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#1
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Phil wrote:
We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Andre Jute Visit Jute on Amps at http://members.lycos.co.uk/fiultra/ "wonderfully well written and reasoned information for the tube audio constructor" John Broskie TubeCAD & GlassWare "an unbelievably comprehensive web site containing vital gems of wisdom" Stuart Perry Hi-Fi News & Record Review |
#2
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() "Andre Jute" ( snip drivel from unfortunate namesake) Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. ** Dynamic phase shifting of audio signals is all around us, all the time. The fact that cones move, continuously alters the origin and hence time of arrival of any higher frequencies being simultaneously radiated. Phase shift in degrees ( at any point in time) is simply 360 x cone excursion / wavelength of the high frequency. Some call this effect " Doppler Distortion" - a misnomer. Dynamic phase shift in hi-fi amps and pre-amps is a myth - closely related to the old slew rate myth. Remember TIM and SID ??? Long dead now, poor fellows. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. ** Quad founder, Peter Walker, was widely quoted saying in relation to amplifiers: " If you don't like what you hear coming out, pay more attention to what is going in." In reality, it was a polite dig at the many and ongoing misdeeds of recording industry, which he held in deep contempt. The regular, live FM broadcasts of classical music by the BBC from venues in in London was his idea of reference signal quality. ....... Phil |
#3
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() Andre Jute wrote: Phil wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. First he needs to define what is occuring. I think he means phase modulation by the dynamics of the amplitude changes, so that during increase and decrease of signal amplitudes, the phase of a signal is tweaked to lag or lead during the amplitude change. Let's see what he really means to say. Patrick Turner. Andre Jute Visit Jute on Amps at http://members.lycos.co.uk/fiultra/ "wonderfully well written and reasoned information for the tube audio constructor" John Broskie TubeCAD & GlassWare "an unbelievably comprehensive web site containing vital gems of wisdom" Stuart Perry Hi-Fi News & Record Review |
#4
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Posted to rec.audio.tubes,rec.audio.opinion
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Andre Jute wrote:
Phil wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Andre Jute Visit Jute on Amps at http://members.lycos.co.uk/fiultra/ "wonderfully well written and reasoned information for the tube audio constructor" John Broskie TubeCAD & GlassWare "an unbelievably comprehensive web site containing vital gems of wisdom" Stuart Perry Hi-Fi News & Record Review Pretty much what Patrick said, although I need to reply in the original thread. What Matti Otala PROVED -- and Dr. Ottala was up there with Richard Heyser, not just a professor, but the Director of the Technical Research Center of Finland, and the guy who published many of the original papers on TIM (transient Intermodulation Distortion), so what he said should at least *intially* be taken seriously -- was that the amplitude distortions of the open loop are transformed into phase distortions of the closed loop, where the low frequency signals phase modulate the high frequency signals. I *think* this means that the high frequency signals move back and forward in time in the presense of low frequency signals, something that is actually quite difficult to measure, although all too easy to hear. It has some rather interesting implications about feedback, and the optimum configurations(?) of feedback (what's the word that refers to the various types of feedback electrical circuits?), which I wil hopefully get to sometime soon. In "The Audio Critic," Vol 2, #2, 1979, p 37, Peter Aczel said Otala's technical paper was to be delivered on Feb. 25, 1980, at the 65th convention of teh AES in London. I assume that the paper was published soon afterward in the JAES, but I'm not certain. Guess I should go by the UT library and have a look! (Hey, what's the use of living in the "liberal armpit" of Texas -- Austin -- if you don't make use of its assets? ;-) Phil |
#5
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() "Phil" Andre Jute wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Pretty much what Patrick said, although I need to reply in the original thread. What Matti Otala PROVED -- ** Otala never proved one, single commercial hi-fi amp suffered from TIM in a way that was audible. Many others have proved conclusively that TIM ( ad his cousin SID) is a furphy. and Dr. Ottala was up there with Richard Heyser, not just a professor, but the Director of the Technical Research Center of Finland, and the guy who published many of the original papers on TIM (transient Intermodulation Distortion), so what he said should at least *intially* be taken seriously ** It was - then got utterly debunked by others in the field wordwide. The debunking unfortunately did get the NOT the same publicity as Otala's hypothesis. So, ignorant ****WITS like you never heard about it. -- was that the amplitude distortions of the open loop are transformed into phase distortions of the closed loop, where the low frequency signals phase modulate the high frequency signals. I *think* this means that the high frequency signals move back and forward in time in the presense of low frequency signals, something that is actually quite difficult to measure, although all too easy to hear. ** Such and effect would be extremely easy to measure It just don't exist when musical programme signals are being reproduced. It has some rather interesting implications about feedback, ** ******** it does. In "The Audio Critic," Vol 2, #2, 1979, p 37, Peter Aczel said Otala's technical paper was to be delivered on Feb. 25, 1980, ** That was a very long time ago - ****WIT. ........ Phil |
#6
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Posted to rec.audio.tubes,rec.audio.opinion
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Phil Allison wrote:
"Phil" Andre Jute wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Pretty much what Patrick said, although I need to reply in the original thread. What Matti Otala PROVED -- ** Otala never proved one, single commercial hi-fi amp suffered from TIM in a way that was audible. Many others have proved conclusively that TIM ( ad his cousin SID) is a furphy. Could you name an article showing that slewing induced distortions do not exist? I can use an op-amp with 1 V per millisecond maximum slew rate with no problems with signals at 10 V per millisecond? Surely someone as intelligent as yourself (although it pains me to admit it) wouldn't use that old, tired, debating trick of throwing out a "general criticism without any supporting examples." and Dr. Ottala was up there with Richard Heyser, not just a professor, but the Director of the Technical Research Center of Finland, and the guy who published many of the original papers on TIM (transient Intermodulation Distortion), so what he said should at least *intially* be taken seriously ** It was - then got utterly debunked by others in the field wordwide. The debunking unfortunately did get the NOT the same publicity as Otala's hypothesis. So, ignorant ****WITS like you never heard about it. -- was that the amplitude distortions of the open loop are transformed into phase distortions of the closed loop, where the low frequency signals phase modulate the high frequency signals. I *think* this means that the high frequency signals move back and forward in time in the presense of low frequency signals, something that is actually quite difficult to measure, although all too easy to hear. ** Such and effect would be extremely easy to measure Here I am genuinely curious; how would you measure it? Phil It just don't exist when musical programme signals are being reproduced. It has some rather interesting implications about feedback, ** ******** it does. In "The Audio Critic," Vol 2, #2, 1979, p 37, Peter Aczel said Otala's technical paper was to be delivered on Feb. 25, 1980, ** That was a very long time ago - ****WIT. ....... Phil |
#7
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() Phil wrote: Phil Allison wrote: "Phil" Andre Jute wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Pretty much what Patrick said, although I need to reply in the original thread. What Matti Otala PROVED -- ** Otala never proved one, single commercial hi-fi amp suffered from TIM in a way that was audible. Many others have proved conclusively that TIM ( ad his cousin SID) is a furphy. Could you name an article showing that slewing induced distortions do not exist? I can use an op-amp with 1 V per millisecond maximum slew rate with no problems with signals at 10 V per millisecond? Eh ? Surely someone as intelligent as yourself (although it pains me to admit it) wouldn't use that old, tired, debating trick of throwing out a "general criticism without any supporting examples." Firstly your slew rate figures are surely V/us ( microsecond ) For any sinewave ( see Fourier theory for applicability ) the max slew rate ( at zero crossing btw ) is 2.pi.f.Vpeak. For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Graham |
#8
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Posted to rec.audio.tubes,rec.audio.opinion
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Eeyore wrote:
Phil wrote: Phil Allison wrote: "Phil" Andre Jute wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Pretty much what Patrick said, although I need to reply in the original thread. What Matti Otala PROVED -- ** Otala never proved one, single commercial hi-fi amp suffered from TIM in a way that was audible. Many others have proved conclusively that TIM ( ad his cousin SID) is a furphy. Could you name an article showing that slewing induced distortions do not exist? I can use an op-amp with 1 V per millisecond maximum slew rate with no problems with signals at 10 V per millisecond? Eh ? Surely someone as intelligent as yourself (although it pains me to admit it) wouldn't use that old, tired, debating trick of throwing out a "general criticism without any supporting examples." Firstly your slew rate figures are surely V/us ( microsecond ) No, I'm making a point that, contrary to what Phil said, slew induced distortion can indeed be a problem. For any sinewave ( see Fourier theory for applicability ) the max slew rate ( at zero crossing btw ) is 2.pi.f.Vpeak. For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Graham Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, and not all power amps, with their big, slow, output transistors, are going to be as fast as even a 741. Plus, the point of Matti's work is that problems begin to appear at all levels below the theoretical "breakthrough" point of TIM/SID. In any case, the topic here is not whether most amps have sufficient slew rate -- I assume that *most* good amps do -- but rather about Otala's proof that a feedback amp's "correction" of an amplitude distortion of the open loop phase shifts the high frequency components in the closed loop. I am currently discussing this in the "Negative Feedback in Triodes: The Logical and Experimental Proof" thread from 8/15, so if you're interested, look there (articles posted on 9/6). Phil Allison had a "response" here -- his usual slams and blams with no supporting evidence -- but I actually would like to see his simple test that can show whether low frequency signals in a feedback amp do or do not cause high frequency phase shifting, as it would be useful test, and I'm having a hard time coming up with a simple way to test that myself. Apparently, Otala incorporated a lot of ideas/solutions into his Citation XX power amp, and maybe, if I can find papers by him on that amp, there will be some useful information and tests there, but if PA can come up with something in the meantime, hell that's fine by me! He'll probably think of something really simple and easy, and then refuse to tell me, the ****head ... Anyway, what PA was saying/yelling is that feedback amps DO NOT EITHER CAUSE PHASE SHIFTING OF THE HIGH FREQUENCIES LIKE THAT DUMMY DR. OTALA SAID! My response was simply to ask whether (1) he knew of references that would back up his claim, that Otala's analysis was flawed, and (2) whether he knew of a good, simple test that can be used to test whether LF signals in a feedback amp cause phase shifting of the HF signals, like Otala said they do. Phil To email me directly, cut off my head |
#9
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() Phil wrote: Eeyore wrote: For any sinewave ( see Fourier theory for applicability ) the max slew rate ( at zero crossing btw ) is 2.pi.f.Vpeak. For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Graham Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, Utter drivel. Learn some science before posting such ********. bye bye. Graham |
#10
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Posted to rec.audio.tubes,rec.audio.opinion
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Eeyore wrote:
Phil wrote: Eeyore wrote: For any sinewave ( see Fourier theory for applicability ) the max slew rate ( at zero crossing btw ) is 2.pi.f.Vpeak. For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Graham Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, Utter drivel. Learn some science before posting such ********. bye bye. Graham I'll tell you what, why don't you take one point that you can attack, ignore everything else, and then put out yet another unsupported criticism. In audio, where you have a huge mixture of signals, the waveforms periodically add together, momentarily increasing the slew rate over what a typical -- read, very small amplitude -- audio 20 KHz signal would have. Do you disagree with that? Would you like to state here, for the public record, that you think that when several signals are mixed, that there are no momentary peaks in slew rate which exceed the maximum slew rate found in individual signals? By all means, show everyone here how well you think, and reason, and how much you actually understand about audio. Oh, BY THE WAY, do you have anything intelligent and useful to say about the rest of my comments? Useless cheap shots do not qualify ... Phil |
#11
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() "Phil" ** Otala never proved one, single commercial hi-fi amp suffered from TIM in a way that was audible. Many others have proved conclusively that TIM ( and his cousin SID) is a furphy. Could you name an article showing that slewing induced distortions do not exist? ** Try to comprehend what you read - you ****ing MORON. ........ Phil |
#12
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Posted to rec.audio.tubes,rec.audio.opinion
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Well Andre, it looks like it's you and me, since everyone else has
either pussied out on me (Phil Asshole, Graham), dealt with other issues, although probably honestly (Scott), or is waiting for further developments (Patrick). I went by the UT library today, and looked through several years of JAES, 1980 onward, but only found one article by Matti Otala. I think maybe he published some papers in IEEE, I'll have to check. However, in the Jan. 1980 JAES issue, there is an article by Vanderkooy and Lip****z called "Feedforward Error Correction in Power Amplifiers" that looks *very* interesting! I haven't thoroughly looked at it yet, but they review all the various types of feedforward schemes, starting with the one invented by Harold Black in 1923! They also review the Quad 405, which is a different type of feedforward, which I can state from personal experience sounds pretty bad compared to any decent tube amp. My interest in this is due to my initial conclusion from Otala's paper, namely that a method of feedback -- which we have to have if using solid state devices to obtain low output impedance-- that is quite different from the usual feedback scheme, would avoid the dynamic phase shifting problem. I had worked out an obvious overall topology, which turned out to be almost exactly the same as Black's feedforward scheme of 1923! Whoa ... Briefly, the problem as I see it, is that when a standard feedback amp with, say, 40 dB of feedback is initially hit with a step signal, the initial output is 100 times too large as seen by the feedback circuit. Note that even if the gain is constant, and the load is a perfect resistor of just the right size to give exactly 100 x open loop, the feedback loop sees an ERROR of 99%. This, I suspect, is what kills high feedback amplifiers. It would be one thing if only the minor errors due to slight gain variations in the transistors and impedance variations in the load were converted in to phase distortion, but as far as I can see, the feedback error loop cannot possibly distinguish between those minor errors, and the 100 x gain error that is inherent to the loop! In other words, the excess gain that is used to obtain 40 dB of feedback is itself seen as an error, and converted into phase distortion, just like the "real" errors in the load and output devices. Of course, this error only exists when a signal is present, but every perfect sine wave signal must be 99% corrected, and since this "correction" is actually a *conversion* into phase distortion, we wind up with the horrid sound that high feedback SS amps so often have. The alternative, which is basically Black's original feedforward amp design, is to have one amp that is basically just a transconductance amp, x amps out per y volts in, and the output voltage from this amp is then compared to the input to produce an error signal. This signal is then sent to a second, parallel amp, whose output is then added to the first amp's output. Ideally, you have two time delay circuits to offset the delay through the two amps, and it might be possible to use a parallel resistor in the output to supply some of the damping, but these are mostly details. The main point is that almost all of the "correction" signal would in fact be due to load variations (the second amp's gain is adjusted to match the first amp's gain), and *not* from the excess gain of the feedback amp, thereby greatly reducing the amount of dynamic phase distortion. Furthermore, since this feedback signal can "concentrate on the true errors" (whereas a normal feedback amp "concentrates" 98% to 99% on the excess gain, and only 1% to 2% on the true errors), it may be much more effective at counteracting the errors from the inherent crappy SS capacitors that come with any SS device, and also the SS thermal variations, which are much greater than the thermal variations of tubes. These capacitance and thermal defects are, I suspect, the other reason (in addition to feedback phase distortion) why power SS amps seem to be worse at amplifying a mixture of high and low amplitude signals -- i.e., music -- than tubes. These errors "mess up" the low level signals, causing SS amps to have less life and air than tube amps. Finally, the "error amp" in Black's feedforward design, which sends a signal to the second parallel amp, can be a vacuum tube, and since this *is* able to amplify high and low signals, it can more easily correct the destruction of the low level signals by the main amps! In other words, we can use a vacuum tube to insure quality amplification of the entire music signal, high and low level, and use this to correct the inability of power SS amps to do the same. In a normal feedback amp this would be largely impossible, since the typical level of 26 dB of feedback constantly "corrects" 95% of the output, converting this "error" (which is just the normal open loop gain) into phase distortion, something which normally would overwhelm the ability of a tube to restore low level information. Of course, I don't know if this will actually work, but in theory, it sounds VERY promising! Now, if we can just convince a few people to try my improved double-blind test (which mimics the actions of the Boulder amp people, only at rather higher speed) to select those components that really are capable of greater resolution, and then "dumb-down" the results until it is musical, we may have a truly musical, relatively inexpensive, almost all SS amp! If we could just get some low output impedance (2 ohms) power JFETS to go with it ... Phil Andre Jute wrote: Phil wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Andre Jute Visit Jute on Amps at http://members.lycos.co.uk/fiultra/ "wonderfully well written and reasoned information for the tube audio constructor" John Broskie TubeCAD & GlassWare "an unbelievably comprehensive web site containing vital gems of wisdom" Stuart Perry Hi-Fi News & Record Review |
#13
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() Phil wrote: Eeyore wrote: Phil wrote: Eeyore wrote: For any sinewave ( see Fourier theory for applicability ) the max slew rate ( at zero crossing btw ) is 2.pi.f.Vpeak. For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Graham Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, Utter drivel. Learn some science before posting such ********. bye bye. Graham I'll tell you what, why don't you take one point that you can attack, ignore everything else, and then put out yet another unsupported criticism. In audio, where you have a huge mixture of signals, the waveforms periodically add together, momentarily increasing the slew rate over what a typical -- read, very small amplitude -- audio 20 KHz signal would have. Do you disagree with that? Yes. It's utter ********. A 4V pk-pk signal is not small amplitude either btw. It's typical of pro-audio 'line level' which is why I chose it. Here's a 'big' signal. 15V peak. That's nearly 2V/us @ 20kHz. That's why you shouldn't use 741s for audio btw. Would you like to state here, for the public record, that you think that when several signals are mixed, that there are no momentary peaks in slew rate which exceed the maximum slew rate found in individual signals? There aren't. If you had a scientific education You'd understand. By all means, show everyone here how well you think, and reason, and how much you actually understand about audio. Oh, BY THE WAY, do you have anything intelligent and useful to say about the rest of my comments? Useless cheap shots do not qualify ... Learn some science before posting such ********. Graham |
#14
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Posted to rec.audio.tubes,rec.audio.opinion
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![]() Phil Allison wrote: "Phil" Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, ** The simplest test shows this is UTTERLY FALSE. Mr Fourier would concur with PA too. Graham |
#15
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Posted to rec.audio.tubes,rec.audio.opinion
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Phil said:
Well Andre, it looks like it's you and me, since everyone else has either pussied out on me (Phil Asshole, Graham), dealt with other issues, although probably honestly (Scott), or is waiting for further developments (Patrick). I went by the UT library today, and looked through several years of JAES, 1980 onward, but only found one article by Matti Otala. I think maybe he published some papers in IEEE, I'll have to check. However, in the Jan. 1980 JAES issue, there is an article by Vanderkooy and Lip****z called "Feedforward Error Correction in Power Amplifiers" that looks *very* interesting! I haven't thoroughly looked at it yet, but they review all the various types of feedforward schemes, starting with the one invented by Harold Black in 1923! They also review the Quad 405, which is a different type of feedforward, which I can state from personal experience sounds pretty bad compared to any decent tube amp. That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. Others after him corrected and modified his findings. There are more ways that lead to Rome. BTW a modified 405-II can sound very good, at least to these ears. -- "Due knot trussed yore spell chequer two fined awl miss steaks." |
#16
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![]() Phil wrote: Eeyore wrote: Phil wrote: Phil Allison wrote: "Phil" Andre Jute wrote: We're not talking about "-30 degrees at 20 KHz," we're talking *dynamic* phase shifting, the kind that makes a Crown preamp bite your ears off, while testing at 0.0001% THD. Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. I'm not overimpressed with vanishign THD but this is an amazing explanation for why so many silicon amps, and not a few tube amps, sound like ****. Pretty much what Patrick said, although I need to reply in the original thread. What Matti Otala PROVED -- ** Otala never proved one, single commercial hi-fi amp suffered from TIM in a way that was audible. Many others have proved conclusively that TIM ( ad his cousin SID) is a furphy. Could you name an article showing that slewing induced distortions do not exist? I can use an op-amp with 1 V per millisecond maximum slew rate with no problems with signals at 10 V per millisecond? Eh ? Surely someone as intelligent as yourself (although it pains me to admit it) wouldn't use that old, tired, debating trick of throwing out a "general criticism without any supporting examples." Firstly your slew rate figures are surely V/us ( microsecond ) No, I'm making a point that, contrary to what Phil said, slew induced distortion can indeed be a problem. For any sinewave ( see Fourier theory for applicability ) the max slew rate ( at zero crossing btw ) is 2.pi.f.Vpeak. For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Graham Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, and not all power amps, with their big, slow, output transistors, are going to be as fast as even a 741. Not all output bjts are big and slow. Some do however dislike turning OFF quickly and some display the truly horrible habit of cross conduction at HF, ie, the two bjts in a typical complementary pair are BOTH turned on during a wave cycle during large signal excursions at above 10kHz, and the power supply has to supply a lot more current that is simply passing from rail to rail and its hang onto your hat time for the ride. Plus, the point of Matti's work is that problems begin to appear at all levels below the theoretical "breakthrough" point of TIM/SID. In any case, the topic here is not whether most amps have sufficient slew rate -- I assume that *most* good amps do -- but rather about Otala's proof that a feedback amp's "correction" of an amplitude distortion of the open loop phase shifts the high frequency components in the closed loop. Both amplitude distortions and phase distortions of the open loop response are BOTH corrected by the NFB. Typical open loop phase lag in open loop at 20kHz is 90 degrees, and the 40dB of applied global NFB at 20kHz reduces this typically to less than 5 degrees. I am currently discussing this in the "Negative Feedback in Triodes: The Logical and Experimental Proof" thread from 8/15, so if you're interested, look there (articles posted on 9/6). Phil Allison had a "response" here -- his usual slams and blams with no supporting evidence -- but I actually would like to see his simple test that can show whether low frequency signals in a feedback amp do or do not cause high frequency phase shifting, as it would be useful test, and I'm having a hard time coming up with a simple way to test that myself. Just apply 70Hz and 5kHz signals to the input of an amp in a 4:1 ratio. Filter out all below 1kHz from the output signal. Then you will see what the effect of the 70Hz large signal is upon the fidelity of the 5kHz signal and whether there is any phase modulation in addition to the expected intermodulation. With most well made SS high NFB amps, the IMD is not visible on the CRO and a careful peak detector must be used to measure amplitude variations in the 5kHz, or else filter out the IMD products at 4,930Hz and 5,070Hz. Apparently, Otala incorporated a lot of ideas/solutions into his Citation XX power amp, and maybe, if I can find papers by him on that amp, there will be some useful information and tests there, but if PA can come up with something in the meantime, hell that's fine by me! He'll probably think of something really simple and easy, and then refuse to tell me, the ****head ... But all these investigations have been done many times before. What exactly do you hope to gain by goading the ungoadables on the group to find out what you should be willing to find out for yourself? Do you suspect to find some hitherto unused uninvented techniques of making amplifiers perform better? Anyway, what PA was saying/yelling is that feedback amps DO NOT EITHER CAUSE PHASE SHIFTING OF THE HIGH FREQUENCIES LIKE THAT DUMMY DR. OTALA SAID! My response was simply to ask whether (1) he knew of references that would back up his claim, that Otala's analysis was flawed, and (2) whether he knew of a good, simple test that can be used to test whether LF signals in a feedback amp cause phase shifting of the HF signals, like Otala said they do. Be like me, find out by building one's own test gear and testing. It took me months to do it all but after reading all the conflicting opinions about all this in Electronics World copies from the 1970s to 1980s BEFORE the internet was mainstream, I decided to look myself at what happened in amps that i should be worried about. Patrick Turner. Phil To email me directly, cut off my head |
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![]() "Sander de******" That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. ** Otala never published any actual "findings" whatever. His wacky and always controversial hypotheses briefly polluted the technical press in the mid and late 70s, actually. Reminds me a lot of the bizarre "Cold Fusion " fiasco. How embarrassment. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. ** Absolute ******** !!!!!!!!! Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. ** Never has - even when applied in time honoured, traditional ways. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. ** Highly doubtful that a room temp cretin like * YOU* thinks at all. Just reacts to its environment - like any slimy reptile. Others after him corrected and modified his findings. ** ****ed on them from a VERY GREAT height and buried them, basically. There are more ways that lead to Rome. ** Depends how completely LOST you are. BTW a modified 405-II can sound very good, at least to these ears. ** So will an unmodified one & and an original 405. Now - **** the HELL OFF Bloody Jerk OFF !!!!!!!!!! ........ Phil |
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![]() Phil wrote: Eeyore wrote: Phil wrote: Eeyore wrote: For any sinewave ( see Fourier theory for applicability ) the max slew rate ( at zero crossing btw ) is 2.pi.f.Vpeak. For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Graham Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, Utter drivel. Learn some science before posting such ********. bye bye. Graham I'll tell you what, why don't you take one point that you can attack, ignore everything else, and then put out yet another unsupported criticism. In audio, where you have a huge mixture of signals, the waveforms periodically add together, momentarily increasing the slew rate over what a typical -- read, very small amplitude -- audio 20 KHz signal would have. Do you disagree with that? Would you like to state here, for the public record, that you think that when several signals are mixed, that there are no momentary peaks in slew rate which exceed the maximum slew rate found in individual signals? By all means, show everyone here how well you think, and reason, and how much you actually understand about audio. Oh, BY THE WAY, do you have anything intelligent and useful to say about the rest of my comments? Useless cheap shots do not qualify ... Phil But the worst case additions of many different varying frequencies and amplitudes of a musical signal cannot give rise to a faster rise time than that of a full power sine wave signal at the frequency limit of the music bandwidth. This bandwidth is about 20 kHz these days, and whatever you do with other waves below this F the rise time will never be faster than that in a 20kHz full power sine wave. If anything happens at a faster rate, ie, the slope of the wave graph is steeper than that of a 20kHz wave then there are higher F present which are above 20kHz. If what you are suggesting is possible, then such manifestations of higher frequencies above 20kHz could easily be filtered out and proved to exist. Foe example if the amp produces 2H and 3H of say 20kHz, then indeed the 40hHz and 60kHz would become real, and the IMD product between say 8kHz and 18kHz of 26kHz would become real, providing the amp has the capacity to pass the higher F. Usually most amps can do this because such spuriae above 20kHz are low in level. Patrick Turner. |
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"Phil" wrote in message
Eeyore wrote: For a 20kHz signal of say 2V peak amplitude that means a slew rate of 0.25 V/us ! Even a rubbish 741 or 1458 can manage that ! Agreed. Given that modern audio op-amps are capable of slew rates of ~ 10V/us - you're never even remotely close to slew limited anything. Agreed. I was just looking at the specs for a modern op amp that I think TI or National is trying to popularize, and the SR was more like 100 V/uS Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, OK, let's totally lose our minds and say that it is 10 times as much. That makes 2.5 V/uS, which is still only 25% of the capabilities of ca.-1980s op amp. BTW here's a hint - the slew rate at the output of any digital player is limited to voltages along the lines of what Graham came up with in his calculations. Why we laugh when people "upgrade" the outputs of CD players by putting in op amps with rediculously high slew rates? ;-) and not all power amps, with their big, slow, output transistors, are going to be as fast as even a 741. Wrong again. It's hard to find a modern power amp that slews even just 10 times faster than a 741. Most do far better than that. Plus, the point of Matti's work is that problems begin to appear at all levels below the theoretical "breakthrough" point of TIM/SID. See my previous ludicrously tough example that included a 10:1 safety margin and was still hitting only 25% of the capabilities of 25-year-old technology. In any case, the topic here is not whether most amps have sufficient slew rate -- I assume that *most* good amps do -- but rather about Otala's proof that a feedback amp's "correction" of an amplitude distortion of the open loop phase shifts the high frequency components in the closed loop. Which is wrong. I am currently discussing this in the "Negative Feedback in Triodes: The Logical and Experimental Proof" thread from 8/15, so if you're interested, look there (articles posted on 9/6). I can't bear to look at that thread again Phil, because you took such a merciless licking from just about everybody. |
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"Phil Allison" said:
"Sander de******" How did you find out? ;-) That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. ** Otala never published any actual "findings" whatever. Look up some AES papers some time. His wacky and always controversial hypotheses briefly polluted the technical press in the mid and late 70s, actually. Reminds me a lot of the bizarre "Cold Fusion " fiasco. How embarrassment. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. ** Absolute ******** !!!!!!!!! We've learned *nothing* since the '70s? Surprising, and not according to my findings. Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. ** Never has - even when applied in time honoured, traditional ways. Many seem to disagree, and not only simple techies like me. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. ** Highly doubtful that a room temp cretin like * YOU* thinks at all. Just reacts to its environment - like any slimy reptile. Others after him corrected and modified his findings. ** ****ed on them from a VERY GREAT height and buried them, basically. In PA-speak, probably. There are more ways that lead to Rome. ** Depends how completely LOST you are. BTW a modified 405-II can sound very good, at least to these ears. ** So will an unmodified one & and an original 405. Now - **** the HELL OFF Bloody Jerk OFF !!!!!!!!!! ....... Phil Are you feeling well, Phil? Your tone isn't that harsh and shrill as usual. -- "Due knot trussed yore spell chequer two fined awl miss steaks." |
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![]() "Sander de******" ** Highly doubtful that a room temp cretin like * YOU* thinks at all. Just reacts to its environment , slowly - like any SLIMY reptile. Now - **** the HELL OFF YOU ****wit Audiophool JERK !!!!!!!!! ....... Phil |
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![]() Sander deWaal wrote: Phil said: Well Andre, it looks like it's you and me, since everyone else has either pussied out on me (Phil Asshole, Graham), dealt with other issues, although probably honestly (Scott), or is waiting for further developments (Patrick). I went by the UT library today, and looked through several years of JAES, 1980 onward, but only found one article by Matti Otala. I think maybe he published some papers in IEEE, I'll have to check. However, in the Jan. 1980 JAES issue, there is an article by Vanderkooy and Lip****z called "Feedforward Error Correction in Power Amplifiers" that looks *very* interesting! I haven't thoroughly looked at it yet, but they review all the various types of feedforward schemes, starting with the one invented by Harold Black in 1923! They also review the Quad 405, which is a different type of feedforward, which I can state from personal experience sounds pretty bad compared to any decent tube amp. That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. The only time large amounts of additional distortion harmonic or intermodulation harmonic products are generated in a FB amp are when the 1, The open loop THD and IMD is high, perhaps 10% and before clipping is occuring, 2, the open loop bandwidth is poor, allowing higher F distortion products to not be reduced by the FB because applied FB is dependant on open loop gain and phase shift. 3, the open loop phase shift is poor. 4, the amount of NFB is small, tyically less than 14dB. There have been several articles in Wireless World and Electrincs World where the authors have explored the phenomena of applied NFB around a rather non linear amp. Where say 3H is abundant in the open loop thd, when mild NFB is used there is 2H and 4H generated by IMD process because the 3H fed back is modulated by the fundemental to make the sum and difference IMD products of 2H and 4H. However, these additional distortion products are low in level, but are reduced if enough NFB is applied and the bandwidth phase shift of the open loop permits stability. So we have SS amps with typically 60 dB of applied global NFB at say 500Hz which is where the maximumm open loop gain is. the open loop -3dB point is at say 1kHz, and at 10kHz the OLG is -20dB, so that only 40 dB of applied NFB is used at 10kHz but its enough to reduce measured distortions of all kinds to triflingly small levels. The problem eith many old amps is that where you did try to test a full power signal at 20kHz or above, the input stages had to perform serious electronic gymnastics to force the output stage to produce a level response, ie, the input stages saturated, and the sine wave at HF became a triangular wave, and limiting commenced before full power and mid frequency clipping levels were reached. Some tube amps are not imune to such ****e happening at HF. SE amps using pentodes do have open loop problems of high open loop THD and IMD. When used with only 10dB of global NFB the outcome is sonically not much better than with no NFB although at least the Rout has been reduced to about the same as a triode without NFB. So with pentodes, one must use a lot of NFB to get anywhere good at least as measurements go, and still a large spray of extra harmonics are made and remain but at a low level when 20dB of NFB is used. SE triodes when used without expecting too much average power, ie, less than 10% of the clipping power will sound better than the pentode with mild NFB and the same max power capability. That's because the triode manages lower Rout than the pentode with too little NFB and the triode THD/IMD without loop NFB is less than the pentodes with mild NFB. Whether the class A SE pentode with 20 dB loop NFB is better sounding than the SE triode without NFB is an often bitterly argued topic, but I have to say I prefer the sound of a 2A3 with a paltry 6dB of NFB compared to a 6V6 or EL84 with same 4 watt maxima and 20 dB of NFB. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. Others after him corrected and modified his findings. There are more ways that lead to Rome. BTW a modified 405-II can sound very good, at least to these ears. There is no accounting for taste. Patrick Turner. -- "Due knot trussed yore spell chequer two fined awl miss steaks." |
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"Phil Allison" said:
"Sander de******" ** Highly doubtful that a room temp cretin like * YOU* thinks at all. Just reacts to its environment , slowly - like any SLIMY reptile. Now - **** the HELL OFF YOU ****wit Audiophool JERK !!!!!!!!! ...... Phil Thanks Phil, for a moment there I was afraid that you were not well. This post is just what I expected in the first place, thanks for reassuring me that all is normal again! How are your Quads doing these days BTW? -- "Due knot trussed yore spell chequer two fined awl miss steaks." |
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![]() Phil Allison wrote: "Andre Jute" Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. ** Dynamic phase shifting of audio signals is all around us, all the time. The fact that cones move, continuously alters the origin and hence time of arrival of any higher frequencies being simultaneously radiated. Phase shift in degrees ( at any point in time) is simply 360 x cone excursion / wavelength of the high frequency. Some call this effect " Doppler Distortion" - a misnomer. Hi Phil, Here's my take on this interesting topic. I say that this is precisely an acoustical frequency modulator. If you input two sinusoids, one low and one high, then the spectrum of the upper one will be spread out about its center. And the greater the amplitude of the bass signal, the greater the modulation index. From the modulation index one could predict what the side bands will look like. I found the term "Doppler Distortion" helpful. The situation here is not exactly like the sound of the horn of a train passing a station. Rather, it is the sound of the horn of a crazy train oscillating back and forth across the station. Joe |
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![]() "Joseph Meditz" wrote in message ups.com... ** Groper Alert ! ** Dynamic phase shifting of audio signals is all around us, all the time. The fact that cones move, continuously alters the origin and hence time of arrival of any higher frequencies being simultaneously radiated. Phase shift in degrees ( at any point in time) is simply 360 x cone excursion / wavelength of the high frequency. Some call this effect " Doppler Distortion" - a misnomer. I say that this is precisely an acoustical frequency modulator ** It is an acoustic phase modulator. I found the term "Doppler Distortion" helpful. The situation here is not exactly like the sound of the horn of a train passing a station. Rather, it is the sound of the horn of a crazy train oscillating back and forth across the station. ** Nope. .......... Phil |
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![]() "Sander deWog******" **** the HELL OFF YOU ****wit Audiophool JERK !!!!!!!!! ....... Phil |
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On Fri, 8 Sep 2006 10:43:58 +1000, "Phil Allison"
wrote: "Sander deWog******" **** the HELL OFF YOU ****wit Audiophool JERK !!!!!!!!! ...... Phil Gee, Phil look at you---multi-tasking! And I thought you could only be disgusting in one place at a time. |
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Phil wrote:
Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, Yes and no, Phil. Take 20kHz at full amplitude to define the required slew rate. Add another identical signal, in phase. You now have twice the slew rate, as you think. But the signal is also twice full amplitude, so it is not comparable. To make it comparable, you must reduce it to full amplitude. In so doing, you halve the slew rate, returning it to its original value. Hence adding these signals together doesn't alter the slew rate, as long as the total signal remains within the defined full amplitude. Does this logic hold for the sum of a full amplitude 20kHz and some other, lower frequency? Intuitively yes, to me. A bit of simple trig would confirm. What about smaller signals? Well, they will never have a higher slew rate than the 20kHz at full amplitude, surely? Thanks for your input, Phil cheers, Ian |
#29
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"Ian Iveson" wrote in
message k Phil wrote: Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, Yes and no, Phil. Take 20kHz at full amplitude to define the required slew rate. Add another identical signal, in phase. You now have twice the slew rate, as you think. But the signal is also twice full amplitude, so it is not comparable. To make it comparable, you must reduce it to full amplitude. In so doing, you halve the slew rate, returning it to its original value. Hence adding these signals together doesn't alter the slew rate, as long as the total signal remains within the defined full amplitude. Agreed. Does this logic hold for the sum of a full amplitude 20kHz and some other, lower frequency? Intuitively yes, to me. A bit of simple trig would confirm. The math supports your intuition. What about smaller signals? Well, they will never have a higher slew rate than the 20kHz at full amplitude, surely? As long as they are band-limited to 20 KHz. The agenda that seems to be hidden from Phil relates to the vast improvement in the bandwidth of power transistors over the years. In the 60s and early, large power devices used in power amps usually ran out of gas below 1 MHz. Today for about the last 20 years, parts that beat that by a factor of 10 or more are plentiful and inexpensive. |
#30
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Sander deWaal wrote:
Phil said: Well Andre, it looks like it's you and me, since everyone else has either pussied out on me (Phil Asshole, Graham), dealt with other issues, although probably honestly (Scott), or is waiting for further developments (Patrick). I went by the UT library today, and looked through several years of JAES, 1980 onward, but only found one article by Matti Otala. I think maybe he published some papers in IEEE, I'll have to check. However, in the Jan. 1980 JAES issue, there is an article by Vanderkooy and Lip****z called "Feedforward Error Correction in Power Amplifiers" that looks *very* interesting! I haven't thoroughly looked at it yet, but they review all the various types of feedforward schemes, starting with the one invented by Harold Black in 1923! They also review the Quad 405, which is a different type of feedforward, which I can state from personal experience sounds pretty bad compared to any decent tube amp. That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. Others after him corrected and modified his findings. There are more ways that lead to Rome. BTW a modified 405-II can sound very good, at least to these ears. The Quad I heard was not modified, so I couldn't say, although the one I heard sounded very good unless compared to a good PP tube amp. I want to emphasize that the original subject here is not what Phil Allison implies, namely TIM or SID, but rather the question of whether negative feedback causes audible problems even when there is no TIM. I'll go ahead and quote the review of Otala's paper I gave in the other thread, from "The Audio Critic," Vol 2, #2, p 37, regarding Matti Otala's analysis of feedback (made after he, the editor Peter Aczel, Mitch Cotter, Stew Hegeman, Andy Rappaport, Max Wilcox, and Bruce Zayde had a "BS" session in TAC); "The paper presents rigorous mathematical proof, for the most generalized, all-inclusive case, that feedback cannot make amplifier distortions go away; all it can do is to change one kind of distortion into another. By the application of feedback, the amplitude nonlinearities of the open loop are converted into phase nonlinearities of the closed loop. That's all. The garbage cannot, by definition, be made to disappear; it's simply swept into another corner. In the typical feedback amplifier, the amplitude of the audio signal phase-modulates the high-frequency components of the signal. Furthermore, any amplitude intermodulation distortion in the open loop is converted into phase intermodulation distortion in the closed loop. What about TIM, alias SID? It turns out that it (he?) is a limit case of this feedback-generated phase modulation effect, with all shades of gray possible before the actual black eruption occurs. None of this shows up on standard tests." I still can't find this paper, despite several trips to the UT library, but a little thought shows that it actually is consistent with much of what you and the others ae saying. When an amp with, say, 40 dB of feedback is hit with a step, the output initially has an "error" of 100 x, *independent* of any gain or load non-linearities, which must be "corrected" by the feedback loop. For every single change in the input voltage, the gain is off by a factor of 95 to 105, depending on gain and load non-linearities, and this error must be corrected by the feedback loop. Intuitively, it seems obvious that Otala's proof must in *some* way be correct, that this constant "correction" must play havoc with low level and high frequency signals. I don't think anyone would deny that, given an amp with variable feedback followed by a pot to equalize the overall gain, turning up the feedback will eventually make an amp that, like the Crown preamp, will "bite your ears off," even if the amp never gets into TIM territory or other obvious problems. The question is how much of an effect does Otala's "dynamic phase shifting" have. Here again, it seems obvious that part of the problem was the S-L-O-W power transistors of the late '70's, when Otala's various articles were written. I suspect that high speed devices reduce the problems created by feedback, the amount of phase distortion produced, and of course Otala himself came up with several ideas to reduce these effects in his Citation XX design, although I also haven't been able to find any literature on that design. Nevertheless, it is a given, in my mind, that a very high open loop gain, with its need to constantly "correct" every input signal by 99% (in the case of 40 dB feedback), *regardless* of the inherent linearity of the amp's devices and circuit, MUST cause problems for signals 60 dB to 80 dB below the main signal, and perhaps also phase shift the high frequency components, as Aczel's summary of Otala's paper states, thereby robbing the circuit of much of its "life" and "air," the criticisms one normally hears about high feedback amps, and also solid state amps, in which the solid state capacitances and high thermal variations also interfere with low level signals. This will not show up as TIM or SID, unless the amp has been very poorly designed, and I'm still not sure how one would measure it. My best guess has been to use a 20 Hz signal and a much smaller (-60 to -80 dB) 10 KHz signal, filter out the 20 Hz signal from the output with a notch filter plus high pass filter, and either look directly at the 10 KHz signal for signs of distress, or filter it out with another notch filter, and see if phase shifting causes "sidebands" to appear and disappear when the 20 Hz signal is put in and out of the test. Assuming that normal feedback causes problems -- and as Patrick says, with low feedback and tubes it isn't too bad, but SS amps have more problems and generally need more feedback -- it would be nice if we could figure out a way to tremendously reduce the need for feedback to "correct" every normal signal by 99% even when there is no device distortion, meaning allow the feedback to "focus" *only* on actual device and load non-linearities. Here is where Black's "feedforward" circuit may allow for a real advance in SS amps, especially if tubes, with their (generally) superior ability to handle a mix of high and low level signals without messing up the low level information, are used to provide the error signal. Properly applied, Black's feedforward scheme (but not the feedforward designs by many others!) does exactly this, it allows feedback to appear and affect the signal *only* when actual deviations caused by device or load non-linearities appear. It may even be possible to correct the effects of a typical transistor's parallel capacitances -- which, being made of silicon, are of *very* poor quality -- and thermal variations! Normally, this would be a "why waste the time, just use tubes," situation, but good output transformers are heavy, big, and expensive, and if the amps used in compact disk players, as well as TV's, could be considerably improved, that would be nice! And of course, inexpensive amps that sound very good are always in demand. Phil |
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![]() Phil wrote: Sander deWaal wrote: Phil said: Well Andre, it looks like it's you and me, since everyone else has either pussied out on me (Phil Asshole, Graham), dealt with other issues, although probably honestly (Scott), or is waiting for further developments (Patrick). I went by the UT library today, and looked through several years of JAES, 1980 onward, but only found one article by Matti Otala. I think maybe he published some papers in IEEE, I'll have to check. However, in the Jan. 1980 JAES issue, there is an article by Vanderkooy and Lip****z called "Feedforward Error Correction in Power Amplifiers" that looks *very* interesting! I haven't thoroughly looked at it yet, but they review all the various types of feedforward schemes, starting with the one invented by Harold Black in 1923! They also review the Quad 405, which is a different type of feedforward, which I can state from personal experience sounds pretty bad compared to any decent tube amp. That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. Others after him corrected and modified his findings. There are more ways that lead to Rome. BTW a modified 405-II can sound very good, at least to these ears. The Quad I heard was not modified, so I couldn't say, although the one I heard sounded very good unless compared to a good PP tube amp. I want to emphasize that the original subject here is not what Phil Allison implies, namely TIM or SID, but rather the question of whether negative feedback causes audible problems even when there is no TIM. I'll go ahead and quote the review of Otala's paper I gave in the other thread, from "The Audio Critic," Vol 2, #2, p 37, regarding Matti Otala's analysis of feedback (made after he, the editor Peter Aczel, Mitch Cotter, Stew Hegeman, Andy Rappaport, Max Wilcox, and Bruce Zayde had a "BS" session in TAC); "The paper presents rigorous mathematical proof, for the most generalized, all-inclusive case, that feedback cannot make amplifier distortions go away; all it can do is to change one kind of distortion into another. By the application of feedback, the amplitude nonlinearities of the open loop are converted into phase nonlinearities of the closed loop. That's all. The garbage cannot, by definition, be made to disappear; it's simply swept into another corner. In the typical feedback amplifier, the amplitude of the audio signal phase-modulates the high-frequency components of the signal. Furthermore, any amplitude intermodulation distortion in the open loop is converted into phase intermodulation distortion in the closed loop. What about TIM, alias SID? It turns out that it (he?) is a limit case of this feedback-generated phase modulation effect, with all shades of gray possible before the actual black eruption occurs. None of this shows up on standard tests." I still can't find this paper, despite several trips to the UT library, but a little thought shows that it actually is consistent with much of what you and the others ae saying. When an amp with, say, 40 dB of feedback is hit with a step, the output initially has an "error" of 100 x, *independent* of any gain or load non-linearities, which must be "corrected" by the feedback loop. For every single change in the input voltage, the gain is off by a factor of 95 to 105, depending on gain and load non-linearities, and this error must be corrected by the feedback loop. Intuitively, it seems obvious that Otala's proof must in *some* way be correct, that this constant "correction" must play havoc with low level and high frequency signals. I don't think anyone would deny that, given an amp with variable feedback followed by a pot to equalize the overall gain, turning up the feedback will eventually make an amp that, like the Crown preamp, will "bite your ears off," even if the amp never gets into TIM territory or other obvious problems. The question is how much of an effect does Otala's "dynamic phase shifting" have. Here again, it seems obvious that part of the problem was the S-L-O-W power transistors of the late '70's, when Otala's various articles were written. I suspect that high speed devices reduce the problems created by feedback, the amount of phase distortion produced, and of course Otala himself came up with several ideas to reduce these effects in his Citation XX design, although I also haven't been able to find any literature on that design. Nevertheless, it is a given, in my mind, that a very high open loop gain, with its need to constantly "correct" every input signal by 99% (in the case of 40 dB feedback), *regardless* of the inherent linearity of the amp's devices and circuit, MUST cause problems for signals 60 dB to 80 dB below the main signal, and perhaps also phase shift the high frequency components, as Aczel's summary of Otala's paper states, thereby robbing the circuit of much of its "life" and "air," the criticisms one normally hears about high feedback amps, and also solid state amps, in which the solid state capacitances and high thermal variations also interfere with low level signals. This will not show up as TIM or SID, unless the amp has been very poorly designed, and I'm still not sure how one would measure it. My best guess has been to use a 20 Hz signal and a much smaller (-60 to -80 dB) 10 KHz signal, filter out the 20 Hz signal from the output with a notch filter plus high pass filter, and either look directly at the 10 KHz signal for signs of distress, or filter it out with another notch filter, and see if phase shifting causes "sidebands" to appear and disappear when the 20 Hz signal is put in and out of the test. Assuming that normal feedback causes problems -- and as Patrick says, with low feedback and tubes it isn't too bad, but SS amps have more problems and generally need more feedback -- it would be nice if we could figure out a way to tremendously reduce the need for feedback to "correct" every normal signal by 99% even when there is no device distortion, meaning allow the feedback to "focus" *only* on actual device and load non-linearities. Here is where Black's "feedforward" circuit may allow for a real advance in SS amps, especially if tubes, with their (generally) superior ability to handle a mix of high and low level signals without messing up the low level information, are used to provide the error signal. Properly applied, Black's feedforward scheme (but not the feedforward designs by many others!) does exactly this, it allows feedback to appear and affect the signal *only* when actual deviations caused by device or load non-linearities appear. It may even be possible to correct the effects of a typical transistor's parallel capacitances -- which, being made of silicon, are of *very* poor quality -- and thermal variations! Normally, this would be a "why waste the time, just use tubes," situation, but good output transformers are heavy, big, and expensive, and if the amps used in compact disk players, as well as TV's, could be considerably improved, that would be nice! And of course, inexpensive amps that sound very good are always in demand. You do talk a shocking amount of drivel ! Graham |
#32
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"Phil" wrote in message
Nevertheless, it is a given, in my mind, that a very high open loop gain, with its need to constantly "correct" every input signal by 99% (in the case of 40 dB feedback), *regardless* of the inherent linearity of the amp's devices and circuit, MUST cause problems for signals 60 dB to 80 dB below the main signal, and perhaps also phase shift the high frequency components, as Aczel's summary of Otala's paper states, thereby robbing the circuit of much of its "life" and "air," the criticisms one normally hears about high feedback amps, and also solid state amps, in which the solid state capacitances and high thermal variations also interfere with low level signals. Could you be more presumptious or wrong, Phil? This will not show up as TIM or SID, unless the amp has been very poorly designed, and I'm still not sure how one would measure it. There's really no way to measure your imagination, Phil. My best guess has been to use a 20 Hz signal and a much smaller (-60 to -80 dB) 10 KHz signal, filter out the 20 Hz signal from the output with a notch filter plus high pass filter, and either look directly at the 10 KHz signal for signs of distress, or filter it out with another notch filter, and see if phase shifting causes "sidebands" to appear and disappear when the 20 Hz signal is put in and out of the test. You're looking for the unholy Grail, Phil. People don't use notch filters that much any more. They just apply the test signal and analyze the amp's output with a very good spectrum analyzer. Assuming that normal feedback causes problems That takes a lot of ignorance or paranoia. -- and as Patrick says, with low feedback and tubes it isn't too bad, but SS amps have more problems and generally need more feedback SS amps don't have more problems, if well-designed. -- it would be nice if we could figure out a way to tremendously reduce the need for feedback to "correct" every normal signal by 99% even when there is no device distortion, meaning allow the feedback to "focus" *only* on actual device and load non-linearities. The error here is that there are any unavoidable problems with the application of lots of feedback. Here is where Black's "feedforward" circuit may allow for a real advance in SS amps, especially if tubes, with their (generally) superior ability to handle a mix of high and low level signals without messing up the low level information, are used to provide the error signal. In fact tubes have no such advantages. Properly applied, Black's feedforward scheme (but not the feedforward designs by many others!) does exactly this, it allows feedback to appear and affect the signal *only* when actual deviations caused by device or load non-linearities appear. This is nuts. It may even be possible to correct the effects of a typical transistor's parallel capacitances -- which, being made of silicon, are of *very* poor quality -- and thermal variations! Making really good power amps with silicon transistors is an old art that is quite well perfected at this time. Normally, this would be a "why waste the time, just use tubes," situation, but good output transformers are heavy, big, and expensive, and if the amps used in compact disk players, as well as TV's, could be considerably improved, that would be nice! And of course, inexpensive amps that sound very good are always in demand. Thats why so many of them are made and sold - there's lots of demand for them. |
#33
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"Eeyore"
wrote in message Phil wrote: Sander deWaal wrote: Phil said: Well Andre, it looks like it's you and me, since everyone else has either pussied out on me (Phil Asshole, Graham), dealt with other issues, although probably honestly (Scott), or is waiting for further developments (Patrick). I went by the UT library today, and looked through several years of JAES, 1980 onward, but only found one article by Matti Otala. I think maybe he published some papers in IEEE, I'll have to check. However, in the Jan. 1980 JAES issue, there is an article by Vanderkooy and Lip****z called "Feedforward Error Correction in Power Amplifiers" that looks *very* interesting! I haven't thoroughly looked at it yet, but they review all the various types of feedforward schemes, starting with the one invented by Harold Black in 1923! They also review the Quad 405, which is a different type of feedforward, which I can state from personal experience sounds pretty bad compared to any decent tube amp. That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. Others after him corrected and modified his findings. There are more ways that lead to Rome. BTW a modified 405-II can sound very good, at least to these ears. The Quad I heard was not modified, so I couldn't say, although the one I heard sounded very good unless compared to a good PP tube amp. I want to emphasize that the original subject here is not what Phil Allison implies, namely TIM or SID, but rather the question of whether negative feedback causes audible problems even when there is no TIM. I'll go ahead and quote the review of Otala's paper I gave in the other thread, from "The Audio Critic," Vol 2, #2, p 37, regarding Matti Otala's analysis of feedback (made after he, the editor Peter Aczel, Mitch Cotter, Stew Hegeman, Andy Rappaport, Max Wilcox, and Bruce Zayde had a "BS" session in TAC); "The paper presents rigorous mathematical proof, for the most generalized, all-inclusive case, that feedback cannot make amplifier distortions go away; all it can do is to change one kind of distortion into another. By the application of feedback, the amplitude nonlinearities of the open loop are converted into phase nonlinearities of the closed loop. That's all. The garbage cannot, by definition, be made to disappear; it's simply swept into another corner. In the typical feedback amplifier, the amplitude of the audio signal phase-modulates the high-frequency components of the signal. Furthermore, any amplitude intermodulation distortion in the open loop is converted into phase intermodulation distortion in the closed loop. What about TIM, alias SID? It turns out that it (he?) is a limit case of this feedback-generated phase modulation effect, with all shades of gray possible before the actual black eruption occurs. None of this shows up on standard tests." I still can't find this paper, despite several trips to the UT library, but a little thought shows that it actually is consistent with much of what you and the others ae saying. When an amp with, say, 40 dB of feedback is hit with a step, the output initially has an "error" of 100 x, *independent* of any gain or load non-linearities, which must be "corrected" by the feedback loop. For every single change in the input voltage, the gain is off by a factor of 95 to 105, depending on gain and load non-linearities, and this error must be corrected by the feedback loop. Intuitively, it seems obvious that Otala's proof must in *some* way be correct, that this constant "correction" must play havoc with low level and high frequency signals. I don't think anyone would deny that, given an amp with variable feedback followed by a pot to equalize the overall gain, turning up the feedback will eventually make an amp that, like the Crown preamp, will "bite your ears off," even if the amp never gets into TIM territory or other obvious problems. The question is how much of an effect does Otala's "dynamic phase shifting" have. Here again, it seems obvious that part of the problem was the S-L-O-W power transistors of the late '70's, when Otala's various articles were written. I suspect that high speed devices reduce the problems created by feedback, the amount of phase distortion produced, and of course Otala himself came up with several ideas to reduce these effects in his Citation XX design, although I also haven't been able to find any literature on that design. Nevertheless, it is a given, in my mind, that a very high open loop gain, with its need to constantly "correct" every input signal by 99% (in the case of 40 dB feedback), *regardless* of the inherent linearity of the amp's devices and circuit, MUST cause problems for signals 60 dB to 80 dB below the main signal, and perhaps also phase shift the high frequency components, as Aczel's summary of Otala's paper states, thereby robbing the circuit of much of its "life" and "air," the criticisms one normally hears about high feedback amps, and also solid state amps, in which the solid state capacitances and high thermal variations also interfere with low level signals. This will not show up as TIM or SID, unless the amp has been very poorly designed, and I'm still not sure how one would measure it. My best guess has been to use a 20 Hz signal and a much smaller (-60 to -80 dB) 10 KHz signal, filter out the 20 Hz signal from the output with a notch filter plus high pass filter, and either look directly at the 10 KHz signal for signs of distress, or filter it out with another notch filter, and see if phase shifting causes "sidebands" to appear and disappear when the 20 Hz signal is put in and out of the test. Assuming that normal feedback causes problems -- and as Patrick says, with low feedback and tubes it isn't too bad, but SS amps have more problems and generally need more feedback -- it would be nice if we could figure out a way to tremendously reduce the need for feedback to "correct" every normal signal by 99% even when there is no device distortion, meaning allow the feedback to "focus" *only* on actual device and load non-linearities. Here is where Black's "feedforward" circuit may allow for a real advance in SS amps, especially if tubes, with their (generally) superior ability to handle a mix of high and low level signals without messing up the low level information, are used to provide the error signal. Properly applied, Black's feedforward scheme (but not the feedforward designs by many others!) does exactly this, it allows feedback to appear and affect the signal *only* when actual deviations caused by device or load non-linearities appear. It may even be possible to correct the effects of a typical transistor's parallel capacitances -- which, being made of silicon, are of *very* poor quality -- and thermal variations! Normally, this would be a "why waste the time, just use tubes," situation, but good output transformers are heavy, big, and expensive, and if the amps used in compact disk players, as well as TV's, could be considerably improved, that would be nice! And of course, inexpensive amps that sound very good are always in demand. You do talk a shocking amount of drivel ! I think "Phil" is just Moncreiff or Jung posting under an alias. ;-) |
#34
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![]() Another dose of Krooglish gets Mr. **** girded for his weekend joust with Rev. Poop-Head at the Goose Puke Baptist church. Could you be more presumptious Nobody in the history of the world has ever been "presumptious", you dumb ****. Stop lying, please. -- "Christians have to ... work to make the world as loving, just, and supportive as is possible." A. Krooger, Aug. 2006 |
#35
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"George M. Middius" cmndr [underscore] george [at] comcast
[dot] net wrote in message Another dose of Krooglish gets Mr. **** girded for his weekend joust with Rev. Poop-Head at the Goose Puke Baptist church. Could you be more presumptious Nobody in the history of the world has ever been "presumptious", you dumb ****. Stop lying, please. Note Middius' amazing ability to herniate in public himself over a typo. |
#36
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![]() More lies from LiarBorg. I'm not surprised. Shall we notify Rev. Poop-Head that Arnii is renewing his vows to be a good "chrisitan"? ;-) Could you be more presumptious Nobody in the history of the world has ever been "presumptious", you dumb ****. Stop lying, please. Note Middius' amazing ability to herniate[sic] in public himself over a typo. "I apologize for being presumptious." A. Krooger, 20 July 2000 "Since I'm not claiming to be able to read your mind, the act you consider presumptious did not happen." A. Krooger, 17 June 2000 "Presumptious little minx aren't you, Jenn?" A. Krooger, 4 April 2006 "... any reasonable male would be a lot more worried about the "Presumptious" part." A. Krooger, 7 September 2001 -- "Christians have to ... work to make the world as loving, just, and supportive as is possible." A. Krooger, Aug. 2006 |
#37
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In article ,
George M. Middius cmndr [underscore] george [at] comcast [dot] net wrote: More lies from LiarBorg. I'm not surprised. Shall we notify Rev. Poop-Head that Arnii is renewing his vows to be a good "chrisitan"? ;-) Could you be more presumptious Nobody in the history of the world has ever been "presumptious", you dumb ****. Stop lying, please. Note Middius' amazing ability to herniate[sic] in public himself over a typo. "I apologize for being presumptious." A. Krooger, 20 July 2000 "Since I'm not claiming to be able to read your mind, the act you consider presumptious did not happen." A. Krooger, 17 June 2000 "Presumptious little minx aren't you, Jenn?" A. Krooger, 4 April 2006 I'd forgotten about that post LOL "... any reasonable male would be a lot more worried about the "Presumptious" part." A. Krooger, 7 September 2001 -- "Christians have to ... work to make the world as loving, just, and supportive as is possible." A. Krooger, Aug. 2006 |
#38
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Eeyore wrote:
Phil wrote: Sander deWaal wrote: Phil said: Well Andre, it looks like it's you and me, since everyone else has either pussied out on me (Phil Asshole, Graham), dealt with other issues, although probably honestly (Scott), or is waiting for further developments (Patrick). I went by the UT library today, and looked through several years of JAES, 1980 onward, but only found one article by Matti Otala. I think maybe he published some papers in IEEE, I'll have to check. However, in the Jan. 1980 JAES issue, there is an article by Vanderkooy and Lip****z called "Feedforward Error Correction in Power Amplifiers" that looks *very* interesting! I haven't thoroughly looked at it yet, but they review all the various types of feedforward schemes, starting with the one invented by Harold Black in 1923! They also review the Quad 405, which is a different type of feedforward, which I can state from personal experience sounds pretty bad compared to any decent tube amp. That you didn't find much after 1980, is because mr. Otala published his findings in the early '70s. At that time, the problems as described by him, were a reality in may commercial amplifiers, and we've learned a lot since then. Later, people like Daugherty and Greiner proved that (large factors of) feedback isn't the evil that may seem to think it is, and that it doesn't necessarily generate additional distortion, when applied correctly. I happen to think that Otala has played a major role in getting more insight in what happens in an amplifier stage with feedback. Others after him corrected and modified his findings. There are more ways that lead to Rome. BTW a modified 405-II can sound very good, at least to these ears. The Quad I heard was not modified, so I couldn't say, although the one I heard sounded very good unless compared to a good PP tube amp. I want to emphasize that the original subject here is not what Phil Allison implies, namely TIM or SID, but rather the question of whether negative feedback causes audible problems even when there is no TIM. I'll go ahead and quote the review of Otala's paper I gave in the other thread, from "The Audio Critic," Vol 2, #2, p 37, regarding Matti Otala's analysis of feedback (made after he, the editor Peter Aczel, Mitch Cotter, Stew Hegeman, Andy Rappaport, Max Wilcox, and Bruce Zayde had a "BS" session in TAC); "The paper presents rigorous mathematical proof, for the most generalized, all-inclusive case, that feedback cannot make amplifier distortions go away; all it can do is to change one kind of distortion into another. By the application of feedback, the amplitude nonlinearities of the open loop are converted into phase nonlinearities of the closed loop. That's all. The garbage cannot, by definition, be made to disappear; it's simply swept into another corner. In the typical feedback amplifier, the amplitude of the audio signal phase-modulates the high-frequency components of the signal. Furthermore, any amplitude intermodulation distortion in the open loop is converted into phase intermodulation distortion in the closed loop. What about TIM, alias SID? It turns out that it (he?) is a limit case of this feedback-generated phase modulation effect, with all shades of gray possible before the actual black eruption occurs. None of this shows up on standard tests." I still can't find this paper, despite several trips to the UT library, but a little thought shows that it actually is consistent with much of what you and the others ae saying. When an amp with, say, 40 dB of feedback is hit with a step, the output initially has an "error" of 100 x, *independent* of any gain or load non-linearities, which must be "corrected" by the feedback loop. For every single change in the input voltage, the gain is off by a factor of 95 to 105, depending on gain and load non-linearities, and this error must be corrected by the feedback loop. Intuitively, it seems obvious that Otala's proof must in *some* way be correct, that this constant "correction" must play havoc with low level and high frequency signals. I don't think anyone would deny that, given an amp with variable feedback followed by a pot to equalize the overall gain, turning up the feedback will eventually make an amp that, like the Crown preamp, will "bite your ears off," even if the amp never gets into TIM territory or other obvious problems. The question is how much of an effect does Otala's "dynamic phase shifting" have. Here again, it seems obvious that part of the problem was the S-L-O-W power transistors of the late '70's, when Otala's various articles were written. I suspect that high speed devices reduce the problems created by feedback, the amount of phase distortion produced, and of course Otala himself came up with several ideas to reduce these effects in his Citation XX design, although I also haven't been able to find any literature on that design. Nevertheless, it is a given, in my mind, that a very high open loop gain, with its need to constantly "correct" every input signal by 99% (in the case of 40 dB feedback), *regardless* of the inherent linearity of the amp's devices and circuit, MUST cause problems for signals 60 dB to 80 dB below the main signal, and perhaps also phase shift the high frequency components, as Aczel's summary of Otala's paper states, thereby robbing the circuit of much of its "life" and "air," the criticisms one normally hears about high feedback amps, and also solid state amps, in which the solid state capacitances and high thermal variations also interfere with low level signals. This will not show up as TIM or SID, unless the amp has been very poorly designed, and I'm still not sure how one would measure it. My best guess has been to use a 20 Hz signal and a much smaller (-60 to -80 dB) 10 KHz signal, filter out the 20 Hz signal from the output with a notch filter plus high pass filter, and either look directly at the 10 KHz signal for signs of distress, or filter it out with another notch filter, and see if phase shifting causes "sidebands" to appear and disappear when the 20 Hz signal is put in and out of the test. Assuming that normal feedback causes problems -- and as Patrick says, with low feedback and tubes it isn't too bad, but SS amps have more problems and generally need more feedback -- it would be nice if we could figure out a way to tremendously reduce the need for feedback to "correct" every normal signal by 99% even when there is no device distortion, meaning allow the feedback to "focus" *only* on actual device and load non-linearities. Here is where Black's "feedforward" circuit may allow for a real advance in SS amps, especially if tubes, with their (generally) superior ability to handle a mix of high and low level signals without messing up the low level information, are used to provide the error signal. Properly applied, Black's feedforward scheme (but not the feedforward designs by many others!) does exactly this, it allows feedback to appear and affect the signal *only* when actual deviations caused by device or load non-linearities appear. It may even be possible to correct the effects of a typical transistor's parallel capacitances -- which, being made of silicon, are of *very* poor quality -- and thermal variations! Normally, this would be a "why waste the time, just use tubes," situation, but good output transformers are heavy, big, and expensive, and if the amps used in compact disk players, as well as TV's, could be considerably improved, that would be nice! And of course, inexpensive amps that sound very good are always in demand. You do talk a shocking amount of drivel ! Graham Here's a suggestion, Useless: Why don't you share with us some of YOUR insights and analyses so we could what "non-drivel" looks like. Unless, of course, a useless pussy like yourself (or Arny) has none, or is too much of a coward to stick your neck out. The only useless drivel I see is the CONSTANT use by you and Arny of that same old, OLD, tired, debating trick of putting out a general criticism, with no specific examples, and no supporting evidence. And no matter how many times it gets pointed out to you, we can all count on one thing; your next post will do it again. I've seen useless pussies before, but until you two came along, I had not seen petrified useless pussies. You two are unique, that I'll admit. Phil |
#39
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Patrick Turner wrote:
[snip] Well, remember, the maximum slew rate found in audio signals is much greater than what a theoretical 20 KHz signal is going to supply, and not all power amps, with their big, slow, output transistors, are going to be as fast as even a 741. Not all output bjts are big and slow. Some do however dislike turning OFF quickly and some display the truly horrible habit of cross conduction at HF, ie, the two bjts in a typical complementary pair are BOTH turned on during a wave cycle during large signal excursions at above 10kHz, and the power supply has to supply a lot more current that is simply passing from rail to rail and its hang onto your hat time for the ride. Plus, the point of Matti's work is that problems begin to appear at all levels below the theoretical "breakthrough" point of TIM/SID. In any case, the topic here is not whether most amps have sufficient slew rate -- I assume that *most* good amps do -- but rather about Otala's proof that a feedback amp's "correction" of an amplitude distortion of the open loop phase shifts the high frequency components in the closed loop. Both amplitude distortions and phase distortions of the open loop response are BOTH corrected by the NFB. Typical open loop phase lag in open loop at 20kHz is 90 degrees, and the 40dB of applied global NFB at 20kHz reduces this typically to less than 5 degrees. Patrick, for god's sake, if you disagree with me, fine, but how many times do I have to say that what Matti was talking about was not the usual lag at 20 KHz, but rather phase-SHIFTING, as in MOVEMENT, not lead or lag, that as you yourself correctly guessed, is a function of low frequency amplitude, not overall frequency. Yes, feedback allows greater bandwidth, thereby improving "phase distortion," where distortion is defined as CONSTANT phase shift as a function of frequency. Otala, the technical director of the Finnish Institute for whatever, was not that stupid! Come on, you know that "dynamic phase distortion" refers to something else, you said so yourself. I am currently discussing this in the "Negative Feedback in Triodes: The Logical and Experimental Proof" thread from 8/15, so if you're interested, look there (articles posted on 9/6). Phil Allison had a "response" here -- his usual slams and blams with no supporting evidence -- but I actually would like to see his simple test that can show whether low frequency signals in a feedback amp do or do not cause high frequency phase shifting, as it would be useful test, and I'm having a hard time coming up with a simple way to test that myself. Just apply 70Hz and 5kHz signals to the input of an amp in a 4:1 ratio. Filter out all below 1kHz from the output signal. Then you will see what the effect of the 70Hz large signal is upon the fidelity of the 5kHz signal and whether there is any phase modulation in addition to the expected intermodulation. With most well made SS high NFB amps, the IMD is not visible on the CRO and a careful peak detector must be used to measure amplitude variations in the 5kHz, or else filter out the IMD products at 4,930Hz and 5,070Hz. Thank you, but there may be more to it than that. For one, let's see what happens when the 70Hz signal is 80 dB higher than the 5KHz signal, and let's make certain that our equipment is sensitive to rapid forward and backward shifts in time of the 5KHz signal. A 'scope would almost certainly catch that IF triggered with a constant timer, not the 5KHz signal. Apparently, Otala incorporated a lot of ideas/solutions into his Citation XX power amp, and maybe, if I can find papers by him on that amp, there will be some useful information and tests there, but if PA can come up with something in the meantime, hell that's fine by me! He'll probably think of something really simple and easy, and then refuse to tell me, the ****head ... But all these investigations have been done many times before. Could you name one? What exactly do you hope to gain by goading the ungoadables on the group to find out what you should be willing to find out for yourself? Who said I was trying to "hire for free" the ungoadables? I am discussing a subject, including possible problems and solutions. Ideally, we all have a complete workbench, but I think it's a bit bigoted to imply to those of us who do not that we should get our own bench and do all investigations ourselves, before we are permitted to discuss a subject here. Do you suspect to find some hitherto unused uninvented techniques of making amplifiers perform better? I can almost say, "Duh, of course!" However, I do not necessarily EXPECT to do anything, because I AM discussing a subject, which I have a right to do on this list, as it is basically what the list is all about! Besides, you have yet to give an honest answer to a method I suggested that should improve things, my higher resolution double-blind suggestion. I gave you a PROOF -- which is no more and no less reliable than the premises -- that if you use that method, you can find differences between components that the normal double blind cannot reveal. That will not automatically enable you to make a better amp and make more money, but it should help! You're welcome, by the way ... Anyway, what PA was saying/yelling is that feedback amps DO NOT EITHER CAUSE PHASE SHIFTING OF THE HIGH FREQUENCIES LIKE THAT DUMMY DR. OTALA SAID! My response was simply to ask whether (1) he knew of references that would back up his claim, that Otala's analysis was flawed, and (2) whether he knew of a good, simple test that can be used to test whether LF signals in a feedback amp cause phase shifting of the HF signals, like Otala said they do. Be like me, find out by building one's own test gear and testing. It took me months to do it all but after reading all the conflicting opinions about all this in Electronics World copies from the 1970s to 1980s BEFORE the internet was mainstream, I decided to look myself at what happened in amps that i should be worried about. And I am impressed by that. You are not, like Graham or Arny, either useless or a pussy. But not everyone can do that, and it is unfair of you to suggest otherwise. Phil Patrick Turner. Phil To email me directly, cut off my head |
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Posted to rec.audio.tubes,rec.audio.opinion
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Joseph Meditz wrote:
Phil Allison wrote: "Andre Jute" Let's hear some more about this dynamic phase shift that pours a pint of vinegar into a Crown preamp. ** Dynamic phase shifting of audio signals is all around us, all the time. The fact that cones move, continuously alters the origin and hence time of arrival of any higher frequencies being simultaneously radiated. Phase shift in degrees ( at any point in time) is simply 360 x cone excursion / wavelength of the high frequency. Some call this effect " Doppler Distortion" - a misnomer. Hi Phil, Here's my take on this interesting topic. I say that this is precisely an acoustical frequency modulator. If you input two sinusoids, one low and one high, then the spectrum of the upper one will be spread out about its center. And the greater the amplitude of the bass signal, the greater the modulation index. From the modulation index one could predict what the side bands will look like. I found the term "Doppler Distortion" helpful. The situation here is not exactly like the sound of the horn of a train passing a station. Rather, it is the sound of the horn of a crazy train oscillating back and forth across the station. Joe So, is an acoustical frequency modulator some type of equipment? It does sound like what I *think* Otala is saying happens when a feedback amp gets hold of two sinusoids, as you say. What is a good methos for seeing this spread? Someone suggested (oh hell, I think it was Arny; a USEFUL idea???) using a spectrum analyzer, maybe it's that simple? By the way, I *think* PA is wrong, in a way. It may be that a cone moving forward at a bass frequency can Doppler shift a high frequency signal, but doesn't the mic that recorded the two frequncies to begin with invert this process, thereby cancelling it out? Just thinking ... Phil |
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