| lumanauw |
I observed (maybe wrong), that a power amp with single differential gives more focus in sound reproduction. It is not about nice to listen to, but more focus.
I bought Douglas Self's book and asked him why he didnt write at all about power amp with dual differential (transistor pairs, 1pair npn and 1 pair pnp). He answered that he suspected that dual differential power amp is not as linear as single differential amp.
Is it true that single differential amp (like aleph) is better than dual differential amp (most of the pro-audio amp, like crown, crest, uses this dual differential).
What is really the different between building power amp with single differential and dual differential? Maybe both can produce music that is nice to listen to, but is single differential more focus (more linear, like Douglas Self said?) |
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| lumanauw |
The first schematic (Kaneda) is single differential (TR1,2), with upper cascode (TR3,4). The second schematic (schdetail.gif) is dual differential (maybe I should say complementary differential) the upper is NPN (TR1,2), and lower is PNP(TR3,4).
Thanks for the example. This is exactly what my question is all about. Single differential (like kaneda) VS complementary differential (like schdetail.gif). |
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| PRR |
> single differential gives more focus in sound reproduction.
This has to be an over-generalization. Either type can be good or bad.
The choice is mainly about the fact that a NPN-PNP design can give outputs at both rails, which affects design of the second stage. It may be better, it can be worse.
Anyway: The difference you describe probably has a lot to do with driver and output stage linearity, more than input topology.
> Douglas Self... he didnt write at all about power amp with dual differential (transistor pairs, 1pair npn and 1 pair pnp). He answered that he suspected that dual differential power amp is not as linear as single differential amp.
I'm not sure I agree that it is worse, but for the levels used in audio input stages it should be about the same.
If the effective input voltage, under feedback, rises from a milliVolt to over 20mV, some types of NPN-PNP complementary diff-pair can hold "linearity" a little further than a single pair. But operation in that range isn't the "very-good linearity" we expect in audio. It may be useful in comparators and other mainly non-linear devices which need a semi-linear range for smooth settling. |
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| lumanauw |
It is true that complementary differential (pair of npn for +voltage and pair of pnp for -voltage) makes more sense if we see them. It makes the whole circuit more reasonable, and more symmetrical. Upper differential is working for all upper transistors, and lower differential is working for the negative half. Especially if we look at high power amplifier (pro-audio) where the rail can reach more than +/-100VDC, maybe this complementary differential is the only answer, to divide the working voltage (Vce) only from +rail to ground, not to -rail. This is due to the dissipation limitation with certain current we must handle (and also limitation from maximum Vce for transistors)
But when I notice the comment from Mr.Douglas Self, and see the design of Mr.Nelson Pass, this question rises in my mind. Mr Pass once told me to read the article he wrote about how mosfets works. In his paper, I learned that N mosfet and P mosfets have very different Vgs thereshold. And when I read the history of bipolar transistors, I also learned that PNP and NPN are very different. NPN is what we found first. It is the nature of transistors. PNP is found long time after NPN, and using very different moulding from NPN. We can see from datasheets, especially mosfets (like IRF540 and IRF9540), the N mosfets always superior in datasheet than P mosfets (with the same shape).
In power amp, differential stage is what I think the most important stage in shaping the output. It works with small signal, and everything began here. So every little difference or mismatch will be sighted as all the signal amplifies. We can cover some of this mismatch by putting feedback, forcing to fix the errors. But the question is "which is better?"
So if the NPN and PNP are not the exact oposite (with exact behavior) will my original question here makes sense? |
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| PRR |
> high power amplifier (pro-audio) where the rail can reach more than +/-100VDC, maybe this complementary differential is the only answer, to divide the working voltage (Vce) only from +rail to ground, not to -rail.
Nope. Same voltage either way.
> the history of bipolar transistors, I also learned that PNP and NPN are very different. NPN is what we found first. It is the nature of transistors. PNP is found long time after NPN
Sounds like a highly simplified history. From 1954 to 1964, THE most common transistor was the Germanium, and 99% of those were PNP, from computers to telephone switchgear to car and pocket radios. The reason "all" cars (not just Cadillac) went to 12V Negative Ground was so we could bolt the collector of a monster Ge PNP to the radio case for cooling. Then came Silicon, which is slightly easier to make in NPN. But both polarities of both materials were available from the beginning, and there is an exhaustive survey of complementary topologies in a book I have that was published just a few years after the transistor was invented.
But history does not matter. Yes, a PNP BPJ has slightly higher ohmic loss and Vbe than an NPN BJT of the same size, but the difference is very small and you can find pairs of devices where the difference from NPN to PNP is less than the difference from one transistor to the next of the same type. In MOSFETs the difference is larger, enough so you "never" use a P-type for switching if you can avaoid it. But in linear or AB amplifiers, you always add some fixed resistance (or other feedback) to set bias current and reduce nonlinearity. This has to be larger than the devices' own internal resistance, so the difference between the P-side and the N-side becomes very small.
> differential stage is what I think the most important stage in shaping the output. It works with small signal
There are many ways to look at amplifiers, and you do have to look at the design from all directions.
But in one big way, the input stage is the "easiest". The output devices have to swing current from 0 Amps to 5 Amps, and large voltage swing too. The input devices can be designed (by having plenty of current gain in the whole amp) to swing only from 99µA to 101µA with nearly constant voltage (and low voltage if you want). While they have a very critical job, they do not have to carry the strain of pushing big current or even large current changes. In principle you can make the signal across the input pair(s) as small as you like, and push distortion down to the vanishing point and accuracy up as high as you could ever hope for. |
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| john curl |
You folks are worrying too much. The complementary differential has lower distortion, all else being equal. If you build a design like the comp symmetry example shown, just put a large cap across the second stage base to the nearest supply. Either side will do. I found an increase of distortion of 5 times in an example that I measured more than 30 years ago.
Think it through. First you lose GAIN (6db) because you have only one working input stage. Second, you INCREASE even harmonic production, because you are not equally driving the second stage transistors. Try it and see. Don't worry about the intrinsic mismatch in N vs P transistors or fets, it still is better to use both together. |
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| PRR |
> You folks are worrying too much.
Who is worrying? lumanauw has an idea that one topology is better than another. In the hands of a skilled experienced designer, this may be true (though all-else is never equal). But at lumanauw's apparent level of learning, it is important to realize that no topology is "unavoidably" better than another. We can easily design a "dual differential" that is bad (heck, I've done that), and a "single differential" that is better. Many sing-diff inputs work very well. Complementary diff has some "obvious" advantages, but they can be botched with careless detailing. lumanauw's theory may be right. But he needs to understand that the devil is in the details.
That's where Doug Self is useful. He has worried the Classic Topology to the bone, and pointed out many "small" details where most designs could be significantly better. The concept and many of the details can be applied to other topologies. But you have to learn them before you can apply them. Self teaches this well. He gets fabulous performance out of something that looks like a 1972 Fisher. We can criticize that he works mostly by measurements, not by ear. But his details look like things that "should" be better to the ear, and others with golden-ears can judge his work for themselves. There are details where I disagree with his theory, but he is honestly investigating and sharing which is all anybody can do.
> you lose GAIN (6db)
Too true. Though gain in BJT is cheap. And with overall feedback and slow output devices, gain has to droop inside the audio band to keep the loop stable. That -tends- to lead to a design where almost all gain is made in one stage, usually the second stage, so the total gain can be compensated to one-pole response. With the huge potential gain of BJTs, this usually means deliberate reduction of input stage gain with emitter resistors.
FETs is different. Gain is harder to come by. In a sense, they already have the "emitter resistors" built-in that we add to many BJT stages. If you can find semi-matched complements (I sure agree that small mismatching is no big deal) then all-complementary is one preferred topology.
> you INCREASE even harmonic production, because you are not equally driving the second stage transistors.
Less if you detail so the standing current is very-large compared to the signal current (base current in the output stage). This is one place where many BJT designs come up a dollar short.
I'm also, at my current level of understanding, becoming less concerned about THD and more about distortion spectrum. A spectrum with no even-order is un-natural. If the slope of the spectrum is steep and/or all products are far below system noise, that won't matter. And I'm not sure I can equate lumanauw's observation "more focus" with distortion spectrum. |
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| john curl |
I would like to make the case for complementary differential topology:
It isn't the INPUT stage that is lowered in distortion, it is the 2'nd stage which is usually single ended and has to develop almost all the gain for the amp, which improves. This problem was first addressed with 'bootstrapping' using a cap connected to the output of the amp to give positive feedback and increase the driver load impedance. The next approach was to use a constant current source as a load, favored even today by Doug Self. Finally, the equal driving of both driver transistors, either with a current mirror, or with a complementary differential input.
I have used each of these approaches over the last 35 years, and personally, I prefer the complementary differential fet input. Don't tell Doug Self, but fets actually work darn well as input stages, and have many advantages, such as no need for an input capacitor, and very high slew rate operation, without any noise tradeoff. Also, they tend to be more RFI resistant, because their input diode is off, rather than conducting.
While I have the greatest respect for what Doug Self has published, please don't box yourself in a corner by thinking that that his input is the only or necessarily the best approach to circuit design. |
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| AKSA |
This is an extremely well-informed thread. Comments from Jonathan, PRR and John are outstanding.
I agree strongly with PRR's comment about distortion spectrum and its affect on musical perception. Distortion spectrum is, in my opinion, the crucial factor; you want something with maximal H2, progressively decreasing with increasing harmonics, and least, if not zero, from H6 onwards. The total distortion figure is not too important, though intermodulation performance is VERY important. The significant thing is to have a distortion spectrum which is not too dissimilar from that found in the natural world. High levels of odd order, with evens missing, or very small, is not natural sounding.
A complementary dual diff will greatly minimize H2, H4, H6 - even order - while not much increasing H3, H5, H7 - odd order. A single diff will have quite high H2, some H3, some H4, and some H5. This observation is counter-intuitive because a complementary dual diff 'looks' right. The point concerning driving a fully symmetrical voltage amplifier (more correctly described as a transresistance amp as it converts current input to voltage output) is well taken. There are linearity advantages to a fully complementary VAS, but, as before, even order harmonic generation is nulled.
I agree with John that pretty much any topology can be made to sound good. The schematic is just the beginning; it takes careful dimensioning, component choice and layout to produce good sonics, but it is possible voice almost anything to sound good. Good sonics are probably equal part topology, dimensioning, component choice and layout, and a huge amount of work is required to make it happen. You will NEVER know by simply examining the schematic, any more than you can definitively gauge the emotional impact of a symphony by studying the score. My personal take is for a progressively reducing harmonic spectrum, adjusted so H2 predominates, but with very careful 'voicing' which is really only achievable with lots of listening. It is also true that some people like the SE sound, while others the PP sound. This is even true of Class A versus AB; for myself, I prefer Class AB as it seems somehow more dynamic and lifelike, at least in my designs. You can't please 'em all, and you'll die trying!
We tend to concentrate on the technology in straight electrical terms, but much work remains to be done on the psycho-acoustic phenomenon, the subjective aspects of why we like what we like, and why we tend to become religious zealots about it. More work is also required in the area of harmonic spectrum, particularly as it relates to musical scales, and profiling musical instrument tones. Much of this work is actually known, but for some reason it does not seem to be found in the audiophile and designer communities.
Cheers,
Hugh |
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| lumanauw |
Mr.John Curl, I agree that anyone cannot take only 1 resource as the only right thing, especially in audio electronics. With more discussion and learning from various experienced people we can get closer to what is the right fact. Actually I always wanted to try to built fet differential, like using k389 or j109. But they are difficult to get here. The best I can get is k30, but that is only 30vmax. Do you have any example of your preferred fet differential input for me to see?
Mr.PRR. I'm interested in using PC's computer card as Harmonic distortion analyzer. In another thread you have mentioned it, but for me it is not clear enough. Can you give me any url on how can I built myself a distortion analyzer with PC's card? Is there any additional software needed?
I'm getting confused when people talks about 2nd order, 3rd, 4th, 5th order harmonics, etc.
Are they (every single harmonics) visible at the pc if we use your arrangement of PC distortion analyzer? Or is it just numbers from calculation? |
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| Steve Eddy |
| quote: | Originally posted by john curl
Let me help with this even-odd topic:
'Science and Music' Sir James Jeans 1937 p. 87 This book is available through Dover.
"... The seventh harmonic, however, introduces an element of discord; if the fundamental note is c', its pitch is approximately b (flat) ''', which forms a dissonace with c. The same is true of the ninth, eleventh, thirteenth, and all higher ODD-numbered harmonics; these add dissonance as well as shrillness to the fundamental tone, and so introduce a roughness or harshness into the composite sound. The resultant quality of tone is often described as METALLIC"
Well folks what do you think that this means? |
What it doesn't mean is that we're more sensitive to odd-ordered harmonics, which is what Hugh had claimed. Please read what he wrote:
This argument is largely based on the fact that the ear is not sensitive to large levels of H2, up to about 2%, yet hypersensitive to odd, higher orders, such as H7 (around 0.05% according to studies I have seen of the differences between soft and hard trumpet sounds.)
He's not saying that odd-order harmonics sound different compared to even-order harmonics. He's saying that we're more sensitive to odd-order harmonics compared to even-order harmonics.
| quote: | | This is not the first time that I have quoted this passage over the years, but it just gets ignored by those who would not learn from it. |
Well I'm ignoring it here because it hasn't anything to do with the issue in question. And the issue isn't whether odd-order harmonics sound different from even-order harmonics, but rather whether odd-order harmonics are more audible compared to even-order harmonics.
| quote: | | SE has told me, in print, to sell my books, since my personally having them is a waste of time and space. Here is one exception to his assertion. :cool: |
Yes, I'd said that they're a waste of space because of your habit of supporting your arguments by telling people how many books you have on the shelf (i.e. "You're wrong! I've got 200 books on my shelf so I know what I'm talking about!") instead of actually quoting something from those books to support your argument.
So here I'll give you credit for actually quoting something from those books, but unfortunately what you quoted hasn't anything to do with the question at hand.
se |
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| jcx |
| quote: | Originally posted by Sawzall
But anyway, accepting the evidence as presented, what effects would the various topology's have on the error spectrum? That really is the underlying question here - and the one this neophyte would like to understand better. |
I'm afraid that this question has too many dimensions to be answered in any useful way, I would emphasize PRR's and others comments that implementation "details" totally trump any generalizations drawn from simplistic heuristics and visually pleasing symmetries in the schematic representation
Over the years many designers have made different choices of details in the differential input stage alone; Fet vs Bjt, degeneration, cascodes (Fet, Bjt, bootstrapped, folded), current source vs R bias, current mirror vs resistive load, device part #, manufacturers, matching – any of which could be considered a “detail” of implementation that doesn’t rise to the level of the question of whether to use them in a single or complementary differential input
for a "light" (if not ultimately enlightening) discussion:
http://www.diyaudio.com/forums/show...t=&pagenumber=1 |
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| PRR |
> harmonics created by the machine perform minimal spectral shift on the sounds which are being processed.
Not that simple.
The real harmonics are in the ear. Huge levels of 2nd harmonic. It is irrelevant how much 2nd an amplifier makes. Even on clarinet. And as Steve points out, musical instrument harmonics are not perfect multiples of the fundamental or each other (piano is worst, but even a clarinet has inharmonic overtones).
I don't think that cancelling the even-order harmonics is good or bad in itself. Every-harmonic, or every-other-harmonic, doesn't seem to make a lot of difference. Curl suggests that he used to null even-order, probably because "he could", and has shifted away from that, perhaps because it makes little difference to his designs and his ears, or he finds other benefits in not-nulling the evens.
What does seem to matter is getting the spectrum to slope steeply, and/or get it well below noise level.
No-feedback triodes have a steep slope of distortion spectrum. It roughly parallels the ear's own distortion and tends to be masked.
High-feedback amps (any technology and topology that allows really high feedback) can get distortion well below system noise where it vanishes. (It has to be well below system noise; we hear tones 10dB or so below broadband noise.) That works well enough in "small" amps which can be run well below clipping. In large power amps, transient overload non-ideal behavior may color the sound even when "perfect on test tones".
> 40dB per octave
That's a steeper slope than I think is needed. I'm not sure the masking threshold is the right guide. But for comparision: simple THD measurements apply NO weighting and are well-known to be inadequate. IMD applies a first-order weighting and often compares better to "sound". Olsen's tests suggest more like a second-order weighting (12dB/8ve, 40dB/decade), and RDF 4th cites a study where this is shown to corellate well, but this seems to be forgotten and certainly needs re-examination on current "very clever" amplifers.
As to whether hollow-state, BJT, FET, etc is "best": aside from some real-life issues (you can't take huge feedback around a tube with the transformer it often wants), I think very-good designs are possible many ways. John argues well for FETs; a hard-core BJT designer could argue against his points. I'm inclined to think that details are more important than devices. And also: that many existing designs have flawed details. Case in point: the many straight and elaborated "Classic BJT power amplifiers" that don't work as well as Doug Self's designs. There may be better amps than Self's, but there are certainly a lot of worse ones out there.
Bootstrapping an AB output.... shudder. I think it was whatzhisname at SWTP who first woke me up to this folly. I mentioned the 1972 Fisher because it was a clean example of the post-bootstrap era.
> what effects would the various topologies have on the error spectrum?
Less effect than the detailing. If a BJT diff-pair is not kept perfectly balanced, distortion won't cancel. Not keeping enough effective current gain does nasty things to BJTs. And while Class AB output stages are a mess, there are a lot of bad Class A designs too. Some of these ills are "easier to avoid" with FETs, but you can't just stick some FETs in a "topology" and get great results.
As for "getting stuck in Doug Self"... While I only discovered Self a few years ago, I've been mucking-around with The Classic Topology for (hmmm...) 30 years, and still find tidbits to chew on in his writings. More brilliant designers have noted all this before and moved on, but for many of us a few years of pondering Doug Self is a good foundation for design in any device or topology.
> 'Science and Music' Sir James Jeans 1937.... "... The seventh harmonic, however,..."
Same thing pointed out by Helmholtz in On the Sensations of Tone written in 1875. Before there were amplifiers and harmonic meters. It is amazing how much he demonstrated, measured and extrapolated without electricity. It is heavy reading, and some of the terminology has changed since then, but if you really are interested in what and how we hear you must start reading it.
Another classic, mostly for musicians, is Benade's Fundamentals of Musical Acoustics.
All these books are shockingly cheap via Dover's reprints. (I don't want to know how much my near-mint 1882 copy of Helmholtz is worth...)
> some folks should just be ignored
At least not over-reacted-to. |
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| Fred Dieckmann |
"Less effect than the detailing"
What an excellent post. I may nominate it for post of the week. There so many considerations in designing a good amp that the choice of the front end topology is a small part of the equation. I have heard several really nice sounding amps that had very different front end topologies. I am building an amp (a friends design) with a very simple and classic topology but with great attention to details. I am very curious what an "oldy but goody" design sounds like with attention to details..... |
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| Steve Eddy |
| quote: | Originally posted by PRR
High-feedback amps (any technology and topology that allows really high feedback) can get distortion well below system noise where it vanishes. (It has to be well below system noise; we hear tones 10dB or so below broadband noise.) |
Yes, we can hear pure tones a good way down below broadband noise. But the music itself is rather broadband and will be dramatically greater than the system noise so the question becomes how deep into the music can we hear under practical conditions?
Under practical conditions we have not only system noise, but the "noise" of the music itself, as well as the ambient noise in the listening room which even in a recording studio is typically around 30dB SPL, not to mention the fact that the typical listening room is anything but anechoic so you also have a rather reverberant acoustic environent to contend with.
And then there are our ears themselves which when exposed to sound have an autonomic "clinching" response to sound such that the very act of listening to music at typical listening levels dramatically reduces their sensitivity which is already reduced due to the ambient noise levels we're constantly exposed to.
For example, entering an anechoic chamber from even a relatively quiet ambient environment, it takes some time before our ears "relax" and fully acclimate to the reduced noise level before you can start hearing all those things you don't otherwise hear.
An excerpt from Everest's Master Handbook of Acoustics illustrates this well:
The delicate and sensitive nature of our hearing can be underscored dramatically by a little experiment. A bulky door of an anechoic chamber is slowly opened, revealing extremely thick walls, and three-foot wedges of glass fiber, points inward, lining all walls, ceiling, and what coul dbe called a floor, except that you walk on an open steel grillwork.
A chair is brought in, and you sit down. This experiment takes time, and as a result of prior briefing, you lean back, patiently counting the glass fiber wedges to pass the time. It is very eerie in here. The sea of sound and noises of life and activity in which we are normally immersed and of which we are ordinarily scarcely conscious is now conspicuous by its absence.
The silence presses down on you in the tomblike silence, 10 minutes, then a half hour pass. New sounds are discovered, sounds that come from within your own body. First, the loud pounding of your heart, still recovering from the novelty of the situation. An hour goes by. The blood coursing through the vessels becomes audible. At last, if your ears are keen, your patience is rewarded by a strange hissing sound between the "ker-bumps" of the heart and the slushing of blood. What is it? It is the sound of air particles pounding against your eardrums.
So now how sensitive are our ears after being exposed for a time to music at average levels on the order of 80-90dB SPL, on top of the ambient noise, and in a reverberant environment?
In any case, I don't believe there is any singular "best" way to go about designing an amplifier. At least in the practical sense, outside of purely technical arguments. Most every approach has its adherents and the subjective tastes and preference of listeners varies considerably which is why everything from highly non-linear single-ended tube amps and highly linear solid state amps such as the Halcros are able to thrive in the same market.
I think it ultimately just boils down to trying things for yourself and decide which you ultimately prefer.
se |
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| john curl |
How about a harmonic weighting factor of: A=(n-1)!/2
This will make higher order harmonics important very quickly. This was found in a '72-73 'Wireless World' article by Bob Stewart, now of Meridian. Works for me! |
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| sully |
| quote: | Originally posted by Steve Eddy
The silence presses down on you in the tomblike silence, 10 minutes, then a half hour pass. New sounds are discovered, sounds that come from within your own body. First, the loud pounding of your heart, still recovering from the novelty of the situation. An hour goes by. The blood coursing through the vessels becomes audible. At last, if your ears are keen, your patience is rewarded by a strange hissing sound between the "ker-bumps" of the heart and the slushing of blood. What is it? It is the sound of air particles pounding against your eardrums.[/i]
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Did some speaker work in an anechoic chamber undergraduate..
That account is accurate, from my experience.. I didn't wait long enough for the hissing sound. But, boy the blood sound is really neat..
Cheers, John |
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| PRR |
> I'm interested in using PC's computer card as Harmonic distortion analyzer. Can you give me any url on how can I built myself a distortion analyzer with PC's card?
RMAA 5.1 release with manual on "Basics of Audio Measurements"
> 2nd order, 3rd, 4th, 5th order harmonics, etc. Are they (every single harmonics) visible at the pc...?
Yes, and audible to the ear. Get a sine wave generator. Put it through an amplifier. At normal levels the output sounds a lot like the input. When you crank it "too high", it sounds like another higher-pitched instrument has joined-in. If that instrument is playing an octave higher (2X frequency) than the fundamental, it is Second Harmonic. If it sounds like 1.5 octaves higher (3X frequency), it is Third Harmonic. Two octaves (4X freq) is Fourth Harmonic.
(Note that Physicists count the Fundamental as "First", while musicians often speak of "partials" counting the octave-up overtone as "first". Most technical literature uses the Physicists' numbering, but you may see it otherwise in musician's and instrument-makers' writings.)
In a single-ended tube amplifier, you typically hear the 2nd come up first, then the 3rd, and if you listen close you can identify 4th, 5th, and higher harmonics. Some push-pull or heavy-feedback amplifiers won't make the even-numbered harmonics no matter how hard you smack them. BUt many-many "push pull" amplifiers have non-push-pull driver stages. Curl cites this as a reason to go all push-pull. If you use the simple push-pull tube amplifier with voltage-amp, split-load phase inverter, and P-P output, with triodes, the driver stage is often making lots of 2nd harmonic while the output's 3rd harmonic is still small.
While it is easiest to hear these harmonics when you smack an amp into clipping, lesser amounts of harmonics are present all the time, and you hear them even if you can't pick them out. Once you hear them "big", it is easier to identify small amounts of distortion. |
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| PMA |
| I would prefer the complementary differential topology. With added current mirrors and cascodes. |
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| SY |
| quote: | | I think it was whatzhisname at SWTP who first woke me up to this folly. |
Dan Meyer?
In any case, it's pretty easy to get the amp-induced harmonics an order of magnitude lower than the audibility numbers being bandied about. Topology is an important factor, but as PRR has pointed out, it's just one factor. If we're designing for ultra-low distortion figures, well, all of this becomes important. If we're doing something easier, like merely designing a box of gain that cannot be detected by ear, we are free to forget about the mice and concentrate on the elephants marching around in the room. |
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| bocka |
.| quote: | | What does seem to matter is getting the spectrum to slope steeply, and/or get it well below noise level. |
I highly agree with this statement.
| quote: | | How about a harmonic weighting factor of: A=(n-1)!/2 |
John,
can you make this formula a little bit clearer to me? It looks nice, but I don't know where to start. Is second harmonic n=2? And what do you think is the max. amount of k2?
In my opinion there are several other important things.
The following statements are just an assumption:
1. The higher the output voltage the higher the distortion may be
2. Distortion should be independant from the load
3. Reversed error spectrum (higher output voltage/power) with decresed distortion is bad. This is typical for many class AB-amps.
4. The higher the output voltage the more higher harmonics may be present in the spectrum
5. Distortion should be independant from frequency
6. Distortion spectrum should not vary when two or more signals appear
7. With coupled rf-noise the the input distortion spectrum and levels should not vary
It's just my opinon but I think this therory is more complex than only viewin' to one (technical and therefore simplified!) signal |
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| john curl |
| How about A[H(n)]=amplitude of the individual harmonics (n) The [ ! ] is a factorial. |
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| hitsware |
I think NP makes a good point in one of his papers for the viability of the single ended approach. (He's speaking of output stages, but the same should hold true throughout the chain):
Sound is single ended I.E. we don't live at 0 p.s.i. The natural state of an acoustic wave is to vary around a positive value rather than to be bipolar...........mike |
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| PMA |
| quote: | Originally posted by hitsware
I think NP makes a good point in one of his papers for the viability of the single ended approach. (He's speaking of output stages, but the same should hold true throughout the chain):
Sound is single ended I.E. we don't live at 0 p.s.i. The natural state of an acoustic wave is to vary around a positive value rather than to be bipolar...........mike |
But this has already been received by the microphone, you do not need to add any more non-symmetrical acoustic pressure distortion ;) |
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| bocka |
The problem is we cannot build a distortion-free amp (with only one exception...)
In this way we should keep the audible distortion level below the hearing threshold level. Unfortunaltely no one know exactly what this means. So we use static signals in order to map the "audible quality" to measurable signals... |
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| hitsware |
>But this has already been received by the microphone, you do not need to add any more non-symmetrical acoustic pressure distortion
But for the electronic device to act in an 'airlike' manner, should it not run the same way. Granted sort of metaphysical but ?????? |
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| PMA |
| quote: | Originally posted by bocka
The problem is we cannot build a distortion-free amp (with only one exception...)
In this way we should keep the audible distortion level below the hearing threshold level. Unfortunaltely no one know exactly what this means. So we use static signals in order to map the "audible quality" to measurable signals... |
Agree, we should examine dynamic spectral distortions and and intermodulations .... |
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| bocka |
| quote: | | Agree, we should examine dynamic spectral distortions and and intermodulations .... |
Yes, I think so, but I fear it's a multi-dimensional problem...
Spectral distortions and and intermodulations at nearly every frequency, aly load, any level. It's gettig (too?) scientific. |
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| PMA |
| That's why we still use our ears, isn't it? ;) |
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| bocka |
| And use MUSIC as test signal? :eek: |
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| PMA |
| Digitize input and output (at the speaker) with say 20-bit resolution, high sample rate, store into great memory and try to compare? |
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| bocka |
Seems a good way. Maybe we can use a (calibrated) differential/instrumentation amp (fed by input and output of the power amp), sample the output of the diff-amp, do a FFT (or better continiously FFTs) over the spectrum and normalise it. Could be very interesting. Maybe the spectrum varies from FFT to FFT. If it does it could be a first clue due to different sound by several amps.
Maybe more tomorow. It's late in Europe |
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| john curl |
Trust me, it is difficult to get anything except a tube or class A FET amp to have an extremely low level of higher order harmonics, especially open loop and over extended frequency.
The reasons are:
Transistors are pretty darn nonlinear, and they have several different distortion producing components. These include: very non-linear Gm (voltage gain), non-linear BETA, and non-linear input capacitance (changes with voltage level on both the collector and base, referred to the emitter).
When you TRY to linearize them with local feedback (series resistor) you convert the even order harmonics into higher order odd harmonics.
If you try to use loop feedback, then you get TIM or FM modulation distortion, i.e. FIM from modulating the open loop bandwidth with amplitude changes with signal level. This is just with class A, Class AB or B is much worse.
It is a difficult problem. This is why we have developed sophisticated topologies in order to minimize the generation of distortion, over the decades.
In any case, the generation of higher order harmonics are not a good idea. |
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| AKSA |
I agree with using music as the ultimate test. It's woolly, in that it's not objective, but since assessment of amp quality is often made en masse in the marketplace by consumers (who are usually less interested in the technology than the 'sound'), it seems logical to design with simple topologies then 'voice' with careful listening tests over large samples to achieve the goal.
You could argue that intense, spec driven engineering has created amps of vanishingly low distortion, but little apparent correlation with 'acceptable' sonics, at least from the point of view of listener (read: consumer) popularity.
PRR's point about a steadily decreasing harmonic structure, maybe around 12dB/octave, sits very well with me. It might also imply that distortion profiles increasing with frequency are not too important as all the harmonics beyond about 10KHz fundamental are inaudible anyway. Further, any harmonic content less than about 80dB below is probably inaudible in the average urban sitting room. I've enjoyed some success with very simple topologies - even including boostrapping the VAS load - by careful attention to layout, diff pair balance, component choice and dimensioning. Speed of the VAS is important, the dead zone at crossover is critical, and lag compensation is crucial.
The very controversy of just why certain designs sound as they do would indicate design detail and empirical refinement are the major factors, just as PRR opines, and music is the real test.
And Fred, I'll be most interested in the results with your 'oldie but goody'. Keep us posted!
Cheers,
Hugh |
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| Nelson Pass |
I'm a bit surprised that no one has mentioned one of the
advantages of single-pair inputs biased by a constant
current source and driving an SE 2nd stage which is loaded
with a constant current source - namely the bias stability that
such arrangements provide. It is easy to make a stable CCS,
and this stabilizes the operation of the gain devices.
Usually the output stage bias is derived from the 2nd stage
DC bias, so further down the chain we also see more output
stage stability as a result.
I'm not necessarily advocating single vs dual diff pairs, as I
use either when I feel like it, but having a convenient CCS
by which you anchor the circuit helps explain one of the virtues
of some earlier solid state amplifiers. I bet a few more Tigers or
Zillas would still be working today if they had taken that
approach.
:cool: |
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| PMA |
| I agree that higher order harmonics should be avoided, even if they are very very low in magnitude. The other important issue is an intermodulation with HF signals and spikes, like D/A conversion residuals at the output of CD players. This is perfectly audible and can be quite effectively minimized by filtration and bandwidth reduction to a reasonable limit, say some 100 kHz. |
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| Elso Kwak |
| quote: | Originally posted by Nelson Pass
I'm a bit surprised that no one has mentioned one of the
advantages of single-pair inputs biased by a constant
current source and driving an SE 2nd stage which is loaded
with a constant current source - namely the bias stability that
such arrangements provide. It is easy to make a stable CCS,
and this stabilizes the operation of the gain devices.
Usually the output stage bias is derived from the 2nd stage
DC bias, so further down the chain we also see more output
stage stability as a result.
I'm not necessarily advocating single vs dual diff pairs, as I
use either when I feel like it, but having a convenient CCS
by which you anchor the circuit helps explain one of the virtues
of some earlier solid state amplifiers. I bet a few more Tigers or
Zillas would still be working today if they had taken that
approach.
:cool: |
Hi Nelson,
You mean Ampzillas?
Mine has constant current sources for the input differential pairs.
It's still working after 28 years! |
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| bocka |
I highly interesting thread...
John, it's interesting that your results are nearly the same as mine. Distortion is mainly produced by the output stage when VAS and diff-pair is designed as Doug Self would call blameless. Most distortion comes nearly from crossover when leaving class A and moving into class AB. This introduces high order harmonics. In this case a class AB produces at least 10 or 20 times the distortion of a well designed class A stage.
Worsed in this case is a CFP, followed by a bipolar EF in class AB. Better is a MOSFET in class A and best the bipolar class A because of the higher transconductance of the bipolars. Unfortunately this is only true when a bipolar is driven by a low impedance. Obviously a VAS stage has a high output and typical bipolar stage a very non-linear input impedance witch is the main mechanism of crossover distortion in a bipolar output stage.
The only way to come out of this problem is to implement an additional "driver-stage" to convert the high impedance from the VAS to a low-impedance stage which drives the output stage. This can be done with a simple emitter follower. Or by a tripplet as output. Or something more sophisticated like Nelson does ;)
Has anybody measures what happens to the distortion spectrum when introduced an additional signal? If we have an amp witch has a distortion spectrum at 600Hz like A=(n-1)!/2 or which decreases with 12dB/octave, will the spectrum vary if we have another signal maybe of 7kHz? I do not mean something like intermodulation, the first order intermodulation products should be masked by the ear. Will an amp in presence of two signals also have the same spectrum of A=(n-1)!/2? |
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| PMA |
bocka,
and what to do when there is 2nd harmonic of some -115dB, 3rd of some -120dB, IMD of similar order, no higher harmonics, class A, and you find that it is not THD and IMD what makes a difference, but filtration, CD signal treatment, cable termination etc? That most of the standard approaches to solve sound problems are simply wrong? ;) |
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| AKSA |
That was a tantalizing post, PMA....
Hugh |
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| john curl |
Well stated, Bocka. Most of our distortion is in the output, followed by the high voltage driver that has to develop the complete voltage swing for the amp. The input stage used to be a big problem when we used maximum Gm input stages. This was because the input stage would work harder and harder with increasing frequency, ultimately causing slew rate limiting, and even earlier, TIM (or SID). Walt Jung and Matti Otala have published reams of info in this, beginning in the 70's.
Today, we all degenerate our bipolar input stages, increase our gain-bandwidth of our amp, or both. Fets usually don't have to be degenerated in order to get a very high slew rate, because they are always lower Gm than non-degenerated bipolars, and they are more linear as well. |
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| dimitri |
boska wrote:
Obviously a VAS stage has a high output and typical bipolar stage a very non-linear input impedance witch is the main mechanism of crossover distortion in a bipolar output stage.
Sorry this is true _only_ when the _open_ loop VAS is taken into account. When you speak about Self/blameless design you should have in mind that because of the local loop through frequency compensation capacitor the vas output resistance will _lower_ with frequency. Definitely it wouldn’t be the “main mechanism”. |
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| fdegrove |
Hi,
What keeps on surprising me is how tube and transistor problems are similar....
Not knowing all that much about semis, you can still read JCs post as if it were tubes he talked about .
It's perfectly understandable to both parties...absolutely brilliant.
Than there's Matti Otala's work regarding TIM and other distortion artefacts that no oscope shows up...
Suddenly I feel at home...great!!!:)
Cheers,;) |
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| fdegrove |
Hey Jam,
| quote: | | Please......you talk like you might build a solid state amp soon. |
I do? I wouldn't count on it if I were you...
Let's just say that I'm at awe with JC's knowledge and I really find myself in his thinking and your sense of humour at the same time...:D
There's still way too much for me in the tube department to be learned, a field that I'm really passionate about, before I even think of looking into semis.
The more I learn, the more I realise I don't know nothing at all...don't even know if that's a proper English expression...
Be my guest and educate this hopeless Rita...:D |
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| lumanauw |
I have another "dumb" question. Look at sch1.jpg. I got this design from Randy G Slone book, about the "Optimum Power Amp". The schematic is very nice to see, complementary differential with current mirror, the VAS is cascoded with 2 transistors.
But when I make experiment of it, this design just doesn't work. Up until now I cannot figure what is the voltage in the base of Q11 and Q12 (first VAS transistor). I need it to determine the VAS's standing current. But the base of Q11 and Q12 is located between collector of differential and collector of current mirror. How can I calculate the voltage between 2 collectors?
When I just change the current mirror with ordinary resistor, this amp work. But how come the author have measurement figures, if the amp doesn't work?
Question no2. comes in sch2.jpg. This is the power amp from motorola. Why is the collector pin of Q2 and Q4 doesn't tied up to +/-VCC with it's own resistor? Q1 and Q3 tied up to +/-VCC via 2.21k resistor. In ordinary design, the Q2 and Q4 will go straight to +/-VCC or via another 2.21k. In that design, both collectors are tied to the VAS's resistor, the 22,1ohm resistor. What is the point of this design, and what is the advantages and disadvantages? |
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| lumanauw |
This is sch2.jpg
*I wonder, yesterday this thread is 5 pages, but this morning is only 3 pages. Why is that? |
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| Tube_Dude |
Hi Lumanauw!
| quote: | | Up until now I cannot figure what is the voltage in the base of Q11 and Q12 (first VAS transistor). |
The voltage in the base of of Q11 is 1,2volts (two diode drops) under the rail positive voltage!
Ex. If the rail is + 40 volts the voltage at the base of Q11 will be 38,8 volts!
In Q12 will be the some but with negatif voltage!(two diod drops from the minus rail)
| quote: | | Question no2. comes in sch2.jpg. This is the power amp from motorola. Why is the collector pin of Q2 and Q4 doesn't tied up to +/-VCC with it's own resistor? Q1 and Q3 tied up to +/-VCC via 2.21k resistor. In ordinary design, the Q2 and Q4 will go straight to +/-VCC or via another 2.21k. In that design, both collectors are tied to the VAS's resistor, the 22,1ohm resistor. What is the point of this design, and what is the advantages and disadvantages? |
This is for some inside feedback loop...it can increase stability and linearity..
Advantages..as expressed formely more stability and linearity in open loop...
Disadvantages...less open loop gain for more overall feedback for less overall distortion...
See one example of this technic in a design of our forum friend Jonh Curl:
http://marklev.com/marklev/JC3/jc3schematics.jpg
Regards! |
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| bocka |
| quote: | | Sorry this is true _only_ when the _open_ loop VAS is taken into account. When you speak about Self/blameless design you should have in mind that because of the local loop through frequency compensation capacitor the vas output resistance will _lower_ with frequency. Definitely it wouldn’t be the "main mechanism". |
dimitri,
yes, you're completely right about the Doug Self stage with compensation capacitor as it gives local feedback to the VAS stage. I'd be a little more precise about this. If we'd use an uncompensated or open loop VAS stage we can see, that a typical bipolar output stage will load the VAS stage witch results in heavy high order distortion of the output stage.
I think in this point of view tubes and bipolar are quite different. Tubes in class AB will have lower distortion levels and lower high order distortion without any applied feedback.
To my opinon and experiance an amp with low inherent (feedbackless or open loop) distortion sounds better with applied feedback. One possibility to archieve this is to use an output tripplet or a MOSFET output stage. |
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| PRR |
> I have another "dumb" question. ...design from Randy G Slone book, about the "Optimum Power Amp". ...when I make experiment of it, this design just doesn't work. Up until now I cannot figure what is the voltage in the base of Q11 and Q12 (first VAS transistor).
I think this was discussed before, and the consensus is that it will not work as shown. As your "dumb" (not dumb!) analysis shows, the current in the second stage is not clearly defined. As Tube Dude says, to "work" it has to be something like 2Vbe. But circuits do what you make them do, not what you think they will do. In this case there is nothing that "makes" the second stage current any particular value.
This goes back to Nelson's comment about how nice it is to have one side setting currents throughout the amp: "the bias stability that such arrangements provide." You can do both sides, but there are ways to go wrong that you may not notice until you smoke a boardful of transistors. Old-hands like Pass and Curl are either smart enough or smoke-experienced enough to see these on paper, but I can still do equally dumb things despite years of smoking parts.
> how come the author have measurement figures, if the amp doesn't work?
I suspect the wrong diagram got printed, or the publisher's artist re-drew the author's sketch wrong.
Actually: if the input quad is perfectly matched, the second stage can only draw a "very small" base current. With luck, the stage may wind up somewhere between cut-off and smoking, and actually amplify. It is not impossible that it would work. But it is hypersensitive to part parameters and generally won't work except by luck.
> power amp from motorola. Why is the collector pin of Q2 and Q4 doesn't tied up to +/-VCC with it's own resistor?
Returning the other side of the diff-amp to the next stage emitter gives some of the benefit of a current mirror without some of the problems. To my eye it is "ugly", but often useful. |
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| Bricolo |
| quote: | Originally posted by Nelson Pass
I'm a bit surprised that no one has mentioned one of the
advantages of single-pair inputs biased by a constant
current source and driving an SE 2nd stage which is loaded
with a constant current source - namely the bias stability that
such arrangements provide. It is easy to make a stable CCS,
and this stabilizes the operation of the gain devices.
|
Isn't this an aleph with a ccs? |
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| PRR |
> What keeps on surprising me is how tube and transistor problems are similar....
They are identical. Just a different balance of problems.
BJT has a "junction" that current won't cross without a certain energy, and the amount of current can be calculated.
FETs have a bulk resistance that can be "squeezed" by an external field, but the edge of the squeezed area is a junction.
Thermionic vacuum tubes have a little of both. The cathode surface is a junction, just not a solid-state junction because one side isn't solid. But when you get down on the atomic level and look how electrons come off the cathode, you have to use the exact same physics as transistor theory. However, most of the energy applied to this junction is "heater", not Signal. Then the electrons pass through a "bulk" which is squeezed and stiffled by electric field.
Philosophically, "all devices are the same".
Yes, practical devices differ enough to hide the underlying unity of all electronics.
Differences:
* atomic-size junction or bulk effect
* parasitics
The practical vacuum tube is far from perfect. Cathodes hang onto electrons very hard. Somehow the cathode matter must be fluid enough to loosen electrons while solid enough to hold a shape. In operation, a monatomic layer of the surface is above melting-point while the rest of the cathode is solid. Very tricky (and poorly understood) doping is needed for great results. Then the bulk of the tube would control poorly if designed to classic "ideal cylindrical triode" geometry, real tubes use gimicky geometry to improve gain in the normal operating range. The theory books talk about 3/2-Law: you won't catch a practical triode amplifier doing any such thing.
The modern (since 1965) BJT is nearly "ideal", in the sense that the theory is clear (and simple) and practical devices follow theory extremely closely. Also the atomic-sized junction with minimal bulk effect gives the absolute best transconductance of any device. (Though it is worth noting that if we had very clean Germanium, Gm would be twice that of Silicon.) A significant drawback is that all BJT devices require a significant and not clearly defined input current.
The FET is a bulk device and all about geometry and clever device design. Within the range that the device is intended for, we can use simple theory for small signals, but the large signal parameters are all empirical. And the bulk geometry means the Gm tends to be lower than a BJT (though oddly this is untrue at super-low currents). The bulk geometry also means that FETs have significant capacitances and thus can draw large input currents in AC amplifiers, and for large FETs sometimes more current than a BJT even inside the audio band.
You can treat tubes and FETs as BJTs that buffer (more or less) their inputs but have big variable resistors hanging on them. A BJT-nut will argue: take the gain without the losses, use BJT with precision fixed resistors. BJTs do have some bulk effects but we can cascode them away. On the other hand, if you keep adding things to fix the big errors, you end up with a growing pile of "small" errors that can get out of hand. There are a lot of many-BJT plans that are more clever than smart. This even infects tube-heads.
It's not just the device. It's not just the geometry. It's not just the detailing. Everything has to play together nice.
If a certain device or geometry is forced upon you (personal preference, price, existing board), you can probably detail it to a very high standard (higher than commonly found in real life).
If you have a clean-sheet, it is hard to screw-up a simple tube amplifier. At a higher standard of precision, a reasonably simple and easily understood BJT design can work extrememly well (and a lot cooler than tubes). When your specs seem to suggest very complicated BJT designs, you may find that an FET design will do the job without as many "small side effects", because they are a nice compromise between the fairly low gain and precision of tubes and the hyper-high and hard-to-control gain of BJT.
But when the "spec" is the musical ear, some imprecision is ignored and other faults are pinpointed. The ear is a flawed but very clever thing. We can stand in a field of chirping crickets and hear the toe-step of the tiger that is about to eat us (if our ancestors did not hear this well, we would not be here). The ear makes 10% distortion and is unflat 40+dB, but can pick-out a tone 20dB below the general noise level. The measurements that are easy are often not that important. We are nearing the point of machines as clever as the ear, but we are not using them in the same clever way as the ear. |
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| Steven |
| quote: | Originally posted by PRR
>
I think this was discussed before, and the consensus is that it will not work as shown. As your "dumb" (not dumb!) analysis shows, the current in the second stage is not clearly defined. |
This has been discussed in
http://www.diyaudio.com/forums/show...5526#post195526.
I wonder whether these current limiters in the VAS cascode stage (small print, but it seems to be Q14 and Q17 with 33Ω resistors) have something to do with the undefined bias point. At least the VAS current will be limited to approximately 20mA. If so, could that mean that Slone knew it was undefined and used this as a way to 'define' it? Not very sophisticated.
Steven |
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| mlloyd1 |
I've got two questions:
1. Did anyone build a version to see what would happen?
2. Did anyone ask Sloan to see what he had to say (" there is a diagram error" or "it really does work as long as you maintain the global feedback loop" or ... etc.)
mlloyd1
| quote: | Originally posted by lumanauw
.... this design just doesn't work.... |
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| Steve Eddy |
| quote: | Originally posted by PRR
But when the "spec" is the musical ear, some imprecision is ignored and other faults are pinpointed. The ear is a flawed but very clever thing. We can stand in a field of chirping crickets and hear the toe-step of the tiger that is about to eat us (if our ancestors did not hear this well, we would not be here). |
Or the toe-step of the tiger that's not there at all.
One of the reasons our ear/brain system isn't quite the highly accurate detection system that some tend to portray it is because nature has wired us to OVERdetect. In other words, we're more prone to perceive a toe-step when there is none than to not perceive one when there is.
This unfortunately makes it rather trivially easy to cause people to perceive differences in audio even when there are none.
| quote: | | We are nearing the point of machines as clever as the ear, but we are not using them in the same clever way as the ear. |
Like hearing tigers that aren't there? :D
Anyway, yeah, the ear/brain system is pretty amazing. But I think in the world of audiophilia, it often gets oversold and there's a tendency to ignore its very real limitations.
se |
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| PRR |
> nature has wired us to OVERdetect. In other words, we're more prone to perceive a toe-step when there is none than to not perceive one when there is.
On one hand, that's a good point. It is certainly safer to flinch at a hundred non-tigers than to not-hear just one actual tiger. If the tigers or wolves or fruit-tree branches don't get you, your fellow man is probably sneaking up behind you with a rock. Through most of mankind's history, life has been dangerous and some of those dangers can be heard coming. OVER-detect!
On the other hand, this one simple thought could tear-away the cherished foundations of both the "I know what I hear" and the "measurements don't lie" schools of hi-fi philosophy. |
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| lumanauw |
Yes, actually I have made it. The output is just big voltage plain DC, which is not affected at all with music. First I got confused with it. But when I replace the current mirrors with resistor with calculated value, it works. But the sound is not very good.
Is there anyway to determine voltage between two collectors (like between collector of differential with collector from current mirror? |
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| hitsware |
you don't need no followers.
sort of a REAL class AB |
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| john curl |
| You are close, hitsware. This design concept is about 30 years old, as it was first used in the JC-2. Now, how can we improve it? |
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| Bricolo |
| by adding current sources? |
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| hitsware |
| quote: | Originally posted by john curl
You are close, hitsware. This design concept is about 30 years old, as it was first used in the JC-2. Now, how can we improve it? |
I know it can be unstable. In at times unpridictable ways. Also at times it will blow the P channel output (using Hitachis). Perhaps it blows the outputs by pulling the gate too low and burning the input zener so gate resistors may solve that. The whole thing may be better implemented with bipolars (putting the - input devices on the heatsinks for thermal compensation. AND carefull selection of the outputs (rugged enough to withstand the SOA problems this type output scheme (common emitter(source)) seems to engender.........mike |
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| jam |
Mr.Curl,
What if we replace the resistors on the tails of the differential with adjustable current sources, we would be then able to adjust bias and dc offset at the same time.
My question , how the bias stability of such a circuit would be?
Regards,
Jam |
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| john curl |
| First, think about removing ALL the current sources and just using 1 resistor between the source pairs on the input. Second, this is actually a very stable design, because it has almost no gain, but 1/2A Hitachi devices can be problematic, because they are not well matched in this situation. The P channel looks like a triode, but the N channel looks like a pentode. Too much 2'nd harmonic. How do we fix this? |
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| jam |
| Cascode the output pair? |
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| john curl |
| Yep, makes a world of difference, in this case. This is essentially the 2'nd stage for my Vendetta Research phono preamp. |
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| fdegrove |
Hi,
| quote: | | Cascode the output pair? |
Good thinking, that would change the harmonic spectrum reducing the second harmonic.
Oops... I just see you had a reply already.
Cheers,;) |
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| jam |
Mr. Curl,
If you were to update the JC-3, would this be the topology you would use? It would be fairly simple to add a Vbe multiplier and outputs to the circuit.
If this is is correct would you choose bipolars (with drivers) or mosfets for the outputs, and would you still use the inverting configuration?
Regards,
Jam |
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| jam |
Frank,
Trolling the solid state section again? .......................;) You will soon be joining us. :D :D :D
Regards,
Jam |
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| Bricolo |
| quote: | Originally posted by john curl
First, think about removing ALL the current sources and just using 1 resistor between the source pairs on the input. Second, this is actually a very stable design, because it has almost no gain, but 1/2A Hitachi devices can be problematic, because they are not well matched in this situation. The P channel looks like a triode, but the N channel looks like a pentode. Too much 2'nd harmonic. How do we fix this? |
Mr Curl,
Cascoding could work good, but would'nt be easier to use other output devices? (better matched ones)
Could you explain me why it has low gain? I thought that dif pairs had a very high gain. Are dual differentials different?
So, can hitsware's schematic be used as if, without feedback? |
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| fdegrove |
Hi,
| quote: | | Trolling the solid state section again? ....................... |
Guilty as charged, sir....;) |
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| hitsware |
| quote: | Originally posted by john curl
First, think about removing ALL the current sources and just using 1 resistor between the source pairs on the input.
|
Wouldn't that make it class B ? I.E. at quiessence there would be a virtual ground at the center of the resistor and grounds on the gates (assuming the gates are tied together (+to+ & -to-) |
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| hitsware |
| quote: | Originally posted by hitsware
Wouldn't that make it class B ? I.E. at quiessence there would be a virtual ground at the center of the resistor and grounds on the gates (assuming the gates are tied together (+to+ & -to-) |
Duuh ..... I keep forgetting jfets are depletion devices ... Still sort of a nebuosity(sp?) about the 'virtual gnd' between the halves, but ..... (1/2 way between + and -) = 0 |
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| hitsware |
| As an output stage ..... NOT a line amp ..... |
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| Fred Dieckmann |
| Change the resistor connecting the sources of N pair and P pair of JFETs to three delta connected resistors with the bottom resistor between the sources of the P JFET pair. Adjusting the value of this resistor will change the degeneration and gain for that half of the circuit to compensate for the N channel MOSFET's greater transconductance than the P channel MOSFET. The parallel combination of the other two resistors will set the bias current of the JFETs independently of the value of degeneration resistor for the P pair of JFETs. You also need to cascode the JFETs for supply voltage above about +/- 20 volts. R1 and R3 will set the bias for the the front end and output stage and will be something different than value shown on the schematic below. |
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| john curl |
Actually, this makes a pretty lousy power amp, BECAUSE there is no loop gain with a low Z load. Makes a darn good line amp, however.
I even made the Grateful Dead line driver to drive the final mix from the mixing board to the stage with a similar configuration. All else being equal, fets are better than bipolars, but sometimes bipolars are useful in this circuit. |
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| Fred Dieckmann |
| Well, maybe not no loop gain. The Hitachi K1058/J162 pair biased at half an amp would give an output stage gain of about 5 for 4 ohms I believe. With two pair of output devices and some fairly low value drain resistors for the JFET front end to drive the MOSFET Gate to source capacitance, I bet you could get 16 or 20 dB of negative feedback. This wouldn't be a great measuring amp but might not sound too bad. I have the parts around...... maybe I will simulate it and build it if the numbers look reasonable. |
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| hitsware |
| quote: | Originally posted by Fred Dieckmann
This wouldn't be a great measuring amp but might not sound too bad. |
Yes! There is a certain quality to the common emitter(source) type configuration. Sort of a fullness in the low end.......
BUT beware (maybe a sim would catch it) of sudden 'mode shifts' for lack of a better term. Since the gain of the output stage does vary so does the feedback (if used). ALSO I think some sort of "shootthrough" (too much conduction from rail to rail through the outputs) sometimes occurs........mike |
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| john curl |
| Fred's biasing has problems. Not in concept, but in execution. The resistor values are too large, and the differential gain is asymmetrical |
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| hitsware |
| somebody's doin' it ...... |
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| Fred Dieckmann |
DO NOT RUN K389 AND J109 AT 32 VOLTS.
There also needs to be gate resitors on the mosfets and preferably the jfets as well I would cascode or series zener drop the drain voltages for the JFETs. I wonder if an amp from this schematic was actually built...... |
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| Fred Dieckmann |
| quote: | Originally posted by john curl
Fred's biasing has problems. Not in concept, but in execution. The resistor values are too large, and the differential gain is asymmetrical |
The point was to make the gain asymmetrical to compensate for the the N channel mosfet having a greater transconductance. I believe you would want indentical open loop gains for each half of the circuit for lowest 2nd harmonic distortion. I believe the transconductance of the J109 is also a little greater than the K389 at the same current if I remember correctly. I don't remeber seeing this done before but I am sure it has been, perhaps even by Mr. Curl. The bias resisitors value will depend on the Idss of the Jfets and the value of the drain load resistors. I believe I stated that the would probably not be 75 ohms. I would try for at least 2 or 3 mA of bias current for each jfet and pick ones with Idss several mA above the bias current that they will be used at. |
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| Fred Dieckmann |
| The J109s are good for 30 Volts....... There are reasons for a cascode circuit besides just the drain to source voltage being too high. |
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| Elso Kwak |
Hi hitsware,
JFets get noisy at high supply voltage because of leakeage, though not destructed. See the datasheet of f.a 2N5912 at www.vishay.com and Horowitz.
I learned it the hard way. Some form of cascoding will prevent this from happening.
:cool: |
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| Fred Dieckmann |
| Give that man a gold star. I was also considering power dissipation which is about 200 mW for these devices. |
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| john curl |
| The circuit should work, just as it is. Gate resistors are most probably optional on this design. Why? I don't know why you like gate resistors of jfets. All it does, in this case, is to make the input more noisy. |
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| PMA |
| Simple is good, but not the best. This amp (post 81) will have pretty high distortion, driving gate-source capacitance of the Mosfets. |
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| hitsware |
| quote: | Originally posted by PMA
Simple is good, but not the best. This amp (post 81) will have pretty high distortion, driving gate-source capacitance of the Mosfets. |
I've done something simular (single ended 2sk170 driving Hitachis with no feedback) and got ~0.6% THD 20 to 20k. Not as bad as one would expect. (or get with IR types) IMO .... mike |
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| Nelson Pass |
Every type of device has it's own sweet spot(s). Don't
substitute devices into the same circuits and imagine that
you have a real comparison. |
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| hitsware |
| quote: | Originally posted by Nelson Pass
Every type of device has it's own sweet spot(s). Don't
substitute devices into the same circuits and imagine that
you have a real comparison. |
Ya Mon .... Forgot to mention used the Hitachis as followers ... |
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| hitsware |
| quote: | Originally posted by john curl
The circuit should work, just as it is. Gate resistors are most probably optional on this design. Why? I don't know why you like gate resistors of jfets. All it does, in this case, is to make the input more noisy. |
If'n you use a gate resistor with a jfet (at the input) ..... You get a rolloff that completely negates any kind of TIM .... I use 10K with 2sk170 ....... |
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| Steve Eddy |
| quote: | Originally posted by hitsware
If'n you use a gate resistor with a jfet (at the input) ..... You get a rolloff that completely negates any kind of TIM .... I use 10K with 2sk170 ....... |
Hmmmm. But since FET input capacitance is so non-linear, wouldn't such a high value input resistor just acerbate the problem and produce greater amounts of high frequency distortion which could fold back down into the audio band?
A shunt RC network would ultimately be preferable, yes? Assuming of course that one's goal is optimal objective performance.
se |
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| hitsware |
>But since FET input capacitance is so non-linear, wouldn't such a high value input resistor just acerbate the problem and produce greater amounts of high frequency distortion which could fold back down into the audio band?
It seems that would be the case, but in practice it works. It's a differant ball game than with something like the mundo IR type fets where the output (drain) reflects back to the gate. Using a 10kHz squarewave as referance, I get nice smooth response.
>A shunt RC network would ultimately be preferable, yes? Assuming of course that one's goal is optimal objective performance.
Yes, theorieticly. But sometimes to me super simple elegance in the schematic can outweigh other factors. I must say however I've not aurally A-B 'd the 2 approaches. Usually my squarewave approach is pretty dependable though...........mike |
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| PMA |
| A question is if we consider THD of 0.6% to be low or high. For me that is high. How is the THD at 10 - 20kHz? Did you measure IMD - probably more problems with Mosfet's non-linear input capacitance. |
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