[Andre Jute]

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

[Phil Allison]

"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

[Patrick Turner]

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

[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

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

[Phil Allison]

"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

[Phil]

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

[Eeyore]

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

[Phil]

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

[Eeyore]

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

[Phil]

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

[Phil Allison]

"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

[ScottW]

"Phil" wrote in message
...
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?

If you have a signal with too much amplitude for your amp,
its called clipping.

ScottW

[Phil Allison]

"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.



Plus, the point of Matti's work is that problems begin to appear at all
levels below the theoretical "breakthrough" point of TIM/SID.


** Shame no-one could find evidence of this in commercial amplifiers.


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.


** ********.

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.






....... Phil

[Phil]

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

[Eeyore]

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

[Eeyore]

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

[Sander deWaal]

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."

[Patrick Turner]

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

[Phil Allison]

"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

[Patrick Turner]

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.

[Arny Krueger]

"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.

[Sander deWaal]

"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."

[Phil Allison]

"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

[Patrick Turner]

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."

[Sander deWaal]

"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."

[Joseph Meditz]

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

[Phil Allison]

"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

[Phil Allison]

"Sander deWog******"


**** the HELL OFF

YOU ****wit Audiophool JERK !!!!!!!!!


....... Phil

[[email protected]]

On Thu, 7 Sep 2006 21:54:33 +1000, "Phil Allison"
wrote:



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.


Now there is something I can agree with 100%...


Bob FlintsTone

[paul packer]

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.

[Chris Hornbeck]

On Thu, 07 Sep 2006 13:40:14 +0200, Sander deWaal
wrote:

That you didn't find much after 1980, is because mr. Otala published
his findings in the early '70s.

Right-e-o. He wants AES Rotterdam 22.2.1973, Jan Lohstroh
and Matti Otala "An Audio Amplifier for Ultimate Quality
Requirements", which rigorized the issues and proposed
(later universally accepted) changes to topology.


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.

Also in American audio press Jung, Stevenson, and Todd wrote an
excellent "Overview of SID and TIM" in the Summer of 1979 _Audio_
magazines, which lead to the Spring 1980 _Audio_ articles by Cordell
"Another View of TIM".

Cordell showed clearly, and argued for, what Jung had implied,
that a sufficiently low level of THD at 20KHz (the number .02%
has become accepted) is enough to show that no TIM will exhibit
*with a bandlimited input signal*.

For output amplifiers, end of discussion. For input stages fed
un-bandwidth-limited signal, greater scrutiny is required.

Some common examples of input stages faced with larger than
"CD" bandwidth are microphone preamps, phono equalizers, and
D/A converter summing amps.

Much thanks, as always,

Chris Hornbeck
"Man is the measure of all things. Sigh.
Happy Ears!" -Al

[Ian Iveson]

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

[Arny Krueger]

"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.

[Phil]

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

[Eeyore]

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

[Arny Krueger]

"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.

[Arny Krueger]

"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. ;-)

[George M. Middius]

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

[Arny Krueger]

"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.

[George M. Middius]

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