Terry Demol said:
WRT Hawksfords OP topology, why does it not reduce
distortions of OP devices???
Cheers,
Terry
OK, this reduces!!!
but the question remains: which is speaker's impact on this correction?
i do very like Hawkswards topology!
There were posts about Charles Altmann's SPLIF amplifier topology in which diyers criticised the concept.
http://www.diyaudio.com/forums/showthread.php?threadid=3558&highlight=
You should look at paper Class A Power Modules presented by Erno Borbely in Audio Electronics 4/97; his concept was quite different and was warranted by his many years lasting experience.
http://www.borbelyaudio.com/index15.htm
http://www.diyaudio.com/forums/showthread.php?threadid=3558&highlight=
You should look at paper Class A Power Modules presented by Erno Borbely in Audio Electronics 4/97; his concept was quite different and was warranted by his many years lasting experience.
http://www.borbelyaudio.com/index15.htm
Sam9! I want to view that schematicSome of Slone's designs are essentially "Blameless" with some bells and whistles. The "11.4" is a good example. Some of us here have built it and seem very happy with the sound.
🙂
sajti said:
Hi Graham,
most of the amplifiers use output choke, and Miller capacitor at the VAS stage. And there are some amplifier which sounds good...
I think that Mr. Self get the THD, and try to kill it. But he doesn't care with other parameters, such as IM, TIM, DIM etc. Reduce the THD is not so complicated, just go higher open loop gain, and apply more nfb. This way will increase the dynamic distortions as well. I don't care as much with the THD. My amplifiers not use large nfb, max. 12-14dB. The THD is bit more, say 0.5-1%, but I think this is not so important in practical apllication (listening)
sajti
yes! timbre is made by THD,TIM, noise.....
I have a agreement with sajti!sajti said:
Hi Graham,
most of the amplifiers use output choke, and Miller capacitor at the VAS stage. And there are some amplifier which sounds good...
I think that Mr. Self get the THD, and try to kill it. But he doesn't care with other parameters, such as IM, TIM, DIM etc. Reduce the THD is not so complicated, just go higher open loop gain, and apply more nfb. This way will increase the dynamic distortions as well. I don't care as much with the THD. My amplifiers not use large nfb, max. 12-14dB. The THD is bit more, say 0.5-1%, but I think this is not so important in practical apllication (listening)
sajti
timbre is made by THD,TIM, noise, slew rate.
You try to measure TIM,slew rate of your amp
thanh said:With No global feedback ,how can I solve " DC output" ?
It's very easy to do that without needing to go to the expense of a dc servo.
If you split the normal NFB resistor into two, with each resistor of half the resistance of the old resistor, and then decouple the centre point to ground with a capacitor, you still have NFB at very low frequencies to control the dc gain, but no feedback at higher frequencies. You can then experiment with different types of feedback to control the audio frequency performance.
I think that Mr. Self get the THD, and try to kill it. But he doesn't care with other parameters, such as IM, TIM, DIM etc. Reduce
In general, anything that supresses THD will do the same for IM. TIM & DIM tend to be supressed by by high internal slew rate. At least, in a simulator Slone's are farly high in this regard. I don't think he worries about them particularly with his mirror image designs because good results in this regard are almost "automatic" with those topologies.
On the otherhand, I suspect he puts a lot of work into "tuning" the short circuit protection so that it remains inactive when not needed. If the cicuit becomes active before it should it can add a lot of nonlinearaity to an otherwise good design! This isn't a very exotic or exciting aspect of amp design but one that makes a difference in end result. There is a lot of trial and error testing to get these working just so that one doesn't see or appreciate when you look a the final schematic.
Hi All,
When you have low NFB you do not have a very low output impedance against which loudspeaker system generated back EMF can make a choke react, but then there is not much loudspeaker damping either. Yes some designs are okay with Miller connected C.doms, but only where that component does not reduce loop gain at audio frequencies.
Cheers ......... Graham.
When you have low NFB you do not have a very low output impedance against which loudspeaker system generated back EMF can make a choke react, but then there is not much loudspeaker damping either. Yes some designs are okay with Miller connected C.doms, but only where that component does not reduce loop gain at audio frequencies.
Cheers ......... Graham.
Do you mean Error Correction. Are they "Match made in Heaven" with non-feedback power amp?You can get low output impedance by sophisticated design of the output stage ..
Ouroboros said:
It's very easy to do that without needing to go to the expense of a dc servo.
If you split the normal NFB resistor into two, with each resistor of half the resistance of the old resistor, and then decouple the centre point to ground with a capacitor, you still have NFB at very low frequencies to control the dc gain, but no feedback at higher frequencies. You can then experiment with different types of feedback to control the audio frequency performance.
Which is basically a DC servo with slightly less OLG than an
opamp 🙂
Cheers,
Terry
lumanauw said:
Do you mean Error Correction. Are they "Match made in Heaven" with non-feedback power amp?
Class A output stage or class AB with error correction - both can make very linear output stage with very low output impedance, according to appropriate circuit design.
Pavel, do you think that measuring standing waves can be usefull
I did it and adjusted the coil, but could not perceive sound changes.
the same way we use to measure radio frequency Standing Wave Ratio, to read the mismatches between the Radio Frequency Generator (transmitter) and its Antenna (aerial)...or to measure the entire output..... aerial plus cable...measuring, in this last case, in the transmitter output.
I suppose that my idea may be stupid....because, despite of measurements done....could not perceive nothing with my years.... scope damaged...and PC scope is awfull.
regards,
Carlos
I did it and adjusted the coil, but could not perceive sound changes.
the same way we use to measure radio frequency Standing Wave Ratio, to read the mismatches between the Radio Frequency Generator (transmitter) and its Antenna (aerial)...or to measure the entire output..... aerial plus cable...measuring, in this last case, in the transmitter output.
I suppose that my idea may be stupid....because, despite of measurements done....could not perceive nothing with my years.... scope damaged...and PC scope is awfull.
regards,
Carlos
Carlos,
I do not have exact idea how to use standing waves measurement in audio. I assume that fast signal generator and scope should bring the result as well - provided we speak about oscillations, or reflections caused by impedance mismatch - you know that it is possible to see mismatch of 50 Ohm cable terminated by 75 Ohm resistor, in case you have possibility to generate nanosecond rise time and measure it on a fast scope.
Yes, PC is useless for this.
I do not have exact idea how to use standing waves measurement in audio. I assume that fast signal generator and scope should bring the result as well - provided we speak about oscillations, or reflections caused by impedance mismatch - you know that it is possible to see mismatch of 50 Ohm cable terminated by 75 Ohm resistor, in case you have possibility to generate nanosecond rise time and measure it on a fast scope.
Yes, PC is useless for this.
Hi,
Standing wave problem must be considered if cable lenght is longer than 1/10 signal wave lenght.
Minimum audio wave lenght is for fmax=20kHz ca 15km.
Usually, audio interconects are shorter than 1,5 km.🙂
Regards,
Milan
Standing wave problem must be considered if cable lenght is longer than 1/10 signal wave lenght.
Minimum audio wave lenght is for fmax=20kHz ca 15km.
Usually, audio interconects are shorter than 1,5 km.🙂
Regards,
Milan
Dobro došli Milan,
you are right, but that's not what we are looking for. Cable type, impedance and termination makes difference for HF D/A residuals (output of CD players) magnitude, cable EMI suppression etc. These signals interfere with useful audio band signals and create unwanted frequency compounds, small in magnitude, but unfortunately audible. That's why every cable and every electronic circuit sounds different. Your engineering point of view is correct, but things are more complicated.
Pavel
you are right, but that's not what we are looking for. Cable type, impedance and termination makes difference for HF D/A residuals (output of CD players) magnitude, cable EMI suppression etc. These signals interfere with useful audio band signals and create unwanted frequency compounds, small in magnitude, but unfortunately audible. That's why every cable and every electronic circuit sounds different. Your engineering point of view is correct, but things are more complicated.
Pavel
Dekuji moc Pavel,PMA said:Dobro došli Milan
As you were actually considering how to measure standing waves I merely wanted to tell you that there was no point in doing that since interconnects are way too short compared to the audio signal's wavelength.you are right, but that's not what we are looking for.
Interesting but highly speculative I'd say.Cable type, impedance and termination makes difference for HF D/A residuals (output of CD players) magnitude, cable EMI suppression etc. These signals interfere with useful audio band signals and create unwanted frequency compounds, small in magnitude, but unfortunately audible.
Regards,
Milan
Drodzy Slowianscy Bracia!
what for falling into the new problem: measuring the standing waves, etc.
our goal i think is is to separate speaker's impact on the nfb signal. or maybe not separate but eliminate.
so maybe would be fine to build a simple unit which will be comparing an output signal taken from an "ideal" resistor and from a "real" speaker. a simple unit a'la dc-servo with an opamp for example. then the difference would be mixed with the global nfb signal?
complicated?
what for falling into the new problem: measuring the standing waves, etc.
our goal i think is is to separate speaker's impact on the nfb signal. or maybe not separate but eliminate.
so maybe would be fine to build a simple unit which will be comparing an output signal taken from an "ideal" resistor and from a "real" speaker. a simple unit a'la dc-servo with an opamp for example. then the difference would be mixed with the global nfb signal?
complicated?
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