The title might seem paradoxical, but hear me out. 8)
Distortion is of course any modification to the waveform of the signal; this is caused by a non-linear transfer characteristic. The gain changes depending on the signal level. In the case of clipping the gain would be zero, as there is nowhere to go. If it drops to zero abrutly, the wave will be flat-topped with lots of higher order harmonics; this is generally considered harsh. Valves tend to clip softly with lower harmonics, and the wave is more rounded. This often seen as the holy grail of distortion: soft-clipping.
However, we don't need to clip the signal. :) As stated, distortion happens because the transfer characteristic is non-linear. Well, every device is non-linear to a degree. Pull up a datasheet and look at the tranfer curves; they're (obviously) not straight. This means that gain varies based on level, and the signal will be distorted.
What does this have to do with soft clipping? Well, it will sound a lot like soft clipping. I know RG has brought this up before with the conduction knee of diodes. Keep it in the non-linear area and there you go, sweet sounding distortion. Of course, you don't need to use diodes. If the gain stage is non-linear (of course it is) you'll get some distortion.
How would we apply it? Well, discrete components are a must. Look through datasheets to see which ones are less linear. Pick the operating conditions that maximize that area. Look at the curviest part of the curve and bias is into that area. Put a few in series to add to the distortion.
I hope that you found this interesting, and that it will give some of you some ideas. I also hope that you have more suggestions for increasing device nonlinearities too, I don't know too many. :)
lol, this is exactly where I've been going, especially with the fact that FETs are actually highly non-linear in large signal analysis. Its just that we zoom in so much with small signal changes that it appears more or less to be straight.
For instance, if you could actually push the thing into the scale where it starts taking on a curve function for gain, you could split the signal in half with two push-pull FETs, and then you'd have equal distortion on both sides of the voltage swing, or something like that.
from what I've read of papers on he subject, corners of clipping create more harmonics than non-linear gain.
You can see a few of my tries here.
http://www.diystompboxes.com/smfforum/index.php?topic=102984.0
You should take a look at some of the work being done by audiophiles on single-ended triode amplifiers with no negative feedback. They sound good, but there is definitely some distortion there and you don't need test equipment to hear it. The distortion comes from a non-linear transfer function which never gets to the clipping level. It is largely second and other even harmonics. But anything heard through these amplifiers does not sound like the original signal as provided by a low-distortion amplifier.
Some things come to mind.
Quote from: WaveshapeIllusions on June 11, 2013, 12:37:17 AM
Well, discrete components are a must.
Perhaps. However, waveshapers with opamps and diodes or transistors are a long tradition in the analog community.
QuoteI know RG has brought this up before with the conduction knee of diodes. Keep it in the non-linear area and there you go, sweet sounding distortion. Of course, you don't need to use diodes. If the gain stage is non-linear (of course it is) you'll get some distortion.
The MOS Doubler and JFET Doublers at GEO were an exercise in digging out the distortion products by cancelling the original signal. It is true that every device that isn't truly linear (and none of them are) will cause some distortion. However, the distortion of a bipolar, JFET, or MOSFET within the regions that are not obviously clipped tend to be about 1-4%. You need something curvier than these.
QuoteHow would we apply it? Look through datasheets to see which ones are less linear. Pick the operating conditions that maximize that area. Look at the curviest part of the curve and bias is into that area. Put a few in series to add to the distortion.
I posted some things a few years ago that will interest you. You're on a similar path.
Imagine a curved transfer function, with a straighter, more linear part and a more tightly curved part. If we magnify a diode characteristic up, it makes an OK thing to serve as an example.
Below the start of the bending in the conduction knee, it's pretty linear - well, linear by not conducting much, anyway. From the start of the conduction knee to where it straightens out again, you get varying degrees of distortion. Above that, conduction is much more linear again, and controlled by the resistances more than the junction. Even then, if you use a small enough signal, say 10-25mV, the diode's curviest parts look linear. This is the basis of diode modulators. For signals that are small enough, everything looks linear. So one way to make a tremolo is to force a varying current through a diode, and have its forward resistance to a tiny signal to change. The slope of the V-I curve at any point is the effective resistance at that point.
Hmmm. For tiny signals, it's linear. For signals about the size of the diode's forward conduction voltage or a bit less, it's soft clipping. For signals that are much larger than the diode's conduction region, the harmonic result is indistinguishable from razor-sharp clipping. It's not just where you bias the device, it's how big the signal is compared to the size of the conduction knee or nonlinear area in terms of voltage. If the signal is quite small, say 10% or less of the conduction knee, you get variable resistance. This was used in the Magnatone amps with their soft-curved varistors, which had a 70V knee (!) to vary volt-sized signals. If the signal is comparable to the size of the curved area, you get soft clipping to one degree or another. If the signal is bigger (5-10 times, up to several million times bigger), you get sharp clipping from the same device.
It's the relative size of the signal to the clipping knee that makes a difference. It's a question of scaling.
There are some practical issues with very small signals (noise!) and very large ones (power supply!) that make this an issue. I blathered on about that for a while, I believe. You can construct a curved response with opamps and switching devices like diodes and transistors for signals in the volt-to-ten-volt range well enough.
QuoteI also hope that you have more suggestions for increasing device nonlinearities too,
One that leaps immediately to mind is the Vbe multiplier. This is a transistor setup with two resistors that makes the transistor's collector-to-emitter voltage be a multiple of the Vbe. I always intended to work on that some more to see how valid it is in the knee region. This would let you have "diodes" of any reasonable size of 2-10 times perhaps the native V-I characteristic of the base-emitter junction. Looks promising, needs work.
Here's another thought that I posted a ways back: the iterated clipper.
Imagine a clipper that soft clips at 1.0V, with a "knee" range of 0.2V. We feed it a signal of 0.8V, and get not much. Increasing the signal to 0.9V to 1.0V gets more clipped, but still soft. Feeding it 1.5, 2, 3... volts gets increasingly hard-clipped because the portion of the signal's time spent in the clipping knee gets to be such a small percentage of the cycle time. So if we feed it a particular size signal, it seems to be soft clippable.
But real signals are all over the map. How do we ensure it gets just the right size signal?
One possibility is a compressor of some sort, perhaps even companding to put it in the right signal size and then expand it back.
Another is to use another copy of the soft clipper ahead of the soft clipper. The soft clipper clips when the signal is right. Set up the output of the first clipper to produce just the right size signal for the second clipper. But then the first soft clipper doesn't get into the action. OK, put another one ahead of that.
With iterated soft clippers and some care in designing the gains and attenuations between stages you start getting an all-soft clipped signal.
One reasonable way to do this is with CMOS logic gates used linearly, but with low feedback and careful attention to attenuating after gain stages to control the signal size. After all, the output of a CMOS gate has a well-fixed maximum output voltage, so you can get a handle on signal levels that way.
It's a good area to go looking. Hope those help.
Quote from: R.G. on June 11, 2013, 10:18:37 PM
... the iterated clipper. Imagine a clipper that soft clips at 1.0V, with a "knee" range of 0.2V. We feed it a signal of 0.8V, and get not much. Increasing the signal to 0.9V to 1.0V gets more clipped, but still soft. Feeding it 1.5, 2, 3... volts gets increasingly hard-clipped because the portion of the signal's time spent in the clipping knee gets to be such a small percentage of the cycle time. So if we feed it a particular size signal, it seems to be soft clippable.
BTW, that clipper exists (taken from Pritchard's patent "Semiconductor-emulation of vacuum tubes")
(http://diale.org/img/soft_clippling.png)
Quote from: R.G. on June 11, 2013, 10:18:37 PM
One that leaps immediately to mind is the Vbe multiplier.
ah, that is a very good idea ("rubber diode")!
Quote from: R.G. on June 11, 2013, 10:18:37 PM
One possibility is a compressor of some sort...
I've been playing with the distortion section of the Peavey standard amplifier (turning it to 9V) and it does exactly that, plus the soft clipping and the duty cycle modulation. The compression is not much, typically .25s before it starts to decay. It consists of a discrete op-amp (one NPN and PNP) with a buffer with a peculiar feedback arrangement. This amp is dated from the 70's (?) and we still doing the same thing, amazing!
Quote from: amptramp on June 11, 2013, 05:16:47 PM
You should take a look at some of the work being done by audiophiles on single-ended triode amplifiers with no negative feedback. They sound good, but there is definitely some distortion there and you don't need test equipment to hear it.
And it *really* sounds good with a guitar.
If you want to dork out on the patent stuff, here's a link.
http://www.google.com/patents/US5434536 (http://www.google.com/patents/US5434536)
At the bottom of the page there's a bunch of references to related patents, but the ones by Prichard are the most useful.
Here's a list of all of Prichard's patents - and many seem relevant to pedals.
https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Eric+K.+Pritchard%22 (https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Eric+K.+Pritchard%22)
Anyone have more info on this Prichard guy... does he make pedals? I'd like to see some of this patents in action.
Thanks for getting this all started TCA.
Here's a couple other related patent authors.
https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22W.+Brown+James+Sr.%22 (https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22W.+Brown+James+Sr.%22)
https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Jack+C.+Sondermeyer%22 (https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Jack+C.+Sondermeyer%22)
Peavey Corp has a bunch of good ones
https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=inassignee:%22Peavey+Electronics+Corporation%22 (https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=inassignee:%22Peavey+Electronics+Corporation%22)
From Mr. Peavey
https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Hartley+D.+Peavey%22 (https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Hartley+D.+Peavey%22)
Sorry, I got a little carried away. Good thing i have to work today... or I would probably spend all day looking at these things --- what a blast. .
oops another
https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Randall+C.+Smith%22 (https://www.google.com/search?tbo=p&tbm=pts&hl=en&q=ininventor:%22Randall+C.+Smith%22)
back to work...
Oh man, check out some of L. Fender's patents... so cool!!!
OK really, back to work...
QuoteAnyone have more info on this Prichard guy... does he make pedals? I'd like to see some of this patents in action.
He's the Pritchard in Pritchard amps: http://www.pritchardamps.com/pritchardamps.cfm (http://www.pritchardamps.com/pritchardamps.cfm) ;)
Quote from: tca on June 12, 2013, 04:25:15 AM
BTW, that clipper exists (taken from Pritchard's patent "Semiconductor-emulation of vacuum tubes")
There are many of them in various guises. The limiting section of many radio receiver front ends has an iterated diffamp limiter.
The Pritchard patent is a variant of the waveform shaper with voltage thresholds and diodes setting where the nonlinearities happen. In fact, I wonder that the patent examiner let that one past the "one skilled in the art" test, as most analog EEs would have recognized it immediately.
I was probably not clear about the iterated limiter. This was a a series of identical sections, each of which clips the signal. The trick is that each prepares the signal to fit into the soft clipping threshold of the next without going into the hard clipping area.
Quote from: WaveshapeIllusions on June 11, 2013, 12:37:17 AM
... it will sound a lot like soft clipping.
One of the things that we keep forgetting when discussing this topic is the frequency response, after and before clipping. In this vid, what do you think is the output (green, light green, red)? http://www.diale.org/ogg/clipp.ogv
P.S.
If you have any problems seeing the file use the VLC player: http://www.videolan.org/vlc/
P.P.S.
Here is an image (steady state):
(http://www.diale.org/img/clipp.png)
What is the wave form that is feeding the power amp? That rounded light green, or the dark red one?
as humans we each have our own subjective ideas about what distortion is and what it sounds like to us, to the extent of creating quite a vocabulary of fuzz - some of the descriptions of pedals on this very forum serve as example with a recent thread asking what is creaminess, fizz, etc or the odd russian translations of sand and lotion, etc., aside from hard or soft clipping, etc
psycho acoustics has always been a fascinating subject for me but i've only recently started thinking a little more beyond frequency response (fletcher munson curves etc) and time domain stuff - sound localisation/spatial awareness. (special awareness?! :P)
when musing distortion recently i stumbled upon evoked otoacoustic emissions:
Quote from: http://en.wikipedia.org/wiki/Otoacoustic_emissionAn otoacoustic emission (OAE) is a sound which is generated from within the inner ear. Having been predicted by Thomas Gold in 1948, its existence was first demonstrated experimentally by David Kemp in 1978... ...there are two types of otoacoustic emissions: spontaneous otoacoustic emissions (SOAEs), which can occur without external stimulation, and evoked otoacoustic emissions (EOAEs), which require an evoking stimulus.
which links to:
Quote from: http://en.wikipedia.org/wiki/Maryanne_Amacher"When played at the right sound level, which is quite high and exciting, the tones in this music will cause your ears to act as neurophonic instruments that emit sounds that will seem to be issuing directly from your head ... (my audiences) discover they are producing a tonal dimension of the music which interacts melodically, rhythmically, and spatially with the tones in the room. Tones 'dance' in the immediate space of their body, around them like a sonic wrap, cascade inside ears, and out to space in front of their eyes ... Do not be alarmed! Your ears are not behaving strange or being damaged! ... these virtual tones are a natural and very real physical aspect of auditory perception, similar to the fusing of two images resulting in a third three dimensional image in binocular perception ... I want to release this music which is produced by the listener ..."
apologies for the slight thread derail, wondered if folks had any thoughts or opinions on the subject.
Is there some reason that a VCA style compressor without the sidechain filtering couldn't accomplish this type of thing? I guess maybe you'd need to split the signal, half wave rectify and then compress each (to get both positive and negative swings) and then sum them back. Seems like this would give a whole lot of control over not only the curve itself, but also the symmetry of the soft-clipping.
Seems like if it was that easy somebody would have done it by now, but I can't see why it wouldn't work.
Quote from: artifus on June 12, 2013, 11:24:49 AM
... wondered if folks had any thoughts or opinions on the subject.
Ron's post is very pertinent. The guys from the DIY hi-fi community, namely Nelson Pass, has an idea that I quite follow. Air is a single ended medium, in the sense that it can be highly compressed but rarefaction is limited to zero pressure. In fact there is a simple relation between volume and pressure: VP^1.4=cte (1.4 is the adiabatic index), which means that sounds waves are slightly "soft clipped" on one side, more clipping means less pressure, and for a periodic signal that means more energy, louder sound with a lot of 2nd harmonics. So we *perceive* loudness, in a way, by the amount of even harmonics that we ear. This partially explains the common idea that a 1W tube amp "sounds" louder that an 1W SS amp (I'm thinking of a typical class AB amp, of course one can generate as many as we want even harmonics with SS devices) because asymmetric soft clipping can be obtained with tubes (running in class A).
But I've done some more experiments ... I've build some small wattage class A amps and some typical class AB audio amps. I usually run these amps on the same place where I have the hi-fi speakers. The curious thing is that I only get complains about listen to music to loud when running the class A amp, it *seems* that the sound waves travel better when generated in a single ended device, and again even harmonics play a role on the propagation also. Note that these comments are very, very, subjective...
Cheers.
QuoteThe curious thing is that I only get complains about listen to music to loud when running the class A amp
If Class AB sounds louder to you (because of the even harmonics). Then maybe you run your Class A louder than your Class AB. ;D
^ as I said, very, very, subjective...
The basic TubeScreamer core is actually a log amplifier. The reason it clips quite hard is due to the fact that the minimum gain is 12 and the maximum is 118. Thus, a semi-decent humbucker pushing out +/-1V will natively try to boost to +/-12V at min gain - It clearly cannot do that, given the +/-4.5V supply to the opamp, but it will try, except that the diodes hold that down to a maximum of about +/-0.7V. After that, it doesn't really matter how much you turn the gain up, the output will still be limited in this way. The difference, of course, is that (a) smaller and smaller input signals will still reach that clipping threshold and (b) large input signals will get more and more "square" corners, as the rate of increase of a large signal trying to be amplified 118 times will reach the clipping voltage a lot more quickly.
So, to achieve "pleasant distortion without clipping" one could (a) reduce the gain and, if required, (b) use more diodes.
It all depends upon what input signal level you have and what output you want... In the stock TS, you have a 4k7 resistor from Vb to -ve input and a 51k in series with a 500k VR from output to -ve input. In parallel with those last two, you have the diode pair that force the logarithmic response. (I've ignored the cap in series with the 4k7, which changes the response with frequency - low freqs get driven less than high ones.) So, if we change things to 4k7 "outside", 4k7 and 100k VR "inside", then we have changed the gain to a minimum of 2 and a max of a shade over 22. Double up the diodes and we have an absolute max output of about +/-1.4V or so. That will still give you a bit of "grit" at max gain, especially with larger input sources, but it should be pretty subtle, I think?
In the TS, gain is about unity for anything where input x gain > the diode drop. The opamp doesn't clip with reasonable inputs.
Edit - I guess that's difficult wording... If the "diodes open gain" multiplies the input to exceed the diode Vf, the diode closes, shorts the gain resistors, and brings total gain down to 1. This might be a novel way to control the input - cascade several stages of this, maybe... I guess people have running screamers into other screamers for a while now...
I've been playing around with an 808 clone for a few days now and I've also found that high output humbuckers can scare it a bit. Moving the "kinks" caused by diode conduction further away from zero by using higher forward voltage diodes is a possibility, but for me it's also a "flub" issue caused by the low end content from those humbuckers, so thinning out the sound by replacing the 1uF capacitor at the clipping opamp input with something smaller (starting with mebbe 0.1uF) helps too.
Regarding monkeying with non-linear transfer functions - computer-savvy types might want to check out Bidule. (http://www.plogue.com/products/bidule/) it's a modular audio environment that allows you to (amongst other things) apply maths to audio in real time. If you can express it as a "sum" then you can apply it to a signal by plugging together mathematical functions. There's filters, delays, oscillators and whatnot in there too as well as VST hosting. No affiliation, btw!
> frequency response, after and before clipping
+1
Quote from: tca on June 12, 2013, 01:12:41 PM
Air is a single ended medium, in the sense that it can be highly compressed but rarefaction is limited to zero pressure. In fact there is a simple relation between volume and pressure: VP^1.4=cte (1.4 is the adiabatic index), which means that sounds waves are slightly "soft clipped" on one side, more clipping means less pressure, and for a periodic signal that means more energy, louder sound with a lot of 2nd harmonics. So we *perceive* loudness, in a way, by the amount of even harmonics that we ear.
Yes, air is nonlinear. This is a consideration in the design of horn loudspeakers where the air pressures in the throat are sincerely high.
I wonder if you would compute out for me the sound pressure levels in db corresponding to 0.001, 0.01, 0.1, and 1% distortion of air itself as a result of the nonlinearity of the air itself.
QuoteThis partially explains the common idea that a 1W tube amp "sounds" louder that an 1W SS amp (I'm thinking of a typical class AB amp, of course one can generate as many as we want even harmonics with SS devices) because asymmetric soft clipping can be obtained with tubes (running in class A).
I wonder if you could contrast this theory with Russell O. Hamm's paper on vacuum tube circuits sounding louder than solid state.
QuoteBut I've done some more experiments ... I've build some small wattage class A amps and some typical class AB audio amps. I usually run these amps on the same place where I have the hi-fi speakers. The curious thing is that I only get complains about listen to music to loud when running the class A amp, it *seems* that the sound waves travel better when generated in a single ended device, and again even harmonics play a role on the propagation also. Note that these comments are very, very, subjective...
Weather was once very, very subjective. Lighting was the result of angering Zeus - or was it Thor? Or both? Illness was once very subjective - sickness was a result of evil spirits, and warts were cured by black cats and midnight, etc. As men of science, we have a duty to put numbers on things where we can, don't we?
> sound pressure levels in db corresponding to 0.001, 0.01, 0.1, and 1% distortion of air itself as a result of the nonlinearity of the air itself.
1% ~~ 164dB SPL WTF cares abut 0.001% THD?
Quote from: ashcat_lt on June 12, 2013, 03:37:43 PM
In the TS, gain is about unity for anything where input x gain > the diode drop. The opamp doesn't clip with reasonable inputs.
You're right, ashcat... Serves me right for digging out something I half-worked out about 4 years ago and not taking the time to review it. I'll try to get it right(er) this time. Just considering the first opamp, then as I said, at minimum "Drive" the resistors are trying to give you a gain of 12. With a +/- 1V input signal, they would
try to give you +/- 12V. They clearly cannot do that with only a +/-4.5V supply and, if you took the diodes out, the rail-to-rail clipping would be horrendous to listen to. But the diodes in parallel with the resistors limit the voltage swing available from the output to the -ve input. As soon as they start to conduct, the resistors begin to be bypassed. I made a mistake earlier... the diodes actually limit the amount by which the output can
exceed the input. So, the opamp would, in the absence of the diodes, try to smash from rail to rail. With the diodes, a +/-1V input appears as about a +/-1.5V output and the waveform gets weird in any case.
The other mistake I made 4 years ago was to investigate a similar layout, but wired as an
inverting amp, not, as the TS is connected, in a
non-inverting configuration. If you wire it as an inverting stage, then you get a "proper" logarithmic amplifier (pretty crude and inaccurate) where the output can never exceed +/- one diode drop (obviously more diodes = bigger output). I think judicious choice of gain resistors and number of diodes might serve the OP's needs?
Wow, this picked up quick. I think diodes would probably work pretty well after everything that's been said about them. The issue is keeping them at the knee. I play bass, so there are some rather large peaks followed by lower levels, so it would probably overshoot the clipping threshold on the attack. I'll have to try a VCA beforehand to keep the levels even as suggested.
My main focus has been on amplifying devices though. I like the SET idea, those sound nice. I think log amp was mentioned? That would be interesting too. I've mainly been looking at different loads to put at the collector/drain. I've heard a lamp can bend the curve a bit. I've been thinking that using a diode as a load might be interesting, if it can be kept at the knee area. Perhaps an LED?
I agree that frequency response is very important to the sound. Lowpass filtering on both sides smooths things out significantly. I've got a big muff with a cutoff that must be around 1 -2 kHz and it is rather pleasant sounding. I try to cut some bass before clipping to keep it from cutting off the peaks too much, and more lowpass filtering keeps the harmonics generated from going too high.
The TS ideas seem good too. I think some of the log amps use diodes in the FB loop to get that transfer characteristic. I've thought about using resistance in series with the diodes so that gain decreases with each step up in voltage. Haven't tried it yet though.
Changing the signal level to get it to sound right is good as well. I think tca mentioned that in another thread. If the signal is at the right level all the time you can put the curvier sections to good use. Or if the signal occupies a good chunk of the curve, it would probably have higher distortion.
"keeping them at the knee" is probably the essence of my own diode based single transistor distortion. It makes a beautiful sine wave that, for all intents on purposes in that single frequency (without being able to measure harmonics), appears to mimic the kind of "clipping" that tubes do. With the proper biasing, anything's possible.
Quote from: R.G. on June 12, 2013, 11:34:47 PM
I wonder if you would compute out for me the sound pressure levels in db corresponding to 0.001, 0.01, 0.1, and 1% distortion of air itself as a result of the nonlinearity of the air itself.
I'll do that, need some more time to think... Actually I've some numerical musings about it.
Quote from: R.G. on June 12, 2013, 11:34:47 PM
II wonder if you could contrast this theory with Russell O. Hamm's paper on vacuum tube circuits sounding louder than solid state.
That is not to hard, do that later on.
Quote from: R.G. on June 12, 2013, 11:34:47 PM
Weather was once very, very subjective. Lighting was the result of angering Zeus - or was it Thor? Or both? Illness was once very subjective - sickness was a result of evil spirits, and warts were cured by black cats and midnight, etc. As men of science, we have a duty to put numbers on things where we can, don't we?
You have to agree that the dark side has it charms, I'm often, as any other mortal, attracted to this medieval side of common existence. ;) But you are right, *I should* put some numbers on to things!
General comment; the essence of using diode-connected MOSFETS (gate->drain) as diodes is that this setup gives a very large knee curve that never seems to fully saturate to some very small resistance like bipolars do.
The result for signals of a volt to several volts is that the "information" above the start of diode conduction is never fully lost by being squashed completely flat, and the knee is quite big and round. It's not quite a Vbe multiplier, but has its own charms.
Quote from: tca on June 13, 2013, 06:18:06 AM
Quote from: R.G. on June 12, 2013, 11:34:47 PM
I wonder if you would compute out for me the sound pressure levels in db corresponding to 0.001, 0.01, 0.1, and 1% distortion of air itself as a result of the nonlinearity of the air itself.
I'll do that, need some more time to think... Actually I've some numerical musings about it.
at what temperature, altitude and humidity? (etc.) :P
there are many phenomena in this universe we have yet to attach numbers to and quite some doubt about many of the numbers we have decided upon and use by default. most are just handy models that kinda seem to hold true under certain conditions... the map is not the territory. the weather is still somewhat of a mystery.
thinking only in numbers can be quite restrictive creatively as thinking only creatively can sometimes turn out to be too fanciful once the numbers are applied. but most of the more interesting stuff goes on at the edges. i s'pose creativity is the hunt and the numbers the kill? nailing it down once detected?
if the same purely numerical approach were taken in music creation we'd all just be playing the same 12 bar blues riff over and over again for decades... wait, what? :P
and most musical disection and critique happens post creation.
isn't it all just different languages and thought processes to consider the one same thing? tools i guess.
Quote from: artifus on June 13, 2013, 03:31:10 PM
at what temperature, altitude and humidity? (etc.) :P
You're messing up my next question!! :icon_lol:
Quotethere are many phenomena in this universe we have yet to attach numbers to and quite some doubt about many of the numbers we have decided upon and use by default. most are just handy models that kinda seem to hold true under certain conditions... the map is not the territory. the weather is still somewhat of a mystery.
On the other hand, if you can keep your head straight that the numbers are a useful approximation, they can be a big help. In the works of Lord Kelvin on three separate quotes:
"To measure is to know."
"If you can not measure it, you can not improve it."
"In physical science the first essential step in the direction of learning any subject is to find principles of numerical reckoning and practicable methods for measuring some quality connected with it. I often say that when you can measure what you are speaking about, and express it in numbers, you know something about it; but when you cannot measure it, when you cannot express it in numbers, your knowledge is of a meagre and unsatisfactory kind; it may be the beginning of knowledge, but you have scarcely in your thoughts advanced to the state of Science, whatever the matter may be." [PLA, vol. 1, "Electrical Units of Measurement", 1883-05-03]
He was not hung up on the numbers being exact, but being a means to knowledge. Smart guy. We named the absolute temperature scale in his honor.
Quotethinking only in numbers can be quite restrictive creatively as thinking only creatively can sometimes turn out to be too fanciful once the numbers are applied. but most of the more interesting stuff goes on at the edges. i s'pose creativity is the hunt and the numbers the kill? nailing it down once detected?
In the words of Robert A. Heinlein:
"A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects."
Quoteif the same purely numerical approach were taken in music creation we'd all just be playing the same 12 bar blues riff over and over again for decades... wait, what? :P
Interesting that you should bring that up. You'd like to read something about algorithmic music, I think. See https://ccrma.stanford.edu/~blackrse/algorithm.html (https://ccrma.stanford.edu/~blackrse/algorithm.html) for a starting point. I understand that some early work on this attempted to approximate the style of historical composers by setting certain parameters to tell the computers how to compose. The story goes that they tried it by setting the parameters to compose rags in the style of Scott Joplin. They had some problems, because the program kept turning out tunes that Joplin had, in fact written. They had to back off a bit on the parameters to get it to do similar but not the same. :icon_eek:
Quoteisn't it all just different languages and thought processes to consider the one same thing? tools i guess.
It is, but just when we're OK with that, something odd comes along. It turns out that people do not hear the same melody in the same way, and that it seems to be related to cultural groups or ethnic groups. Your comment about "languages" keyed that one up. :icon_biggrin:
Quote from: R.G. on June 13, 2013, 04:42:31 PM
You'd like to read something about algorithmic music, I think...
thanks for the link. seen this: http://canonical.org/~kragen/bytebeat/ (http://canonical.org/~kragen/bytebeat/) ?
Note that just about any curve that you try to apply, logarithmic or otherwise, is going to have the issues of "staying in the knee" as RG put it. If you zoom in too far, even the curviest part will be close enough to linear to not make a difference. If you zoom out far enough, it pretty much has to approach linearity at some point.
A diode's IV curve looks a lot like a log curve to me, and I'd imagine a log amp will give us the same problems. Without recursion of the process, you end up having to design the thing to work well with a very narrow range of inputs. The problem of making one "soft clipping" box which will work well with any guitar and player you'd want to plug in still remains.
I would submit that if we try too hard to constrain the input to keep it in the knee, we will necessarily begin to limit the dynamic response of the circuit, such that "touch response" and "cleaning up when you roll back the volume" will start to go out the window pretty quick.
I notice in guitar pedals we don't talk about transformer distortion at all. Very small transformers with small cores suffer from very high low frequency distortion - so it generates partials of lower frequencies, but not of upper frequencies. This means less "fizz" and you're not clipping. I've driven a booster from my bass into a passive DI before to exagerate this effect.
There's a circuit for a log amp in Horowitz "The Art Of Electronics" on pages 212 - 218 (2nd Ed) and a better, temp compensated one on page 254 that will happily do 4 decades up to 4V... it sez 'ere.
Quote from: ashcat_lt on June 13, 2013, 06:55:10 PM
Note that just about any curve that you try to apply, logarithmic or otherwise, is going to have the issues of "staying in the knee" as RG put it. If you zoom in too far, even the curviest part will be close enough to linear to not make a difference. If you zoom out far enough, it pretty much has to approach linearity at some point.
A diode's IV curve looks a lot like a log curve to me, and I'd imagine a log amp will give us the same problems. Without recursion of the process, you end up having to design the thing to work well with a very narrow range of inputs. The problem of making one "soft clipping" box which will work well with any guitar and player you'd want to plug in still remains.
I would submit that if we try too hard to constrain the input to keep it in the knee, we will necessarily begin to limit the dynamic response of the circuit, such that "touch response" and "cleaning up when you roll back the volume" will start to go out the window pretty quick.
Now you're on the path. :icon_biggrin:
And the IV curve *is* a log curve. The current through a diode is an exponential function of the voltage across it, right there in the semiconductor equations. Various imperfections limit diodes; a diode connected transistor follows the diode equation better than an isolated diode.
Quote from: Jazznoise on June 13, 2013, 07:44:56 PM
I notice in guitar pedals we don't talk about transformer distortion at all. Very small transformers with small cores suffer from very high low frequency distortion - so it generates partials of lower frequencies, but not of upper frequencies. This means less "fizz" and you're not clipping. I've driven a booster from my bass into a passive DI before to exagerate this effect.
It's hard to tame. The problem is that it's linearly related to the volt-time product, and works on linearly increasing frequencies. Much like pot taper, the "curve" of distortion versus frequency is very abrupt sounding.
> ...a log amp ... Without recursion of the process, you end up having to design the thing to work well with a very narrow range of inputs. The problem of making one "soft clipping" box which will work well with any guitar and player you'd want to plug in still remains.
"Any guitar and player" is compensated by the Drive knob.
A LOG amp makes near the *same* output shape for "any" level (over 30dB to 70dB depending what you pay and how much hiss you accept).
But a single hand+axe still makes a wide range of signal level, from a quiet tease to a wild finish.
To some degree we *need* the overtone structure to change. Hot-signal and Fuzz are ways we "fake" the overtone dynamics that voice (and flute sax trumpet) have and that a naked steel string lacks. When a singer raises her voice, the total power changes little, but the overtones get a lot stronger. For the quite small (because amplified) vibrations of electric guitar, the overtone dynamics are small. Extra "small" because amplified guitar is supposed to be BIG, so we should have "big shout/scream" sound also, to contrast with the mellow sound at lesser levels.
> at what temperature, altitude and humidity?
Say "air you can breathe". It *really* does not matter, 2.0% or 2.1%.... R.G. is asking 2% or 0.02%?
And there is a simple analogy to rough-estimate the magnitude of acoustic distortion from knowledge of electronic distortion. You need a number for SPL of "loud" and a number for air-total-overload.
When done with that (?), find numbers for the distortion of the *ear*. (This is not so easily computed, without destructive disassembly; but it can be tested several ways, numbers are out there, and much higher than air-overload).
I think that some numerical musing is in order. The next figure shows 4 graphs: the characteristic function of a diode (VD as a function of Vi, btw it is not logarithmic), the condition number of such that function, a typical sin wave clipping and the condition number of that wave as a function of time.
(http://www.diale.org/img/diode_char.png)
A few introductory comments. "the condition number of a function with respect to an argument measures how much the output value of the function can change for a small change in the input argument" (ref.: http://en.wikipedia.org/wiki/Condition_number), for a power law x^a is simply a. As you can see this shows that the diode characteristic can only give us power law behavior with exponents between 1 and 0. Secondly I should say that the plots are not numerical, they were obtained by analytically solving the non-linear equation
VD=n*VT*ln[1+Vi/(IS*R)-VD/(IS*R)]
Notice the log contribution of the diode (you can ask how do I managed to solve such an equation :D ).
I think this type of clipping is completely different of what one gets from a vacuum tube although we try hard to make it to sound just like one. These plots just complements all the comments above about scale, amplitude and distortion. Also note the slight hysteresis of the last plot.
Cheers.
Funny how this comes up now..I've been thinking a few weeks about how I could slow down the transition from clean to distorted, pretty much exactly the same thing this thread is about....i have come up with an idea I want to try but have no time for another few weeks. Its diode based clipping with dual pairs of diodes where one pair act as a threshold for lower gain....without my breadboard I'm not bright enough to say it works or dismiss it(and probably should just shut up until I know, so I don't look stupid if it doesnt) , but if anyone is interested in doing the legwork, i will scan my notes and someone else can try it...j
Maybe something opposite to clipping?
For example, a non-linear EXPANDING of a sine-wave?
If we're talking a discreet BJT transistor, what about diode leakage current to apply some current to the base when biased signal would otherwise put the transistor completely into cutoff?
There is a "pocket" of non-linear amplification for BJTs and FETs. Whenever you make a graph for these things, you'll usually see
(http://i42.tinypic.com/xxd6p.gif)
Fig A is a BJT, Fig B is a FET, Fig C is a pulsed gain for a BJT (just for the sake of interest).
Something to notice about the FET is the non-linear amplification when the FET is receiving amps nearly equivalent to vacuum tubes.
FETs are naturally non-linear, although what we do is zoom in to a large degree by using smaller current, so that the curve plot takes on a much more linear appearance, with a negligible .01% or less error.
The BJT has a "pocket" of non-linear gain, if you were to keep the voltages and current around that zone.
My own examples of achieving non-linear gain range from exactly this "pocketing" for BJT through what is usually "hard clipping" diode sets, via proper input voltage and current, to "misbiasing" and adding a FET to a BJT.
^ That pockets that you show are 0.2V (or smaller) in amplitude, it should be difficult to keep the input signal well restraint to get that behavior. In the third case temperature is about 125C! But I do get you point.
I think the distortion in a guitar amp has a big amplitude, i.e., the ratio of the input voltage variation by the voltage power source is almost unity, for certain bigger than 0.707 What I think is missing is a dynamical way of controlling gain, compression or expanding the signal to keep it in that good region for long enough time. One other thing is that when we think of clipping we almost forget what happens to the other part of the signal, it is not only the soft-clipping but the elongation of the other non-clipped part; the crossing zero voltage points are different for the input signal and the output. Don't know if clear enough.
Yes, although the third graph also has 25C and -40C, which requires greater current to hit a useable small signal pocket, right after line A and Line B cross. Again, it's reminiscent of vacuum tube amps.
Well yeah, when the ratio of your loudest to quietest discernable signal amplitudes exceeds 80dB (i.e. 10,000:1) then keeping your signal in a tenuous "sweet spot" is a challenge. :icon_wink:
I reckon that our tendency to think in terms of imaginary sinewaves on a virtual oscilloscope and all the other circuit simulation stuff makes us forget about the dynamic range of real world signals and the principle that any curve approaches a straight line if you're only acting on a small part of it. No-one ever seems to mention grid conduction ( a power 3/2 phenomenon) when talking about valve stuff either, but that's another can o' worms...
This subject has vexed me for decades, but experimentation suggests to me that it's about far more than simple static curves. Pre and post eq plays a huge role, as does time-variant whatnot. Sooner or later you have to stop looking at wiggly lines in a computer simulation and go build / measure / listen!
yeah but what happens when i bend down from a flattened ninth to an augmented seventh over a passing fourth? does it go to eleven? :P
Quote from: tca on June 14, 2013, 06:53:04 PM
What I think is missing is a dynamical way of controlling gain, compression or expanding the signal to keep it in that good region for long enough time.
Hmmm. Almost sounds like a case for companding before and after distortion, doesn't it?
:icon_biggrin:
Quote from: gritz on June 14, 2013, 07:36:57 PM
Well yeah, when the ratio of your loudest to quietest discernable signal amplitudes exceeds 80dB (i.e. 10,000:1) then keeping your signal in a tenuous "sweet spot" is a challenge. :icon_wink:
Hmmm. Almost sounds like a case for companding before and after distortion, doesn't it?
:icon_biggrin:
Quoteexperimentation suggests to me that it's about far more than simple static curves. Pre and post eq plays a huge role
I've been suggesting for years if not a decade or more that distortion experimenters go get a surplus/cheap stereo graphic equalizer and put one channel before their distortion, one after. It's an easy way to spend a whole weekend making ever-more beautiful distortion sounds like snowflakes - no two the same, and probably no way to get back to the last one you loved, however fleetingly.
:icon_biggrin:
EQ stack pre-distortion = selective band distortion, more or less distortion for frequency range.
possibly.
Companding sounds like a great way to get it to sound right. Compress the signal to keep it in the right region; impress the input signal amplitude on the distorted side. Of course, with enough stages of a few percent distortion the desired effect would be fine. I think four might be enough. My current pre has three (two common source and a follower) and it sounds like its almost there. It clips on every low note attack though, so I think it needs more interstage attenuation. Just enough to minimize clipping though. Those clipped transients add to the sound.
I actually don't get the soft clipping tube vs transistor comparisons. On the amp comparisons, tube amps traditionally have minimal feedback. Tranistor amps have lots, so of course they clip suddenly. If you use tubes with that level of feedback their clipping performance is almost identical. A single transistor is quite a bit rounder when it saturates.
On the EQ idea, I've been meaning to put something like that together. I was thinking that a state variable filter with variable frequency would be good on the input. Possibly two peaking parametric EQs for the mids. I'd say the output shaping has a bigger effect on the tone than the input, so it would need more dramatic shaping options. Probably two shelving, maybe three. Three or four peaking. All parametric of course. If I wanted to be really crazy maybe one peaking that tracks frequency so that you could accentuate certain harmonics of the note.
I need to draw up a schematic for this. Call it the most tunable distortion.
Quote from: R.G. on June 14, 2013, 08:06:37 PM
I've been suggesting for years if not a decade or more that distortion experimenters go get a surplus/cheap stereo graphic equalizer and put one channel before their distortion, one after. It's an easy way to spend a whole weekend making ever-more beautiful distortion sounds like snowflakes - no two the same, and probably no way to get back to the last one you loved, however fleetingly.
Specially the mids... from 300Hz to 2kHz, I had the same experience with a simple two pots position/cut mids.
Quote from: R.G. on June 14, 2013, 08:06:37 PM
Hmmm. Almost sounds like a case for companding before and after distortion, doesn't it?
:icon_biggrin:
Have you ever actually tried it?
I have and it sounded pish (to me). :icon_lol:
It's all subjective, of course - we are subconciously "trained" by years of experience and cultural influence to expect guitar to sound / behave in a certain way and dynamic range compression is all part of the distortion sauce. By contrast, in electronic music land one can apply all sorts of "wrong" (flabby, fizzy or just plain broken) distortion to a synth or sample and in the right context it can sound awesome.
Anyway, companders. These will add time-variant amplitude discontinuities (and noise, most likely). In software we have the luxury of tricks like Hilbert transforms and lookahead that makes level detection for (fairly) blameless companding less of an ordeal, but in analogue hardware? Still, perhaps the results of the compander "getting it wrong" on transients might make it more interesting. Harmonic distortion is just harmonic distortion, after all. More or less of the same odds and / or evens, with maybe more (or less) up top. Is there a Holy Grail? I'm not sure. Are there better places to look for one? Quite possibly, I'd say - unless you want to make a guitar sound less like the guitar we all know, or you're just curious. But there ain't no point in being curious if you're not going to do anything about it, so maybe it's time to stop speculating and get building... :icon_wink:
Quote from: Thecomedian on June 14, 2013, 08:27:56 PM
EQ stack pre-distortion = selective band distortion, more or less distortion for frequency range.
possibly.
Like the RAT's reduction of bass in the distortion cell? I suspect that this was designed to try to stop the bass strings swamping treble freqs as the chord dies away some seconds after pick-strike.
As to the compression inate to "classic" distortion circuits, how about an envelope detector at the input to alter the gain of the output buffer/gain recovery stage? You would get the gritty sound we all know and love, but playing dynamics would be in there as well...
Quote from: gritz on June 15, 2013, 06:29:11 AM
Have you ever actually tried it?
I have and it sounded pish (to me). :icon_lol:
Yes, I have. It is quite tricky to get normal companders to get the amount of compression and expansion right for a good guitar sound. And as always "good" and "sounded like" are very, very subjective.
Quote
It's all subjective, of course
... as you note. :icon_biggrin:
QuoteAnyway, companders. These will add time-variant amplitude discontinuities (and noise, most likely).
A well-done compander will match its compression and expansion time constants to the source and the listener's preferences. This is (a) tricky and (b) subjective again.
And noise. I found I needed to bring guitar up to line level first, then compand to get the compander I was using - the 571 - to work well. Trying to use the compander all by itself injected noticeable noise, pre-amplifying much less so. Of course, since distorting the signal is level dependent, that required reworking the distortion, too.
And, after all, what I said was that the post I replied to sounded like a case for compansion, no? :icon_biggrin:
QuoteStill, perhaps the results of the compander "getting it wrong" on transients might make it more interesting.
...
Harmonic distortion is just harmonic distortion, after all. More or less of the same odds and / or evens, with maybe more (or less) up top. Is there a Holy Grail? I'm not sure. Are there better places to look for one? Quite possibly, I'd say - unless you want to make a guitar sound less like the guitar we all know, or you're just curious.
Practically everything we do to guitar today makes it sound less like the guitar we know. Distortion lops off most of the excursions and transients by its nature. The holy grail of most metalheads is "sustain for days" and "more gain", meaning "more distortion" as we now know. In fact, we have a paucity of new places to look for effects these days. We distort the waveforms, mess with the attack time and decay time with attack delays and compression sustainers, fool with the harmonic content with pre-filters, EQs and post filters, time delay, wobbling frequency peaks and notches, even deliberately adding noise in two cases I remember.
In retrospect, the idea seems to be to change something about a guitar's sound to make it sound less like the guitar we all think we know, but to do that in an interesting and arguably new(ish) way that's not too far from the bounds of "guitar as we know it", but far enough to sound novel.
So - are there better ways? I'm sure there are. But the DIY community is an expanding wavefront in all directions today. The trick is to be new, but not too new, different but not too different, a true "don't care whatever happened before" rebel, but with a "vintage" sound somehow. It's a little schizoid, no?
And "transparent". I forgot "transparent". :icon_lol:
Quote
But there ain't no point in being curious if you're not going to do anything about it, so maybe it's time to stop speculating and get building...
I heartily recommend that. Been doing that for nearly 40 years now.
Hi R.G. - I suppose the answer (to the original poster as well as everyone else) is "well, it depends on that you want to do."
Answers might include:
!) "I want it to sound like x"
2) "I don't want it to sound like x"
3) "I don't have the faintest idea what I want it to sound like, but if I'm gonna have to spend weeks / months on it then it better not suck"
4) "I'm just curious and I'll get around to it one day"
5) "I'm just curious, but I doubt I'll ever do anything about it"
6) "I want analogue, dammit (with proper, old - fashioned components)"
7) I'm happy to rough it out in digital first, just to see if it's worth pursuing"
etc.
All are valid and will help narrow down the approach (or lack of it!)
re the NE570 / 571 and beyond. A 2:1 compression / expansion ratio isn't going to get us very far at all. As a very clunky comparison the Tubescreamer uses a maximum of over 40dB of gain to send the signal past that "crossover kink" for a reasonable duration. Keeping a signal range of 40dB (or more likely 60+dB) within the sexy part of a transfer function will require something far smarter - maybe based on the THAT 218x VCA and 2252 RMS detector with the adaptive capacitor time constant hack - details available on the THAT website. They're not the easiest chips in the world to breadboard though - simply because they're so sensitive.
I monkeyed with companding as part of research into harmonic excitement - feeding a compressed signal into a bendy transfer function in order to create a (hopefully) constantish level of harmonic content that could be eq'd, expanded and then mixed back in. Great for adding a bit of shimmer to certain sounds, but imo not stellar with guitar or bass in most situations. IMO, of course. It didn't start out as a guitar project, but ineveitably it went that way and I tried allsorts...
A hot dog is still a hot dog, no matter how much mustard you put on it. It's still a hot dog if you bury it in chilli sauce or tzatziki... Perhaps the question is "should I try another brand of hot dog? What do I want this hot dog to taste like anyway?"
Hungry now.
Still, I recently spent an evening re-amping a TS808 clone with seven different flavours of opamp and didn't experience any kind of epiphany whatsoever (just odd harmonics, mainly). So I guess I just don't get it. :icon_lol:
Quote from: gritz on June 15, 2013, 03:11:52 PM
re the NE570 / 571 and beyond. A 2:1 compression / expansion ratio isn't going to get us very far at all.
Nope. That was one of the issues - needing far more companding. Hacking the 571 got a lot more compressed than the stock 2:1, and in the end it was complicated and only used the VCA. And it still needed more work.
Still, it did make the problem smaller. :icon_lol:
QuoteAs a very clunky comparison the Tubescreamer uses a maximum of over 40dB of gain to send the signal past that "crossover kink" for a reasonable duration. Heeping a signal range of 40dB (or more likely 60+dB) within the sexy part of a transfer function will require something far smarter - maybe based on THAT 218x VCAs and THAT 2252 RMS detectors (with the adaptive capacitor time constant hack - details available on the THAT website. They're not the easiest chips in the world to bradboard though - simply because they're so sensitive.
Actually, I think the problem is not so much squeezing the signal range into the sensitive area of a soft clipping curve. I believe it's the practicalities of making a compressor and expander work quietly and track well enough through the whole process, including with time constants. I messed with the THAT chips a bit for another project, and they are fairly complex to work with.
QuoteI monkeyed with companding as part of research into harmonic excitement - feeding a compressed signal into a bendy transfer function in order to create a (hopefully) constantish level of harmonic content that could be eq'd, expanded and then mixed back in. Great for adding a bit of shimmer to certain sounds, but imo not stellar with guitar or bass in most situations. IMO, of course.
On the other hand, I think most guitarists aren't particularly wedded to getting a no-compression result. Real distortion heads LIKE the "sustain" that they get from just smashing the amplified signal through a clipper. It may be that a more usable compander would have non-matching compression and expansion.
It's one of those areas that are ripe for experimentation. :icon_biggrin:
QuoteStill, I recently spent an evening re-amping a TS808 clone with seven different flavours of opamp and didn't experience any kind of epiphany whatsoever (just odd harmonics, mainly). So I guess I just don't get it. :icon_lol:
The company I work for did that with an audience of pro guitarists, then made the vote for which opamp they liked best - but blind, not knowing what opamp was what. The result was that no one opamp, including the JRC4558 and a video opamp as a ringer, had any particular advantage. I think you just reconfirmed our experiment. Not that True Believers would believe it, of course.
"But Duuuude... it hasn't got a 4558 in it - it can't be any good!"
"OK, I'll put a 4558 in it. That'll be $20 extra."
Kerching!
I know that we're all about analog stuff in this particular forum, but it seems to me that if one wanted to experiment with this type of stuff there are a number of decent wave shaping plugins out there. If you've got a decent computer you'd ought to be able to experiment with everything discussed in this thread without all of the fiddling to get the analog circuit to behave as intended. Once you find a combination of digital modules which make a sound that you like, you can look into making it happen on the breadboard, and then...
I built some JS effects for Reaper specifically for the purpose of playing with these sorts of things. I'm hack, of course, so they are only kind of half assed. They don't do proper oversampling, and I don't think that the "diode curves" are technically mathematically accurate. One of them, though, allows you to adjust the curve. I used gain = 1/(x^n), where n is the input and and you can set x as you please. I would be delighted if it actually did anybody any good.
My First JS's (http://forum.%^&*os.com/showthread.php?t=117293)
And of course if anybody can see a major flaw in the program, or offer suggestions for improvements, I'd love to hear it!
Quote from: R.G. on June 14, 2013, 08:06:37 PM
Quote from: tca on June 14, 2013, 06:53:04 PM
What I think is missing is a dynamical way of controlling gain, compression or expanding the signal to keep it in that good region for long enough time.
Hmmm. Almost sounds like a case for companding before and after distortion, doesn't it?
Actually when I wrote that I was thinking also (and mainly) to move the relative position of the knee voltage, using a peak detector and getting some kind of duty cycle distortion. Instead of manipulating the input signal amplitude (by companding) to alter the clipping level based on that amplitude value.
Quote from: R.G. on June 15, 2013, 03:25:22 PM
On the other hand, I think most guitarists aren't particularly wedded to getting a no-compression result. Real distortion heads LIKE the "sustain" that they get from just smashing the amplified signal through a clipper. It may be that a more usable compander would have non-matching compression and expansion.
It's one of those areas that are ripe for experimentation. :icon_biggrin:
Yes, Definitely! IMO some form of dynamic range compression is generally desirable - if only so we can make ourselves heard over the drummer :icon_lol: but many guitarists eschew "transparent" compression (the sort that doesn't leak transients or apply tonal variation or harmonic content) because
it doesn't sound like it's doing anything. So we tend to desire "something" to give us the impression of dynamics when the actual signal has a very squashed dynamic range.
Putting on my synth hat for a moment (if only to cover my bald head) I have to say that static waveforms are deadly dull and don't cut it in a mix*, which is why we apply envelopes, filters and sometimes distortion and all manner of other whatnot.
* The one exception to this is sub-bass which - while enveloped in an on / off manner - tends just to be a sine. But it's more "felt" than heard. A buttress, if you will - rather than a timbre in it's own right.
So, progress. But how to accentuate the good and toss out the bad? And how to decide what is good and bad in the first place? We're allowed to struggle here, because I think that the whole concept of timbre in a psychoacoustic sense (i.e. how the human brain perceives the messages that the ear sends it when confronted by musical noises) is still very much open to speculation. A lot of reseach
suggests that identification of a particular instrument relies heavily on the note onset and if that note onset is sawn off then test subjects find it harder to identify an instrument from the sustain portion of a note alone. However, as musos we treat this stuff as a way of life and perhaps are more attuned to those little differences. Or more subject to the power of suggestion. Or both. A quick story:
Years ago I swore I could hear even order harmonics burping out of my Danelectro Fabtone distortion pedal. I thought this absolutely couldn't be the case, because I knew the gain part of the pedal was just a couple of opamp gain stages followed by a symmetrical diode clip to ground. So I ignored it, but it just kept nagging at me. Eventually I attacked it with the spectrum analyser - and those even order harmonics were indeed there. Puzzled, I pulled the pedal apart, attacked it with the scope and found that when your average bipolar
output opamp (i.e. almost every opamp we use in pedal land) rail-to-rail clips, it gets about half a volt closer to V+ than to V- and that's enough to produce audibe even order harmonics - even when the asymetric swing is brutally clipped by a pair of silicon diodes afterwards. I pulled out the opamp (a JRC4558D) and socketed, so I could try some of the other usual suspects (a TL072, LM833, NE5532, OP275, etc...). Not a blind bit of difference between 'em in this most testing application- not even in the frankly woeful noise floor.
This is why I have a (possibly unreasonably) low cork - sniffing tolerance. If (for instance) two opamps sound vastly different in circuit, then either:
1) at least one of them is complaining about
something and research should be aimed at
why in order to extract maximum goodness, rather than just crediting Mr Mojo, or:
2) It's a product of of a flawed test, or one's own imagination.
I did however find that the rather cruel addition of a 4k7 resistor between clipping opamp output and V+ gave an audible (and measureable) increase in even order products (and was quicker than lowering Vref). It's still there - and it's still one of my go-to pedals, having had a bit of tuning to the frequency response of various parts.
Sorry, went a bit off-topic there...
@ ashcat_lt: I just saw your post as I hit "submit". I agree that realtime approximations on computer are worth far more than guesswork and static spice-type stuff for getting a feel of the problem - I use Bidule a lot for applying maths to audio in real time. I haven't done Reaper in a long while and I'm rather up to my @ss in alligators at the moment, so I can't promise I'll be able to take a look any time soon, but I would encourage anyone with a computer to take a lead from the approach of ashcat_lt and make some noise / bust some myths. :)
Quote from: R O Tiree on June 15, 2013, 09:02:44 AM
Quote from: Thecomedian on June 14, 2013, 08:27:56 PM
EQ stack pre-distortion = selective band distortion, more or less distortion for frequency range.
possibly.
Like the RAT's reduction of bass in the distortion cell? I suspect that this was designed to try to stop the bass strings swamping treble freqs as the chord dies away some seconds after pick-strike.
As to the compression inate to "classic" distortion circuits, how about an envelope detector at the input to alter the gain of the output buffer/gain recovery stage? You would get the gritty sound we all know and love, but playing dynamics would be in there as well...
spitballing off that, what about a pass filter to ground that is kept closed during the attack, and as decay sets in, the voltage level drops, which starts allowing more and more bleed-off of the lower frequencies, for a dynamic reduction in lows during decay while mid/high retain full strength? I guess like a gate?
Quote from: Thecomedian on June 15, 2013, 08:08:23 PM
Quote from: R O Tiree on June 15, 2013, 09:02:44 AM
Quote from: Thecomedian on June 14, 2013, 08:27:56 PM
EQ stack pre-distortion = selective band distortion, more or less distortion for frequency range.
possibly.
Like the RAT's reduction of bass in the distortion cell? I suspect that this was designed to try to stop the bass strings swamping treble freqs as the chord dies away some seconds after pick-strike.
As to the compression inate to "classic" distortion circuits, how about an envelope detector at the input to alter the gain of the output buffer/gain recovery stage? You would get the gritty sound we all know and love, but playing dynamics would be in there as well...
spitballing off that, what about a pass filter to ground that is kept closed during the attack, and as decay sets in, the voltage level drops, which starts allowing more and more bleed-off of the lower frequencies, for a dynamic reduction in lows during decay while mid/high retain full strength? I guess like a gate?
Funnily enough, I'm working on noise reduction (in software) at the moment - which is pretty much the opposite of what we're talking about here - i.e. I'm
expanding the dynamic range of the signal before the distortion stages. As the signal drops below a certain level, lows below about 500Hz and highs above about 1000Hz are progressively rolled off, then the signal is "faded out"as it drops further. Because software amp sims don't create noise (in the form of hiss and hum) of their own, but simply amplify what you feed them, it works well and is pretty transparent, so that suggests that "going the other way" - i.e. compressing the signal heavily before the non-linear stage might not give any Holy Grail moments.
If the amp sim in question is prone to flab then your take on rolling off the bass as the input signal drops cleans it up nicely, but tbh it's just as effective to filter the lows out the old fashioned way (with a capacitor and resistor) and it doesn't hugely affect the note attack, because that contains a high proportion of high frequencies anyway.
fig.2 (http://www.pat2pdf.org/patents/pat4899115.pdf) read page 8 (us-pat.#4899115)...
other food for thought... (http://www.pat2pdf.org/patents/pat4571548.pdf) (us-pat.#4571548)
I was imagining some diode or biased transistor that keeps the door closed if there's enough voltage, but then opens up more and more to a "lows-to-ground sink" as voltage starts dropping.
Quote from: Thecomedian on June 16, 2013, 03:51:07 AM
I was imagining some diode or biased transistor that keeps the door closed if there's enough voltage, but then opens up more and more to a "lows-to-ground sink" as voltage starts dropping.
as opposed to the "series-diodes simple noisegate-thingy"?
Quote from: puretube on June 16, 2013, 03:57:28 AM
Quote from: Thecomedian on June 16, 2013, 03:51:07 AM
I was imagining some diode or biased transistor that keeps the door closed if there's enough voltage, but then opens up more and more to a "lows-to-ground sink" as voltage starts dropping.
as opposed to the "series-diodes simple noisegate-thingy"?
I have honestly no idea how either would work, but I feel like using a transistor would allow much finer control. They are essentially like auto-potentiometers.
I have to withhold saying yes or no to either until I probe the limits of both designs. ;D
edit:
a glance at what noise gates are: the support circuitry in the noise gate with the IC probably simplify the process of frequency selectivity and bleeding. It would take a fair number of support components to build an analogous discreet transistor-based device that behaves similarly.
Quote from: puretube on June 16, 2013, 03:57:28 AM
Quote from: Thecomedian on June 16, 2013, 03:51:07 AM
I was imagining some diode or biased transistor that keeps the door closed if there's enough voltage, but then opens up more and more to a "lows-to-ground sink" as voltage starts dropping.
as opposed to the "series-diodes simple noisegate-thingy"?
using this snippet (http://www.diystompboxes.com/smfforum/index.php?topic=102821.msg913643#msg913643) as a sensor,
to find out where/when the desired low-voltage -region is situated...
^ Or something like this?!?
(http://diale.org/img/diode_triode.png)
R.G.
I seem to remember you posted a circuit fragment as a kind of challenge to this forum some years ago.
It was a limiter circuit from a Vox solid state amp IIRC.
I can seem to find it.
It was clever simple circuit.
Yeah, it was the limiter out of the Thomas Vox "big head" amplifers, notably the Royal Guardsman and Beatle. It had a pair of diodes to do the clipping, each with something like 100R in series with it, but biased with a variable DC current through a couple of other resistors. By adjusting the DC in the string, you moved the limiting point to let the signal be bigger or smaller.
Thomas used it to keep from blowing the output transistors, I think. But it was useful as it added a soft clipping to a solid state power amp that kept it from doing the razor-edged clipping that most SS amps of the era did.
I've done something similar with both diode-connected MOSFETs and a resistor-diode ladder setup to give a broadly curved transfer function to a power amp.
It's hard to get this right in a 9V power supply just because the voltages and signal levels are so low. 'S why I'm enjoying reading the other travelers on the trail. :)
Quote from: Gus on June 16, 2013, 02:21:03 PM
R.G.
I seem to remember you posted a circuit fragment as a kind of challenge to this forum some years ago.
It was a limiter circuit from a Vox solid state amp IIRC.
I can seem to find it.
It was clever simple circuit.
This one?
(http://www.premierguitar.com/education/images/pic_200709_techviews_1.gif)
Ref.: http://www.premierguitar.com/Magazine/Issue/2007/Sep/Adjustable_Clipping_in_VOX_Amps.aspx
How many people looked at that pic and thought, "J201? Where is it?" and then face-palm, "Junction 201"
I've heard a bit about the Vox limiter. First time I've seen it though. That's a really clever setup. It looks like the negative peaks are limited at the same level though? The bias voltage only goes to the one diode... Wait, nevermind, I see it.
Since this is moving to distortion methods in general, I have another thing to add. What if we took that limit level pot and replaced it with... let's say an LDR and then run an LFO through the corresponding LED? For a somole version, just two diodes clipping to a reference voltage that is modulated by an LFO. I'm sure that has been done before? I imagine that it might sound a bit like PWM?
Quote from: R.G. on June 16, 2013, 02:36:32 PMIt's hard to get this right in a 9V power supply just because the voltages and signal levels are so low. 'S why I'm enjoying reading the other travelers on the trail. :)
What if you replaced r257 with a large value cap, so that node becomes signal ground but can still be biased with enough dc to keep the diodes almost open?
J
edit, nevermind. I didn't pay close enough attention to the direction of the diodes...
Has anyone adjusted this for 12v? Id like to play with it in front of a j201.
I can go 15v if necessary. What transistors would work?
http://www.premierguitar.com/education/images/pic_200709_techviews_1.gif
perhaps this could be of interest?
http://www.aronnelson.com/gallery/main.php?g2_view=core.DownloadItem&g2_itemId=50220&g2_serialNumber=1
its something Im working on and have not built yet, just simmed. its in Another thread that gets next to no attention due to not precenting a finished design ( poweramp for guitar concept)
note that allthou the combined gain is 40db(voltagegain 100) the output doesnt flat top untill the signal Before the limiter is allready clipped
blue trace is input, green is just after the first opamp and red is output.
my intent is to replace the second opamp with a Power ic such as lm1875 or similar ( a bridged lm1875 amp actually ~50watt at +-15volt)
any thoughts
Johan