An experimental 4-th order linear audio power amplifier - Page 1
Forum section: electronics / projects

author: GK
author_rank: Super Contributor
timestamp: May 14, 2013, 01:50:59 pm
content: Hello all,Here are some details of an experimental audio amplifier design I have come up with that I hope will set a precedent for low distortion.It is essentially a conventional audio power amplifier wrapped up into a conditionally stable 4-th order feedback loop to maximise feedback in the audio frequency spectrum.A detection circuit that automatically senses whenever the amplifier enters a non-linear region of operation (such as during power up / down or when approaching/entering clipping/saturation) electronically and seamlessly switches the feedback loop into an unconditionally stable, 1st order system for the duration that the amplifier remains in a non-linear region of operation. This maintains stability under all operating conditions (well, that is the plan anyway)I actually derived this idea from the Hypex patent by Class-D amplifier designer Bruno Putzeys for a 5-th order self oscillating class D amplifier. See here:http://www.google.com/patents/US20110068864However Class D doesn’t excite me much and I’d like to have a go at applying the principle to a linear amplifier. Here is a simplified/preliminary schematic:The majority of the loop gain throughout the audio frequency spectrum and three of the poles of the 4-th order system are provided by the three series-connected integrator stages U2 through U4. There are also three zeros which cause the open loop response to revert to a single pole roll-off as the loop gain goes through 0dB, thus making the system conditionally stable. Here is the fourth-order loop gain and phase response:Now, the capacitor of each integrator is connected in parallel with an analogue switch. When these three analogue switches are activated, the integrators are turned from integrators into basic inverting stages. When this happens the feedback loop transforms into a 1st order system. Here is the open loop gain and phase response with the switches activated:The trick to the whole concept of course, is the detection circuit that activates the analogue switches, switching the control loop into a 1st order system whenever the amplifier begins to enter saturation. This is achieved by a window comparator based on IC’s U6 and U7 that effectively monitors the “error voltage” at the virtual earth of the feedback resistor network. Any major +/- voltage deviation here indicates that the amplifier is coming “out of regulation” and this is when the detection circuit activates to switch to a 1st order response. So then, does it work? Well, here is the amplifier clipping at 20 kHz with almost 6dB of overdrive: (EDIT - the schematic has since been modified - see reply#8).The green trace the amplifiers voltage output, the purple trace is the detector output and the red trace is an amplified version of the virtual earth signal. As can be seen, there isn’t a hint of global loop oscillation as the system automatically reverts from a conditionally stable 4-th order one to an unconditionally stable 1-st order one during the saturation/clipping interval.And finally, here is the (unsaturated, 4-th order-operating) square wave response at 20 kHz:Note that the overshoot of the conditionally stable 4th order loop is successfully suppressed by the signal input filter (R18, C7) which implements a pole slightly below the frequency at which the three integrators are zeroed out.I hope to get a real life experimental prototype built up an operational in the week or so.

author: GK
author_rank: Super Contributor
timestamp: May 15, 2013, 12:15:51 am
content: Hmpff. I guess Nth-order control loop theory and design is a boring topic then.Unless I’m a dunderhead and missing something in a serious way I think the concept is sound, but success in real life I think will be dependant on the “saturation” detector being made sufficiently sensitive. A more refined circuit than what I posted above may be required. My current plan is to capacitively couple (to eliminate DC offsets) the buffered virtual ground (summing junction) signal to a moderate-gain limiting amplifier prior to the window comparator. This way the circuit could be configured to easily trigger on mV's of “error” signal. A better window comparator design having hysteresis to eliminate oscillation will also be implemented in the initial prototype.Any ideas? Sounds good? If the Hypex “N-core” amplifier modules were not so expensive I would have bought one already to reverse engineer it, with particular interest in the saturation detection circuitry.Also, can anyone suggest a good performing analogue switch that operates on +/-15V supply rails and has lower parasitic capacitances than the DG4XX series? To simplify board layout my preference right now is for “single” devices.

author: robrenz
author_rank: Super Contributor
timestamp: May 15, 2013, 12:51:03 am
content: Not boring, just way beyond me being able to make any valuable input

author: c4757p
author_rank: Super Contributor
timestamp: May 15, 2013, 12:53:16 am
content: Quote from: GK on May 15, 2013, 12:15:51 amNot at all! Damn, I missed this post yesterday. And...Quote from: robrenz on May 15, 2013, 12:51:03 amBut I can still salivate, can't I?

author: GK
author_rank: Super Contributor
timestamp: May 15, 2013, 03:07:31 am
content: LOL. Well, we will just have to wait until the first soldered prototype materialises before drooling too much. However hoping that the concept works as seamlessly in real life as it does in SPICE I am quite excited already about the potential. An amplfier with a distortion residual at any power level and any audio frequency indistingushable from the <-120dB THD+N residual floor of my THD analyser (http://www.users.on.net/~glenk/thd/thd.htm) is a very real possibility.There is also potential scope for increasing the order beyond the 4th just by adding additional integrator stages.If it wasn't for the precedent established by the 5th-order "N-core" amplifiers by Hypex I don't think this is a concept that I would have ever considered workable or worth experimenting with.

author: digsys
author_rank: Unknown
timestamp: May 15, 2013, 04:15:39 am
content: Quote from: GKMore that many people have no idea what it means :-) Way over my head. I did dabble with valve amps and early silicontypes in my younger years. In a "specialist" case like this, I'd bump the thread occasionally for the next few days. I knowthere are several audio-freaks that habit here, you just need to ferret them out !Quote from: GKI used to remember using a TI one ages ago, when I did mixing desks. I'd have to see if I can find the old drawings.On another note, can you repair Valve amps :-) ?

author: GK
author_rank: Super Contributor
timestamp: May 15, 2013, 09:12:42 am
content: I think I am going to end up selecting a DG4XX series after all. The "off" in/out capacitance of 8pF for the devices I am looking at really are small enough and the parts are readily available in easily soldered SOIC. There are some other nicer parts available but teenie little 8-pin SOT-23's kind a suck for a home brew PCB.I've repaired a few valve amps that I have owned myself, but never anyone elses

author: GK
author_rank: Super Contributor
timestamp: May 15, 2013, 10:19:41 am
content: Quote from: GK on May 15, 2013, 12:15:51 amOK, I've done a fair bit more experimentation now and I appear to be getting this "saturation" detection under control. It turns out that a significantly more complex overall detection system is going to be required for the best overall control of the clipping event. Just detecting the "error voltage" at the virtual earth appears to be adequate for the taming of stability, but not for the cleanest possible recovery from clipping, particularly at low frequencies and at only very slight/minimal onset amplitude clipping/saturation.The amplitude of the "error" signal is proportional to frequency, due to the decreasing open loop gain as frequency increases. This means that the threshold of detection has to be set high enough to accommodate the amplitude of the error signal at the highest audio frequencies under normal operating conditions. For example, if the amplifier has a loop gain of 54dB (500 Av) at 20kHz, and the output of the amplifier clips against the rails at +/-30V, then the amplitude of the error signal will be 30V/500 = 60mV peak at 20kHz, at the onset of clipping. For a reasonable margin of safety the detection threshold could therefore be set to, say +/- 100mV.This makes the detection system rather sensitive and effective to a clipping event at 20kHz (as demonstrated in my simulation as shown in my opening post), but a great deal less so at 20Hz. This is because the loop gain is a great deal higher at 20Hz than it is at 20kHz (especially in an n-th order system) and the amplitude of the error signal is so low that it essentially lives down in the noise. At 20Hz, an error signal amplitude as large as +/-100mV represents a significant aberration in the output signal, and the clipping recovery performance is nowhere near as good as it is at 20kHz. In fact my sims show significant (ugly) output voltage waveform distortion both entering and leaving clipping to a degree that I do not consider acceptable.One cure would be to deliberately kill the open loop gain such that the open loop bandwidth is equal to 20kHz. This would make the the loop gain throughout the audio frequency spectrum essentially constant. This means that the amplitude of the error signal from 20Hz to 20kHz would also be essentially constant, and the threshold of detection constantly optimal throughout the audio band. However this would mean limiting the loop gain at all frequencies to only that which can be safely applied for stability reasons and adequate phase margin at 20kHz; which is essentially hobbling the amplifier and NOT making full use of negative feedback.Fortunately, however there is a better cure. I am currently working on a complementary detector which senses a clipping event by monitoring and triggering on the slightest conduction of the anti-saturation clamp transistors applied to the stage that those who are familiar with basic amplifier topologies will recognize as the "VAS" in my simplified simulation schematic.I say "complementary" as this detector will not make the one monitoring the virtual earth error signal obsolete. Their digital outputs will be ORed together to the analogue switch control line. The "error" signal still needs to be monitored to cater for "saturation" events that are not related to clipping, such as when the power output stage protection circuitry limits the load current due to a short or excessively low impedance load being connected to the speaker terminals.My revised detection scheme appears to be working rather well in the simulator and I will post updated "preliminary" results after another evening of experimentation and circuit revision / concept evolution.

author: GK
author_rank: Super Contributor
timestamp: May 15, 2013, 01:22:04 pm
content: OK, a bit earlier than expected. I just posted the revised schematic and edit the opening post to show it. The schematic now incorporates the clipping detector comparators as mentioned, having their outputs ORed with the error signal detection comparators. The circuit is still rather primitive and none of the comparators have any hysteresis, so there is a little comparator oscillation on state changes to very low dv/dt signals, but besides that the clipping performance at all frequencies is now almost perfect. He it is at 1kHz. Again the "saturation" comparator output / analogue switch control line (red trace) is plotted along with the amplifiers voltage output, demonstrating how the loop response reverts to 1st order during the clipping interval.

author: ftransform
author_rank: Frequent Contributor
timestamp: May 16, 2013, 04:08:54 am
content: If you tell me what the hell this thing is useful for I will etch a board and make it myself.It sounds pretty awesome, I know a little bit about dbs, degrees, poles and zeros, butA sonic welder? A boom box? Is this kind of circuit useful for driving some kind of transducer? I feel like doing anything mechanical with what you are building is totally over kill.

author: DrGeoff
author_rank: Frequent Contributor
timestamp: May 16, 2013, 05:00:22 am
content: Missed this thread, interesting.Have used MAX333's before (Off C = 5pF, +-15V rails), although not in this application.Is your detector going to detect the clipping event or will it detect the onset of clipping, allowing a margin to react before the clipping event?Or rather than a clipping detector, a distortion detector, comparint the input and output signals (with scaling) to sense the onset of clipping or non-linearities in output?

author: GK
author_rank: Super Contributor
timestamp: May 16, 2013, 08:48:02 am
content: Quote from: ftransform on May 16, 2013, 04:08:54 amJust a speaker for playing tunes. The aim is to get the distortion extremely low, simply as anexercise in design. Mind you a typical forum audiophile would fall over himself to pompously inform you with petty glee just how horrible such a contraption would supposedly sound due to the use of all that evil negative feedback and "blind faith" in measurement, so depending on who you listen to driving a speaker with it may or may not be a good thing.I am currently laying out a PCB for the first real life prototype to test the feasibility of the concept. If it actually works out, I will design a complete, functional and practical amplifier module along the lines described, with full details including PCB Gerber files to be published on my website like my other projects and projects pending.

author: GK
author_rank: Super Contributor
timestamp: May 16, 2013, 09:01:25 am
content: Quote from: DrGeoff on May 16, 2013, 05:00:22 amThe clipping detector shown works by detecting current conducted by the VAS anti-saturation clamp, so clipping has to begin for the comparator to react. However, as far as my simulations go this seems to be adequate. I'll know for sure though after conducting some thorough testing on the up-coming prototype. I have also considered a "preemptive" rail-tracking comparator that triggers on voltage thresholds just below the actual threshold of clipping. That's currently my plan B.Your suggestion of a distortion detector may also be a feasible solution, but I'm thinking rather as a more sensitive detector of saturation not related to clipping (instead of the current scheme of simply triggering on excessive amplitude of the virtual earth "error" signal). Pretty much any good idea or detection concept in on the cards at this early stage of the experimental design

author: GK
author_rank: Super Contributor
timestamp: May 16, 2013, 09:13:52 am
content: Quote from: robrenz on May 15, 2013, 12:51:03 amHey, just on a previously discussed topic, that may be of interest, one of the ~30 ancient texts on analog computing that I have tracked down from secondhand booksellers over the net has a really cool chapter on mechanical computing elements. See the attached pics for a tasterThe book isAnalog Methods, second edition, Karplus & Soroka.

author: robrenz
author_rank: Super Contributor
timestamp: May 16, 2013, 11:29:07 am
content: Just like the electronic versions those things would work great if you could turn off friction and gravity and have infinitely rigid material that had no mass. Plus machine the curves with no error, but that should be easier on material with no mass

author: GK
author_rank: Super Contributor
timestamp: May 16, 2013, 01:08:14 pm
content: You know back in the day the inaccuracies of these kind of computing machines wasn’t that much of a limiting factor in a lot of circumstances, as the variables for the problems they were used to solve were often representative of physical phenomenon which could not be measured or ascertained by indirect methods with a great deal of precision in the first place.

author: SeanB
author_rank: Super Contributor
timestamp: May 16, 2013, 01:40:44 pm
content: Mechanical ones were used for decades as autopilots, the Dakota used one that ran on engine vacuum to power it, and it was reasonably reliable, enough so that you could let go of the controls and have "Charlie" fly the plane for hours. Add a bit of electrical parts and you got the later autopilots that could go cross continent with minimal error.

author: GK
author_rank: Super Contributor
timestamp: May 17, 2013, 10:29:19 am
content: Coincidentally, this morning I picked up my second Korn & Korn book;Electronic Analog Computers, second edition, 1956, from the post depot. Just had a chance now to put my feet up and have a flick through. It contains a whole section on the simulation of airplane flight and auto pilot systems. How cool is that?!

author: toli
author_rank: Frequent Contributor
timestamp: May 17, 2013, 11:30:26 am
content: Hello GK,First I must admit I didn't read all of the posts in the thread, just the first few so what I'm asking might have been discussed already.The idea is very interesting indeed, and I must say I really like it. I see you took into account the inductance of the load with the zobel network, however, did you also check the response caused by the clipping detectors under different conditions/frequencies? While I didn't think about it too much so I might be completely wrong, I'd assume the way it enters/recovers from clipping can have a very large effect on the stability of this feedback loop.In a way, to me, it seems similar to some of the current limiting circuits in regulators (as both of them change the operation of the feedback loop during operation), making the current limiting too fast can result in stability issues under certain load conditions. I know you did a simulation and posted the results in the first post, however I think different load conditions and frequencies should be tested to be certain there isn't an issue.Again I might be completely wrong, and in that case you can just ignore it

author: lewis
author_rank: Frequent Contributor
timestamp: May 17, 2013, 11:49:26 am
content: Is the goal of the design to produce an amplifier with the lowest THD possible? If so, you might want to look into Hawksford Error Correction. Bob Cordell has some great material in this regard, and Dougls Self offers some excellent insights for reducing the THD of LIN-topology amplifiers, some of which I'll mention below.My experience with buggering about with the feedback network of a high-gain amplifier is that 'you're in a world of shit', especially when you connect the amp to a reactive load and feed it a music signal, but my knowledge of control theory here is very limited. The key is when you test the amp under real world conditions, ensure you test the amp under real world conditions! A simulated reactive load gives a good indication of stability without damaging voice coils.http://www.aikenamps.com/spkrload.htmlThe output stage looks pretty darn good, emitter degeneration on the differential pair loaded with a degenerated current mirror is a very good idea, as is the overcurrent protected and buffered VAS (although there's no conventional output current protection so it may not be needed). The miller compensation is slightly unusual, at first sight the 470R looks too high and/or shouldn't be needed, but if it works what the hell. You might want to consider 2-pole miller compensation, it increases open loop gain in the audio bandwidth giving you more feedback, but my knowledge is limited here.Replacing the series diodes in the current sources with a green LED tracks the Vbe thermal drift a bit better; a 2-transistor current source improves accuracy and stability. Biasing the current source with two series resistors with a heavy capacitor down to the negative rail at the midpoint improves PSRR. Decoupling the differential pair with a simple RC or a capacitor multiplier on each rail improves PSRR dramatically. A cascode on the diff-pair has a similar effect. You may also need a few uH series output inductor (8-10 turns of thick wire around a pencil works well) with a 1-10R resistor in parallel with it before the output zobel but after the feedback take-off point to improve stability on transients. All these little techniques can reduce the distortion of the amplifier before feedback is applied resulting in substantial gains in performance when the loop is closed.But definitely look into Hawksford.

author: GK
author_rank: Super Contributor
timestamp: May 17, 2013, 02:07:15 pm
content: OK; the goal at the moment is to develop an verify the feasibility of the "Nth-order" control scheme. The design shown is only a basic/simplified one knocked up in the simulator for the purpose of evaluating the feasibility and operation of this high-order feedback and stability control scheme; the actual discrete power amplifier part isn't at all intended to be an advanced, an ultra low distortion, or even a complete design in its own right. I am currently working on the PCB for a 5th-order implementation of the scheme detailed (four op-amp integrators rather than three) based on a small, 8W/8R discrete op-amp design. At this stage there is no point in prototyping anything more elaborate to proof or otherwise the concept.To be honest I'm not particular enamored with Hawksfords "error-correction" scheme. It's just another method of applying an increased amount of negative feedback around the power output stage which can be done just as effectively by other means without the need for a critical distortion "null" adjustment.The 470 ohm resistor in series with the Miller compensation capacitor introduces a small amount of phase lead to improve the phase margin at the unity loop gain frequency. 2-pole compensation applied here would not be entirely compatible with the overall concept as implemented - which as shown thus far is in fact a 4-pole system.

author: GK
author_rank: Super Contributor
timestamp: May 17, 2013, 02:26:02 pm
content: Quote from: toli on May 17, 2013, 11:30:26 amYes, you are quite right about the clipping. In a "conditionally stable" feedback system like this clipping is one of those scenarios which can result in the amplifier breaking into uncontrollable and unquenchable global loop oscillation. However this is something I hope to have overcome with my detection scheme which automatically switches the feedback order to a safe 1st order system whenever an unsafe state of operation is encountered. My text books tell me that so long as the amplifier is operating in a linear region, the total phase shift can go well through 360 degrees, but so long as the open loop gain reverts back to a 6dB per octave (single pole) slope before crossing through unity gain, the system as a whole will be "conditionally stable".

author: TheRevva
author_rank: Regular Contributor
timestamp: May 17, 2013, 10:53:04 pm
content: Quote from: robrenz on May 16, 2013, 11:29:07 amSo you're saying that if we switch the source material from unobtainium to nonexistium, we'll be sweet?Some of the best aspects of this nonexistium include:1: The pirce per kG is ideal!2: Travel at the speed of light (c) finally becomes easyThe downside is that I haven't found a supplier that stocks the nonexistium. All my normal suppliers suggest the unobtainium as a 'possible replacement' (probably because they achieve a better sales margin?)I'll keep searching and let y'all know.

author: GK
author_rank: Super Contributor
timestamp: June 01, 2013, 12:34:37 pm
content: Hello all.Attached is a picture of my initial prototype amplifier built just to test the concept. It is in fact a fifth-order ("5-pole") implementation of the scheme (having four op-amp-based integrator stages) based around a low-power discrete power amplifier that clips at 15W into a 4 ohm load.At the moment the circuit board shown contains all of the circuitry bar the detection/comparator circuitry required to automatically switch the control loop between a 1st order system and a 5th order system. My initial concern was just to verify that a 5th order control loop could really work. To be honest in theory I pretty much new that it would work, but I just had to actually build a real life prototype and see it work with my own eyes. The way the shown prototype works in its current state of construction is that I can manually switch it between a 1st order system and a 5th order system by asserting the logic control line for the analog switches which short the capacitors of the four series-connected integrator stages.I can happily report that, so long as the amplifier is operated within its linear region of operation (ie not driven into clipping), it is 100% stable in the 5th order mode of operation with any signal stimulus I have (exhaustively) tested it with. The prototype implementation has a relatively conservative unity loop gain "crossover" frequency of 650kHz and slightly over 60dB of global loop feedback at 20kHz. The prototype has a modified gain distribution amongst the integrator stages for better noise performance. I have achieved an input-referred noise equivalent to an input resistance of 4k. As far as I have been able to measure distortion thus far, in the 5th-order mode there doesn't seem to be any discernible distortion products breaking through the noise floor at any operating frequency.At the moment, if the amplifier is driven into clipping in the 5th order mode, it predictably breaks out into (non-destructive) global loop oscillation, and remains in an oscillatory state until I manually switch it into 1st order mode, as I have not fully yet implemented the error-sensing control circuity to do the switching automatically before global loop oscillations even have the chance to begin.However, from my very positive (and somewhat exhaustive) experimental results thus far I am left with little doubt that the basic scheme is sound. Over the next several nights I will complete a prototype implementation of the control circuitry and report perhaps next weekend with some oscilloscope trace photos or videos to demonstrate the stability of the completed system if anyone is interested.From this point forward though I am not willing to reveal any actual circuit details or expound upon topological modifications I have made to the basic scheme outlined in my opening post. There are also issues concerning the control circuitry that deal with stability during power up and power down which I have not elaborated on, but are necessary for a functioning and practical real-world implementation.The reason for this "secrecy" is because I have since potentially made a deal with an audio electronics magazine to have an advanced design for a high power, ultra-low distortion amplifier module, based on my high-order control loop scheme roughly outlined in this thread, published; pending on the success of my initial prototype as shown here.However, having already started a thread on the experiment, I figured I could at least update it with a report on its success or otherwise without a conflict of interest if anyone is interested.

author: digsys
author_rank: Unknown
timestamp: June 01, 2013, 01:03:11 pm
content: I wish you good luck with your publishing possibility. Even though I haven't done much on audio / amplifier design for many years,(except some valve stuff), I certainly follow your R+D progress. You have at least ONE listener :-)

author: GK
author_rank: Super Contributor
timestamp: June 02, 2013, 02:18:34 am
content: LOL, OK, thanks.... One listener is better than none, I guess.Here is the simulated versus measured response of the 5th-order loop to a square wave stimulus. Those familiar with 2-pole compensation would be aware of the compensation schemes inherent overshoot to transient stimulus due to the necessary zero in the forward gain path. Well, a high-order control loop like this no different, just that the degree of overshoot is inherently greater due to the higher order of the control loop.The first picture shows the (20kHz) square wave response to an non-bandwidth-limited square wave input signal. Just like with 2-pole compensation, the transient overshoot can be suppressed by bandwidth limiting the input signal. This is shown in the second picture. In this particular case, a low pass input filter comprised of a 1k resistor and a 1nF capacitor is sufficient to completely suppress the transient overshoot. Note that the time constant is in fact lower in the scope photo shown as the total resistance is 1600 ohms due to the 600-ohm output resistance of my signal generator (I also increased the simulation resistance to 1600 ohms to match the measurement).A quite successful demonstration of basic control theory and simulation matching real-world implementation, methinks........Oh..... another thing to mention - in the simulation schematic shown in my opening post it can be seen that the DC gain of the integrators was limited by the 1M resistors in parallel with the integration capacitors. I only needed to include these resistors to get the simulation to converge. They are not included in my real-life prototype. This is really an experiment in applying global loop feedback to a crazy degree without any form of pseudo-scientifically-justified hobbling ......... the prototype system is up and running 100% stable withhundredsof dB of open loop gain at DC (effectively the full open loop DC gain of the four series-connected LME49710 op-amps of the integrator stages plus another 100db or so of gain provided by the discrete power amplifier section).

author: IanB
author_rank: Super Contributor
timestamp: June 02, 2013, 04:59:14 am
content: Quote from: GK on June 01, 2013, 12:34:37 pmA question here from a novice: does not amplifier stability depend on the nature of the load? What happens if you give it a strongly reactive and "difficult" load?

author: megajocke
author_rank: Regular Contributor
timestamp: June 02, 2013, 06:37:00 am
content: Cool!

author: GK
author_rank: Super Contributor
timestamp: June 02, 2013, 10:50:26 am
content: Quote from: IanB on June 02, 2013, 04:59:14 amA capacitive load can decrease the phase margin of the amplifier due to the pole formed by the load capacitance in conjunction with the finite open loop output impedance of the power output stage. If the amplifier has a marginal phase margin to begin with a moderate capacitive load can force it into instability. However that is not the case here. I've run it (square wave stimulus) into 2.2uF in series with 2.2 ohms and the amplifier happily continued as normal, burning out the resistor. That is without, at this stage, an off-board capacitive-load-isolating inductor directly at the output of the amplifier.

author: GK
author_rank: Super Contributor
timestamp: June 02, 2013, 10:54:32 am
content: Quote from: megajocke on June 02, 2013, 06:37:00 amHey, thanks.The control circuitry construction is coming along nicely, but I now need to give it a rest for the night. It looks like I'll be listening to tunes through this thing hooked up here in my study by late tomorrow evening........ stay tuned for some scope shots of stability-controlled clipping events that match the simulation results.............

author: robrenz
author_rank: Super Contributor
timestamp: June 02, 2013, 02:37:44 pm
content: I am still following GK, I am hoping some of this will sink in by osmosis

author: c4757p
author_rank: Super Contributor
timestamp: June 03, 2013, 02:34:47 am
content: Quote from: robrenz on June 02, 2013, 02:37:44 pmYeah, me too. This is quite interesting. I'm still trying to find a decent way to teach myself control theory like this - I can stabilize an SMPS feedback loop with some effort, but this is far beyond me. The SPICE/scope comparison shots are especially impressive. Clearly you know what you're doing.

author: GK
author_rank: Super Contributor
timestamp: June 03, 2013, 03:46:54 am
content: Quote from: c4757p on June 03, 2013, 02:34:47 amWell, I think I know what I’m doing for the most part. I deleted my previous post as under further investigation I turned out to have made some incorrect assumptions. One thing that has me baffled right now, thinking about the whole project a bit more, is the systems complete stability to transient input signals that are too fast for the system as a whole to respond to, resulting in slew-rate limiting. This causes stages of the amplifier to saturate, temporarily.However, the temporary state of saturation during a period of slew rate limiting, both in SPICE and in the actual prototype, does not appear to be a trigger for instability of the high-order loop. So long as the 5th order system is operated within it linear limits (i.e. not clipping), no matter what kind of input signal I drive it with, I have been unable to excite any form of oscillatory burstat all– not just in SPICE, but, more importantly, in the actual real life realisation sitting on my bench top.That “saturation” during clipping is (completely predictably) a trigger for instability while “saturation” during slew-rate-limiting doesn’t appear to have any negative impact on system stability at all is something that I do not fully understand at this point in time. This immunity of the system (stability wise) to “transient saturation” during slewing is demonstrated by the (non-bandwidth-limited) square wave response scope photos that I posted.The experiment thus far leads me to believe that some of the theoretical “dangers” of conditionally stable control systems that I have read about are not necessarily true or at all insurmountable - the thing on the bench works!

author: C
author_rank: Super Contributor
timestamp: June 03, 2013, 05:11:31 am
content: Something to keep in mind.Tweeter's will some times burn up from the increased energy generated in the high frequency spectrum when a signal clipped. The speaker systems I have seen do not expect a lot of energy to be in the tweeters frequency band.If your amplifier chain is capable or your limiter action has increased the high frequency energy enough, your amplifier may be fine, but the tweeter on the output may have died.C

author: GK
author_rank: Super Contributor
timestamp: June 03, 2013, 05:32:27 am
content: Well sure, but that is true for any solid state power amplifier. One reason to use an amplifier that is powerful enough so that you are not regularly driving it into clipping.

author: GK
author_rank: Super Contributor
timestamp: June 03, 2013, 06:48:16 am
content: Quote from: GK on June 03, 2013, 03:46:54 amAhhhh haaaaa..........I think I'm starting to get a handle on this now............ For the high-order, conditionally-stable system to be triggered into sustained oscillation by a saturation event related to clipping, the saturation event itself must be ofsufficient duration.When the system is successfully triggered into global loop oscillation, the frequency of oscillation is confined to be somewhere within the range that the total phase shift around the loop exceeds 360 degrees. This frequency range is in fact not particularly high and lays quite a bit below the amplifiers unity loop gain frequency (the frequency at which the loop gain goes through unity at a single-pole roll-off).It appears that the duration of the saturation event required to trigger global loop oscillation is roughly commensurate with the frequency at which oscillation can occur.As it turns out, this duration is apparently orders of magnitude greater than the duration of time that any stage of the amplifier can actually remain in a state of saturation during slew-rate limiting.If my preliminary observations as just detailed are correct, then they would explain why slew-rate limiting of the amplifier in response to fast rise-time input signals has so far been demonstrated to have absolutely no adverse effect on global loop stability; that is without the capacity to trigger any kind of observable oscillation, either temporary or sustained, of the high-order loop. This, if correct, is, I have to say, a very pleasing and reassuring observation indeed.Is there a "light bulb" smiley avaliable anywhere? (that is with the light bulb above the head rather than sticking out of the mouth)

author: digsys
author_rank: Unknown
timestamp: June 03, 2013, 09:49:52 am
content: <munching on popcorn and waiting for the bad guys to appear>

author: robrenz
author_rank: Super Contributor
timestamp: June 03, 2013, 11:09:31 am
content: There are signs of osmosis actualy working

author: GK
author_rank: Super Contributor
timestamp: June 03, 2013, 11:43:48 am
content: What bad guys?I am now sitting here, listening to tunes through an audio power amplifier that:1) Has a 5th order control loop2) Is completely stable3) Clips as clean as a whistle4) Has approximately 650dB of open loop gain at DC5) Has 64dB of loop gain at 20kHz (actually verified by measuring the amplitude of the virtual earth error signal)I'm sorry to say it, but I really don't think that I would have a great deal of success trying to measure the amplitude of the error signal at 20Hz or so!I also have a big smile on my face right now.Attached are photos showing the amplifiers clipping performance at 20kHz, 1kHz and 20Hz, along with a close up of the beast itself.The yellow trace is the amplifiers output signal, driving a 4 ohm load. The blue trace is the 5V logic control signal to the analogue switches. When it is low, the system is switched into 1st order mode; 5th order mode when high.The next step? To build a proper amplifier, with a decent PCB layout, capable of delivering a decent amount of power (100-200W).

author: GK
author_rank: Super Contributor
timestamp: June 04, 2013, 02:05:42 am
content: Quote from: GK on June 03, 2013, 11:43:48 amJust to put some kind of perspective on things here, this amplifier has a loop gain characteristic that rises at a 100dB/decade rate (5-pole) with decreasing frequency from a starting point of 64dB at the top of the audio frequency spectrum (20 kHz).This means that, at the lower end of the audio frequency spectrum (20 Hz), the loop gain reaches approximately 360dB. This is egual to a loop gain, expressed as Av, of 1000,000,000,000,000,000.That is the "loop gain" - the actual open loop gain of the amplifier, at this point, is this figure multiplied by the feedback factor, which is ~20. So at 20Hz the open loop gain is 20,000,000,000,000,000,000.The prototype amplifier clipps at 15W average ("rms") into a 4 ohm load, which equals ~11V peak. So, at the onset of clipping at 20 Hz the virtual earth error signal will be approximately equal to 11V / 20,000,000,000,000,000,000Av = 0.00000000000000000055V peak.If I have my math right and haven't missed a zero here or there, that is equal to 550 zetta volts. Does anyone know if HP or Keithely ever made a zetta-voltmeter?LTspice doesn't even have the computational resolution to compute the loop gain as DC is approached. It bombs out even before getting down to 20Hz:

author: JuKu
author_rank: Frequent Contributor
timestamp: June 04, 2013, 07:44:22 am
content: In other words, noise and parasitic effects dominate. Now, those do make control theory "interesting". have you measured your prototype THD+N vs frequency and other conventional audio quality characteristics? and, btw,!

author: GK
author_rank: Super Contributor
timestamp: June 04, 2013, 09:06:40 am
content: Well, bandwidth and rise time are determined by the input filter having a corner frequency of 159kHz. There is effectively no slew rate limiting, as the input filter is the limiting factor. I have not been successful at measuring any distortion products out of the amplifiers noise floor. However keep in mind, this initial prototype is only a low power thingie, with a closed loop gain (20 Av) relatively high compared to its power output. The input sensitivity is 350mV and the input-referred noise is approximately 8nVsqrtHz. in a 20kHz noise bandwidth that gives a S/N ratio of approximately 109dB.8nV of input noise is pretty good for a power amplifier, but not exceptional. However I did not design the prototype for ultimate noise performance as that wasn't my main concern at the time. It would be relatively trivial though to get it down to 5nV or so, mostly by lowering the feedback impedance's and further modifying the gain distribution in the integrator stages.5nV is I think sufficient and is what I am aiming for now, as that will give me a quite healthy S/N ratio of 126dB with an input sensitivity of 1V rms in a 20kHz noise bandwidth (some dB better A-weighted). However that is part of a new design based on my high-order control loop concept, further details of which I am rather obliged to keep under my hat

author: GK
author_rank: Super Contributor
timestamp: June 06, 2013, 10:24:01 am
content: Well it has been a few evenings now and the amplifier is still pumping out tunes here in my study. Funny enough though, after many hours of soak testing I actually did finally identify an intermittent instability, funnily enough though, only occurring when the amplifier was cold and just switched on. However it absolutely had nothing to do with the 5th control loop - forcing the amplifier into 1st order mode by overriding the logic control signal to the analog switches did not cause the oscillation to stop. The oscillation itself was a rather low level one (a very clean sine wave of about 50mV peak at the speaker output) at a fraction over 10MHz.With the amplifier taken back into the "lab" I tracked the cause down to a relatively high Q open loop gain peaking in the integrator bank at slightly above 10MHz. I did this by breaking the global feedback loop and sweeping the amplitude response of the integrator bank with an RF signal generator and 'scope. This peaking in the open loop response turned out to contribute just enough of a degradation to the amplifiers gain margin to cause the intermittent, "on the verge" instability issue.Fortunately though the cure was a very simple one - I just incorporated a pole into the loop by fudging in an RC filter formed by a 1K resistor and a 15pF cap to neuter the peaking with its attendant advancement in phase, to ensure instead that the loop gain continues to roll off at a deliberate and controlled rate. Exactly why the integrator bank exhibits this high frequency peaking (even in 1st order mode when the integrators are transformed into basic inverters) I'm not exactly sure at the moment, but I suspect it is due to a parasitic interaction with some characteristic of the analog switches. This is something that will require further and thorough investigation.While this isn't exactly an "ordinary" power amplifier design, this little issue does illustrate the importance of having to characterize the amplitude and phase responses of all gain blocks within a closed negative feedback loop out to frequencieswellbeyond the point at which the loop gain crosses through unity.Just thought that might be a interesting tidbit for all those control engineers out there who might be reading.

author: Tepe
author_rank: Frequent Contributor
timestamp: June 06, 2013, 10:59:40 am
content: Quote from: digsys on June 01, 2013, 01:03:11 pmYes, in fact more than one

author: GK
author_rank: Super Contributor
timestamp: June 07, 2013, 06:42:59 am
content: Quote from: GK on June 06, 2013, 10:24:01 amFound the problem. I was that silly ground plane separation I put between the analogue switches operating on the integrators and the actual integrators themselves. At the time I layed out the proto PCB I had not yet considered how I was going to later bodge on the experimental control circuitry, so I put in this split (not necessary for the purpose anyway) in an attempt to make sure the ground returns for the control circuitry and the signal circuitry are separated down to the hierarchical star grounding back to where the power enters the PCB.I should have known better than to have split the grounds between the two intimately coupled sections of analogue circuitry. The ~10MHz peaking was caused by some kind of regeneration between the integrator stages due to feedback through the effective parasitic capacitances to ground of the analogue switches feeding into the high inductance of the effective ground loop. Bridging the ground plane split between each analogue switch/integrator combination eliminated the peaking.

author: robrenz
author_rank: Super Contributor
timestamp: June 07, 2013, 11:52:39 am
content: That's exactly what I was thinking right off the bat, but I wanted to see if you found it

author: GK
author_rank: Super Contributor
timestamp: June 09, 2013, 01:22:49 am
content: OK then, perhaps I should commission you to layout the PCB for the magazine article then?Id be able to put my feet up for the next several evenings.There were still a lot of unknowns when I began laying out that proto PCB and it was done with expediency as the highest priority. However as a foundation for a "proof of concept" it has served its function. Still pumping out tunes here in my study too

author: GK
author_rank: Super Contributor
timestamp: October 11, 2013, 08:15:49 am
content: Some cross-forum pollination here - I've noticed that this thread has become a topic of conversation over here:http://www.diyaudio.com/forums/solid-state/240712-cfa-topology-audio-amplifiers-152.html#post3661711Might as well use this as an opportunity to spruik the magazine article I mentioned previously. The magazine in question is this one:http://www.linearaudio.net/For the basis of the article I've designed a complete and functional audio power amplifier based on the high order feedback principle developed during the course of this thread. It clips a fair bit above 100W into 4 ohms. I am currently in the final stages of testing the prototype. A rather unrevealing and small teaser picture of the amplifier under test, connected to my distortion analyser, is attached. There is just a little bit of optimization of the frequency compensation to be completed this weekend, that I wish to do just to satisfy myself that the amplifier as implemented is performing the best that it can. Then I just have to find a quiet free day (a bit difficult lately) to finish formatting the article and diagrams. To the best of my knowledge it will be published in either the upcoming issue of LA or the one after.The amplifier developed has:1) A fourth-order global feedback loop2) a ULGF pushed as high a practical given stability constraints and3) OIC in the form of TMC applied to the OPS.There has been much heated debate about the merits of TPC versus TMC, but no one so far seems to have had the imagination to realize that the two schemes are not mutually exclusive, which is unfortunate. The amplifier developed demonstrates that this is true; though I have gone one or two steps better, of course, with "4PC" rather than just TPC.That's about all I can say really, without spoiling the article. If the acronyms used above confuse anyone over here, I'm afraid that you'll just have to purchase the article when it finally comes out

author: Experimentonomen
author_rank: Regular Contributor
timestamp: October 11, 2013, 08:29:54 am
content: I have found that low THD amplifiers tend to be so clinical and accurate that the sound ends up a bit harch/boring and unpleasing, as the human ear kinda expects the distortion of the average amp circuit found in most consumer amplifiers which makes the sound warm and pleasing.

author: GK
author_rank: Super Contributor
timestamp: October 11, 2013, 08:38:03 am
content: Quote from: Experimentonomen on October 11, 2013, 08:29:54 amThat's a good one - haven't heard that before and must add it to the list. BTW, why is it always the "Human ear"? Is that distinction necessary? Would the intended audience be otherwise unduly confused if it was just "ear"?

author: EEVblog
author_rank: Unknown
timestamp: October 11, 2013, 08:53:57 am
content: Quote from: robrenz on May 15, 2013, 12:51:03 amYes, the more obscure the topic gets, the less likely people will be able to give meaningful input.This is the kinda stuff Doug Ford has wet dreams over (former head audio amp designer at Jands)

author: jaxbird
author_rank: Frequent Contributor
timestamp: October 11, 2013, 10:24:30 am
content: This is a very interesting project.Looking forward to getting a chance to listen to this design.

author: Whuffo
author_rank: Regular Contributor
timestamp: October 16, 2013, 02:36:42 pm
content: It's a wonderful exercise in design and construction. Without making any judgement on your work, let me offer some observations that might interest you.First, the best loudspeakers manage 3% distortion; most are 5% to 7%. And the average person can't hear less than 0.5% distortion; it's just not audible. I know that there are many who would argue these figures longly and loudly, but they're the real truth. The limits to audio fidelity aren't the amplifiers (usually) and definitely not the wiring; the transducers (speakers) are where real improvement could be found - but not easily. Remember that speakers are essentially linear motors, and they have back EMF and varying impedance with frequency and the resonant characteristics of the driver(s). Not an easy load to drive precisely.As long as your amp doesn't clip during normal operation - that's not always an easy one, audio has a lot of dynamic range and transients can push almost any amp into clipping, it'll sound great - as long as its frequency response and phase response are reasonably flat. A good DC coupled class AB amp is more than adequate for this purpose and many of the most highly regarded amps are built that way. A very low impedance power supply will insure that the speaker is tightly controlled.

author: madshaman
author_rank: Frequent Contributor
timestamp: October 16, 2013, 03:36:14 pm
content: Quote from: Experimentonomen on October 11, 2013, 08:29:54 amI'll play devil's advocate: when you listen to music live, do you miss the lack of harmonic distortion?I see the point; much like most home stereo equipment's default eq'ing is the "disco smile" (low and high end are boosted), THD isn't the only thing you want, you want only even order harmonics for that "warm" sound.Also, imho, other than a guitar/bass amp, an amplifier should *amplify*, other equipment is for colouring the sound.If you need that "warm" sound, put a class-A vacuum tube biased pre-amp in front of everything and drive it barely into distortion. I'm sure there are audiophile boxes made just for this.

author: GK
author_rank: Super Contributor
timestamp: October 16, 2013, 11:15:34 pm
content: Quote from: Whuffo on October 16, 2013, 02:36:42 pmA bit presumptuous, don’t you think?QuoteA threshold of audibility of 0.3% for crossover distortion has been demonstrated in blind tests. I don't have the ref at hand right now but it is in Selfs book.QuoteThe design presented is class AB, DC coupled. What is your point?QuoteNot on its own. An amplifier with good PSR and adequate LF damping factor will ensure that the speaker is "tightly controlled". It's an ass-over-head design approach to neglect these factors and instead concentrate on a stupidly over-engineered PSU; allthough that appears to be the norm in "high end".

author: c4757p
author_rank: Super Contributor
timestamp: October 16, 2013, 11:18:33 pm
content: Quote from: GK on October 16, 2013, 11:15:34 pmHigh end audio generally seems to prefer either "ass over head" or "head in ass" as design theories.

author: GK
author_rank: Super Contributor
timestamp: October 16, 2013, 11:21:56 pm
content: No argument there.

author: Jebnor
author_rank: Regular Contributor
timestamp: October 17, 2013, 05:53:19 am
content: I love reading about people who do interesting projects even I don't fully comprehend. I like the challenge of getting up to speed.Thanks GK. Keep it up.

author: jaxbird
author_rank: Frequent Contributor
timestamp: October 22, 2013, 06:26:50 pm
content: Quote from: Whuffo on October 16, 2013, 02:36:42 pmIt is not unusual for high quality speakers to achieve less than 0.5% distortion at normal levels above a few 100Hz. It's very hard to hear distortion at lower frequencies.But I agree that speakers are the weakest link in music reproduction, but IMO it's more to do with coupling, dispersion and nonlinearity than actual distortion.Not saying that all amplifiers sound the same, that is clearly not the case.

author: Whuffo
author_rank: Regular Contributor
timestamp: October 24, 2013, 10:35:58 pm
content: Quote from: jaxbird on October 22, 2013, 06:26:50 pmI suspect those speakers that are 0.5% distortion were built by virgins on a mountain and carried to the masses on the backs of unicorns. They probably also require special oxygen-free directional wiring to achieve that impossible level of perfection. Forget those manufacturer marketing specifications - break out the instrumentation and check it out for yourself. I'd be willing to wager that those "not unusual" "high quality" speakers are on the high side of 5%, not 0.5%.And let's not over complicate this - as far as speakers are concerned, any difference between the signal coming in and the sound coming out is distortion. That's the definition of the word, right?I'll never claim that all amplifiers sound the same; that's not the case at all. But I will claim that all well-designed and constructed amplifiers sound the same; the differences (if any) are inaudible. The one characteristic that is lacking in many amplifier designs is "headroom" - the peak to average ratio of music is pretty large, and without adequate headroom, clipping results. The amount of output power needed depends on the efficiency of the speakers and the size of the listening room. The average power will be moderate and can be easily handled. The peak power will be at least an order of magnitude greater; I'd recommend 100x the average power being available upon demand.That whole "tube vs. transistor" debate revolves around how amplifiers using those components behave when they clip. Avoid clipping, and it's an academic debate. You state that people can hear 0.3% distortion; that may be true for some, but the average listener won't hear 0.5%. In any event, any decent amplifier will produce far less distortion.If we were closer, I'd invite you to come over and hear what audio can sound like. In any event, put the time you'd spend getting that last hundredth of a percent of distortion out of an amplifier into boosting its output power and the result will be much better.

author: GK
author_rank: Super Contributor
timestamp: October 24, 2013, 11:30:57 pm
content: Quote from: Whuffo on October 24, 2013, 10:35:58 pmWho are you talking to now? I stated nothing of the sort. I simply made reference to a study that demonstrated an audibility threshold of 0.3% THD for crossover distortion in a solidstate amplifier with average test subjects - to counter your "factual" assertion that no one can hear even 0.5% THD.In any case, speaker distortion is pretty much completely irrelevant when it comes to the potential audibility of distortion generated by amplifiers. Speaker non-linearity is typically benign, low-order and generaly completely negligible at low power levels (where they operate most of the time). Contrastingly a badly designed power amplifier can have the exact opposite characteristic - can generate high levels of high (harmonic) order crossover distortion at low signal levels.QuoteWho said otherwise? What is your point?

author: Whuffo
author_rank: Regular Contributor
timestamp: October 25, 2013, 08:40:51 pm
content: Quote from: GK on October 24, 2013, 11:30:57 pmWho said what is usually easy to determine from the embedded quotes; I was just having a discussion with someone who drank the audiophool kool aid; nothing more.That said, I'd still recommend that the effort / cost involved in getting that last hundredth of a percent of distortion out would be much better spent on boosting the output power. To prove amp design excellence, build one that will survive running at full power into a dead short with no component failure - not even a blown fuse.Low level distortion is almost always due to the output transistors not being biased properly. There's a natural variation in components that means that no amp will be biased correctly unless each one is adjusted manually or elaborate automatic biasing circuitry is included. Matched pairs on the output devices helps, but you'll still need to insure that the bias is correct and that the idling current through the outputs is sufficient to get them into the linear part of their curve. You've got to waste some power supply current here; that idling current is critical.Low frequency distortion is almost always due to clipping - there's a lot of energy in those low bass notes and an underpowered amp will clip here first.General "rules" for quality amps: a very low impedance, very well filtered power supply to start; the amp should be able to draw any "legal" amount of current at any time without the rails sagging. In my day, we did this with large power transformers, big rectifiers, and big filter capacitors. That might still be the optimum solution - improve on it if you can. Many references to amp design exist - all I'll add to what they may say is to insure that your amp has an odd number of stages in it. Why? Audio amps have large amounts of gain, and if there's any (even a tiny bit) of coupling from the loud end to the sensitive end, if it's out of phase (due to odd number of stages) it'll reduce the gain. If you have an even number of stages, it'll be in phase and you might just create a high powered oscillator.And always remember that while the design and math is precise, the components you build with are not. Imagine a world where resistors are 10% and capacitors are 20% or worse. If your design can function properly with any combination of tolerances in the active and passive components, it'll be solid when you use the nicer components you can buy cheaply today. And never forget: simpler is better, the fewer things to go wrong, the fewer things that will go wrong.PS: Regulation of the main power rails is not too important. What's important is that both rails track each other precisely. A transformer makes this easy; is your approach more cost effective while providing the same functionality?

author: GK
author_rank: Super Contributor
timestamp: October 25, 2013, 11:17:05 pm
content: Quote from: Whuffo on October 25, 2013, 08:40:51 pmThen why do you have so much difficulty replying to things said with any kind of logical coherence? You're just trolling now, right?QuoteWhoooaa! Talk about moving goal posts! Your presumptuous evangelism doesn't give me any confidence that you could design a high performance audio amplifier if your very life depended on it. Just a few post ago you said "A very low impedance power supply will insure that the speaker is tightly controlled" and now you say "Regulation of the main rails is not too important". So which is it?If you knew what you were talking about, you'd be aware that those two things go hand in hand.My power supply has a transformer (a substantially over sized one). What makes you think otherwise?

author: GK
author_rank: Super Contributor
timestamp: October 27, 2013, 01:27:42 am
content: Since the issue of power supply design was brought up……………When it boils down to the aim, in and of itself, of achieving an ever better objective performance from an amplifier design, going overboard on the power supply is the least efficient and the most expensive way of going about it. Big transformers and multiple 10’s of thousands of uF of filter capacitance (especially if the rail voltages are high) can cost the bulk of the money expended.Our hearing response and perceived loudness does not respond in a linear fashion to power. It is generally accepted that a 4 to 10 times increase in power is required to result in a doubling of perceived loudness. Going overboard on power supply regulation and bulk filter capacitance in order to squeeze out, for a given nominal rail voltage, the last possible volt of dynamic headroom prior to clipping on bass transients isn’t going to have much scope whatsoever for increasing the perceived loudness before clipping becomes objectionably audible.A much smarter approach is to start with a design having a decent power supply rejection; one that doesn’t pass on, to an excessive degree, variations of, and crud on, the rail voltages to the speaker.The n-th order feedback method and ultra-low open-loop bandwidth / maximal feedback design “approach” that I have proposed in this thread makes it theoretically and practically possible to construct amplifier modules having absurdly good and far more than adequate power supply rejection - particularly at LF; literally hundreds of dB. The challenge then becomes, particularly for the ultimate minimization of main hum, in adequately isolating earth loops, arranging hierarchical grounding and designing the chassis layout and PSU component placements to minimize the measurable effects of electromagnetic induction.

author: ElectroIrradiator
author_rank: Frequent Contributor
timestamp: October 27, 2013, 06:43:58 am
content: Note that if we are talking simple passive power supplies, then there always exist a limit on filter capacitance beyond which it no longer makes sense to add more. The loss resistances in transformer, rectifiers and ultimately the mains wiring puts a hard limit on the peak charging current. So you will always have a certain minimum of ripple and/or average DC voltage drop, even for an infinitely large filter cap.It very quickly becomes the case that a simple CRC or CLC filter has very much better ripple suppression, compared to the single cap filter, though those solutions have other inherent problems.An excessively large filter cap also hugely increases the apparent VA load seen by the transformer, potentially making a mockery of any attempt at choosing an oversize unit. Which frequently explains why DIY people sometimes runs into problems with their mains transformer overheating, despite the massive overkill factor employed for all PSU components.

author: Whuffo
author_rank: Regular Contributor
timestamp: October 27, 2013, 12:39:09 pm
content: Quote from: GK on October 25, 2013, 11:17:05 pmI'm not trolling, I'm just trying to pass some useful knowledge on to you. And regardless of what you may think, I've designed and built audio amplifiers before. The last one would do full power over 10-30 KHz, and with less than 0.01% distortion at any frequency or output power. Measured with real instruments, and it'd survive full power output into a dead short, too. That was before you were a gleam in your father's eye, and it was much, much simpler than what you propose.One opamp and 4 transistors per channel, basic transformer / rectifier / big filters power supply. If you can do better, then trot it out. Otherwise, get a clue from those who know much more than you do. Your nonsense about the rails doesn't matter - as long as the two rails are the same and opposite voltages, their exact voltage is irrelevant as long as it doesn't exceed the safe voltage of the output devices. Consider that with a center tapped supply, the ripple on the rails cancels out.This is important: as long as the rails are opposite and equal voltages, the amp will perform well no matter how much you load it. Sags in the + voltage are matched by sags in the - voltage and the DC center point remains stable. I won't offer suggested component or filter cap values, because you're so smart and you can figure it out all by yourself. If the power transformer is sized to provide peak load, it won't be a problem.The more complex you make it, the worse the ultimate performance will be. And for those talking about diode and capacitor loading - you're thinking about this wrong. The power supply isn't only needed to supply the average demand, it also has to supply the peak demand which is at least a magnitude higher. If the components are sized to handle the peaks, normal operation won't stress them at all.Quality audio amplification is a solved problem. Until better technology comes along, there's no opportunity for big improvements. Check out the schematics for amplifiers by your favorite brand; they're almost always available online. Check several of them - you'll find that they're more identical than you imagined. Most of the differences quoted in their spec sheets are due to marketing, not electronics.

author: GK
author_rank: Super Contributor
timestamp: October 27, 2013, 01:24:00 pm
content: Quote from: Whuffo on October 27, 2013, 12:39:09 pmYou certainly are trolling here. You never properly address comments made or back up assertions when challenged. You just rant away and pontificate on multiple tangents and different directions instead. What I wrote about the power supply design definitely isn't nonsense, and the points I made aren't refuted or even addressed by the additional (again tangential) "useful knowledge" that you have once again so kindly provided.I've made it quite clear right from the beginning of this thread that the design presented here is an over-the-top, esoteric exercise/study in the practical implementation of a complex, high order control loop. Your continual banging on about the objective levels of performance and minimal complexity needed to make a subjectively transparent amplifier are irrelevant and besides the point. If it bothers you so much that I should embark on such a project and share some of the design theory I suggest that you share your angst with your shrink instead - I'm not particular interested. Or perhaps if you really feel that you have so much important wisdom to impart and are hankering for attention and acknowledgement, why don't you start your own thread: "Whuffo's quide to the ultimate power amplifier" and stop derailing this one?And thanks for the further presumption, but I really don't need advice of how to work out an adequate value of power supply filter capacitance. As detailed already, I have built, tested and measured (on equipment designed and built by myself) the design (not my first). It measures great deal better than 0.01%. For a large part I'm struggling to see harmonic components out of a <0.0001% THD+N measurement floor. Some might even regard that to be by some degree an engineering achievement.

author: GK
author_rank: Super Contributor
timestamp: October 27, 2013, 01:39:53 pm
content: Quote from: ElectroIrradiator on October 27, 2013, 06:43:58 amTell that to the audiophiles and tweakers! I've seen reports/boasts of 200,000uF per rail for an LM3886 "chip amp"!

author: Whuffo
author_rank: Regular Contributor
timestamp: October 27, 2013, 02:02:22 pm
content: Quote from: GK on October 27, 2013, 01:24:00 pmFine. I'll just ignore your insults and name calling and leave you to your moment of glory.

author: ssassen
author_rank: Newbie
timestamp: January 11, 2016, 07:43:45 pm
content: @GK,I've been reading through your 4th order power amplifier topic with great interest. You mention that this design has been featured in Linear Audio magazine, however I can't seem to find in what edition? Now it so happens I own every edition, and have looked through a number of them, but can't seem to find your article, could you perhaps point me in the right direction?Thanks and best regards,Sander.

author: med6753
author_rank: Super Contributor
timestamp: January 11, 2016, 09:26:38 pm
content: I was reading this very interesting and informative old thread. I curious as to what the final results were. But I was also wondering when it was going to "happen". And true to form, it did.Why do discussions about high end audio always, ALWAYS, turn into a monkey poo flinging contest?

author: jsmn4vu
author_rank: Newbie
timestamp: January 11, 2016, 09:29:15 pm
content: > Why do discussions about high end audio always, ALWAYS, turn into a monkey poo flinging contest?Pretty sure it's because in the field of high-end audio, there are so many passionate opinions that are factually unsupportable. I could list a couple of notorious ones, but it'd really amount to no more than trolling.

author: dom0
author_rank: Super Contributor
timestamp: January 11, 2016, 09:33:59 pm
content: Doesn't have to be about high end audio. Try convincing someone (most people anyway) working pro audio that differential signaling isn't about levels or using three wires, but about impedances.QuoteMe too. Thanks to ssassen for bumping the thread and of course a big thanks to GK for his patience

author: fivefish
author_rank: Frequent Contributor
timestamp: January 11, 2016, 10:35:47 pm
content: QuoteI think you meant "balanced lines"-- I agree, balanced lines is all about impedances.To OP:I'd like to see an actual board stuffed with components! What's the latest on this?And ignore the naysayers.

author: T3sl4co1l
author_rank: Super Contributor
timestamp: January 12, 2016, 12:01:04 am
content: Hmm, I didn't see this. Fascinating as usual, GKOnly problem I have, which is rather fundamental in my opinion, and I'm sure you're aware of the drawback:why screw it up by introducing explicit discontinuities?!The analog switches can probably be replaced by OTAs, or even just clamp diodes (back to back zeners from OUT to -IN) on the integrators really, so that a more modest nonlinearity can be used to transition from integrator to inverter behavior.The system is still nonlinear either way, so I wonder if it could be abused to exhibit chaotic behavior. (I'm kind of suspecting not, since it will tend to be more damped, rather than less, as it goes around.) An exploration of that regime (if any) might be informative, in regards to minimizing it (maybe turn "conditional" into "unconditional" stability -- in a wider, nonlinear whole-system sense, rather than the linearized LTI sense being used here).Tim

author: dom0
author_rank: Super Contributor
timestamp: January 12, 2016, 12:19:32 am
content: I guess technically you are right, although differential signaling isn't commonly used without balanced lines, even in digital circuits.As I understand GKs explanations the dominant pole fallback is around to force the amplifier into stable operation even if it is strongly clipping, because an oscillation once occured in this system doesn't die down. Really sounds more like a safety net, nothing that should be active in regular use.....

author: Richard Head
author_rank: Frequent Contributor
timestamp: January 12, 2016, 08:04:09 am
content: I wonder what the transient response would look like?

author: T3sl4co1l
author_rank: Super Contributor
timestamp: January 12, 2016, 10:20:32 am
content: Should be a little slow to get going, then an increasingly rapid slope (you might not be able to see the slopes transition as each integrator saturates), then as it settles down, a very long settling time, to an extremely precise level (though ultimately still limited by the offset and GBW of the amps involved).A fascinating implication of such a network is it seemingly implies that, if stages are transitioned continuously, then one would get a continuously variable (i.e., fractional) integral. In analogy to a saturating chain of ampilfiers, the sum of which exhibits a log(Vin) curve (log base being the gain per stage).The biggest downside is, the 'plant' (last integrator / gain stage / output) needs to be incredibly fast, because each integrator takes off a factor of 3 or more from the GBW. I think. "Very long settling time" is relative to this final bandwidth, which I suppose would have to be in the several MHz range, for an audio application (20kHz+). Which doesn't really sound so bad, at least.Also, the settling would be in terms of not just voltage, but "jerk" (derivative of acceleration, V/s^3)! Actual voltage settling may be good or poor (I'm not sure; depends how much voltage over/undershoot is required to adjust those derivatives; and the compensation*, of course), and will take all that extra time, but we're talking asymptotic reproduction of ramps and second order curves, which means fantastic AC reproduction.*A 4th order controller plus single pole 'plant' implies we can solve for coefficients (hopefully -- if possible) that satisfy a particular filter polynomial, such as a 5th order Bessel. That would be a sweet amplifier indeed.Tim

author: dom0
author_rank: Super Contributor
timestamp: January 12, 2016, 10:40:47 am
content: https://www.eevblog.com/forum/projects/an-experimental-4-th-order-linear-audio-power-amplfier/msg240562/#msg240562

author: wiss
author_rank: Frequent Contributor
timestamp: January 13, 2016, 10:25:41 pm
content: Just a thought (subscribe), THD must be dominated by the input stage in this design. In my experience it gets difficult to get that below -150 dB

author: nano
author_rank: Contributor
timestamp: January 14, 2016, 02:00:08 pm
content: Quote from: GK on May 15, 2013, 12:15:51 amQuote from: T3sl4co1l on January 12, 2016, 12:01:04 amThisvery nice publication describes a fairly simple saturation circuit which improves upon the zener method.As far as I know this solution is very very similar to what hypex uses in the ncore amplifiers.

author: GK
author_rank: Super Contributor
timestamp: January 15, 2016, 12:48:22 am
content: Quote from: T3sl4co1l on January 12, 2016, 12:01:04 amYou obviously need to think this through some more.Quote from: dom0 on January 12, 2016, 12:19:32 amCorrect.Quote from: T3sl4co1l on January 12, 2016, 10:20:32 amNot a single integrator even comes close to "saturating" during slew-rate limiting.Quote from: dom0 on January 12, 2016, 10:40:47 amI get the impression that hardly anyone has properly read through my posts.Quote from: nano on January 14, 2016, 02:00:08 pmInteresting paper, but the scheme is totally unworkable/inapplicable in this design which, having multiple cascaded integrators, is topologically vastly different in operation to the discussed class-d designs (which have ~constant loop gain throughout the audio band).

author: GK
author_rank: Super Contributor
timestamp: January 15, 2016, 01:03:51 am
content: Quote from: ssassen on January 11, 2016, 07:43:45 pmThanks for the interest. I sent the pictured amplifier overseas for independent measurement on an AP system 2, to someone who was going to co-author (simply write up their independent testing) the article for LA.The prototype amplifier when dud somewhere along the line - the independent linearity measurements did not match the final (comprehensive) measurements I made on the prototype before sending it off.The amplifier was returned to me and I eventually tracked the problem down to a dud driver transistor with a measured ~25 hfe. The amplifier PCB was dissected in the process and is not rebuildable.A major PITA all around and I've mostly lost incentive. I have started on a MKII prototype which is significantly improved again but the endeavour is on the backburner.

author: T3sl4co1l
author_rank: Super Contributor
timestamp: January 15, 2016, 06:54:39 am
content: Quote from: GK on January 15, 2016, 12:48:22 am..........................And?You could be less cryptic and more, Idunno, human or something.Tim

author: nano
author_rank: Contributor
timestamp: January 15, 2016, 07:13:30 am
content: Quote from: GK on January 15, 2016, 12:48:22 amI always wondered why Class-D amps use contsant loop gain. Putzeys mentioned he does this for audiophile reasons to have constant distortion over frequency. But are there any technichal drawbacks to place the poles at the origin (like you did)?I’m pretty new to all this control theory stuff and I’m still learning. Would the scheme from the Kemp Thesis work if you added (low gain) local feedback around the conventional power amplifier?

author: ssassen
author_rank: Newbie
timestamp: January 15, 2016, 09:46:05 am
content: Quote from: GK on January 15, 2016, 01:03:51 amThat's indeed unfortunate, I would however be interested in learning more about this MKII prototype, is there anything you can disclose, or would you rather this is kept off the forum?

author: ssassen
author_rank: Newbie
timestamp: January 15, 2016, 01:15:41 pm
content: Quote from: nano on January 15, 2016, 07:13:30 amIt needs to have a controlled behavior, as without an elegant mechanism to reduce the global loop's order you could end up in a situation where the MOSFETs at the output start conducting at the same time, effectively shorting out the power supply. I guess it needs no further explanation why that is a bad thing?

author: dom0
author_rank: Super Contributor
timestamp: January 15, 2016, 04:07:48 pm
content: There are actually Class D controllers around that have neither a controlled dead time nor cross-conduction protection? Ugh!

author: GK
author_rank: Super Contributor
timestamp: January 16, 2016, 12:22:55 am
content: Quote from: nano on January 15, 2016, 07:13:30 amThe short answer is that it is simpler; and in a high-order feedback amplifier quadruply so. I presume that you have read the "F word" article. I would take Putzeys claims regarding constant loop gain with a grain of salt. The article was essentially a precursor to the release of the "N-core" amplifiers to the audiophile market place. If you know much about that domain you know about the plethora of boutique audio designers and popular reviewers that differentiate their products with the "negative feedback is bad" religion.The original gripe with high levels of negative feedback go back to the thoroughly discredited postulations of Otala, which those without a good grasp of control theory continue to perpetuate to this day. It seems intuitively obvious to some that an amplifier with a close enough to theoretically ideal transfer function such that the open loop bandwidth is only perhaps a few tens of Hz is completely inadequate for faithfully reproducing 20Hz-20kHz audio. Personally, I found it a bit cringe worthy that Putzeys, in his article ironically tag lined "there is no such thing as too much feedback", essentially found a different angle to basically argue the exact same thing. Putzeys constant loop gain thing in my view was just a way to make "high level" negative feedback sound palatable to the audiophile crowd. He certainly hasn't managed to produce a credible psycho-acoustic reference for his subjective assertions.But anyway, a good a reason as any for sticking to constant loop gain throughout the audio frequency spectrum in a high-order feedback amplifier is simplicity, and there is nothing wrong with that. If you want to make, say a fourth or a fifth order design as I have shown here, with an open-loop transfer function approaching the ideal, you need to provide a bucket load of gain. You can't provide for that by simply tweaking some circuit constants. As far as I have managed to figure the most intuitively practical and mathematically obvious way of doing this is to cascade multiple integrator stages with zeros. But that adds significant complexity even before contemplating how to keep all of those integrators under control when the control loop enters saturation.There is no reason that I can see why you couldn't implement a control scheme as per the discussed class-d designs around a class-ab linear amplifier. You could even retain the output filter. That would be another technically interesting avenue to pursue just for the sake of it.