[Jenn]

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

[Phil]

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

[Phil]

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

[Phil]

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

[Eeyore]

"George M. Middius" wrote:

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.

Ejsy fof upi dsu ?

Graham

[Eeyore]

"George M. Middius" 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

"... any reasonable male would be a lot more worried about the
"Presumptious" part."
A. Krooger, 7 September 2001

How's that new sub-editor position going btw Georgie Porgy puddikins ?

Graham

[Eeyore]

Phil wrote:

Eeyore wrote:
Phil wrote:
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!

Bwahahahahahhaha !

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.

Well........ since it would be based on sound engineering science, I suspect it
might have limited appeal.


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.

If you knew anything about electronic design and solid-state physics you
wouldn't make such stupid comments about parallel capacitance in transistors for
example.

Tell me more about this and how it works and them I'll correct you OK ?

Graham

[Eeyore]

Phil wrote:

Patrick Turner wrote:

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.

Bwahahahahahahhah !

A 'scope would almost
certainly catch that IF triggered with a constant timer, not the 5KHz
signal.

Do you have a degree in stupidity or simply just denseness ?

Graham

[Eeyore]

Phil wrote:

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 ...

If the concept was even right in the first place then logically it would indeed
'cancel' out.

Graham

[Phil Allison]

"Eeyore"

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 ...

If the concept was even right in the first place then logically it would
indeed
'cancel' out.


** Nonsense.

Shifting the origin point of a steady wave changes its phase at the position
of a test mic - as would shifting the mic.

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.

Dymamic mic diaphragms make only tiny excursions ( condenser mics even
less ) compared to loudspeaker cones - so they react to the pressure at a
fixed point.




........ Phil Allison

[Chris Hornbeck]

On Fri, 08 Sep 2006 19:21:10 GMT, Phil
wrote:

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);

You really want to read Otala's own work. It's an important
historical piece, and has *nothing* to do with the stuff
quoted.

All good fortune,

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

[Chris Hornbeck]

On Fri, 08 Sep 2006 23:59:14 GMT, Phil
wrote:

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.

No.

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?

Or a conventional IM distortion analyzer.

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?

If they were perfectly coupled, yes. But symmetry is broken.

All good fortune,

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

[Ian Iveson]

Phil said:

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.

Army may not be interested in reinventing the wheel, but I quite like
the idea of applying transistor-era thinking back to valve designs. So
thanks for the provocation.

As regards Doppler. I look at Wikipedia and see that the relative
velocity is assumed (by Wikipedia at least) to be constant. The
frequency shift is therefore also constant and so clearly defined. A
changing shift in frequency is an awkward concept, especially if it
changes sinusoidally.

Phase and frequency are two sides of the same kettle of geese. A
constant shift in phase is a change in frequency. Whether one is
changing or the other depends on how you look at it. Possibly, in our
case, how we actually hear it.

Now, apparently some useful information can be gleaned from observing
how the velocities of binary stars, with respect to us, vary as they
orbit each other. The changing velocities are calculated from changing
colours, assuming that the Doppler effect is the sole cause of the
change. That is an example very close to your speaker idea, where a
source moving at a relatively low frequency modulates the higher
frequency of an emitted signal.

But. In the case of the binary star, the difference between the two
frequencies is huge. So great that, for the time it takes to detect a
shift in the higher frequency, the source can be assumed to be moving
at constant velocity.

How far must the frequencies be apart in order to result in something
that can be properly called the Doppler effect? Well, Wikipedia says
frequency shift, so at what point, as you move the frequencies closer,
does it stop resulting in a frequency shift?

Let's take the obvious example of making the two frequencies the same.
Does it make sense to say that the resulting tone, being exactly the
same but with an amplitude shift of anywhere between 1 and -1, is a
consequence of the Doppler effect? Is a shift in amplitude the same
thing as a shift in frequency?

The audio band isn't many orders of magnitude wide. It is probably
more convenient to characterise the effect you speak of in terms of
phase, rather than frequency. Then it stops being a proper Doppler
effect, perhaps.

Your oscillating train example has much greater difference between the
source and signal frequencies. Perhaps you know that already.

Real-time spectrum analysis is another tricky concept.

Ultimately, it's how your ear appreciates it that counts, of course,
etc.

cheers, Ian

[Ian Iveson]

I wrote

A constant shift in phase is a change in frequency.

er...constantly shifting...I know what I mean...

[Robert Morein]

"George M. Middius" wrote:
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.


"I'd like to have as many men and women as possible **** me in the ass
before I die"
George Middius, 8th Grade Graduation Ceremony

[Phil]

Eeyore wrote:

Phil wrote:


Eeyore wrote:

Phil wrote:

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!


Bwahahahahahhaha !


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.


Well........ since it would be based on sound engineering science, I suspect it
might have limited appeal.



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.


If you knew anything about electronic design and solid-state physics you
wouldn't make such stupid comments about parallel capacitance in transistors for
example.

Tell me more about this and how it works and them I'll correct you OK ?

Graham

Sure. You know the gate to source, gate to drain, and the drain to
source capacitances? In a solid state transistor, they are made out of a
dielectric called "silicon" -- not coincidentally, the same thing the
FETS are made of -- and it has a dielectric constant of 12, quite high.
Good, low dielectric constant materials, with constants of 2 to 3, have
enough dielectric absorption to require precision instruments to use
several parallel RC units to offset the noise dumped by the main
capacitors (see p 220 of Horowitz and Hill, "The Art of Electronics").
High dielectric materials are much worse (note the usual warnings for
electrolytic caps, with a mere dielectric constant of 7 to 8), and
NOBODY in their right mind would even dream of using a silicon capacitor
in any circuit required to pass signals in which many important signals
are 80 dB or more below the main signals; you know, like you find in music?

Take just the Cds cap: as the signal moves up and down, the dielectric
absorption of this cap stores and releases energy all the time,
polluting the signal. This pollution is at a very low level, and if
music only used about 40 dB or so, it *might* not be noticeable, or at
least, just barely noticeable. But with music the range is closer to 100
dB, even with analog (humans can easily hear 40 dB into noise, something
the old telegraph operators normally achieved). In contrast, a vacuum
tube has capacitances made out of, well, a vacuum, the medium of space,
which has, as far as we know, no dielectric absorption at all. In
addition -- as you, of course, already know, as can be proven in some of
your old posts here -- the transistor's capacitances *vary* as a
function of both voltage and heat, and the heat varies as a function of
the device's recent signal history, meaning it does *not* follow the
musical signal in a harmonic way, but rather as an rms average over the
most recent second or so. When a capacitance increases or decreases, and
the voltage remains the same, guess what? That's right, the charge
increases or decreases, meaning it gets sucked from or dumped into the
signal in a non-harmonic way. Now, if we have a device with vacuum
capacitors to compare the input versus the silicon capacitor gunked-up
output, we might, just might, be able to offset the crap dumped into the
signal by those capacitors. Clear enough?

Now, you can go into debate mode, in which truth, honesty, and
intelligence are meaningless, and the only goal is to attack your
"opponent's" statements, or you can be honest here. Gee, what do I think
a useless pussy like you will do? Golly, that's a hard one ... Let's
see, "Well, see, modern transistors are free of this." Excuse me, have
you LOOKED at the cap data for MOSFETS (or even power bipolars) lately?
For very small signals, using very small transistors, you can reduce
this effect to -60 dB or so, with great care maybe even -80 dB, or with
tremendous care and many offsetting components, maybe 120 dB, which I
suspect would then be inaudible. But I do not know of a preamp design
which achieves anywhere near 120 dB, although I obviously have not seen
all the circuits out there. Wake up, useless; I will unhappily admit
that I am all too frequently wrong, but I am NOT stupid, and only a
useless, sanctimonious snot like yourself would even try to convince
himself otherwise. But maybe I'm just kidding myself, because maybe you
can list some of your old posts here that not only discuss the subject
of the effect that the solid state parasitic capacitances have on audio
circuits, but which are free of the "many errors" that you will claim
exist, but as always, without actually giving even one example of these
errors. Oh, and no word -- of course -- concerning how many of my points
are correct; that would violate debate rules, which are the only rules
that anyone should ever use! You know, I have terminal cancer, and maybe
heart problems (waiting for the results), but I would rather be me, and
dead soon, than an utterly useless pussy like you, for decades. And no,
my physical problems should NEVER excuse me from INTELLIGENT criticism.

Disgusted Phil

[Phil Allison]

"Phil"


You know, I have terminal cancer, and maybe heart problems (waiting for
the results),


** Don't be so modest with your medical boasting -

you also have terminal autism and bi-polar disorder.



but I would rather be me, and dead soon,


** Then we all have something to look forward to.





........ Phil

[Tom A.]

I only know Matti Otala from being the designer of a particular set of
top of the range Harman Kardon pre/power amp, that got great reviews in
the 80s. I might be mistaken, but isn't he the designer of several well
regarded Electrocompaniet amps?

Here is a link to pictures of the HK set: http://www.popolo.com/citation.html

Tom



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

[Phil]

Phil Allison wrote:
"Phil"


You know, I have terminal cancer, and maybe heart problems (waiting for
the results),



** Don't be so modest with your medical boasting -

you also have terminal autism and bi-polar disorder.




but I would rather be me, and dead soon,



** Then we all have something to look forward to.





....... Phil



Now THIS is cute!

Impressed Phil

[Phil]

Chris Hornbeck wrote:

On Fri, 08 Sep 2006 19:21:10 GMT, Phil
wrote:


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);


You really want to read Otala's own work. It's an important
historical piece, and has *nothing* to do with the stuff
quoted.

All good fortune,

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

I would love to, but I REALLY want to read the paper he wrote, or at
least presented, referred to here, and not just his previous papers.
I've looked in JAES, and there's nothing (although I think the older
papers are there). Any ideas on how to find something like a collection
of his papers?

Thanks, Phil

[Phil]

Tom A. wrote:

I only know Matti Otala from being the designer of a particular set of
top of the range Harman Kardon pre/power amp, that got great reviews in
the 80s. I might be mistaken, but isn't he the designer of several well
regarded Electrocompaniet amps?

Here is a link to pictures of the HK set:
http://www.popolo.com/citation.html

Tom

That is indeed the Citation XX, the one he designed after he did the
analysis on feedback. Supposed to sound fantastic, if you can find one.

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]

"Tom A." wrote:

I only know Matti Otala from being the designer of a particular set of
top of the range Harman Kardon pre/power amp, that got great reviews in
the 80s. I might be mistaken, but isn't he the designer of several well
regarded Electrocompaniet amps?

I've had to repair one those Electrocompianets several times. It may have been a
good design but the implementation was shockingly bad. It hummed a lot too.

Graham

[Phil]

Patrick Turner wrote:


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.



In the first place, this is a minor point which states that you cannot
take a single, typical 20 KHz wave, and assume that that contains the
highest slew rate you will see. But any text will show that when you mix
two sinusoids of the same or different frequencies, you get additive and
subtractive effects, so this is really, really old news. Will the
maximum rates exceed that of a full-power 20 KHz wave? I have to admit,
I don't know for certain, but I *think* the answer is actually yes,
which was one of the reasons why some older high feedback transistor
amps got into trouble; they made assumptions about the maximum slew rate
they would get hit with that were false. Perhaps Otala's older papers on
TIM will provide measurements of actual, rather than theoretical, max
slew rates. Still, as much as I hate to admit it, you may be right on
what we can *safely* assume the max slew rate to be, and I was indeed
thinking that LP audio signals occasionally exceeded this value.

Humbled Phil

[Eeyore]

Phil wrote:

Tom A. wrote:

I only know Matti Otala from being the designer of a particular set of
top of the range Harman Kardon pre/power amp, that got great reviews in
the 80s. I might be mistaken, but isn't he the designer of several well
regarded Electrocompaniet amps?

Here is a link to pictures of the HK set:
http://www.popolo.com/citation.html

Tom

That is indeed the Citation XX, the one he designed after he did the
analysis on feedback. Supposed to sound fantastic, if you can find one.

Probably very good for its time. Many better ideas have happened since. I have
actually adopted some of Otala's thinking in my own designs actually.

Graham

[Phil]

Arny Krueger wrote:

"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.


It is not hidden from me, and I have mentioned the same thing somewhere
in this thread, but since I was never claiming (as implied by PA) that
modern amps produce TIM, it is *almost* irrelevant. The "almost" part
comes from something I believe to be true, namely that the dynamic phase
shifting and other possible abuses of signals 80 dB below the main
signal in high feedback amps are *reduced* as the speed of the closed
loop increases, something that has indeed improved greatly not just
since the 60's, but since Otala wrote his papers in the 70's. However,
"improved" does not equal "perfect," or even "adequate," and since any
SS amp short of a Krell (if that) still is sonically challenged --
*especially* when it comes to the really low-level information --
compared to the best low or zero feedback triode amps, one has to
suspect that it is a problem that has only been reduced, not eliminated.
Given that there is room for improvement, it would be helpful if we
could figure out what is wrong.

Here you have not only my theory as to what is wrong (based mainly on
Otala's paper) -- problems caused by the higher feedback of SS amps,
among other SS problems -- but also a basic outline for a possible
solution. By all means, if you are aware of someone else who has pursued
my line of reasoning, and proved with actual experiments that it is
useless, post the information here (but don't waste our time with TIM
papers, since that is *not* what either I or Otala am/were talking
about). I will then publicly state, here, that you were RIGHT, and that
both my theory about one of the diseases of SS amps, and therefore my
proposed cures, were WRONG, and we can all move on in an attempt to
figure out the true reasons why all but the best SS amps suck when
compared to even very good tube amps, let alone the best ones.

Pragmatic Phil

[Andre Jute]

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

On the one side of this argument we have Dr Otala and Phil "Toob". Dr
Otala is a distinguished scholar and designer of highly rated audio
gear. Phil T is an earnest student who in the manner of earnest
students occasionally overstates his case.

On the other side of the argument we have Arny "I spoke in error"
Krueger and Graham "Poopie" Stevenson. Arny Krueger assembles computers
for a living and had a part in creating a now superceded and never very
useful ABX test to determine which computer sound cards did the least
damage to the sound; he pretends to be a recording engineer. Graham
Stevenson's employment appears to be writing to the Usenet about
politics and boasting about his self-proclaimed part in the design of
the least well-reputed Neve deck; when that wears thin he brags about
his education a generation ago.

From the violent attacks "I spoke in error" Krueger and Poopie
Stevenson have made on Phil T, one would be justified in concluding
that these minor players and permanent losers claim Dr Otala is simply
wrong. In fact, careful reading reveals that their entire argument is
that components have improved so much in the intervening generation as
to obviate Dr Otala's important point. The subtext of their manner of
stating this weak case is that these two loud fools are smarter than
Otala.

It is not difficult for the rest of us to decide who in a crunch we
will believe, the distinguished Dr Otala and a cross section of spec
sheets of modern components showing the problem has moved on, or the
hysterical screechings of the self-important "I spoke in error" Krueger
and his limp sidekick Poopie Stevenson.

This entire long thread has done little to illuminate my main question,
so I put it again in different form in another thread.

Andre Jute
Our legislators managed to criminalize fox-hunting and smoking; when
they will get off their collective fat arse and criminalize negative
feedback? It is clearly consumed only by undesirables.

[Andre Jute]

ADDENDUM: The threads in which the argument referred to below was
carried on are "Negative Feedback in Triodes: The Logical and
Experimental Proof" which starts he

http://groups.google.ie/group/rec.au...f8d8ed 263a35

and "Dynamic phase shift" which starts he

http://groups.google.ie/group/rec.au...34cd31 13f1ab

--AJ

Andre Jute wrote:
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

On the one side of this argument we have Dr Otala and Phil "Toob". Dr
Otala is a distinguished scholar and designer of highly rated audio
gear. Phil T is an earnest student who in the manner of earnest
students occasionally overstates his case.

On the other side of the argument we have Arny "I spoke in error"
Krueger and Graham "Poopie" Stevenson. Arny Krueger assembles computers
for a living and had a part in creating a now superceded and never very
useful ABX test to determine which computer sound cards did the least
damage to the sound; he pretends to be a recording engineer. Graham
Stevenson's employment appears to be writing to the Usenet about
politics and boasting about his self-proclaimed part in the design of
the least well-reputed Neve deck; when that wears thin he brags about
his education a generation ago.

From the violent attacks "I spoke in error" Krueger and Poopie
Stevenson have made on Phil T, one would be justified in concluding
that these minor players and permanent losers claim Dr Otala is simply
wrong. In fact, careful reading reveals that their entire argument is
that components have improved so much in the intervening generation as
to obviate Dr Otala's important point. The subtext of their manner of
stating this weak case is that these two loud fools are smarter than
Otala.

It is not difficult for the rest of us to decide who in a crunch we
will believe, the distinguished Dr Otala and a cross section of spec
sheets of modern components showing the problem has moved on, or the
hysterical screechings of the self-important "I spoke in error" Krueger
and his limp sidekick Poopie Stevenson.

This entire long thread has done little to illuminate my main question,
so I put it again in different form in another thread.

Andre Jute
Our legislators managed to criminalize fox-hunting and smoking; when
they will get off their collective fat arse and criminalize negative
feedback? It is clearly consumed only by undesirables.

[Andre Jute]

And, after all that, we still seek an answer, in the new thread "The
sound of excessive negative feedback" which starts he

http://groups.google.ie/group/rec.au...01e147 c56a8b

Andre Jute wrote:
ADDENDUM: The threads in which the argument referred to below was
carried on are "Negative Feedback in Triodes: The Logical and
Experimental Proof" which starts he

http://groups.google.ie/group/rec.au...f8d8ed 263a35

and "Dynamic phase shift" which starts he

http://groups.google.ie/group/rec.au...34cd31 13f1ab

--AJ

Andre Jute wrote:
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

On the one side of this argument we have Dr Otala and Phil "Toob". Dr
Otala is a distinguished scholar and designer of highly rated audio
gear. Phil T is an earnest student who in the manner of earnest
students occasionally overstates his case.

On the other side of the argument we have Arny "I spoke in error"
Krueger and Graham "Poopie" Stevenson. Arny Krueger assembles computers
for a living and had a part in creating a now superceded and never very
useful ABX test to determine which computer sound cards did the least
damage to the sound; he pretends to be a recording engineer. Graham
Stevenson's employment appears to be writing to the Usenet about
politics and boasting about his self-proclaimed part in the design of
the least well-reputed Neve deck; when that wears thin he brags about
his education a generation ago.

From the violent attacks "I spoke in error" Krueger and Poopie
Stevenson have made on Phil T, one would be justified in concluding
that these minor players and permanent losers claim Dr Otala is simply
wrong. In fact, careful reading reveals that their entire argument is
that components have improved so much in the intervening generation as
to obviate Dr Otala's important point. The subtext of their manner of
stating this weak case is that these two loud fools are smarter than
Otala.

It is not difficult for the rest of us to decide who in a crunch we
will believe, the distinguished Dr Otala and a cross section of spec
sheets of modern components showing the problem has moved on, or the
hysterical screechings of the self-important "I spoke in error" Krueger
and his limp sidekick Poopie Stevenson.

This entire long thread has done little to illuminate my main question,
so I put it again in different form in another thread.

Andre Jute
Our legislators managed to criminalize fox-hunting and smoking; when
they will get off their collective fat arse and criminalize negative
feedback? It is clearly consumed only by undesirables.

[paul packer]

On Sat, 9 Sep 2006 15:02:13 +1000, "Phil Allison"
wrote:


"Phil"


You know, I have terminal cancer, and maybe heart problems (waiting for
the results),


** Don't be so modest with your medical boasting -

you also have terminal autism and bi-polar disorder.



but I would rather be me, and dead soon,


** Then we all have something to look forward to.





....... Phil


You have a heart as big as all outdoors, Phil.

[Phil]

Eeyore wrote:

Phil wrote:


Tom A. wrote:


I only know Matti Otala from being the designer of a particular set of
top of the range Harman Kardon pre/power amp, that got great reviews in
the 80s. I might be mistaken, but isn't he the designer of several well
regarded Electrocompaniet amps?

Here is a link to pictures of the HK set:
http://www.popolo.com/citation.html

Tom


That is indeed the Citation XX, the one he designed after he did the
analysis on feedback. Supposed to sound fantastic, if you can find one.


Probably very good for its time. Many better ideas have happened since. I have
actually adopted some of Otala's thinking in my own designs actually.

Graham

You have designs? I would love to see them, even if it requires
promising to NOT comment on them ... ;-)

Phil

[Chris Hornbeck]

On Sat, 09 Sep 2006 11:27:49 GMT, Phil
wrote:

I would love to, but I REALLY want to read the paper he wrote, or at
least presented, referred to here, and not just his previous papers.
I've looked in JAES, and there's nothing (although I think the older
papers are there). Any ideas on how to find something like a collection
of his papers?

Email me a USPS address.

All good fortune,

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

[Sander deWaal]

Tom A. said:


I only know Matti Otala from being the designer of a particular set of
top of the range Harman Kardon pre/power amp, that got great reviews in
the 80s. I might be mistaken, but isn't he the designer of several well
regarded Electrocompaniet amps?


Yep. A horror to repair, and they failed quite often.

--
"Due knot trussed yore spell chequer two fined awl miss steaks."

[Eeyore]

Sander deWaal wrote:

Tom A. said:

I only know Matti Otala from being the designer of a particular set of
top of the range Harman Kardon pre/power amp, that got great reviews in
the 80s. I might be mistaken, but isn't he the designer of several well
regarded Electrocompaniet amps?

Yep. A horror to repair, and they failed quite often.

You've been there too ?

I totally agree.

Graham

[Sander deWaal]

Eeyore said:


..... Electrocompaniet amps?


Yep. A horror to repair, and they failed quite often.


You've been there too ?


Yup. They sounded OK though, and I liked the looks of the
"anniversary" series with the Norwegian marble front panels.

Many of them went to Asia, as a collector's item.

Good riddance ;-)

--
"Due knot trussed yore spell chequer two fined awl miss steaks."

[ScottW]

"Phil" wrote in message
...
Patrick Turner wrote:


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.



In the first place, this is a minor point which states that you cannot take a
single, typical 20 KHz wave, and assume that that contains the highest slew
rate you will see. But any text will show that when you mix two sinusoids of
the same or different frequencies, you get additive and subtractive effects,
so this is really, really old news. Will the maximum rates exceed that of a
full-power 20 KHz wave? I have to admit, I don't know for certain, but I
*think* the answer is actually yes,

Not without exceeding "full-power"...and then you'd clip anyway.

ScottW

[Chris Hornbeck]

On Sat, 9 Sep 2006 18:28:52 -0700, "ScottW"
wrote:

In the first place, this is a minor point which states that you cannot take a
single, typical 20 KHz wave, and assume that that contains the highest slew
rate you will see. But any text will show that when you mix two sinusoids of
the same or different frequencies, you get additive and subtractive effects,
so this is really, really old news. Will the maximum rates exceed that of a
full-power 20 KHz wave? I have to admit, I don't know for certain, but I
*think* the answer is actually yes,

Not without exceeding "full-power"...and then you'd clip anyway.

Right. Maybe a better way for the OP to approach this is to
begin with a true step input (a signal with *zero* risetime)
beginning at very small levels, then increasing levels until
slewing is observed.

There will be a threshold level where even an amplifier
without feedback will slew.

There will also be a threshold where an amplifier *with* feedback
will slew. If there are no other differences between the
amplifiers, these two levels will be *the same*.

For the second case, as you've said, clipping may already
have occurred. In a modern amplifier this *must* be the
case; it's too easy to get right.

The issue of slew-limiting in amplifiers arose because
amplifiers with feedback must sacrifice some slewing ability
to the altar of stability. (Many, many qualifiers need be
inserted here; whole 'nother can 'a worms. But we're
talking about a thirty year gone issue.) In the early dark
days of semiconductor amplifiers, compromises for stability
were so demanding as to damage signal in the passband at
ordinary levels.




And, FWIW, the highest rise-time possible in a band-limited
signal is a square wave of just under bandpass frequency run
through the bandpass filter. IOW, a sinewave of the highest
frequency within the bandpass.

This is 101 stuff and the OP really should bone up as possible.

Also, there's *zero, none, nada* correlation between feedback
and anything remotely approaching "dynamic phase shift". The
term itself is snake oil of the highest grade.

All good fortune,

Chris Hornbeck
"When you look long into the abyss, the abyss also looks into you."
-Friedrich Nietzsche, _Beyond Good and Evil_

[Phil Allison]

"Phil" toob-headman

" You know, I have terminal cancer, and maybe heart problems (waiting for
the results), "


** Don't be so modest with your medical boasting - you also have terminal
autism and bi-polar disorder.


" .... but I would rather be me, and dead soon, "


** Then we all have something to look forward to.



Now THIS is cute!

Impressed Phil



** To save everyone further grief - drop dead now.





......... Phil

[Phil Allison]

"Eeyore"

I've had to repair one those Electrocompianets several times. It may have
been a
good design but the implementation was shockingly bad. It hummed a lot
too.



** This web page has some startling inside revelations on the saga of the
Electrocompaniet.

http://home.online.no/~tsandstr/OtalaStory.htm


Quote:
So, we realised than the balance of distortions were the essential factor to
consider. Not only do you have to balance THD against TIM, but also low
frequency distortion against high frequency distortion, frequency and phase
response against non-linear distortion in general, and so on.

This insight triggered the Great Change:

One night (it always happened at night times!) we increased the feedback
10dB, to a total amount of 30dB feedback. The sound improvement was
staggering!!! And, contrary to common belief in our own community !





** Their great discovery = more NFB sounds better !!

ROTFLMAO !!!!!





........ Phil

[Phil]

Phil Allison wrote:

"Phil" toob-headman

" You know, I have terminal cancer, and maybe heart problems (waiting for
the results), "


** Don't be so modest with your medical boasting - you also have terminal
autism and bi-polar disorder.


" .... but I would rather be me, and dead soon, "


** Then we all have something to look forward to.




Now THIS is cute!

Impressed Phil




** To save everyone further grief - drop dead now.





........ Phil


Sorry, I have to give you grief as long as possible. But look at it this
way; you, Phil Allison, give meaning, true meaning, to someone's life. -:

Yours truly,
Phil

[Phil Allison]

"Andre Jute"


Dr Otala is a distinguished scholar and designer of highly rated audio
gear.


** Completely false.

Here is the man's CV.

http://www.kauppakorkea.fi/tutu/FUTU...nglCVotala.htm

" His specialities are Management of Change, Business Process
Re-Engineering, International Competitivity, total Quality Management, total
Quality Management, and Learning Organisations. "


The amp designs attributed to him ( for marketing purposes) were the work of
many people, unfortunately, none of them were competent audio designers.




........ Phil