Steve, you just asked for a complete theory of operation for semiconductrs in a few short sentences. This is something one usually needs years to understand, and after 34 years, I'm still not sure I understand it completely.
Hi Dejan,
See, that wasn't so bad was it?
You think so? Look below ...
So, just to emphasize (or clarify):
1. If you set some initial bias on the output transistors and didn't use any feedback or DC servo, you would have speaker damaging DC getting through. Am I correct in making this statement? Aren't some amps designed without feedback?
No, you are not. As far as I know, it is not possible to make any amplifier, of any sort, without feedback. It's just a matter where and how you apply it. "Zero feedback" does not exist, it is a flippant and incorrect way of saying "no overall feedback", but to have it work, you had to apply local feedback.
Let me put it this way - a typical active device (tube, transistor, etc) has a certain gain (amplification) factor under certain circumstances. This can be anything from say 20 to well over 10,000. Now, a typical say 50W amp will need say 1V of input signal to make it to 50W, which is 20V continuous, and therefore an overall amplification factor of 20:1. Tie up any two transistors or tubes, and you will have an amplification factor of at least (20x20) 400:1, even up to several million to one. Furthermore, your distortion would be record beating at this level. So you have to reduce both gain and distortion. Reducing gain will also reduce distortion, but only up to a point, so after that point, you either feedback it, or you end up with unholy muck on your hands, good for nothing. Less gain, but still good for nothing.
So you apply feedback. It can be local or global, but the best way is to use both - judiciously. But we're moving into the feedback theory now ...
Anyway, however you do it, DC drift will also be reduced. With proper design and execution care, DC offset will be low, typically below 50 mV with the least measures taken. Even in the dark days of audio, it was rarely over this value, except on the junkiest of equipment.
So, exactly what is crossover distortion? I had always heard that it was the distortion of a device turning on and off. But because of a bias setting, the device is never really off (in either class A or class A/B). Knowing this, I would then assume that crossover distortion doesn't have to do with a device going from conducting to not conducting. But rather has to do with a device going from hadling its portion of the audio signal to not handling its portion of the signal.
You almost got it. Crossover distortion is really caused by a miniscule lag, a time gap which appears when the NPN device stops conductiong and the PNP device has not yet started conducting (or vice versa). What you refer to above is called switching noise, which is broadly speaking the random noise created by a device when it changes states, from conducting to non-conducting and vice versa.
Crossover distortion is a necessary and an unavoidable evil of any push-pull configuration. However, it can be reduced by various techniques to really academic levels, the only problem being the price of doing it.
It depends on many things, from mode of operation, to the speed of the devices you are using.
If I am correct in my previous statements. I am still confused as to how class A operation produces such small amount of crossover distorion as compared with class A/B? In your example above
once the power demanded of it exceeds 2.9/5.8 W into 4/8 ohms
in the low power range (class A), why are the lower power outputs yielding less crossover distortion? The devices are still handing off to one another.
Sorry for the somewhat repetative nature of my questions. This is a subject that I have been contemplating for some time, and have never really been able to get a handle on. Your fantastic explanations have gotten me closer than ever to full understanding. One more little step, and I think I got it.
Thanls a lot Dejan -- so far awsome,
Steve
Very simple, Steve. In class A, the devices are always in their ON state, they conduct their full current all of the time irrespective of whether there's any signal at all or not. Since they are operating on a constantly open basis, and do not switch on and off, their switching distortion is naturally reduced to almost nothing.
If both devices are in their full ON state, there will be no time lag when one switches off and before the other switches on. This is the key benefit of full class A operation, everything else are its downsides.
Remember that both switching and crossover distortions are compunded in amplifiers which use multiple pairs of output devices, because one simply cannot match perfectly even several NPN devices among themselves, let alone each of them with a corresponding PNP device. Of course, there are techniques to overcome this problem as well, believe me, I've been using no less than 2 pairs of output devices per channel for over 20 years now, because this approach has its tremendous virtues as well as downsides.
Lastly, both problems are really appearent only at low level listening, because both tend to be masked by music. In other words, they don't go away at high power levels, they are just much less obvious as such, but can still cause the loud sound to be smeared and muddy, for example. This is why I use the high bias approach - incidentally, the 600 mA figure I quote is "my" sweet number, that's what I always go for - this all but kills the problem dead at critical low levels, but without the extreme penalties of full class A operation.
A year and a half ago, I wrote a text published on TNT (
http://www.tnt-audio.com ) titled "Put A Tiger In Your Amp", in which I described this kind of tuning mass produced audio gear, with some nice shots of the carnage I wreaked in my own amps. This is an easy tweak, and one that becomes obvious inside of 30 minutes after doing the evil deed. But it never ever fails. Point made.
Cheers,
DVV