. These spurs, just as high as any typical TIM spur, had what appeared oddball frequencies symmetric around the fundamental test signals. It was FM modulation, or PIM.
In verification of this, Mitch Cotter has just found Matti's original IEEE paper on PIM, sent by Matti, denoting a message that Mitch was right. He didn't give Mitch an acknowledgement, however. I hope that this minimizes some confusion. Quibbling over minor details won't be very effective.
 
KSTR said:
Advanced signal processing techniques also can take care of non-linear, non-time-invariant systems. You just need to know more about the system (which is the hard part, of course -- finding a good model for the system you want to correct).


So, it is about "how to correct the model" instead of "how to correct the system".

BTW, for a fixed position in space, cancelling of diffraction perfectly is doable, the common echo-cancellation stuff (each time you make a telephone call DSP-based echo-cancellation is at work, although it's only in 1D space, not 3D).

...and my friend John Pursel don't use cellphone at all because it hurts his professional musician's ears.

Damping true resonances is rather trivial, compared to that (notch out both the freqs where the resonance is and also those fractions of it which give rise to that ringing frequency, as their harmonics).

Making equalizing frequency response is trivial, but damping of resonances such a way is impossible: the problem is, when mechanical systems resonate they produce tails of oscillations. You can make such tails less loud, but you can't damp them.
And, as mentioned, we all completely agree that it is futile to try to correct major design flaws (the "we'll fix in the mix, later" attitude known in recording business). Just like with feedback... it cannnot fix a mediocre system.

The main problem is, the difference between great and mediocre systems is analog. My PA system is great against some mediocre one that can be bought in Guitar Store, including those with digital equalization, while that digitally equalized systems age great in respect to some other systems from the same store that is mediocre.
 
Wavebourn said:
Making equalizing frequency response is trivial, but damping of resonances such a way is impossible: the problem is, when mechanical systems resonate they produce tails of oscillations. You can make such tails less loud, but you can't damp them.
Anatoliy, you are right in that the resonance mechanism itself is still there, but if you don't excite the resonance then there is no problem. But we should stop the discussion about these things in this thread, as it is completey off-topic (sorry that I picked up Scott's DSP sidenote).

- Klaus
 
KSTR said:
Anatoliy, you are right in that the resonance mechanism itself is still there, but if you don't excite the resonance then there is no problem. But we should stop the discussion about these things in this thread, as it is completey off-topic (sorry that I picked up Scott's DSP sidenote).

I absolutely agree with you Klaus that we must stop our resonance in this thread excited by Scott, but in audio reproduction we have to reproduce all signals that present so no matter how digital is our system we would excite resonances of speakers that are mechanical.
 
Wavebourn said:


I absolutely agree with you Klaus that we must stop our resonance in this thread excited by Scott, but in audio reproduction we have to reproduce all signals that present so no matter how digital is our system we would excite resonances of speakers that are mechanical.

More damping needed. 😀 The resonance (and other non-minimum phase) issues of course have nothing to do with the original comment that DSP is probably better for removing first order distortion. I thought running at SPLs for .1% distortion of air was a joke in the first place.
 
scott wurcer said:


More damping needed. 😀 The resonance (and other non-minimum phase) issues of course have nothing to do with the original comment that DSP is probably better for removing first order distortion. I thought running at SPLs for .1% distortion of air was a joke in the first place.

That's right; that's a good analogy. A digital system can put stars instead of letters in some words, but in order to stop resonances sometimes moderators' damping factors need to be involved.
Speaking of your joke, the truth is that such distortions that mimic mechanical media distortions may be left without any compensation since they are not audible at all (I mean, they are not perceived as distortions). Our perception apparatus is much more superior filter to any our electronic creations. Similarly, no matter how often I repeat this simple truth people usually filter it out discovering later for themselves independently.
The model is not the reality; the map is not the territory. That's why some asymmetrical PIMs of high degrees do not sound as distortions if they mimic Doppler effect.
 
john curl said:
I am pretty sure that 50 volts RMS is 625 W. Anyone want to verify this? What is the RATED output of the JC-1 at 4 ohms? 800W. Now that is kind of close, but Bob is slandering the JC-1 with outrageous conclusions without ever measuring one. I protest to the moderators!


John,

I'm not slandering the JC-1, not even close. I've said over and over it is a great amplifier.

I wasn't the one who posted the 50 Hz THD. You asked me what I would get with a calculator, and I said the simple-minded approach would say that 1 kHz THD would be just as bad. I then said I didn't think it was. It is indeed a curiosity that the 50 Hz THD would be worse than the 1 kHz THD.

It is even fair to ask if the posted 50 Hz THD spectrum is representative of what you expect and have measured. Who knows, maybe there was something in the Stereophile test setup.

When you avoid answering reasonable questions, especially in a very curt way, you invite speculation.

Cheers,
Bob
 
Re: 50Hz THD

PMA said:
50Hz THD

I am not sure that Bob Cordell has a point regarding the THD 50Hz importance. I would say he has not, and IMO 1kHz spectrum reveals more about PSU interraction with amplifier like in this case :

509Pasfig7.jpg


In fact, 50Hz measurement masks a lot.

Also, IMO the JC-1 result at 625W/4ohm is very good, we have to assess the spectrum and distortion residual shape, and this would be inaudible.


PMA,

If 50 Hz THD is worse than 1 kHz THD, something bad is happening that we would rather not have happen.

Bob
 
john curl said:
The JC-1 power amp clips at 1154W into 4 ohms. A bit more than 635W where it was measured. I don't think we are bumping into the ripple, but a regulated supply would probably be better. The problem is that the supply would have to dump 4KW on a momentary basis in order to be useful. Perhaps that is why I am not using a regulated output supply?


Hi John,

Thanks for this information and insight. I agree that it is probably not bumping into the ripple, since I would think that would cause spectral components at 6o Hz and some other 50+60 Hz intermod frequencies as well.

Bob
 
john curl said:
Let us try to better understand Matti Otala's philosophy, and not try to second guess him, especially those who have been criticized in print, in the past, for trying to second guess both him and the rest of us who did the research on TIM approximately 10 years before Bob Cordell entered the picture.
Matti Otala was associated with an amplifier that had been accidently miss-wired, back in the 1960's. It sounded better than a similar amplifier that was wired correctly, but it measured relatively poorly. Analysis of why showed that the miss-wired amp had much less feedback than was originally designed in, had a wider open-loop bandwidth, and maybe better slew rate, but that was probably not measured at the time. As he was pursuing his Ph'D degree, he also read the audio paper in the IEEE by Daughority (sp) and Greiner published in 1966. (I read the same paper at the same year in California, and puzzled over it). He wrote his first paper about 1969 on TIM.
Matti always felt, intuitively, that open loop bandwidth was important. First, he thought it was because of TIM, but by 1976, he knew that this was not PROVABLE by TIM alone, so he coined the term DIM at that time, of which TIM was a subset.
Once Matti, EERO, and I presented our paper on measuring TIM at the AES in the fall of 1976, a graph in our paper was noted by Mitch Cotter. He contacted Matti at some point and pointed out that the graph measuring the ua741 op amp had both TIM and PIM components. Matti (and I) had ignored these added components, because while they were real, there was no immediate obvious mathematical relationship to TIM that we could document. These spurs, just as high as any typical TIM spur, had what appeared oddball frequencies symmetric around the fundamental test signals. It was FM modulation, or PIM.
In verification of this, Mitch Cotter has just found Matti's original IEEE paper on PIM, sent by Matti, denoting a message that Mitch was right. He didn't give Mitch an acknowledgement, however. I hope that this minimizes some confusion. Quibbling over minor details won't be very effective.


Thanks for the historical insights John.

You are absolutely right; Matti was involved with TIM a good ten years before I came on the scene. Others were too, including Walt Jung.

You've said in the past that TIM was the distortion itself and that DIM was the particular measuring method. What you are saying now seems to be different.

I'm interested to know how Mitch inferred PIM from what I assume was a spectrum plot. Absent phase information, AIM and PIM are largely indistinguishable - they both produce sidebands about the fundamental. I suppose, however, that if AIM and PIM were both present, perhaps one could infer the presence of PIM if the upper and lower sidebands were of significantly different amplitude.

Cheers,
Bob
 
Re: Re: 50Hz THD

Bob Cordell said:


If 50 Hz THD is worse than 1 kHz THD, something bad is happening that we would rather not have happen.


Speculating further I may assume that it may be the result of servo distortions, as well as temperature-induced distortions in other than output devices that have much smaller dies. I believe the later may be caught up observing CMRR curve with frequency.

However, it is hard to guess without seeing the schematic, but knowing way of John's thinking I may assume how it looks.
 
Bob Cordell said:



John,

I'm not slandering the JC-1, not even close. I've said over and over it is a great amplifier.

I wasn't the one who posted the 50 Hz THD. You asked me what I would get with a calculator, and I said the simple-minded approach would say that 1 kHz THD would be just as bad. I then said I didn't think it was. It is indeed a curiosity that the 50 Hz THD would be worse than the 1 kHz THD.

It is even fair to ask if the posted 50 Hz THD spectrum is representative of what you expect and have measured. Who knows, maybe there was something in the Stereophile test setup.

When you avoid answering reasonable questions, especially in a very curt way, you invite speculation.

Cheers,
Bob

John only wants people to say good things about his amplifiers and only then will he engage in debate about it . If you don't then he will be very dismissive of any negative issues. How can you then have an objective debate about anything with him ??

regards
trev