diyAudio - Blogs - janneman http://www.diyaudio.com/forums/blogs/janneman/ A community dedicated to helping everyone learn the art of audio. Projects by fanatics, for fanatics! en Sun, 12 Feb 2012 20:34:00 GMT vBulletin 60 http://cdn1.dastatic.com/forums/images/diy/misc/rss.jpg diyAudio - Blogs - janneman http://www.diyaudio.com/forums/blogs/janneman/ The F-word - or, why there is no such thing as too much feedback http://www.diyaudio.com/forums/blogs/janneman/734-f-word-why-there-no-such-thing-too-much-feedback.html Thu, 08 Sep 2011 14:55:37 GMT Well, I did already a blog on feedback, it's uses, misuses and misconceptions. But there is someone who can explain it much better than I can, someone who has proven that he really understands what's going on in a feedback amplifier. Someone who knows what the terms 'fast amplifier' and 'slow amplifier' really mean (and what not). Enter Bruno Putzeys, who wrote the article with the subject name in my bookzine Linear Audio, Volume 1. I feel that this is such an important matter for audio that I decided to place it online for free download. You'll find it at Linear Audio | your tech audio resource under the tab Online resources.
Enjoy the ride, and do let me know what you think about these issues.

jan didden ]]> janneman http://www.diyaudio.com/forums/blogs/janneman/734-f-word-why-there-no-such-thing-too-much-feedback.html Feedback, or how to be late and be on time the same time, all the time. http://www.diyaudio.com/forums/blogs/janneman/454-feedback-how-late-time-same-time-all-time.html Wed, 03 Nov 2010 05:19:44 GMT Just a couple of days ago I posted something to try to debunk that tired old myth that 'feedback always comes too late and therefor can't work'. Apart from the fact that obviously it does work, which makes the first statement pretty stupid to begin with, here's my take on it.

The myth may result from an often repeated misconception that feedback comes 'after the fact' and therefore always comes too late.
This has been shown to not be the case over and over again but if you have no engineering background it may be difficult to grasp the concept. Let me try to help.

Obviously, there is a signal delay in an amp from input to output and back to the input through the feedback loop. Since the feedback loop is generally a pair of resistors, the bulk of the delay is in the amp. That is the case both in non-feedback as well as in feedback amps. Such delays are very small, often fractions of a microsecond, and in this context can be ignored.

What is often confused with delay is the phaseshift of the signal while traversing the amp (again, phaseshift through the feedback network can safely be ignored in all but pathological cases). Now if you put a sine wave signal and its phase-shifted version on a scope, it appears that the lagging signal is delayed with respect to the other signal. But it is important to note that it is not strictly the case. Take the case of a sinusoidal current into a cap. The voltage that appears on the cap as a result of the current looks like that current and lags the current by 90 degrees. But, and this is important, any change in that current will have an immediate effect on the voltage! It is not that a chance in current makes the voltage chance some time later, no, the voltage on the cap immedately starts to chance when the currrent starts to change. In hindsignt, it is obvious: the current and voltage are phase-shifted with respect to each other but there is never a 'dead band' where the voltage 'waits' for the current to change. OK, so we have that out of the way.

So, the signal fed back to the input immediately changes if the input signal (or the output signal) changes; feedback does not 'come too late'. But because of phase shift in the amp, the fed back signal is phase shifted (and generally lags) the input signal. But the change in input signal immediately leads to a change in output voltage as well as feedback voltage. I know, it doesn't sound intuitive, but is it basic circuit theory nevertheless.

There are two effects from that phaseshift through the loop.
1 - The feedback mechanism depends on cancellation of (a fraction of) the output signal with the input signal, leaving only the difference, the distortion. That distortion (in opposite phase), is amplified and cancels the distortion in the output, making the amp more linear. With increasing frequency, the phaseshift between the (fraction of) the output and the input increases. Therefor, that cancellation gets less and less complete, making the feedback work less and less well with rising frequency. That is the reason that almost all amps show rising distortion with frequency: the feedback gets less effective.

2 - Another effect of increasing phase shift with frequency concerns stability. If you go so high in freq that the phaseshift approaches 180 degrees, your nfb turns into pfb. That means that the amp sort of generates it's own input signal that gets send around and around, much like howling in a system where the mike picks up the speaker signal, sends it back through the amp etc. If that happens, AND you still have gain in your circuit, you have an oscillator.

The difference between the actual phase shift and the dreaded 180 degrees, at the point where your gain drops down to one (so it can't oscillate), is called the phase margin. Now, even with some phase margin (less than 180 degrees phase shift and still gain > 1) you start to see ringing on fast signals.

But, it is very straight forward engineering to design your circuit to any stability point you want. Sometimes engineers accept some ringing at say supersonic frequencies because a) there will never be any signal in that region except on the test bench and b) it allows just a bit more nfb in the lower freqs to make the circuit a bit more linear or transparent.

Comments, buts, let me know!

jan didden ]]> janneman http://www.diyaudio.com/forums/blogs/janneman/454-feedback-how-late-time-same-time-all-time.html <![CDATA[Don't be such a scientist!]]> http://www.diyaudio.com/forums/blogs/janneman/297-dont-such-scientist.html Tue, 15 Jun 2010 16:21:05 GMT I didn't get it. There are gifted design engineers on this forum. They get involved in threads. BUT, in most cases, eventually an 'issue' develops and the engineering guy gets binned or banned or asks to be banned. Why why why? Happened to me a few times. Not that I got banned, thank Ohm, but I got close to leaving because I too got enough of it.
Of what?
Let me explain. Most engineering types like to explain things, to tell others with less experience and knowledge what they are doing wrong and how they can do it better. They inundate you with facts, figures, links to engineering papers etc, and expect that the other guy flows over with gratitude. But, funny enough, it doesn't happen that way. The 'other guy' gets pissed off from being corrected all the time. Hell, he didn't come here for that, he came to have fun, discuss his hobby and his latest creation.

[flashback] At the time Al Gore's An inconvenient Truth came out, the same director (!) also made Too hot not to handle. Both on the issue of Global Warming. Do you remember that other movie? Neither did I. Did you know that Al's flick, which is basically a spiced-up PowerPoint presentation, boxed $50million? What made these movies so different? THNTH was 100% accurate, had all the right figures and simulations. It was also boring. AIT has been shown to have some errors in a few places (although accurate in a general sense). Some of the graphs were 'enhanced' to let the trend stand out better. But it was NOT boring, because it appealed not just to the brain, but to the heart, to the gut, and Al's sex appeal (it's the eye of the beholder, of course).

[regular programming] Randy Olson is a guy who left a tenured position (the pot of gold for any academic person) to go to Hollywood. A scientist turned communicator. His short movie A flock of Dodo's earned him praise because it was a scientific movie that got to the heart, the gut etc. It may seem prepostrous to compare this forum with a succesful movie maker, but we can all learn a lot here, especially we the engineering types.
If you, as an engineer, want to explain why this newby has his cascodes wrong, you are in the same business as Hollywood. You are telling a story that you hope will stick and have an effect. So far, Hollywood is much, much better at it.
Facts are important. Engineering is important; none of our audio marvels would exist without solid engineering. Errors should be avoided. BUT, if the story you tell is boring, you're wasting your time. And as soon as you find out you're wasting your time, you get frustrated, angry, you clash with the members and the mods and you've lost.

Tell the engineering story from the heart, from the guts. Tell your own crooked path to learning, making mistakes, the joy of doing it right. By all means, be sure to get the facts straight. But don't be boring, don't be such a scientist!

Randy Olson, "Don't be such a scientist - Talking substance in an age of Style". Buy it. ]]> janneman http://www.diyaudio.com/forums/blogs/janneman/297-dont-such-scientist.html Nelson Pass Interview http://www.diyaudio.com/forums/blogs/janneman/213-nelson-pass-interview.html Tue, 23 Feb 2010 19:05:13 GMT My interview with our own Nelson Pass appeared in MultimediaManufacturer. Read it here: http://www.linearaudio.nl/interviews/np.pdf jan didden My interview with our own Nelson Pass appeared in MultimediaManufacturer.
Read it here: http://www.linearaudio.nl/interviews/np.pdf
jan didden ]]> janneman http://www.diyaudio.com/forums/blogs/janneman/213-nelson-pass-interview.html Tech note: Balanced lines-1 http://www.diyaudio.com/forums/blogs/janneman/182-tech-note-balanced-lines-1.html Wed, 13 Jan 2010 18:54:34 GMT Why would one use balanced interconnects, and how can we make them work well? Balanced lines came about at a time where very long signal lines... Why would one use balanced interconnects, and how can we make them work well?

Balanced lines came about at a time where very long signal lines were coming in use for telephone and later for large audio performance venues. If you use a single screened line for your signal, and the line is long, the ground current through the screen causes a voltage between the ground points of the cable ends. Since the signal send out (and received) is the difference between the voltage on the signal wire and the ground wire, the unwanted signal (noise, hum) is effectively added to the wanted (music) signal. We don’t want that.

The trick is to use TWO signal lines in parallel. You send the signal over the two lines in such a way that the signal you want to transmit is the difference between the signals on these two wires, and then at the receiving end you have an amp that reacts to the difference between the two lines, so your signal at the far end is the difference between the signals you send out. You use a third connection, the cable screen, to connect the grounds of the two pieces of equipment. But, and here’s the trick, the voltage on the ground wire is no longer part of the signal! That ground wire is more like a safety ground, and also to screen the signal wires from induced stuff. The screen is not perfect, so their will be some noise etc on the signal lines, but if we make sure that those are very close together, the induced junk on both will be equal. And since we know that the receiver amp also takes the difference between the noise and hum on the two lines, that noise effectively cancels. Neat, isn’t it?

A few observations.
For the cancellation to work, the induced noise & hum must be exactly the same in both wires. That means they have to be very close together, which can be done by twisting them. Secondly, and this is crucial, they must have the same impedance to ground so that the induced junk is attenuated exactly the same in each. An example: to attenuate the noise & hum at the receiving end by a factor of 1000 (which is 60dB) you must make sure that the impedances of the two lines are also balanced to 1/1000. This is a tall order especially at higher frequencies!

It is also clear from the above that the two signals in the wires need not be exactly equal. In fact, it doesn’t matter even if one wire carries no signal and the other does: still, at the receiving end you get the difference between the two.

There is however a good reason to make the two signals not only opposite in phase but also approximately equal: If your sending amp has a max amplitude of say 10VRMS (often limited by the available supply), and you send two opposite 10VRMS signals, the difference between the lines, which is what the receiver sees, amounts to 20VRMS. So this will let you have larger signals that can improve your signal to noise ratio by a few dB. BUT, for cancelling hum and noise, the signal levels are immaterial!

Sometimes the term ‘pseudo-balanced’ is used for connections that are impedance balanced but not signal balanced. That’s OK; for cancelling hum and noise, that’s all you need…..
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]]> janneman http://www.diyaudio.com/forums/blogs/janneman/182-tech-note-balanced-lines-1.html Tech note: voltage regulators-1 http://www.diyaudio.com/forums/blogs/janneman/174-tech-note-voltage-regulators-1.html Thu, 07 Jan 2010 18:10:41 GMT There are lots of types of voltage regulators, but in this installment I’ll talk about series regulators. What’s a regulator? It’s all in the... There are lots of types of voltage regulators, but in this installment I’ll talk about series regulators.

What’s a regulator? It’s all in the name: it REGULATES the voltage to the circuit to be powered to keep it constant and as free of noise and ripple as practical. The ‘regulation’ means that there is some circuitry that compares a reference voltage, like from a zener diode, to the regulated output voltage, and then uses the difference between the two to adjust another element to null that difference. The ‘compare-and-correct’ is crucial for a regulator, and is done by negative feedback….

Look at Fig 1: is there a regulator in there? No, they are all circuits that try to give a constant, ripple free voltage, but if you start to draw varying currents from them, the output will vary with that current and there is no mechanism that somehow tries to null out that variation. Fig 1c is better than 1b, because Q1 buffers the voltage from the zener reference, so the Vo will vary less with load current, but there still is no ‘compare-and-correct’ mechanism.


But, building on Fig 1c, we can add that ‘compare-and-correct’ element as in Fig 2. Let’s say we want a 14VDC regulated supply. We use a 7V zener diode to give us the reference; since it is only very little loaded, it will be pretty constant and ripple free. The opamp compares the ref to half of Vo (from the two 1k resistor divider) and if that is smaller than 7V, the opamp output will rise and open the transistor further, to increase Vo until it is 14VDC again so half of that will be equal to that 7V reference.


A few observations. The correction element (the opamp) needs an error to work; it needs a (very small) difference between its inputs. So the error is never 100% nulled out, but the DC difference is probably much less than the difference caused by the tolerance of the two Vo divider resistors. If you draw varying currents from this circuit, the opamp will immediately correct the output voltage to keep up with the Vo changes from the load current, again to within a few mV. That means, from the load point of view, it is almost an ideal voltage source that has very low output impedance, just a few milli-ohms for a good regulator.
Why is the output impedance Zo important? Load current will cause a voltage drop across that Zo. It doesn’t matter that Zo is not a physical resistor, Ohms’ law is unrelenting! Therefore, with a varying signal load current, this will cause signal ripple on Vo which may cause unwanted output signals in the amp that is supplied by this thing. Anyway, since the opamp will do its best to keep Vo at twice the zener voltage, any input ripple and hum coming from the rectifier is also greatly reduced.

If that opamp is very fast, the opamp will try to open that transistor quickly to correct any errors it senses, but if the pass transistor is much slower, the opamp actually overshoots the correction. At worst case, the system may start to oscillate: when the transistor finally reacts, it'll keep going and will be slow to turn off, so Vo will become too high, the opamp immediately wants to shut off the transistor, Vo gets too low, the opamp immediately wants to open the transistor, etc, you get the point.

To break that vicious cycle, most regulators have a capacitor at their output to prevent Vo to change too fast which prevents the opamp seeing a large and fast jump in Vo. This output cap is NOT responsible for supplying current to the load; it is in parallel with the few milli-ohms output impedance so it would probably need to be a farad or more to make any difference at all. No, that cap is there for stability, and it actually works better when it is a bit lossy so it can damp out any onset of oscillations better. So, don’t put a boutique foil cap here, as it is not only a waste of money but it actually works less well.

Stay tuned, and happy soldering!

Jan Didden
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]]> janneman http://www.diyaudio.com/forums/blogs/janneman/174-tech-note-voltage-regulators-1.html