Krell Class A/B power amps, do 'anticipator circuits' work?
Yeah, I think it has more to do with setting the bias based on the current requirements (as in amps) of the current situation (e.g. listening level and speaker load requirements). For many speakers and most listening conditions, a fully biased class A amp is just wasting power, but it's nice to have it in reserve. I owned a pair of Krell MDA-300 monoblocks for a while that were room heaters. I think these were made in the days when Krell class A amps were fully biased. I replaced these with an FPB-300 (which used the plateau biasing technology) which I owned for more than 20 years. There was a couple years separating my ownership of these amps, as well as different speakers, so I never really got a chance to directly compare them, but they were both excellent amps. I don't think I would hesitate (at least from an SQ perepective) to buy one of the newer D'Agostino designed Krells (not sure about the ones after he left). I finally got rid of mine after it failed for a second time with bad caps and it was more expensive to recap it then it was worth. In hind-sight, I wish I had kept it and done the work myself since I've recently gotten into doing a lot of DIY audio electronics. |
Thank you Jaytor, that is extremely helpful. I had not read it previously. It may be seen to settle the question I put. I have read the patent. It relates to a methodology of adjusting the bias current in response to measured changes in current flowing in the load. At no point does the patent claim the bias adjustment can be triggered so fast that when a sudden high current event occurs, that event can be presented in either pure Class A or employing a higher bias current than that being employed immediately prior to the occurrence of the event. The system can adjust the bias in quite complex ways but it cannot provide a raised bias current in time to catch sudden dynamic increases in the programme. Such effect is advantageous to SQ and can only be achieved by running full-time in Class A. I am satisfied the natural order is restored. The application should not have been described by some as 'anticipator circuits'. That would entail time travel. |
Here is a link to Dan D'Agostino's patent for plateau bias. This describes, in general terms, how his circuit works. https://patents.google.com/patent/US5331291 |
Douglas Self stated that increasing bias often leads to increased distortions. He also measured it. The reason for that is changing transconductance. Transconductance is output current to driving voltage gain, that is rapidly changing when two transistor current (biased region) changes into one transistor current (outside of biased region). He stated: It is not generally appreciated that moving into Class-AB, by increasing the quiescent current, does NOT simply trade efficiency for linearity. If the output power is above the level at which Class-A operation can be sustained, THD increases as the bias advances into AB operation. This is due to so-called "gm-doubling" (ie the voltage-gain increase caused by both devices conducting simultaneously in the centre of the output-voltage range, in the Class-A region) putting edges into the distortion residual that generate high-order harmonics much as under-biasing does. This vital fact is little known, presumably because gm-doubling distortion is at a relatively low level and is obscured in most amplifiers by other distortions.You can find it here (5.3): http://www.douglas-self.com/ampins/dipa/dipa.htm The main reason of going into class A is to reduce feedback. Once feedback is set damage is already done (TIM distortions) and playing with bias won’t change it. Perhaps there is a way of compensating for the change in transconductance (gm doubling) or adjusting feedback dynamically but it is very complicated. My Benchmark AHB2, class AB amplifier, uses separate "error output stage" to avoid recursive feedback and TIM distortions (AAA patent). |
We used to build quite a lot of very high biased push pull Class-A’s 100w> with around 150w in B My old boss (rip) Steven Deratz of Deratz Electronics back in the 80-90’s in Brookvale Sydney, was the first to have a patent on a "Sliding Bias" type circuit, just to save on the amount of heatsinking were had to use with the normal Class-A we built. My personal A amp was a beast 3 man lift self contained water cooled transistor jacket, pump, radiator/fan. Anyway his patent sliding class-A worked fine, but the problem was the initial leading edge of a larger transient was in B after which all else stayed in A until the bias rolled itself back, or if there was a second leading edge transient larger than the first, it too was B. We could never make it sound as good and just high biased Class-A, it just wasn’t fast enough to keep up with music’s varying transients. And if you held the A up for a longer period, you had the same heat problem as normal high biasing. He let the patent lapse and then Technics bought out something called Quarter-A and AA, and then Krell with "Plateau Bias" Moto? Nothing beats the real thing. Cheers George |
You will note I said "waveform", not specifying which one. You are assuming emitter degeneration resistors, but it would be just as easy to pull the value from a high side or low side current sense on the power supply output. I could see a hybrid that also takes into account output voltage since distortion is around the crossover point. |
The bias cannot be adjusted from the signal alone. Any circuit that slides the bias has to take into account the load presented by the speaker, for that is the only thing that draws current through the output devices. The buffer would, in theory, apply to only one type of speaker impedance vs frequency because it is a known that can be the reference of the buffering scheme. If the buffer is set to maintain the bias voltage to equal the voltage across the emitter resistors at, say, a 6 ohm load @ 1khz and the amplifier is hooked up to a speaker that presents 2.5 ohms at that frequency, what does the amplifier do? What about the other frequencies? In other words, the buffer can never tell what the load presented to the amplifier is going to be and the amplifier will do what it would have done without the buffer at low impedance -- switch to class a/b or b. The problem with class A amplifiers is that the bias voltage has to be set so the quiescent current does not overheat the devices. However, when the load impedance drops, the higher output current causes a higher voltage drop across the emitter resistors. If this voltage drop exceeds the bias voltage, one of the complementary pair transistor (current sink) shuts off and the amp goes class B and shoots out a lot of distortion from the hard shut down. In order to keep class A at the lower impedance, the bias voltage has to be increased above the emitter voltage at that impedance. Now there is a higher quiescent current and more heat sinking is required, not to mention the increase in power supply filtering in order to keep the input/driver stages from being affected. So the trick to sliding bias is to not anticipate the load current but to compare it to a reference, a reference that allows for a manageable Q current. I don’t know what Krell uses, but sliding bias can easily be done by monitoring the voltage across the emitter resistor by setting up a reference voltage with diodes in conjunction with resistors. The output across the resistor is compared with the reference voltage and if there is a high current draw, the resulting voltage drop is sensed by the diodes which then open a transistor whose collector steals current from the output transistor base, removing some of the quiescent current gradually, allowing the conducting transistors to operate in class a instead of being reversed biased, shutting down and operating in class B. |
Thanks for the response. Yes a2d, at the outset Krell said that once raised the higher bias will be held 15 to 20 seconds before it is dropped back if there is no further hi-level signal. But that doesn't answer how the first rise is triggered in time and what happens if there is a sudden new peak after 25 seconds. One might postulate a single heavy stroke on a large bass drum every 25 seconds. And Krell informed me there is no buffer=delay circuit. |
As a total guess a tracking peak-hold with delay circuit that drives the bias stage. The bias may rise with the waveform and then stays there slowly decaying down if the music quiets. Won’t be instantaneous but for slow moving signals it may me good enough and for higher frequencies you need a few waveforms to pick up characteristics. |
