Most high end amps are not class A as I see it. It seems there are more AB than A, although I haven't done a study.
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I do not believe that actually to be the case. As to your question, no , you are unlikely to find an amp that will be unquestionable better than your Mac at that price. You could look at the Pass amps in the 150 range, they will have a different sound than the Mac which you may or may not like better. One of the mistakes most commonly made in audio is to think that you will find an amp that does everything better than the one you have. Unless you have a terrible amp this is almost never the case. Every amp regardless of price is a product of compromises and will possess different virtues. If you like the sound of what you have and it drives your speakers well do not change until you have heard something you like considerably better. I have owned and been a dealer for several class A amps in the past but do not own any now and do not foresee buying any in the future. Class A is not a cure all , it is one technique for achieving good sound, there are others. I would listen to a particular amp regardless of its design and evaluate it on its performance.
As Stanwal wisely states, listen before you change. I have owned a McIntosh 2105 for 10 years, and during that time also owned a Mac MC 300 which I sold after 3 years to buy a Pass Labs X250.5. For most efficient speakers, at modest volumes , that amp will run in class A. I used it with 3 different speakers, and for my taste, the Mac 2105 was better in all aspects. The point of course is that you may feel differently, but do not make the mistake that I made and buy on the opinion of others and their taste - try it first.
Class A is the most linear mode of operation known. But- just like with anything high performance, is the exception rather than the rule, which is to say that AB amplifiers are far more common. On top of that, there can be so many compromises elsewhere in the circuit that they may well obviate the advantage of the class A operation.
Triodes are also the most linear form of amplification known, yet the same caveats apply.
The same goes for fully balanced differential operation, circuit minimalism, etc., etc. When you combine these elements, things tend to lean more in your favor, but execution is still an issue. Cost is a different matter- Class A amps do tend to be more expensive due to the larger power transformers required.
On top of that, equipment matching will still be an issue- see
for more information.
Stanwal's advice is extremely good, and there are amps amps out there that will be a whole new world of resolution compared to the Mac, but not likely for the $$$...
Gp_phan, if you can wait and save an additional $1000, and do some bargaining (at the $3000 range) on a used Pass XA30.5, then I know you will get that good Class A sound described in Audiogon. The XA30.5 works well with smaller speakers. It can do the job with some bigger 4-ohm speakers to give them 60 watts/channel, but I would be cautious. People need to do audition before buying anyway. I am not that familiar with other Class A amps beside Pass Lab as they are the lead in this niche market. Perhaps somebody else could help you with the $1000-2000 range. Unfortunately like anything else, you get what you pay for.
I've long been a fan of 'Class A' sound, specifically from solid state components. I'm a tube fan now, and no SS can compare, IMHO. However, when I look back to my SS days, my favorite amps were Class A amps from Threshold, Clayton, etc.
In your price range, I'd reco an older, used Threshold T200, Forte Model 4a (a real bargain), or the Coda 11.5 currently for sale on AudiogoN.
Disclaimer: I have no relations with any sellers.
Another idea might to get a Plinius SA100, about $1800. They have a switch that goes from AB to pure class A. They put these on the amp to save energy, and then you can flip a switch for "serious" listening. When you flip the switch to class A you start burning 400 watts continuously, nice space heater you have there man.
Most class A amps look to be 25% effecient and have an idle current at 4 times their output. So a 50 watt amp has a 200 watt idle current. The Pass Labs site explains this well. I have not listened to class A, but I think the Plinius could satisfy the curious.
Notice that all Audio Research amps have an idle current at about 4 times rated output, the REF 110 at 100 watts idles at 400 watts, the largest ARC amps like 600 idle at 800 watts. These amps have a really loyal following in terms of their sound, and mixed reviews in terms of reliability.
I have been told that class A amplification simplifies things in that the current is not broken up and then reassembled very fast.
But what does it really sound like? It is expensive and seems to be an audiophile only thing, BEST BUY does not have many class A amps, probably none.
So getting back to Bi-Amping. You could Bi-amp with a low power amp in the mid to get the magic class A sound in the vocal range, and then use SS for bass.
Still another option could be like an ARC VT 50 which is 50 watts and then buy something like a REL powered sub to give you that bass slam. Use the ARC VT 50 with a used set of Maggies to get clarity, transperancy and great mid range and then fill in the bottom with REL. You could do that idea used for $3-4k and have an audiophile set up.
Most of you are thinking of 1 type of class-A power amp - the type that Krell, Plinius, Threshold, Forte, Sugden have made/are making. This type of class-A power amp does dissipate a lot of heat as it is biased to run full power bias current all the time. The heatsinks on these amps are large (they have to be!) & the current draw is high thereby running up your electric bill.
There is another category of class-A amps that use sliding bias. I believe that Symphonic Line & Karan amps (there might be others but I do not know) fall into this category. These amps run cool when there is no signal or when the signal is low (from my experience I would say when the SPL level at the listener's chair ~9-10' away is in the 80dB range). As you crank up the signal input, the dissipation increases markedly. These amps do not run up your electric bill.
As others have pointed out - a well executed class-AB amp can be very satisfying sonically. I can say this from my own personal experience.
However, [b]THERE IS[/b] a certain sweetness to the sound from a class-A amp. I realize that more when I take out the class-A amp & put back the class-AB amp. It's much easier for me to discern what I lost (when I put the class-AB in) than what I gained (when I put the class-A in). In my particular case I say "lost" because I personally like the sweetness of the class-A sonics. YMMV.
It would be interesting to see how the Symphonic Line ($$$$) vs. a Plinius ($$) would sound side by side. Manual vs automatic transmission.
I wonder if all high bias current amps, or the sliding bias current high current as needed is where that extra sweet audiophile sound is coming from. I am sure it is just part of the mix, but the Plinius looks like one of the least expensive SS choices to get 100 watts of class A power.
When did audiophile's start to worry about the electric bill
There are some great A/B Amps but they are not great because they are less expensive to operate or that they cost less to obtain.
To think that way is to say a Honda is better then a Ferrari
because it gets better gas mileage.
I find Class A amps to sound better then the Class A/B or D that I have heard.
This hobby is about SOUND nothing more and nothing less.
Bombaywalla - not only bias but also gain (before feedback)is different between class A and AB. Class AB requires much higher gain (usually about 10x) to linearize output transistors with deeper negative feedback. Large gain causes amplifier to overshoot (and even choke) on fast changing input (pulse) since it is unable to feed signal back in the same phase (delays the signal). This is called Transient Intermodulation Distortion and was dicovered in 1970. Before that first SS amplifiers had excellent THD and IMD but unpleasant sound (overshooting introduces odd harmonics).
Using different classes of amps for woofer and tweeter is fine if we can guarantee that phase shift is the same.
Spatine - I've never had bi-amping setup but noticed that people, most of the time, use identical amps for that. Absolute phase should be the same unless amplifier inverts the phase 180 deg - easy to fix by reversing speaker wires. Problem starts when you get amplifier from different family. My amplifier, for instance, is a class D creature based on Icepower 200ASC module from B&O. Data sheet for this module shows +40deg phase shift for all frequencies (0-20kHz). I'm not sure what to think about it. Rule of operation is different and this amp shouldn't perhaps be bi-amped with an amp from another family.
It would be cool though, to use class D with its excellent power, dynamics and bass control with small class A amp for the highs. I'm afraid that harmonics would add improperly and sound would be strange.
Standard (class A, AB, tube) amps might have slightly different absolute phase shift (and different interaction with the speaker) and I wonder about using different amps (or different family of amps) for bi-amping.
Had anybody done that?
GP_phan, whether or not an amp is ostensibly "class A" is probably the last thing you should be worrying about. True Class A operation is extremely inefficient, and while it works well -- it's expensive because of its inefficiency. Which means that the term "class A" becomes a very abused one in audio nomenclature -- kinda like "digital". That is, it seems like everybody wants to say that their amp is CLASS A in big type, and then put in little type how it's not really, actually a true class A amp.
But specifically for your situation. The MC2105 is a very finely built amplifier and will last many years, but it's Achilles heel is not that it's Class B . . . but it has a quasi-complementary (all-NPN) output stage. The reason for this is that when these amplifiers were designed (in the late-1960s), complementary NPN/PNP power transistor pairs simply didn't exist.
What this means is that in your amplifier, the basic linearity of the output stage around the crossover area is quite poor. In addition, the driver stage is really primitive, with simple resistor networks setting the current through the input differential-amp. I'm not faulting McIntosh for these design choices -- they were simply doing the best they could with the materials and experience they had.
I think a good upgrade for you might be to get a later McIntosh amp, from at least the MC2255-era (early-1980s) or later. By this time, they were using fully-complementary output stages, input stages with current-sources and current-mirrors, and voltage amplifiers with an active load -- meaning they're an order of magnitude more linear. And if you like your 2105, you'll LOVE a 2255.
The C28 is a mixed bag -- its circuitry is very much like a tube preamp, but with transistors . . . so it's fully Class A, single-ended operation. Its weaknesses are that the high impedances used (like a tube preamp) make it a bit noisy, and it's pretty complicated and failure-prone in interconnection and grounding. Specific common failures are dried-out main filter caps (causing oscillation), and that four-section volume control gets noisy and has channel-balance problems when it gets older. You might get it serviced, but if yours seems to be working well, then I'd probably upgrade your amp first, and keep the C28 for a little while - and eventually upgrade the C28 with a late-model unit like a C36 or C38.
My hunch is Kirkus is probably right. In practically all Audiogon discussions to bridge tube and SS amplifiers, Pass Lab is mentioned. I know the XA.5 delivers just what people have talked about, but they are expensive. It thus leaves the question as to what it takes in order to lower pricing. I do know of Clayton as well and have listened to the better Clayton amps. They are nice sounding, but they are not cheap either.
Your system sound is determined by the weakest link in your system, right down to interconnects, cables, electrical and vibrational isolators. There is no question that the real McCoy Class A amplifier design is that good. So if you really want what Class A design can deliver, with some give and take and some negotiation with somewhat better than your price target, it might require smaller speakers and not so loud listening with the music taste that does not demand floor jarring bass. It really is a compromise at this price range or about.
but it's Achilles heel is not that it's Class B . . . but it has a quasi-complementary (all-NPN) output stage. The reason for this is that when these amplifiers were designed (in the late-1960s), complementary NPN/PNP power transistor pairs simply didn't exist. What this means is that in your amplifier, the basic linearity of the output stage around the crossover area is quite poor.Kirkus, I'm having a very hard understanding why an all-NPN output stage is very non-linear at the x-over point. Please explain. Thanks.
In addition, the driver stage is really primitive, with simple resistor networks setting the current through the input differential-amp.There doesn't seem to be anything primitive about this input stage except that it is (very) low gain since resistors are used as the load to input differential pair.
I think a good upgrade for you might be to get a later McIntosh amp, from at least the MC2255-era (early-1980s) or later. By this time, they were using fully-complementary output stages, input stages with current-sources and current-mirrors, and voltage amplifiers with an active load -- meaning they're an order of magnitude more linear.Kirkus, what do you mean by "linear"? And, how do "input stages with current-sources and current-mirrors, and voltage amplifiers with an active load" make the power amp more "linear"?
Thanks in advance for the insights!
Kirkus, I'm having a very hard understanding why an all-NPN output stage is very non-linear at the x-over point. Please explain. Thanks.In a typical all-NPN output stage, the drivers are complementary (NPN/PNP), but the outputs are not (NPN/NPN). In order to make the bias voltages work out, the transistors are connected in such a way so that for the positive pair, there are actually two local feedback loops - one around the driver, and one around the output transistor. But for the negative side, the driver and output are in a single local feedback loop together. This means that in the crossover region, there is an abrupt change in the static gain as the current is transferred from one half of the output stage to the other.
If you ever play around with adjusting the bias on such an output stage (I've actually modded many MC2105s to have adjustable bias), it can be easily seen that as the operating point is shifted, the crossover distortion never goes away, it simply changes shape a bit.
There doesn't seem to be anything primitive about this input stage except that it is (very) low gain since resistors are used as the load to input differential pair.The following voltage amp is a transresistance amplifier (current in, voltage out) - meaning that the signal doesn't appear as a voltage on the collector of the diff-amp transistor that drives it . . . rather, it's a current. So in the diff-amp, it's the transconductance (set by the transistors' beta and their emitter resistors) that determines its open-loop gain.
And, how do "input stages with current-sources and current-mirrors, and voltage amplifiers with an active load" make the power amp more "linear"?Using a current source to supply the diff-amp tail current effectively "makes the tail longer", and eliminates the variation in tail current with common-mode voltage. Also, the performance of any diff-amp is VERY dependent on the static balance of current between the two transistors (or tubes) . . . and with resistors setting the current, this critical balance is affected by tolerances in the (carbon!) resistors, and variations in beta and Vbe in the (old!) input transistors themselves, and also the line voltage when using an unregulated supply (as is almost always the case). Using an active current mirror, instead of the resistors, forces the static currents to remain balanced regardless of these variations.
As far as the active load on the voltage amp - the improvement is the same as for any traditional single-ended collector/plate-loaded voltage amp -- that is, by keeping the quescient current constant across the required voltage swing, the amplifier is more linear. And here is one place where increasing the impedance on the collector does indeed increase open-loop gain, which also reduces distortion.
All of this makes the amplifier much more linear in the traditional sense -- its transfer function. And if you ever compare a MC2105 to a MC2255 (an old amp, but which has these refinements) on the test bench - they are worlds apart in the amount of distortion they produce . . . especially in the higher harmonics.
This may be a more general answer than you're looking for, but I like audio components that both sound good and measure well . . . and do so consistently, under a wide variety of conditions, for many years, without requiring much maintainance or repair. I generally feel that measurements are a good indication as to whether the engineering and execution are competent, and good sound is the indication that the basic design (and measurement methodology) is also competent.
To that end (to consistently sound good and measure well), I think there are some things that the best audio products have in common:
1. They are very well suited to the application
2. They make effective use of current technology and financial resources
3. They effectively solve the problems inherent in their basic design choices
4. They have quality and consistency in their execution
There are many topological choices that can fit this criteria well, both tube and transistor . . . just like there are so many different types of amplifiers that both sound good and measure well.
As far as Class A vs. Class B vs. Class AB . . . it's not that I don't think it doesn't matter, it's just that its very difficult to get a true class A design to meet criteria #1 and #2 above. For most solid-state Class AB or Class B amps, the main difficulty comes in meeting #3. For most Class AB or Class B tube amps, the difficulty comes on criteria #3 and #4.
I personally feel that these days, bipolar solid-state amplifiers show the most promise to meet all the criteria similarly well, but regardless of the basic choice of topology, it's a difficult road to get there. If I was designing a solid-state bipolar amplifier to drive a typical high-end domestic passive loudspeaker, I would choose a Class B design. If I was designing a tweeter amplifier for a high-end active loudspeaker, I would choose a class A design. If I was designing a tube amp, I would choose a Class AB design.
Kirkus, you are assuming rational thinking, whereas audiophile is a hobby. A hobby involves high-drive passionate pursuit for the best quality, such that other considerations seem far remote. For audiophile, that means getting the best sound for whatever money one can afford, or dare I say shouldnt afford. As long as the equipment doesnt break down often, nobody cares about anything else, except for a divorce if you spend too much money or make too much intrusion on the decorative well-being of your spouse.
My purpose of getting on Audiogon lately is to encourage people to check out Class A amplifiers, the real Class A stuff, if their pocket book can afford its price. Perhaps those in the borderline price range should wait for a while until they can own this good amplifier design. The sound gap between Class A and other classes is so wide that I cannot picture myself returning to Class AB. I am talking about how to retain the strengths of SS amplifiers and gain the values of tube amplifiers without their overkill. Music is a very potent medicine that can heal you in many ways, ways that the medical world could not provide. Yet for years I have such a tough time getting that quality sound at home, as opposed to a live concert with, somehow, less peripheral noises and proper sound projection. I am so much at awe with what true Class A amplifiers can do to my life that I just want people to check them out.
I am so much at awe with what true Class A amplifiers can do to my life that I just want people to check them out.Spatine, I really don't disagree with you in any fundamental way - it's just that for me, I could say this about a number of different classifications of audio equipment, not just Class A amplifiers. And I do indeed value a rational, disciplined approach to the craft of designing and building audio equipment over a dogmatic approach . . . mainly because I consistently have positive musical experiences from systems that don't fit into any particular belief system.
A well-executed Class A amp can indeed show the true potential of solid-state amplification . . . I simply feel that it's possible these days for Class B amplification to be just as good (and better in some ways) if properly applied. It's just unfortunate that the market is brimming with too many examples to the contrary.
Kijanki, let's look at it like this . . . and this is kinda what I was getting at when I talked about an amplifier being "well suited to the application".
A realistic amount of juice necessary to drive most of the domestic audiophile loudspeakers to comfortably high domestic volumes is, say 25 volts RMS . . . (approx. 75W/8 ohms or 150W/4 Ohms). I feel that a high-quality amplifier must have a low output impedance in order to have predictable performance into the types of loudspeakers that are likely to get connected to it. If we then to honestly call it a "true Class A" amplifier, it must maintain class A operation to the lowest realistic impedance that the amplifier will see -- let's say 3 ohms.
Our class A amplifier will then operate from 20 volt rails, have a quescient current through the output stage of 13.5 amps, and thus dissipate at least 1100 watts of heat for two channels (you did want stereo, no?). If we want this amplifier to be reliable, consistent, and last many years (because it's hard to enjoy the amazing sound quality of a broken amplifier), we need to keep temperature rise to a minimum (do we include the gentleman who lives in an un-airconditioned flat in Singapore in our calculations?), and we of course don't want a fan . . . it's plain to see the bar-tab for transformer and heat-sink is getting pretty hefty.
When I say a Class B amp could be "better in some ways", I'm thinking about all the other things I could, as an amplifier designer, spend the customer's money on to improve the sound, other than a brute-force approach to linearizing the output stage. If I'm clever enough to get similar performance results from from a Class B output stage (yes I know this is a significant challange), I can definately build a better amp for a given amount of resources.
A great comparison is Levinson ML-2s and the ML-3. I've had both amps in my system for a bit, and enjoyed the latter much more . . . the ML-2s were like a high-maintainance chick that looked gorgeous on your arm, but didn't know how to have fun when you got home and turned the lights out.
the ML-2s were like a high-maintainance chick that looked gorgeous on your arm, but didn't know how to have fun when you got home and turned the lights out.
How about a design that runs Class A to two thirds power? Is that like a gorgeous chick that also knows how to have fun when the lights are out?
Actually, not a bad analogy, Kijanki. It's just a question of . . . when do you want the bad news? To deliver more than you promise is usually more exciting than the other way 'round.
But with most semiconductor amplifier output stages, the linearity problem around the crossover point in Class B is substantially less severe than the one that occurs at the transition between Class A and Class B operation on an AB amp. If class B operation is thoroughly and rigourously optimized, then its performance will be less dependent on load and output level (which I think is very important), but it doesn't give true Class A performance under any conditions. Again, the particular application makes all the difference.
Kirkus - one of the problems with class AB is required gain to get rid of nonlinearity. Class AB amps have gain (before feedback) of couple thousands while class A couple hundred. Huge gain (before NFB) and delays just invite TIM if input slew rate in not limited. It might be possible to bias amp just a little higher (class A has about 150% of max current) to move the "kink" a little further away and set minimum gain to get minimum spects like 0.1-0.5% THD and IMD and bandwidth of 50kHz. The key, I believe, is compromise without going to over specifying it.
TIM is one of the reason of class AB sound, but it wasn't known until 1972. "experts" in denial claimed then that all parameters of SS amp are as good as tubes and therefore they must sound the same while average person could hear otherwise.
In all my previous discussion about linearity, I have been talking about the performance of JUST the output stage in isolation. And whether or not the output stage is biased as Class A, Class AB, or Class B . . . has NO effect on TIM.
The "TIM" acronym these days seems to be frequently flung about as a method to justify virtually any school of thought in amplifier design. But if we to talk about Transient Intermoduation Distortion as described by Matti Otala in his early-1970s AES paper -- the main thrust of this paper (and concept) revolves around frequency-compensation techniques. IIRC, Otala was basically proposing alternatives to the ubiquitous Miller compensation around the voltage-amplifier stage, under the supposition that lag compensation would reduce the loading of the differential amplifier under hard-slewing (transient) conditions, thus reducing a major source of nonlinearity.
While I have great respect for Otala's work, there are a few reasons that I feel the TIM concept, as he described it, is long overdue to be put to rest:
- The specific techniques he describes were based on observations in an era when power transistors were extremely slow, even compared to the small-signal stages that preceeded it - meaning that correctly-applied Miller compensation can be far less heavy-handed in the context of modern power semiconductors.
- Reducing the open-loop gain has absolutely no effect on the fundamental mechanism that causes the problem, it just forces the amplifier to work under conditions where it's difficult to occur. Kinda like strapping yourself to a sofa to avoid having foot pain that occurs when you stand up.
- A far more useful method of analysis of TIM is as a conditional reduction of large-signal open-loop gain and phase margin. This predicts the increase in distortion, and explains why a blanket reduction in open-loop gain can reduce the effect. It also gives valuable insight into how to solve the fundamental issue.
- This also lets us look at TIM distoriton for what it truly is -- a stability problem.
There are vastly more resources available today to more fully understand and predict the actual open-loop behavior of an amplifier than in Otala's day (i.e. high-bandwidth DSOs, FFT spectral analysis, and SPICE). This means that the rigourous engineer can more thoroughly investigate and anticipate all stability issues (including TIM), if he/she chooses to do so.
And if they don't, then TIM is the least of our worries.
Kirkus - The magnitude of TIM is highly dependent on the open loop gain (everything else being equal) up to point where output transistors go to momentary saturation and stay there for a moment (having charge trapped at the junction). We cannot hear it (brain fill the gaps) but it make us tired.
TIM can be easily shown with just sum of two signals and the scope but it doesn't show in normal measurement of THD IMD etc. That was the problem in 1970 and is still now.
In an article in Stereophile "A future without a feedback"
Maritn Colloms claims that sound of 700 amps he reviewed was inversely proportional to amount of global negative feedback. One amp he mentions is a CARY monoblock with a strange feature of negative feedback adjustment. It sounded best at the lowest feedback.
In order to guarantee that amp would be free from TIM designer has to limit input slew rate (or frequency) to levels that output has (slew rate or frequency) before feedback is applied.
The issue here, I believe, is not a lack of resources but lack of discipline. I wouldn't buy class AB amp that has 0.0001% THD - that would be insane. At certain point of open loop gain very low level THD distortions (mostly odd harmonics) will be traded by for higher level TIM artifacts (also mostly odd harmonics). In both cases there will be also more (than in class A) harmonics of the higher order.
Yes TIM is a stability issue - when somebody decides to put gain of 10000 into audio amp and publish perfect spects.
Todays output stages are much faster than at the time Otala published his paper but desire to make class AB amp that is as good as class A amp - still exist.
The magnitude of TIM is highly dependent on the open loop gain (everything else being equal) up to point where output transistors go to momentary saturation and stay there for a moment (having charge trapped at the junction).No, I think you're getting a couple of concepts confused.
Saturation of the output transistors happens at clipping or at reverse bias, the latter of which being a point where the charge carriers are accelerated maximally away from the transistor junction. Whether or not this happens is indeed a function of output stage slewing, but is completely an open-loop phenomonon and occurs independently of loop gain. The amplifier need not have feedback (actually it doesn't need small-signals stages at all!) for it to occur. If for some reason the designer wishes to never reverse-bias the output transistors, this is easily acheived by making minor changes in the driver connections - and the result is a slightly slower output stage.
The concept of slew rate limiting that Otala discusses in his seminal paper on TIM is related to the charging and discharging of the capacitor(s) used to set the small-signal bandwidth of the input and voltage-amp stages, and thus the open-loop response corner frequency. Since these small-signal stages are always biased Class A, their slewing performance (and ultimately that of the whole amplifier) is dependent on the quescient current flowing through them (as used to charge/discharge the capacitors), not the open-loop gain (which BTW I'm assuming means the o/l gain below the corner frequency). Otala advocated the use of capacitors in different places (lag compensation), which basically simply changes which stage in the amplifier is responsible for their charge/discharge current. Both Otala's method and the conventional approach have their pluses and minuses . . . and both approaches can be much less drastic with modern semiconductors than the ones available when he wrote the paper.
My point is that TIM can be understood, analyzed, and avoided - and we don't need to go down the "is feedback good or bad?" road to do it. The latter is of course an unsolveable debate at this time (so let's not go there). The biggest point to me about THD, IMD, and TIM is not so much is not what the numbers themselves are -- but what's causing it, and what the best ways are to fix the problems.
Kirkus - Yes, saturation can appear without any gain in the circuit but it has nothing to do with the issue we're discussing.
High slew rate input signals come back to summing junction thru negative feedback delayed because of signal path delays. For a moment amplifier has no feedback and overshoot appears at the output (or earlier dependent on design). This will happen to any amplifier if slew rate is not properly limited at the input.
Amount of this overshoot is a function of amps open loop gain and in really bad case will take output stage to momentary saturation.
Let forget what is causing it, I agree, and look what to do to fix it. Class AB amp exhibits higher order of mostly odd harmonics at very low signal levels while THD and IMD is measured at substantially higher levels and doesn't show it. In order to lower it - either components have to be very linear or feedback has to be deeper. Careful selection of transistors and better circuit will help to a degree but will never eliminate big "kink". Local feedback will help as well but most of the linearizing will be done in the global NFB (global vs. local is a separate discussion). Bandwidth has to be limited at the input to bandwidth of the amp without the feedback (open loop). That's all. It is tradoff between low level THD and bandwidth on one side and TIM on the other.
High slew rate input signals come back to summing junction thru negative feedback delayed because of signal path delays. For a moment amplifier has no feedback and overshoot appears at the output (or earlier dependent on design). This will happen to any amplifier if slew rate is not properly limited at the input.The only part of this I disagree with is the phrase "at the input". And yes, these are the very fundamentals of proper frequency compensation. The only nit-pick I would add is if this is indeed done "at the input", that implies a passive network before amplification . . . and while slew-rate and bandwidth are two different things, you can't limit one without the other in a passive network. Which makes it a bandwidth discussion, and again takes us right back to the fundamentals of stability in feedback amplifiers, and phase margin.
Also, a Class AB amp cannot "exhibit higher order of mostly odd harmonics at very low signal levels" as compared to a Class A design . . . very simply because at "very low signal levels", it's a Class A amp. That's its raison d'etre. I could maybe see how other non-crossover nonlinearities (this "kink" you describe) may have been a slight contributor at very specific power levels in days of output devices like 2N3055/2955, but is virtually nonexistant in properly loaded constant-beta modern bipolar power transistors.
But to answer Oem's question . . . I don't think that Class B is "the answer", it's one of many valid options. Why is it not used more? I don't know for sure, but I'd speculate that the main reasons are because it requires extremely accurate thermal bias compensation, and the nonlinear base currents demanded by the drivers from the voltage-amp . . . both present significant (but not insurmountable) design challanges for good results. Class AB is significantly less critical in these regards, but the trade-off is a greater variation in performance with load and signal level, both of which are dynamically working to pull the amp from Class A operation towards Class B. And this transition isn't a particularly graceful one.
Kirkus - class A amp at very low levels has both output devices conducting simultaneously doubling their voltage gains. It creates wobble in output linearity - no escape from that. It is known as "gm doubling". Increasing bias won't help since overbiasing creates higher order of odd harmonics (because of gm doubling)as well as underbiasing.
The term "gm doubling" as I understand it refers to the increase in gain that occurs from when one half of the output stage is conducting, to when both halves are conducting. The idea is that there's "double the transistors", so there's "double the transconductance". The static gain is of course no where near double because of the local feedback intrinsic in the output stage.
"Gm doubling" is the main problem with a Class AB amplifier, and manifests itself as a sharp nonlinearity at the point of transition between Class A and Class B operation. This can easily be seen when looking at the distortion output on a 'scope - as the signal is increased into a resistive load, the nonlinearity on the distortion residual slides, in alignment, from the tips of the waveform toward the center, as the amplifier transitions from Class A to Class B. And you're correct in the assertion that increasing bias doesn't improve the problem, it just increases the signal level at which it occurs.
Class A amplifiers don't have this problem, because "gm doubling" occurs all the time - since both halves are always conducting, there's no change in gain to the point where they're not. Class B amplifiers must walk a thin line between cutoff (underbiased) and gm-doubling (overbiased) - this is why bias tracking is so critical on class B designs.
We touched on Bi-amping a bit and using class "A" on the high frequency, and class A/B on the bass. I noticed on a system you helped create you used a Marantz Model 2 with EL34 tubes for high frequency and then a McIntosh MC275 with KT88 for the low frequency.
Is the Marantz running in class A, or do you like the mix of a nice EL34 amp for the high frequency and then the KT88 for the low frequency.
How can you bi-amp economically to try and get nice high notes, leads, vocals, and then get good bass as well. If you bi-amp should you stick to all tubes, or all SS or can you mix it?
Kirkus - I read article that shows harmonics of output stage (class AB) with underbias, proper bias and overbias. It's a little eye opening since many people believe that increasing bias will make class AB sound like class A. It is not so simple - otherwise everybody would have done that.
What is typical bias these days? I remember vaguely something around 100mA.
I also read on the subject of complementary pair distortions that you mentioned once. It pretty much confirms what you said but I just couldn't believe that, according to it, some companies use all NPN transistors because PNP transistor is a little more expensive. Crazy.