Here's some good continuation for the Rane's article about tweeter damage due clipping presented by skey:
http://www.st-andrews.ac.uk/~jcgl/Scots_Guide/audio/clipping/page1.html

I don't see why it shouldn't work but i'm quite sure you have to tweak the component values.

The circuits i simulated presented only a small partion of different power amp circuits. For example, I could replace the VAS stage with either a simple common emitter or with a complementary feedback pair. Both would create different type of clipping thus changing the amplifier tone. I could convert to one rail supply, start shaping the tone within the feedback loop or just change feedback topology to feedback current instead of voltage - or both.

I chose not to. Some people claim that simple circuits sound better - now this claim is quite easy for most of us to believe, especially because most of the good sounding guitar PA circuits seem to be rather simple. I just wanted to know whether this claim holds a truth or not. As can be read from the results, there were some differences but they were undetectable by (at least my) ears. I guess this would be equally true in almost any topology i could test. The reason for that good tone must come from some other design aspects and the simplicity is easily explained: It's simple just because it can be.

Tubes? I don't want to mess with them, everyone seems to have too strong opinions about it. In general, i think their sound is overrated, as is the case with fets too: The circuitry makes more difference than the tube itself. Anyway, how many people actually can distinguish the tube tone? Even my petty Zoom 606 digital multieffect has fooled some people to think that my 12W solid state amp was a tube amplifier. The reason why people prefer tube PA's tone is that the power amplifier has a transformer which does some neat tricks. An inductor as a load behaves way differently than a resistor, you know. I'm quite sure that similar results could be achieved with transformer coupled high voltage fet drivers. Of course tube fanatics would disagree about anything with silicon sounding like valves, even when the hard eveidence would show that they're wrong. This is an endless debate and i rather stay out of it.

Anyway, since my test didn't seem to produce any wild results i think i might try to expand it and convert the simple topology to a "vintage" topology with transformer coupling. I probably will test it with only drivers coupled and with both drivers and the load coupled. That is, when i have time.

More simulation results and some impressions of my first - and non-blinded - listening tests:

- First and only easy-to-detect difference between the two circuits is that the simpler one exhibits a noticeable startup thump while voltages settle.
- Overdriving creates some differences between output signals. (See the first post). However, these differences seem to be detectable only by visual comparison of both output signals and at least i couldn't hear any of them.
- Just as i guessed, temperature variations affect the simple design more: Distortion varies within few decibels and gain varies few millivolts. (Max. 0.005% variation in the output). Modern design seems to be much more stabile. LTspice can not simulate a thermal breakdown, which would probably occure faster than any noticeable difference between output signals could be detected. Interesting point is that when the ambient temperature rises very high (100 C) the frequency response of both amplifiers becomes almost identical. Altogether, the differences caused by temperature changes where a lot smaller than i thought they would have been.
- Subtle differences between same transistor models seem to cause no measurable difference to the performance. I could push this test even further but i feel that would be deliberate biasing of the results. Usually the transistor's should be well matched anyway.

Actually, i'm very amazed that it's really hard to detect any difference between the circuits. I could be missing some important point. Please do comment and recommend some ideas.

Quote from: joecool85 on April 15, 2006, 08:50:37 AM
Hopefully its just the chip and you can solder in a new one on be on your merry way.

This comment expresses the main thing why i really do not like chip amps in general. I mean, there's nothing wrong with them; they are far more advanced than the basic discrete designs and packed with very nice features - plus their thermal coupling is awesome: No need for thermally coupled bias circuits. And fixing them is easy, just put in a new one.

...But what about 15 to 20 years from now (maybe even less) when National has decided to discontinue some of their chip amp products and LMxxxx chips have become obsolete? This has happened to far too many chips already and the petty DIY market is not really a reason for big manufacturers to reconstruct a product line. When you blow a discontinued chip are you going to hunt through junk yards or pawn shops - or just build another power amplifier board with a new chip model? In later case you can never reproduce the tone of the original chip. This is really, well, inconvenient situation for people who should fix equipment for their living.

True, transistors become obsolete too but at least one can always replace them with one's that are close enough to specs.

Anyway, sorry to hear what happened RDV. Blowing up stuff unintentedly is never fun. I once did that two times in a row for a complete powerstage destroying every transistor in it - both times because of a probe slip. You can imagine i was ready to kill myself just out of sheer annoyment. Hopefully you didn't blow anything expensive, like transformers.

Unfortunately i currently have no have time to construct these circuits in real-life so i have to rely on spice simulation that doesn't guarantee accurateness. However, ltspice has a feature that allows feeding the circuit with a .wav file and extracting the output. Unfortunately, this means that i will loose the "touch" to the guitar. In some sense this is a good thing: I do not have a chance to adjust my playing to the circuit. Basically, i will try this and also see if i can create some variation between identical spice components in order to simulate the real-life differencies that they might have. I have to find out how to do this correctly without biasing the results.

As of the test signals... well, naturally i have been planning to try with both "clean" and processed. Clean signal should be a good measure since it exhibits some transients that should overdrive the power stage and test how much the clipping changes the overrall tone. Currently unclipped signals of both circuits seem identical, except for the different frequency response. Processed tone... Well, i was planning to use some kind of distorted tone since it contains sharp edges that should - or should not - be amplified accurately. Also, both signals (clean and distorted) retain a lot of variations that should be tested: Ie. Palm muting, pinch harmonics, using a pickup or just finger picking. Trying both signal types with reverb is also one of my plans since producing reverb seems to be difficult for some systems.

And no. I have not planned to test the circuits with ie. different instrument signals or with a music or speech source. I already know from experience that the purpose of the amplifier dictates a lot of how it should behave. I feel that including any other than guitar amplification purpose to the test would expand the spectrum too much and i plan to keep this project as a way of measuring the production of guitar sound.

Colorization is a good point. Partly because i run a simulation both circuits are quite "flat", meaning the output is free from the colorisation, created by, for example the guitar speaker, that could actually change the tone drastically. (Excluding the simulation of speaker's impedance of course). I have a way to simulate the guitar speaker's frequency response so i can include it to the test. Anyway, i feel this should be done so that it will not bias the results.

I still have to plan how i will compare the results since i probably can not gather a test group. I have to find a way to play the clips in a random order (that i do not know at first). Then during listening i will write down my impressions and later see if they had any commonities to the the circuit's that produced the soundclips. The drawback of this test is that it's results are based on to my ears. They will be dictated by my taste, no matter what i do. If i find a way to upload some sound samples to the internet i will do it.

Edit: I will probably also create some clips where the output signals are mixed to some musical context.

Well, i'm not very good at explaining everything in a simple form but i can try.

The job of Q18 and Q11 (plus the circuitry that's surrounding them) is to provide a constant current for transistors. Why constant current? Well, If we take a basic common emitter circuit the voltage at it's transistor's collector can be calculated from the formula

      Vc = Vcc – IcRc,

which is simply the subtraction of supply voltage and voltage drop over collector resistor. Despite of it's simplicity the formula shows two very important things: 1. Fluctuations in the supply voltage show up in the collector voltage and 2. Smaller voltage drop over a constant resistance means a smaller collector current Ic. The latter means that as the transistor's collector voltage increases the collector current must decrease since the voltage difference between the collector and the supply becomes smaller. If the collector current decreases it means that the emitter current has to decrease as well and less emitter current means lower gain. The exact opposite happens when the output voltage decreases and as a result the waveform will become distorted having a flattened top halfwave and a stretched bottom halfwave.

So, the idea is simply to provide a constant current for the transistors to retain their linearity. If the current over resistor is

      I = U / R,

one can see it will fluctuate within the value of U. Constant current, however, remains constant. The bootstrap (R12, R1 & C4) is a form of constant current source too. It tries to create a constant (DC) voltage drop (= constant current) across the "lower" resistor (R1). (Ideally the AC voltage at the output should equal AC voltage at Q1's collector). The constant current actually allows the "higher" end of the capacitor to swing past rail voltage, which means the voltage at the output will swing closer to supply rail. Modern circuits tend to replace this form of CCS with an active one. Why? It is more linear and doesn't require a bulky capacitor.

Transistors Q17 and Q16 are another form of constant current source. Their job is to balance the loading of the differential circuit, which results into more linear operation.

I hope this was simple enough. If you have more interest, Rod Elliott has written some good articles about the operation of constant current sources and basics of amplifier design:

http://sound.westhost.com/ism.htm
http://sound.westhost.com/amp_design.htm

There are lot of claims that simpler power amplifier circuits sound better than complex modern ones. Could this really be so?

We all know that simplism has some non-arguable benefits:
- The devices are more likely closely matched since there is a smaller amount of them.
- The layout is simpler which can have some benefits: ie. shorter tracks, less interference. (Assuming everything is done right of course).

...As well as drawbacks:
- Simple design is more likely much more unstabile.
- Simple design is more likely much more unlinear.
- Bad layout will more likely cause more problems.



I simulated two circuits in ltspice. (See the thumbnail link). They are almost "identical", meaning the signal path is almost identical. Both circuits use long tailed pair input with the same amount of feedback so the gain is same, at least on 1 kHz. (There's soon more about this). The output topologies are a bit different: The simple, "old-fashioned" circuit has a quasi-complementary output, which was very common until high power complementary transistors became available. The topology has an advantage of closer matching between the output devices, while the "modern" one uses a fully complementary output. Besides output topology and replacing constant resistances or bootstraps with constant current sources these two circuits are identical.

Their performance however is not:
- The "modern" circuit has a more constant gain throughout bandwidth. The gain of the simple circuit drops heavily on higher frequencies.
- "Modern" circuit can swing closer to rails.
- Simple one suffers from higher DC offset.
- Simple one has worse distortion figures, especially on 2nd harmonic. An exception occures when the circuits are hardly overdriven: In the simpler circuit the 2nd harmonic drops 10dB in relation to "modern".

Every aspect of the simulation shows that the modern circuit is far more better. It shouldn't be more sterile or colder sounding, nor does it clip harsher in relation to simpler one. As a matter of fact, the the simpler one clips harsher than the modern! Undoubtedly the real-life indifferencies between devices will add something to the game. Is it enough to make a simple circuit perform better - i don't think so. Could this preferance over simplism be explained by "psychoacoustics": One knows that he is listening to a simpler circuit and regards the distortion as "pleasentness"? I will soon do some listening tests between these two simulated circuits and get back to this.

Meanwhile, be free to comment and present your ideas.

The more i have played with skey's idea the more sense it starts to make: The frequency band split frequency has to be chosen carefully, (so that the low freq will indeed mean low), and possibly the amount of overdrive on the low frequency side has to be limited a bit too. High frequencies do indeed seem to need more limiting than the low frequencies to sound clear but the most important thing is to attenuate both frequency bands before summing them up - otherwise the clipped low frequencies will dominate too much. If the signals are attenuated so that they will not overdrive the next stage the clipped bass actually seems to be harder to detect: The result is somewhat "unclear" tone which  has a "punchier bass" in comparison to the way of limiting the low frequency content, which seems to result into a "clear" low end but overdriven sound in general.  Both limiting tactics could benefit from a post low-pass filter that would attenuate the harsh high frequency noise that clipping the signal creates. Also, the idea seemed to work better on a clean signal, which naturally has a punchier low freq content than an overdriven signal. The results with highly overdriven palm-muted grind riffs weren't very impressive. In the later case limiting the signal equally on every frequency seemed to provide a better result.

I hope you will look further into this idea skey. At this point, the biggest downside i see in it is the added complexity of circuitry.

I usually take a sturdy sheet of metal and bend it to either U or L shape using a wooden hammer and some bars of wood that act as "guides". I wish i had an access to bending machine...

Sounds like an interesting theory. Have you tested it? As far as i know the bass frequencies tend to dominate, and if they clip they will drown the highs. You know that farting sound coming out from the speakers, thats a clipping bass frequency. It's much more difficult to detect a clipping high frequency. This is why most distortion pedals cut down bass frequencies before clipping stages.

I did a quick test on your theory. Unfortunately, i was quite sure that i could predict the result. I took a .wav file and split it in low and high freq content. I clipped both of them in two ways: 1) I clipped more the low frequency content and less the high freq (like in your theory) and 2) I clipped more the high freq and less the low freq. Finally, i summed the signals back together. I did the test with both clean and overdriven guitar sample and in both cases the one with more clipped bass sounded much more horrible than the one where the amplification of low frequencies was limited. The overdriven sample especially sounded really horrible when it clipped on low freqs.

Anyway, i also ran some ltspice tests which showed me that your theory might actually have some point: A constant bass frequency seemed to have less harmonics after clipping when the summed highs were attenuated. You might be able to build something out of this so don't give up. I'm also ready to change my opinion when i'm proven to be wrong.

Rane's document had some good points. It is right about the detectors operating too slowly to detect a clipping bass frequency. Also, you are right about the fact that limiter's should operate on separate frequency bands. Best audio limiters do.

Here is a very definitive collection of all kinds of audio circuits, including transistor guitar amps:
http://www.freeinfosociety.com/electronics/schempage.php?cat=1

Quote from: Stompin_Tom on April 11, 2006, 11:04:47 AM
What determines the output impedance of a circuit?

There are many ways to measure it and Google provides a good help - as usual. One way, but not neccessarily the best, is this: Just measure the resistance between ground and the output while the device is off. From a schematic i would look for resistance between signal node and ground/supply. If you want very reliable results use better ways: Google will list at least few of them with a search term "output impedance". I think that the authors of these sites do a better job in describing the methods than i would.

The preamps and stompboxes might work as is, then again they might not. In order to know that you have to know the input sensitivity of the power amp and the output voltage of the "pre" circuit. You also have to match up the input and output impedances so that you will not have any undesired frequency losses. The concept behind choosing the right amount of gain is actually quite simple to understand: The more gain you have the more closer to it's clipping point the amp stage operates. Amplify too much and you clip the signal. Distortion. Some circuits also become unstabile with too high gain - which usually shows up as oscillation.

Noise is a trickier thing since it comes from everywhere. Everything adds up a little bit of it. Unfortunately, the mean amplitude of guitar signal is quite low; only few tens of millivolts. Run this signal through an attenuating part of the circuit (basically any passive part ie. a tonestack or a resistor) and you will effectively increase the amplitude of noise in relation to signal. This is why you need to amplify the signal before processing it in a passive circuit: To increase Signal to Noise Ratio (SNR).

These two things (SNR) and gain go together: If you plan to process the signal, you need to amplify it in order to keep the SNR acceptable. Typical guitar preamps do not really add up much gain to the signal: Most of the amplification is done just to keep noise to the minimum while processing the signal passively. If processing is done in an active circuit the total gain of a preamp might be close to zero. Many power amps take a maximum input signal amplitude of 1 V RMS to reach the full power, some guitar power amps seem to need even less since they are typically quite high gain. So what is enough gain? If you're still uncertain read the first paragraph. After that read the next one to confuse you even more:

Guitar power (and pre) amps are quite tricky to design. Why? Consider this: In the worst case, an efficient guitar pickup might put out a transient of 2 volts. If the power amp operates at a gain of, say 30, that already equals an output voltage of 60V! You need some serious power to produce a voltage swing like that to a few ohm load. Gladly it's only a transient and your amp might even make it - for a small period of time - but only if the supply voltage and the amp stage's capability to swing close to rail is high enough. Tube amps have inductors at the output and can therefore swing higher than the supply voltage - modern transistor circuits can not do this. In an opamp based preamp circuit, (that can typically swing to about +-14V), a transient as high as this causes some serious distortion! Since the transient voltages can be quite high the gain has to be very low in order to retain a reasonable headroom. Tricky? Not yet, but here's the catch: The typical mean amplitude of a guitar signal is only about 20 mV! With a gain of 30 it means only 600 mV at the output. Quite pathetic, huh?

The circuit details of the original Deacy are not very well known indeed. As far as i know, it was built into an old "bookshelf" 12" x 6" hifi speaker box, yes, using germanium transistors. The circuit was reputedly based to an old 1960's car amp so i'm quite sure that the output topology was transformer coupled. I'd also guess that the circuit has been changed during maintenance more than once after the first build. The output power? Hard to say but that battery looks like the ones used for powering portable tube radios.
http://www.dawsonsonline.com/newlettpics/deacy2.jpg
http://www.guitarraonline.com.ar/set-ups/brianmay/chicas/05MAYfotoE.jpg