[DVV]

Not in my book, but I was lost awhile ago   If you ever find yourself in Cyprus, you're welcome to analyze my SL equipment.  I love the sound and have no regrets as far as the cost but I do wonder what makes it tick.  There's not much written about Symphonic Line that I've found, aside from my own babblings, and I feel there should be.  

cheers,

Dick

Dick, I was on Cyprus last in 1969 (or was it '68?). I usually spend my vacation with my family on mainland Greece. It's about 9 hours' drive from my apartment, close you might say.

As for Symphonic Line, I don't think you are babbling, my admittedly limited experience with their product indicates they should be taken very seriously.

As for being lost, if I can help, let me know. The biggest problem with writing about technology is knowing where to draw the line, since you can't accommodate everybody. Like others, I too overdo it sometimes, sort of go over that line without even realizing it.

Cheers,
DVV

[audiojerry]

DVV, if you are willing to continue producing, I'm an avid reader. I myself am feeling a little guilty that I'm taking advantage of such great information for free...

I'm tickled pink!

What's next? Can you maybe offer some insights on the merits or hype relating to fully balanced, dual differential, designs and whether they are inherently better?

[DVV]

DVV, if you are willing to continue producing, I'm an avid reader. I myself am feeling a little guilty that I'm taking advantage of such great information for free...

I'm tickled pink!

What's next? Can you maybe offer some insights on the merits or hype relating to fully balanced, dual differential, designs and whether they are inherently better?

Jerry, it's almost impossible to go on without illustrations. I mean, why write 500 words to describe something you can draw?

Also, I think some of our members here are starting to lag behind, and that will definitely not do.

What I'm thinking about is synthesizing it all into one, with illustrations, then making a PDF file of it. That way, everybody can download it and read and reread it at their own leisure.

Not that there's much left to cover, mind you. The next step would logically be to crash course you into a designing attempt, but that's deceptively easy.

The only way left to go is to start delving into details, resistor by resistor; the only alternative is to sit back and answer incoming questions.

Cheers,
DVV

P.S. I just spent most of the day sorting out the tons of schematics I have. I want to put them up on my site's download area, sorted out, so anybody can download them. That might help - print out a schematic, look at it, and we can all go over it, part by part.

[hairofthedawg]

Not completely lost, just a totally different area of electronics for me.  I'm used to fixing things, not designing.  Mostly swapping circuit boards but that's changing too.  Getting to be more swapping whole units out here in the field.  Sort of easier, sort of not.  Lots of FM going on lately.

Yeah, they should be taken seriously...just wish they had a better spokesman on this board.  Klaus knows what's up, but as for me, I'm just a guy lucky enough to have enough money  at the right time.  It's nice that  Tyson has the preamp now too so he can give a frame of reference.

Regardless, I appreciate you sharing your knowledge and please let me know if you'd like to head down this way.  Plenty of room!

cheers,

Dick

[karthikn]

Hi DVV,

This is a question that is a few posts behind.  But I had to read and reread and make sure I understand enough before asking a question:

The second voltage gain stage and this generator are connected by a bias tracking network (also called a simulated zener diode). This is typically a transistor whose job is to monitor the temperature rise of the output stage and adjust the bias (quiescent) current as per the requirements. It can also be used to set and/or change the idle current of the amplifier

Is this transistor really a temperature measuring device or a device which gets a measure of temperature based on the current passing through it ?  Either way, wouldn't the ambient air temperature affect its measurement and hence the bias current vary under different temperature conditions ?  

BTW, please do continue and dont be put off by the silence.  I probably speak for a lot of people who read this when saying that it takes time to understand this.  A measure of that would be this thread being alive for a long time. Pictures would definitely help and please let me know if I can help in any way.

Karthi

[audiojerry]

BTW, please do continue and dont be put off by the silence.  I probably speak for a lot of people who read this when saying that it takes time to understand this.  A measure of that would be this thread being alive for a long time. Pictures would definitely help and please let me know if I can help in any way.

Here, Here! Look at the number of reads: 1365 and counting

DVV, have you thought of publishing? I'm going to give some excerpts to my son, who is a junior in computer engineering, but I'm trying to kindle more interest in him for audio engineering. I'm encouraging him to press his school to become more involved in this field, maybe as an audio club or R&D group in amplfier circuitry. I think it would be a worthwhile pursuit for a university to do research on finding empirical ways to define and evaluate ideal audio circuits.

[DVV]

BTW, please do continue and dont be put off by the silence.  I probably speak for a lot of people who read this when saying that it takes time to understand this.  A measure of that would be this thread being alive for a long time. Pictures would definitely help and please let me know if I can help in any way.

Here, Here! Look at the number of reads: 1365 and counting

DVV, have you thought of publishing? I'm going to give some excerpts to my son, who is a junior in computer engineering, but I'm trying to kindle more interest in him for audio engineering. I'm encouraging him to press his school to become more involved in this field, maybe as an audio club or R&D group in amplfier circuitry. I think it would be a worthwhile pursuit for a university to do research on finding empirical ways to define and evaluate ideal audio circuits.

Been there, done that, Jerry. In 1989, I wrote and published a book called "Computers and Words - Text processing". The first round sold out, they reprinted, the second edition sold out, and then the local wars overtook us and stopped most publishing.

But yes, I would think of publishing in an edition called say "Audiophilia for Audiophiles", or some such. Where different people (VERY important, different people, no gurus, thank you) took up different subjects (CD players, DVD players, turntables, loudspeakers, etc) and explained them in not very technical, not very precise, but easy to understand words. Not vulgar, not flippant, but understandable to a layman.

But I'm just warming up so far, I am about to hit you people with what you never thought was possible. For example, did you ever think you would be able to design your own active crossover? Yet, National Semiconductor published a wonderful book way back (1980 - National Semiconductor - Audio/Radio Handbook). It's worth its weight in enriched uranium, believe me. It covers about 90% of what there is to cover in audio. The section on active crossovers is so GOOD, I scanned it and will post it up on my site soon enough.

As for this little series, take your time, no hurry, think about it, and ask. Don't worry about not asking the Ideal Question, ask whatever you want to know.

Cheers,
DVV

[JoshK]

FWIW Dejan, I would buy it!

[DVV]

Hi DVV,

This is a question that is a few posts behind.  But I had to read and reread and make sure I understand enough before asking a question:

The second voltage gain stage and this generator are connected by a bias tracking network (also called a simulated zener diode). This is typically a transistor whose job is to monitor the temperature rise of the output stage and adjust the bias (quiescent) current as per the requirements. It can also be used to set and/or change the idle current of the amplifier

Is this transistor really a temperature measuring device or a device which gets a measure of temperature based on the current passing through it ?  Either way, wouldn't the ambient air temperature affect its measurement and hence the bias current vary under different temperature conditions ?  

...
Karthi

No, it is the actual sensor. It has to be mounted as near to the output devices as possible, so it changes temperature along with them. It's a simple enough circuit in its basic form, but it's not very precise, more of an approximation. However, over time, a few more complex but also more precise circuits have been developed, as well as a few tricks to prevent it from overcompensating too much (i.e. while not elimintaing, at least minimizing the error).

Personally, I do it much differently, but then, most of my pet amp project is different, not in wild ideas, but in the execution of what is necessary. Let's say I think a little laterally. If you should take that to mean that I'm a bit off my rocker, you'd be right.

Cheers,
DVV

[DVV]

Fully balanced topoligies mean that we have in fact two standalone amplifiers working in parallel, one for the positive, and the other for the negative part of any sine wave.

This topology has some advantages, like being able to work off lower voltage supply lines, which is always good for the output devices, and the fact that it doesn't rely on the ground for a reference point. Hence, any junk in the ground cannot influence it, nor can any change of ground potential influence its operation.

But it has some serious drawbacks too. One is that for it to realy work well, you have to have almost perfectly matched amplifiers, which is never easy to do. However, I must note here that some very interesting topologies have been shown to work, and work well - for example, James Bongiorno's Sumo 9+, made way back in 1982. Exemplary work, that one - simple, ellegant and efficient, and above all, it sounds fabulous even after all these years.

The second drawback is that it is usually (but not always) more complex than a single amp. More parts mean less reliability by default, more chance of something going wrong.

Fully complementary is somewhat similar, but not the same. This assumes that is every NPN transistor has a complementary PNP transistor, their inherent differences will tend to even out and significantly reduce even order harmonic distortion (2nd, 4th, 6th, etc harmonic), leaving only odd ordered harmonics (3rd, 5th, 7th, etc) to be minimized by the negative feedback.

To the best of my knowledge, this topology was first used by - hang on, now! - James Bongiorno in 1971.

It has been shown that the basic assumption of reducing even harmonics is, or can be, true.

This is the only place where Dan Banquer and I part company (figuratively speaking, of course), because Dan uses this topology, and I don't, obviously because he thinks it's better, and I don't. I may have the greatest respect for Dan's work as I do, but I do not believe this approach is inherently better than many others. It does reduce the even harmonics, but in my view, the best of amps is the one in which all forms of distortion are equally present. Feedback is essentially non-selective, it feeds back EVERYTHING, although of course, it can be tweaked by altering its characteristics, using dual feedback lines (what Luxman in its day called Duo-Beta) with very high feedback for DC to keep the amp's DC balance in good shape and less feedback for AC, adding non-linearizing elements (the simplest of which is bypassing the feedback resistor with a capacitor, so feedback rises above a certain point, thus enlarging the amp's linear bandwidth), etc. Heck, DC servo is essentially a second feedback path, with tremendous feedback below a certain point, typically below 1 Hz, and zero feedback above say 5 Hz.

I believe that the inherent differences between NPN and PNP transistors will always be such that a fully complementary topology will never really improve matters to the point of being clearly superior. For one thing, it uses more transistors, each one of which is a possible source of problems. To me, it's a classic case of quid pro quo, gain here, lose there.

Mind you, this is just my personal view, and if truth be told, then we must note that some of the best amplifiers of our times have used this topology - just as other best amplifiers of our times did not use it.

Another interesting topic here is feedback - yes or no? I refer to overall feedback, which is taking a part of the signal from the amp's output and feeding it back to the input stage.

Some, and particularly the modern Scandinavian school, maintain that zero feedback is better becouse of superior transient response. The idea is to use local feedback around each stage, the rationale being that distortion should be dealt with where it appears, right there and then. It is possible to obtain large bandwidths, low distortion, good damping factor and high slew rates without overall feedback, and these are usual reasons for feedback, which makes one wonder why do we need it, if we can achieve our goals without it?

Superficially, this is all true as far as it goes - but it doesn't go far enough. True, you can have low overall distortion, high slew rates, good damping factor and a large bandwidth without it by applying local feedback. Therein lies the first potential danger - to get it, you may have to ovedo local feedback, do on micro scale what you are trying to avoid on the grand scale. But if you spoil the sound of any one stage of the amplifier, the end result will invariably be a poor amplifier.

The second problem lies in subtle mismatches between stages, between transistors, components, etc. Since this is dealt with locally, formal distortion is taken care of, but slight mismatches will still be in our output signal, with no overall feedback to cancel them, or at least reduce them.

Personally, i don't like the sound of zero feedback amps. I have heard quite a few, and not one sounded natural to me, all had one fault or another. If they had anything in common, it would be a two dimensional sound stage, sort of flat. Detailed, precise, but flat. And that I just can't stand.

My ideal are the so-called low feedback amps. What is high and what is low is of course relative, but over the last few years, it has been accepted that overall feedback of 20:1 (26 dB) or less is low feedback. In 1993, Harman/Kardon's 6550 integrated amp had an overall feedback of just 14 dB (5:1), and that was a commercial unit not even pretending to be high end, though it was more expensive than most in its power envelope. In 1999, Harman/Kardon's 680 integrated amp had an overall feedback of 10 dB (3:1). Both sound really good, believe me, I have both at home. But in the late 70-ies, feedbacks of 40-60 dB (100-1,000:1) were quite common.

In my view, the PROPER way to use feedback is to make the amp as linear, as wideband and as low distortion as possible without any overall feedback - then add just enough feedback to make something good even better yet. That H/K 680 is a perfect example - without any feedback, its distortion is about 0.5% and it goes above 85 kHz open loop (no feedback). Then they add just 3:1 feedback to get the distortion to below 0.09%, and the bandwidth out to beyond 250 kHz. Just what the doctor ordered.

To know how it's possible, remember that their head designer, Richard Miller, worked with Prof Matti Otala for 12 years, 1974-1986, producing such outstanding designs as Citation XX. Matti Otala identified Transient Intermodulation Distortion (TIM), and proposed methods to measure it, and developed tests for it and showed ciruits rid of the problem in 1972. Then he talked about loudspeaker load complexity, how it works and what to do about it. In short, together with James Bongiorno, he's the most influential person on this planet alive regarding power amplifiers.

Which brings us to the most crucial issue - what do we design the amplifier for? Speakers, you say? Ah-ha, but how do speakers behave? What will the amplifier actually see as its workload?

Will the amp make it? Will the Speaker triumph over our hero? Watch this space for more revelations, same time, same channel.

Revealingly,
DVV

[DVV]

The !@#$%^&* muting circuit is on again. All i get is silence, no channel is working ...

Cheers,
DVV

[seppstefano]

Hi there,
here in Italy there's somebody (not me) who exactly thinks that the amp makes sense only when seen in the context of the whole system: source - amp - loudspeaker. Besides this guy is a fan of zero feedback and, so, claims that any element of the chain should be zero feedback.

Apart from zero feedback considerations, I get the point that an amp by itself is almost meaningless. But if you do match exactly an amp to a given output (or input) you must cope with some limitations (at least!).

IMHO it's all about money.

If you do the best possible amplifier for Loudspeaker Stephanus, you will target mainly Stephanus loudspeaker owners (which may be not so many...) On the other side, by building a reasonably well made amp will make you able to target a wider market!

Just my although obvious ,02

[hairofthedawg]

Sorry if this has already been covered...I was just looking at all the photos  from CES that fatherandsonaudio very graciously posted, thanks guys, and noticed that some of the designers put a lot of thought into the chassis of their amps.  Understandable from a marketing standpoint, but does the chassis have much effect?  I understand the heat sinks.  I guess it should be heavy enough to be immobile...  Are there any other considerations?

[DVV]

Sorry if this has already been covered...I was just looking at all the photos  from CES that fatherandsonaudio very graciously posted, thanks guys, and noticed that some of the designers put a lot of thought into the chassis of their amps.  Understandable from a marketing standpoint, but does the chassis have much effect?  I understand the heat sinks.  I guess it should be heavy enough to be immobile...  Are there any other considerations?

Actually, yes. It is most undesirable to have your capacitors, usually big and lunky things, vibrate. There are a number of methods by which this can be avoided, and making your chassis as inert, or acoustically dead, as possible is very desireable.

Nor is anyone keen to have their electronics vibrate, excited by natural vibrations of the ambinet and airborne vibrations, but these are small things and not as prone to mishap as large and bulky caps.

Cheers,
DVV

[karthikn]

Hi Dejan, in your examples you mention how feedback results in increased bandwidth.  Can you elaborate on how the effect is achieved ? Thanks, Karthi

[DVV]

Hi Dejan, in your examples you mention how feedback results in increased bandwidth.  Can you elaborate on how the effect is achieved ? Thanks, Karthi

By correcting the signal that would otherwise be lost in distortion and level. However, it is NOT a linear function. You cannot say that if you add say 20 times the feedback, your amp will have a 20 times larger bandwidth. It will have a larger bandwidth, but by how much depends on many design factors inside the amp.

There are other considerations as well. For example, your feedback could be defined by a simple resistor voltage divider, the simplest (and usually best) form, but you could also add more elaborate schemes, such as those which increase feedback over some point, to enlrage your useful bandwidth still more. Even multiple feedback paths.

All of this is being done. Each approach has its good and bad sides. My personal view is that the simpler the better, but then, I can afford to be ellegantly simple because I go for a wide response without any feedback anyway. If I get 60 kHz or more of response without any feedback, a feedback factor of say 20:1 (26 dB) will push this over 1 MHz, but only because my topology is such that feedback is rather comprehensively acting. In other topoligies, it might not do so.

An amp by a well known manufacturer has an open loop bandwidth of less than 5 kHz. 26 dB of feedback will ideally enlarge this to 100 kHz, and since he needs more than that, he is forced to use more feedback.

But I think the problem of feedback deserves to be mentioned all by itself, it's a very interesting topic.

Cheers,
DVV

[AKSA]

Folks,

I have studied amps quite intensively for about eight years now.  I have no particularly preference for tubes or SS, and like to build hybrids.  I sell two kitset amps called the AKSA, one of 55W and the other of 100W, a new preamp called the GK-1, and a small tweakable two way speaker.

I am not an engineer, and resolutely develop my circuits by ear, using only a meter and CRO ocassionally.  The 100W AKSA gives 0.045% into 8R resistive at 20KHz and 10 watts, sounds pretty good, and so I offer here my philosophy for designing amps.

****************************************************
· Prevent Interstage Crosstalk – Decouple the supply rail for the low current stages.

A hard working output stage creates heavy, periodic voltage sag on the supply rail, and this has adverse impact on the voltage amplifier and to a lesser extent on the differential input stage.  Interstage crosstalk can be almost eliminated by fitting a diode and small resistor in series from the positive supply rail and a decoupling electrolytic capacitor to ground.  Interestingly, sonic testing reveals that a similar network on the negative rail has no beneficial effect.  This simple decoupling network ensures that supply rail disturbances created by a hard-working output stage never interfere with the operation of the voltage amplifier and the input differential pair.  By ensuring that complex interactions cannot occur between the output and input stages, a potential source of serious intermodulation between the two input stages and the output stage of the amplifier is eradicated.  Listening tests reveal a cleaner sound, particularly on heavy musical passages, with superior resolution, clarity and a notable lack of intermodulation.

·  Foster High Linearity – Operate the voltage amplifier at constant current
Remembering the sterile sound and high frequency complications of the constant current source leads to an old concept called bootstrapping.  The technique uses the amplifier’s low impedance output to dynamically elevate the voltage amplifier power supply in step with the output signal, permitting the voltage amplifier to run at close to constant current.  Constant current operation for any amplifying device improves linearity, minimises distortion and greatly ‘speeds up’ the amplifier by removing loading on the voltage amplifier.  The bootstrap also has implications for source impedance.  The alternative, the constant current source, has been tried many times but always leads to a dry, unexciting sound – and it confers an extreme high frequency bandwidth which needs to be viciously tamed to ensure amplifier stability.  The taming process costs musical vitality.  To its discredit, the constant current source does confer an asymmetric clip, slicing the negative half cycle long before the positive.  However, although a bootstrap does suffer small current variations, it yields good (though deliberately not outstanding) voltage amplifier linearity and intrinsically confers rapid and convenient high frequency rolloff.  Early rolloff helps to make the amplifier stable without resorting elsewhere to draconian bandwidth limitation thus bringing some immunity to digital high frequency artefacts and radio frequency interference, allowing the designer to reduce the size of the output inductor and lag compensation.  Again, listening tests reveal that a current source powering the voltage amplifier creates a ‘dry’ sound.  The bootstrap gives a subjective impression of speed, attack, pace and liquidity – all of which add impressively to the subjective musical experience and take the listener closer to the passion of the original recording.

·  Minimise amplitude/phase Intermodulations – Split DC offset and AC feedback control.
This technique effectively changes the proportions of resistance and capacitance at the feedback node.  This reduces adverse signal damping effects arising from the RC series chain in the network whilst preserving offset control.  It also creates two AC paths to the blocking capacitor – usually an electrolytic – and this sets up a secondary charge path which masks the poor sonics of such capacitors by minimizing electrolytic memory effects.  The result is a superior, longer lasting decay, particularly on fast percussive material such as cymbals, a considerable reduction in the size of the shunt capacitor and an improvement in sonic ambience.

·  Eliminate Switching Transients – Implement charge suckout on the output stage drivers.
The output stage introduces switching non-linearities on the output as one output device hands over control of the speaker voice coil to the other.  Base junction charge suckout, identified by Self in his impressive 1993 series of articles, involves placing a small resistor and capacitor in parallel between the two driver emitters in a conventional darlington output stage.  This parallel RC network ensures that the drivers (and with them, the output devices) turn off more cleanly when speaker current flow passes to their opposite numbers at the crossover of the signal.  This greatly reduces switching distortion, the bane of all Class AB amplifiers, by removing the usual spray of high order switching spikes generated by the handover event.  It smoothes the transfer of the baton in the musical relay.  It is this phenomenon which is partly responsible for the splashy, grainy and often harsh top end of most solid state amplifiers, since the ear readily registers switching disjunctions at high frequency and the feedback loop lacks the necessary speed to correct it.
It is clear that many modern amplifiers deliver poor performance because of short term stability problems which manifest only in musical passages but rarely in steady tone testing favoured by contemporary testing regimes.

·  Choose Semiconductors with care.
The final step in the re-design concerned the choice of semiconductors.  To achieve good slew characteristics, the speed of all active devices is important, particularly the voltage amplifier.  Also important is the output stage current versus voltage linearity (sometimes referred to as transconductance), since this ensures a high feedback factor under conditions of high output at high frequencies.  Note that the speed of the output devices is NOT of the importance one might expect.  Of course, cost and availability are also important factors.
It is important to maintain a consistent current gain in the output stage as the speaker current demand increases.  This parameter relates to large signal linearity;  a vital component of the amplifier’s overall transfer function.  The choice finally settled on one transistor complementary pair which achieves constant gain (hfe or beta) from 100mA to 7A.  This was the 2SC5200 (npn) and the 2SA1943 (pnp).  These devices were developed by Toshiba specifically for push pull audio amplifiers, and are rated at a fast 30MHz.  These are impressive figures for a 12 amp, 230V, 150 watt transistor with excellent SOAR ratings.
The drivers must also exhibit high current capacity, with good linearity and matching speed.  A high current rating ensures they are not destroyed if the output devices are subjected to a momentary short, and a consistent current gain across a wide range fosters linear, stable performance into difficult loads.  The devices initially chosen were the ON Semiconductor (formerly Motorola) MJE15030 and MJE15031, and these are still used in the 100W AKSA.  These devices are commonly used as drivers in the very high power amplifiers of professional audio, and are virtually unburstable in a small, low power audio amplifier.  They can easily withstand a short term current burst of 8 amps, almost 100 times the working requirement in the AKSA circuits.  This rating guarantees a long, reliable life – very important in this application, and especially in a kitset where robustness is mandatory.  Recently new drivers have become available from Toshiba specifically for the chosen output devices, and if anything, their performance is superior, particularly in terms of linearity.  These devices, now fitted to the 55W AKSA, are 2SC4793 (npn) and 2SA1837 (pnp) and their speed is almost triple that of the Motorola devices.
The voltage amplifier is particularly important, since the amplifier’s entire voltage gain is incorporated in this common emitter stage.  Only this transistor configuration gives both current and voltage gain, and this is heavy going for a transistor owing to Miller capacitance, so it must be very fast and linear with high gain.  It must be able to withstand more than twice the rail voltage of the amplifier at 10mA without a heatsink.  Such a transistor with very low Miller capacitance is critical to good performance in a quality SS amplifier.

The input differential pair transistors are perhaps the easiest to choose.  Since they are small TO-92 devices, their die is small with low parasitics and they are fast.  Virtually any PNP device with a rating of at least 20mA and 100MHz is adequate.  However, it is fairly important that they be closely matched for gain, as this confers precise differential pair current balance.  Current mismatches affect the sonics and the DC offset of the amplifier quite significantly.  Finally, the 150V 2N5401 and BF491 were chosen from ON Semi and Philips, as these are inexpensive, quiet, very fast and available in batches with consistent beta, making matching relatively easy.
********************************************************

This article delineates what I have found to be important.  I claim no originality;  just careful, painstaking assembly of a variety of simple designs and meticulous listening.  The interpretations are mine;  some can be disputed on math grounds, and I don't pretend to follow some of the more obscure modes and happily eschew PSpice, which tells me almost nothing about the sonics or even the stability margin.  Some of this material amounts to audio heresy, and I make no apology for that, but neither will I strenuously defend my points because I haven't the time or energy.  The proof of it all lies in a good listen to my amps, nothing less.

I hope it's of some use, and have to say this is a very IMPRESSIVE forum, and my sincere thanks to JohnR, the Borg who set it all up!  

Cheers,

Hugh

[JoshK]

Thanks Hugh for your input!  I was beginning to wonder if you would chime in.  I think this forum is really off to a great start.  I think I am going to purchase a single mono 100 kit from you soon for my center channel to be, if things go well I will likely buy another pair of mono kits for the rears.  

What are you guys' thoughts on digital amps?  I know the technology is still infantile in comparison to tube or SS, but owning a Spectron (because of its sound, not its audiophile theory pleasing ability) I can say that they do a lot of things wonderfully.

[MarinRider]

Glad to see Hugh Dean join this thread.

Most of my opinions relating to amplifier sound quality (expressed earlier in this thread) come from countless experimental modifications to Borbely's DC100 amps. This is why I now avoid current sources (or use with extreme caution), dc servos (fine if you don't want fine detail), dual complemetary topologies (no point) and cascodes (usually), and concentrate on simple designs and proper implementation.

A year or so ago I read Hugh's amp philosophy (which went a bit further than mine - i.e. bootstraps etc) and was so impressed that I made a lash up of what I thought was in his amp. It made me happy and sad - happy with the sound quality and sad that Hugh's amp was better than all my experimental versions of Borbely and Lindsley Hood amps. So I bought the 100W AKSA with Nirvana upgrade. The offical  ASKA was/is hugely better than my lash-up.

I have tried all sorts of ways to improve the ASKA and failed (I had no problem getting the JLHs and Borbely amps to sound better).

I don't build power amps any more, I just listen to them (and spend far too much of my time reading AudioCircle and DIYAudio.com).


BTW, I have seen mention of saturation of mains power transformers in power amps in various threads. Can anyone explain the mechanism by which this happens?

[Dan Banquer]

A few words about feedback and amplifiers. Without global feedback, or to put it more precisely, an amplifier in open loop mode will typically have very high gain at low frequency and have a pretty limited bandwidth. Feedback sets the gain and helps extend bandwidth if the feedback is low enough, which is usually the case. Too much  global feedback and you have an amplifer that will be less stable and may well have such artifacts as slewing induced distortion, which is the result of limited bandwidth. In short, feedback helps set the gain and bandwidth of a power amp.