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#1 |
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diyAudio Member
Join Date: Sep 2005
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Okay, here's a question regarding your basic three stage power amplifier and how to compensate it. Normally, the compensation capacitor is placed from the output of the first stage to the output of the second stage (and thus it envelopes the VAS stage).
==> The slew rate depends upon the bias current in the differential input stage, and the bias current in the VAS stage. What if we arranged it so that one end of the compensation cap is connected to the output of the driver stage? In this capacity, the bias current of the VAS stage does not determine part of the slew rate of the amp; the bias current of the driver stage determines it (and the driver stage current is usually higher than the VAS stage current, which means more current to discharge Ccomp, which means higher slew rate). As seen on the picture, instead of connecting the floating end of Ccomp to the collector of Q3 (as is traditionally done), why not connect it to the emitter of Q4? |
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#2 |
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diyAudio Member
Join Date: Nov 2004
Location: Florida
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Not sure why it's not been tried.
The logic makes sense. I'll try it. |
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#3 |
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diyAudio Member
Join Date: Jun 2004
Location: Warsaw
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As far as I remember long lost mr. Evil experimented with this, but from different reasoning.
I don't think you will improve slew, rather you will make it more asymetric. Not sure though. regards |
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#4 |
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diyAudio Member
Join Date: Jan 2002
Location: Earth
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Hi rtarbell,
I'd say the primary reason this is not usually done is because of stability. Q4 introduces extra phase shift that reduces the phase margin of the miller stage. In either configuration, the slew rate is primarily limited by the supply current of the stage driving Q3's base, rather than the collector current of Q3, so including Q4 in the loop doesn't help that much. Consider the extreme case of the beta of Q3 being infinite. You'll see that the first stage effectively charges and discharges Ccomp...like an op-amp integrator. The fact that Q3 beta is finite only adds to the burden of the first stage. BAM |
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#5 |
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diyAudio Member
Join Date: Feb 2003
Location: ..
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Dr Ed Cherry proposed this long ago, in fact he prefers enclosing the whole output stage
http://www.diyaudio.com/forums/showt...ht=#post228259 http://www.diyaudio.com/forums/showt...rry#post281394 (and next post) MikeS has some objection to enclosing output Qs (apparantly objecting to bjt outputs) which he has yet fully to share with us in the "how many ways to stabilize" thread: http://www.diyaudio.com/forums/showt...197#post647197 compensating from a buffered vas allows the use of 2 pole compensation with with unequal C, potentially removing the Miller multiplication of the compensation C seen at the vas input at audio frequencies - without buffering the vas would be loaded by the larger C, R to (ac) gnd on it's output. limiting audio frequency gain improvement from the 2 pole compensation |
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#6 |
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diyAudio Moderator Emeritus
Join Date: Oct 2002
Location: Bandung
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Fig A is the most common compensation, miller cap, between B/C of T1 (VAS transistor). The value is pF.
DougSelf said that Fig B is "sub optimal", since it is loading collector of T1 by big capacitance (nF value here) I'm not thinking voltage/current phase. But fig B seems to be a lowpass filter (in conjuction with internal impeance of T1 collector). If Fig B is actually a lowpass, Has anybody tried fig C? Inductor there (in the position of base stoppers) should be a lowpass too. |
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#7 |
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diyAudio Member
Join Date: May 2005
Location: Pretoria, South Africa
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The C placed as in A in Lumanauw's sketch is universally used but can pose stability problems since it remains active in a manner of speaking. The method B is more docile. It is indeed a simple low-pass filter affording the desired phase correction. I disagree somewhat with Douglas Self (something that I have never done on any other occasion!), in that for optimal design this compensation should apply ideally outside the audio band, He is correct in saying that it gives rise to distortion, but that is only relevant if compensation starts within the audio band (and how it does in some circuits!) I would also recommend reading of Dr Cherry's relevant articles.
In that sense inductors could indeed be used as in C but I have never seen it - perhaps cost? In a circuit of my own I use topology B, with a capacitor of only 220 pF required, starting effect at about 25 KHz only. If I try to get similar results by A, I get oscillation, or at least a sharp peak in response at about 3 MHz. But there is a preferable way that I have mentioned before on another thread, which was expounded in an article by Dr John Ellis in Electronics World March 2003, called "Audio Power Amplifier Feedback Compensation". There he showed the superiority of a phase-lead-input-lag type of system, which avoided the disadvantages of the commonly used Miller C.dom. (Lumanauw, I think that I posted a copy of that article to you.) In general, the advantages of also using a capacitor in parallel with the feedback resistor must not be overlooked. |
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#8 |
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diyAudio Moderator Emeritus
Join Date: Oct 2002
Location: Bandung
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Hi, Johan,
Thanks again for the article |
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#9 |
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diyAudio Member
Join Date: May 2005
Location: Pretoria, South Africa
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Lumanauw,
I will have to do a Spice analysis regarding that, when I have time. The next couple of days are somewhat tied up - will come back to you. Regards. |
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