traderbam said:
The classic Class AB amplifier is regarded as having 3 stages,
an input differential stage that used properly only has current
gain, a voltage stage that effectively has all the voltage
gain, and a buffer output stage with only current gain.
In this context GM's circuit is a two stage amplifier, as is JLH's
classic class A design. Here the voltage gain and driver stage
are combined with a "phase splitting" transistor, very similar
to standard valve amplifier practise. The White cascode follower
class A push pull output stage is also based on valve practise.
The 3 stage circuit is certainly not inherently stable
without correct compensation arrangements.
According to this :
http://www.tcaas.btinternet.co.uk/jlhtrans.htm
Neither is a two stage design.
http://www.tcaas.btinternet.co.uk/jlhesl.htm
For the 2 ohm Quad ESL version of the JLH, I believe GM's circuit is a development of this.
sreten.Attachments
sreten said:Here the voltage gain and driver stage
are combined with a "phase splitting" transistor, very similar
to standard valve amplifier practise.
there is an amazing similarity between the JLH circuit and the Willliams valve amp. I suspect JLH based its design onthe Williams' design (JLH mentioned a lot of it in his 1969 write-up).
I guess those who don't believe voltage devices can work in the JLH circuit would not believe that the Williams design would work because of its use of all voltage driven devices.
I sometimes wonder that had FETs been available back then, would JLH have used BJTs in his 1969 article.
To LineSource:
You have told us, that you wish to use this amp for a ribbon and you stated that it had 0.9 ohms resistance. I would like to know, if this resistance is constant in its frequency range or does it change? In other words; can you provide us with an impedance curve ( 20 - 20kHz)?
You have told us, that you wish to use this amp for a ribbon and you stated that it had 0.9 ohms resistance. I would like to know, if this resistance is constant in its frequency range or does it change? In other words; can you provide us with an impedance curve ( 20 - 20kHz)?
hi graham, I’m having a hard time seeing "1st cycle distortion" as anything more than
a sneaky way to demand “ridiculous” bandwidth, perhaps you have a methodology that doesn't include the frequency components from the discontinuity at t = 0?
I just put a 10 KHz sine into a 4uS RC low pass filter:
4 uS RC--------.four “total harmonic distortion”
.tran 0 100u 0 10n--------3.505198%
.tran 0 104u 4u 10n-------1.286445%
.tran 0 129u 29u 10n-----0.016112%
400 nS RC
.tran 0 100u 0 10n--------0.056223%
I think it's pretty clear that a RC cannot produce any actual harmonic distortion on its own – the different start times in the .tran above show how quickly the # change as the t = 0 discontinuity is allowed to decay
I’m not suggesting that a 4uS RC would be inaudible, just using it to illustrate the point that what you’re measuring with "1st cycle distortion" is amp bandwidth, not harmonic distortion – or some “slow negative feedback” artifact
Thanks for your patience, don’t mean to sound like we’re “piling on”
ps: anyone know how to get tabs to work?
a sneaky way to demand “ridiculous” bandwidth, perhaps you have a methodology that doesn't include the frequency components from the discontinuity at t = 0?
I just put a 10 KHz sine into a 4uS RC low pass filter:
4 uS RC--------.four “total harmonic distortion”
.tran 0 100u 0 10n--------3.505198%
.tran 0 104u 4u 10n-------1.286445%
.tran 0 129u 29u 10n-----0.016112%
400 nS RC
.tran 0 100u 0 10n--------0.056223%
I think it's pretty clear that a RC cannot produce any actual harmonic distortion on its own – the different start times in the .tran above show how quickly the # change as the t = 0 discontinuity is allowed to decay
I’m not suggesting that a 4uS RC would be inaudible, just using it to illustrate the point that what you’re measuring with "1st cycle distortion" is amp bandwidth, not harmonic distortion – or some “slow negative feedback” artifact
Thanks for your patience, don’t mean to sound like we’re “piling on”
ps: anyone know how to get tabs to work?
Hi pinkmouse,
I find it ironic that non abusive language content will be pulled, and yet erroneous technical content will be left.
A rowdy class is not necessarily bad if we all become educated, myself included.
Manners eventually improve, understanding not necessarily so.
Hi Rtirion,
None of us can realise what we do not know, but I've had enough of those who try to shoot me down by building walls of questions, instead of thoughtfully progressing one step at a time.
I really am wasting my time here.
Please contact me on
graham.maynard1@virgin.net
for the original the GM.25-8 circuit.
I do not wish to place this circuit in the public domain prior to publication.
Hi sreten,
All of my circuits follow the non-complementary bipolar output as first published by JLH.
John knew what he was writing about.
I first built a triple push-pull 50W-8 ohm just after the original 69 article came out. It used two mirrors plus CCS, was rebiased to 6 ohm, and then a rather hot but still reliable 4 ohms. It was very clear sounding and is right beside me as I write this more than 30 years later, though one of the psu elecrolytics has recently left a puddle.
JLH himself wrote about 90 degree phase shifts per device, but whether they have an impact was not covered !
You have attached the circuit suggested by Jos in post#34.
R4 and R8 are the best compromise for the single ended input, but provide insufficient pulldown for really good high frequency response at large amplitude.
Better results are also possible by reducing R6 and R7 to 47R and 560R.
A differential stage however further reduces distortion by a factor of ten, and I am sure that Rtirion has seen that the stability margin of the 'One ohm' circuit remains excellent, with both of the 90 degree stage turnover poles being clearly defined, and no minor poles showing at all, as with the circuit you have just posted.
My use of the words inherently stable does not mean that the amplifier cannot be made to oscillate due to load induced phase change interacting with the amplifier's first 90 degree phase change. I have not implied any meaning that is different to what I have stated.
Electrostatics are hard to drive cleanly. An amplifier which suits them is not automatically best suited to driving all other types of loudspeaker. However any good amplifier should be able to drive electrostics if the transformer has sufficient primary dc resistance, or if a low value series resistor is inserted, say 100 to 220 milliohms.
Note that the response of this JLH circuit is softened by the inclusion of an input filter which might increase stability with electrostatics by reducing the maximum slew rate. I choose to leave all input filtering to the pre-amp or source, and thus enjoy clearer reproduction by not fitting them.
Hi Millwood,
Millwood I was directly invloved in power audio for 25 years both privately and occupationally, also hands-on in my own home interest for forty years. I didn't have the luxury of the simulators which you use. So be careful of the way in which you *interpret* results, stop criticising unnecessarily, and stop trying to imply that I don't understand the information I am clearly failing to communicate to you. I am not alone in holding these specific ideas.
Yes that is an "ideal amp" so that there cannot be any ambiguity of results.
What about the first cycle distortion response of your own circuit which I asked you about ?
Or do you think that that does not count ?
Cheers .............. Graham.
I find it ironic that non abusive language content will be pulled, and yet erroneous technical content will be left.
A rowdy class is not necessarily bad if we all become educated, myself included.
Manners eventually improve, understanding not necessarily so.
Hi Rtirion,
None of us can realise what we do not know, but I've had enough of those who try to shoot me down by building walls of questions, instead of thoughtfully progressing one step at a time.
I really am wasting my time here.
Please contact me on
graham.maynard1@virgin.net
for the original the GM.25-8 circuit.
I do not wish to place this circuit in the public domain prior to publication.
Hi sreten,
All of my circuits follow the non-complementary bipolar output as first published by JLH.
John knew what he was writing about.
I first built a triple push-pull 50W-8 ohm just after the original 69 article came out. It used two mirrors plus CCS, was rebiased to 6 ohm, and then a rather hot but still reliable 4 ohms. It was very clear sounding and is right beside me as I write this more than 30 years later, though one of the psu elecrolytics has recently left a puddle.
JLH himself wrote about 90 degree phase shifts per device, but whether they have an impact was not covered !
You have attached the circuit suggested by Jos in post#34.
R4 and R8 are the best compromise for the single ended input, but provide insufficient pulldown for really good high frequency response at large amplitude.
Better results are also possible by reducing R6 and R7 to 47R and 560R.
A differential stage however further reduces distortion by a factor of ten, and I am sure that Rtirion has seen that the stability margin of the 'One ohm' circuit remains excellent, with both of the 90 degree stage turnover poles being clearly defined, and no minor poles showing at all, as with the circuit you have just posted.
My use of the words inherently stable does not mean that the amplifier cannot be made to oscillate due to load induced phase change interacting with the amplifier's first 90 degree phase change. I have not implied any meaning that is different to what I have stated.
Electrostatics are hard to drive cleanly. An amplifier which suits them is not automatically best suited to driving all other types of loudspeaker. However any good amplifier should be able to drive electrostics if the transformer has sufficient primary dc resistance, or if a low value series resistor is inserted, say 100 to 220 milliohms.
Note that the response of this JLH circuit is softened by the inclusion of an input filter which might increase stability with electrostatics by reducing the maximum slew rate. I choose to leave all input filtering to the pre-amp or source, and thus enjoy clearer reproduction by not fitting them.
Hi Millwood,
Millwood I was directly invloved in power audio for 25 years both privately and occupationally, also hands-on in my own home interest for forty years. I didn't have the luxury of the simulators which you use. So be careful of the way in which you *interpret* results, stop criticising unnecessarily, and stop trying to imply that I don't understand the information I am clearly failing to communicate to you. I am not alone in holding these specific ideas.
Yes that is an "ideal amp" so that there cannot be any ambiguity of results.
What about the first cycle distortion response of your own circuit which I asked you about ?
Or do you think that that does not count ?
Cheers .............. Graham.
Graham Maynard said:
Firstly, read my new sig...
What is erroneous content? I am certainly not a Phd in EE, specialising in analogue electronics, who propheses to know everything, and as far as I know, neither are any of the rest of the mods.
As with any internet, (or non personal contact of any kind for that matter), it is impossible to acertain any truth merely by what sounds good. All information gathered or points made are subject to further research to prove their veracity, mine included. This site is not, and never can be, the whole truth as regards Audio, but it is a great place to start learning. No technical comment should ever be removed because it may be incorrect, unless it could cause personal risk, because sometimes the only way to learn is by getting things wrong, or it may even be something new and wonderful but as yet not repectable
We mods are not your parents, to guide and protect you from making mistakes, technical or otherwise. Think of us more as the old time cop on the corner of the street, not getting involved in arguments, but if you start fighting, you get a clip round the ear...
Hello all,
I just thought I would mention that I have confirmed what is being said about the phase shift, if you can bear with me during this long-winded explanation of my take on the topic. Some of my early high slew simulation circuits had very considerable phase shift. Yet, my last one with the current mirrors had very little of it, yet its slew rate was not double the earlier ones. However, its phase shift was several times better and the early waveforms were not distorted. I do believe that the early designs, though they measure well if the signal is permitted to stabilize, nonetheless, probably would not sound as good even though they actually measure much lower distortion wise. I think it is true because music is complex, and does not afford the chance all the time for a steady state signal to gain stability.
Yet, if the phase shift is inside a feedback loop, if it is a problem it shows itself as instability and oscillation. so, I tend to think that if a circuit is stable, then any apparent phase shift may not be inside the feedback loop. So maybe there is no real problem after all. ??
So, I think it is also important whether the phase shifting components are inside or outside the feedback loop. If outside, they should not actually increase distortion much so long as they do not interact poorly with the feedback loop--at least distortion other than simple harmonic. Admittedly, I just don't quite have it all settled in my mind about how the audibility of the phase shift is dependent on various factors in the sound system. For example, the type of distortion outside feedback loops may be more simple harmonic distortion when using multi-way and or multi-amplified speakers, yet it may not be remediable because simple placement of the speaker drivers in relation to each other and the listener may have even more effect even if the all signals going to them are perfectly in phase. Yet, I have an impression that the phase shift inside a feedback loop is more noxious since it has a chance to churn around the circuit many times as the signal becomes infinitesimally small after a certain number of passes. Well, these ramblings are pretty much my opinion. I can't say it is very accurate in this case, just that I do think it has some type of effect which may be audible.
I just thought I would mention that I have confirmed what is being said about the phase shift, if you can bear with me during this long-winded explanation of my take on the topic. Some of my early high slew simulation circuits had very considerable phase shift. Yet, my last one with the current mirrors had very little of it, yet its slew rate was not double the earlier ones. However, its phase shift was several times better and the early waveforms were not distorted. I do believe that the early designs, though they measure well if the signal is permitted to stabilize, nonetheless, probably would not sound as good even though they actually measure much lower distortion wise. I think it is true because music is complex, and does not afford the chance all the time for a steady state signal to gain stability.
Yet, if the phase shift is inside a feedback loop, if it is a problem it shows itself as instability and oscillation. so, I tend to think that if a circuit is stable, then any apparent phase shift may not be inside the feedback loop. So maybe there is no real problem after all. ??
So, I think it is also important whether the phase shifting components are inside or outside the feedback loop. If outside, they should not actually increase distortion much so long as they do not interact poorly with the feedback loop--at least distortion other than simple harmonic. Admittedly, I just don't quite have it all settled in my mind about how the audibility of the phase shift is dependent on various factors in the sound system. For example, the type of distortion outside feedback loops may be more simple harmonic distortion when using multi-way and or multi-amplified speakers, yet it may not be remediable because simple placement of the speaker drivers in relation to each other and the listener may have even more effect even if the all signals going to them are perfectly in phase. Yet, I have an impression that the phase shift inside a feedback loop is more noxious since it has a chance to churn around the circuit many times as the signal becomes infinitesimally small after a certain number of passes. Well, these ramblings are pretty much my opinion. I can't say it is very accurate in this case, just that I do think it has some type of effect which may be audible.
Graham Maynard said:
I was referring to what you said the second trace was of. And with respect to the second trace, you said two different things. First you said the second trace was "bass driver output" and then you said the second trace was "V at bass driver."
Thinking for myself isn't very helpful when it comes to trying to understand what it is that YOU are trying to say.
And my understanding of what you have written can only be as good as what you have written. And some of what you have said isn't very understandable. And I'm not sure whether that is due to a lack of understanding on my part or a lack of communicating on your part.
That's why I ask questions.
Would you really rather I try second guessing what is is you're trying to say than to get you to clarify what it is you're trying to say?
But it is possible for you to clarify what it is you're trying to say.
Well, when you use words like "reflections" instead of "ringing" I'm left wondering if the gaps in understanding aren't on your end.
Again, that's why I ask questions, so I can try and get a better idea of what you're trying to say.
Actually I think it's your imagining what happens from one nanosecond to the next that's leading you to some erroneous conclusoins here vis a vis your "fist cycle distortion."
se
Graham Maynard said:
Shoot you down by building walls of questions instead of thoughtfully progressing one step at a time?
You mean questioning is NOT part of thoughtful proression? To expect a lack of questioning is NOT thoughtful progression but thoughtless obedience.
If you're looking for a place where all your pronouncements are blindly accepted without question, then I guess you are wasting your time here.
se
Graham Maynard said:
what is erroneous or not is quite subjective so it is hard to get your standards accepted universally.
Graham Maynard said:
I am not sur if anyone is trying to shoot you down: nobody here is that important for anybody else to try to shoot them down. you included.
Graham Maynard said:
I am sure you will pick up a few things here and there if you look hard. And old eastern saying has it that out of any three stupid persons, at least one of them must be able to teach you something new.
Graham Maynard said:
I am sure you are one intelligent person, as the rest of the forumers. But I am one of those who respect more knowledge and give zero benefit to your background or years in the industry. Your words stand alone on their own merits and if they are wrong, they are wrong regardless how many amps you have built or how many articles you have published. I don't admire you or anybody else for that matter but I have tremendous amount of respect for good work.
Graham Maynard said:
you apparently do have access to simulators otherwise how did you simulate the ariel speakers?
Graham Maynard said:
I didn't interpret anything. I was trying to ask questions sot hat I can understand your charts and words.
Graham Maynard said:
sometimes we think we know what we are talking about but that doesn't always turns out to be true.
Graham Maynard said:
It is entirely possible that a group of people are wrong on an given subject. and i don't see how you are immune to that.
Graham Maynard said:
That's interesting. If it is indeed an ideal amp, how did you conclude from this simulation that a class a amp will have problem driving the ariel when in fact your simulation doesn't involve any class a amp at all?
Plus, if it is an ideal amp, its output is perfectly in sync with the input signal. And from where the feedback is picked up, there is no "error" to talk about.
Graham Maynard said:
I am not sure what you meant and mean by "first cycle distortion" as certainly it is not possible to do an fft using just the first cycle waveform. So where do we stop then? the longer we sample the waveform for fft, the less effect we get for the first cycle. I guess you will have to define "first cycle distortion" better for us.
Where does is that knowledge supposed to come from?
"But I am one of those who respect more knowledge and give zero benefit to your background or years in the industry."
Yeah! What in world do you expect to learn by doing something for 30 or 40 years. Knowledge is transcendental * and what good is formal training and job experience. Anyone who can't explain a subject, that took years to gain experience in, to someone with the very basics in 4 or 5 post on a forum is an outright fraud! Shame on you and others of your ilk.
* I wonder if there is such a thing as transcendental stupidity.
"But I am one of those who respect more knowledge and give zero benefit to your background or years in the industry."
Yeah! What in world do you expect to learn by doing something for 30 or 40 years. Knowledge is transcendental * and what good is formal training and job experience. Anyone who can't explain a subject, that took years to gain experience in, to someone with the very basics in 4 or 5 post on a forum is an outright fraud! Shame on you and others of your ilk.
* I wonder if there is such a thing as transcendental stupidity.
Hi jcx.
Sincere thanks for trying. Lateral thinking..........
You appear to indicate that if you start the fourier transform examination at a period later than t=0, then the error reduces.
However the waveform starts to change from t=0 and does not become steady until after approximately 4.5x the group delay period for the input filter, or an amplifier's phase shift equivalent.
If you start the waveform examination after any given delay period you ignore the exponential amplitude distortion that has already occurred, this with a level that varies exponentially with first cycle input frequency or rate of change.
A filter or amplifier uniquely distorts a first cycle waveform, and we listen to these more than others for our auditory cues.
My circuit to LineSource does not distort first cycles more than 0.01%, which is especially useful for high audio frequency driving.
Hi Boholm
Is it not the case that the impedance is purely resistive to beyond AF.
See the Visaton driver spec.
Hi subwo,
As Fred says, there is a ridiculous amount of background noise here that nobody does anything about, so I'll e-maill you directly.
Hi pinkmouse.
When you clean up strings by pulling out the bad attitude stuff, you let the intemperate keyers know that they can get away with writing what they want. There is no reason for them to stop doing it because they know you will wipe their tracks clean.
Your policy is unfair on those who make genuine technical effort.
Is there any way that complaints can be made about individuals ?
Cheers ............ Graham.
Sincere thanks for trying. Lateral thinking..........
You appear to indicate that if you start the fourier transform examination at a period later than t=0, then the error reduces.
However the waveform starts to change from t=0 and does not become steady until after approximately 4.5x the group delay period for the input filter, or an amplifier's phase shift equivalent.
If you start the waveform examination after any given delay period you ignore the exponential amplitude distortion that has already occurred, this with a level that varies exponentially with first cycle input frequency or rate of change.
A filter or amplifier uniquely distorts a first cycle waveform, and we listen to these more than others for our auditory cues.
My circuit to LineSource does not distort first cycles more than 0.01%, which is especially useful for high audio frequency driving.
Hi Boholm
Is it not the case that the impedance is purely resistive to beyond AF.
See the Visaton driver spec.
Hi subwo,
As Fred says, there is a ridiculous amount of background noise here that nobody does anything about, so I'll e-maill you directly.
Hi pinkmouse.
When you clean up strings by pulling out the bad attitude stuff, you let the intemperate keyers know that they can get away with writing what they want. There is no reason for them to stop doing it because they know you will wipe their tracks clean.
Your policy is unfair on those who make genuine technical effort.
Is there any way that complaints can be made about individuals ?
Cheers ............ Graham.
Charles Hansen said:
So I would suggest we return to the original point of this thread. I said that he needed +/- 10 volt rails. Graham suggested he needed higher rails, and proceeded to offer his reasons. Even if I disagree, there is something there for me to learn.
don't you think, if we take your advice, that he should have posted in another thread that it needs higher rail?
Charles Hansen said:
that doesn't mean others cannot.
Graham Maynard said:
You can make complaints about individual posts. Each post has a "Report" button at the bottom, right next to the "Quote" button. When a particular post is dragging a thread off topic, this is usually an effective way to keep things on track. The moderators will often split posts or move them to "Texas".
For example, I posted that the 25 watt class-A amp could be built with +/- 10 volt rails in the output stage. Graham replied that he would use higher rail voltages. When questioned, he gave his reasons. But some have kept challenging Graham's posts in a negative fashion. These type of argumentative posts should be split off by the challenging poster, not the moderator into a separate thread. But if someone persists in degenerating a particular thread, try using the "Report" button.
Graham Maynard said:
jcx was talking about a single pole low-pass filter in his original post. Its response to a sine wave starting at t = 0 (and zero for t < 0) can be computed easily without SPICE at all. Just take the Laplace transform of v(t) = V0 sin(omega*t) and multiply it by the transfer function of a single pole low-pass filter, then do a partial fraction expansion of the product. Taking the inverse transform of the result, you'll find that the output consists of the sum of two terms.
1) A decaying exponential whose time constant is the reciprocal of the cutoff frequency of the low-pass in radians per sec.
2) A constant amplitude sinusoid whose phase is shifted from that of the input by an amount that depends on the filter cutoff frequency and the frequency of the sinusoid.
Your use of the term "amplitude distortion" implies that the amplitude of the sinusoid is being modulated in some way, that is, it's not constant with time. That's not true at all. The fact that distortion shows up in SPICE simulation is just an FFT/SPICE artifact, because it's catching the exponential from the transient response of the circuit.
Millwood, Sreten and Steve,
You have filled LineSource's thread with barrages of mostly non-technical and irrelevent comment personally directed at my presentation. ( You don't kill messengers if you cannot understand their accent )
If you cannot challenge the content of my writing on a technical basis, then you are not helping anyone. I am not obliged to you.
I write thanking other thread writers for more direct e-mails, and in agreement I will have to ignore screens full of silly questions and comment from now on.
Thank you Charles Hansen for your feedback on that.
Millwood,
You keep bringing up Mosfets for the JLH circuit.
I said they will not work.
Due to their gate-drain capacitance, which don't forget wrt resistor fed gate drive is dynamically amplified by the Mosfet stage gain - which is also influenced by the load (loudspeaker) and *its* reactivity, the JLH circuit, which naturally amplifies treble transparently, is turned into an unstable three stage amplifier.
The varying Mosfet gate capacitance upsets the current splitter by causing hf imbalance. The original JLH was slightly affected by bipolars, that is why I use low value resistors in the circuit I submitted to LineSource.
The stabilisation components that become necessary with Mosfets then introduce high frequency first cycle distortion, which audibly affects transient reproduction, and can be measured by Fourier analysis using a simulator.
No.
The Williamson does not work anything like the JLH.
In its phase splitter the anode resitance went to a low impedance voltage supply to optimise voltage splitting.
The collector resistor in the JLH splitter has a high impedance source point, for current splitting.
In the Williamson there is a differential triode stage following the phase splitter to further improve the and match the voltage waveforms - AND - isolate the triode connected tetrode's grid-anode capacitance, which is also influenced by load reactivity, from the push-pull voltage phase-splitter itself.
When you tell me that you can make the JLH circuit reproduce first treble cycles more accurately using Mosfets, than the circuit I have already forwarded to LineSource, only then will you prove me wrong.
Cheers .............. Graham.
You have filled LineSource's thread with barrages of mostly non-technical and irrelevent comment personally directed at my presentation. ( You don't kill messengers if you cannot understand their accent )
If you cannot challenge the content of my writing on a technical basis, then you are not helping anyone. I am not obliged to you.
I write thanking other thread writers for more direct e-mails, and in agreement I will have to ignore screens full of silly questions and comment from now on.
Thank you Charles Hansen for your feedback on that.
Millwood,
You keep bringing up Mosfets for the JLH circuit.
I said they will not work.
Due to their gate-drain capacitance, which don't forget wrt resistor fed gate drive is dynamically amplified by the Mosfet stage gain - which is also influenced by the load (loudspeaker) and *its* reactivity, the JLH circuit, which naturally amplifies treble transparently, is turned into an unstable three stage amplifier.
The varying Mosfet gate capacitance upsets the current splitter by causing hf imbalance. The original JLH was slightly affected by bipolars, that is why I use low value resistors in the circuit I submitted to LineSource.
The stabilisation components that become necessary with Mosfets then introduce high frequency first cycle distortion, which audibly affects transient reproduction, and can be measured by Fourier analysis using a simulator.
No.
The Williamson does not work anything like the JLH.
In its phase splitter the anode resitance went to a low impedance voltage supply to optimise voltage splitting.
The collector resistor in the JLH splitter has a high impedance source point, for current splitting.
In the Williamson there is a differential triode stage following the phase splitter to further improve the and match the voltage waveforms - AND - isolate the triode connected tetrode's grid-anode capacitance, which is also influenced by load reactivity, from the push-pull voltage phase-splitter itself.
When you tell me that you can make the JLH circuit reproduce first treble cycles more accurately using Mosfets, than the circuit I have already forwarded to LineSource, only then will you prove me wrong.
Cheers .............. Graham.
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