I was wondering if anyone knows of a fuzz or distortion pedal that that produces only EVEN ORDER HARMONICS. Like the process used on Aphex Exciters but carried even further?IF not then any ideas on its development would be cool! ::)
You can create even harmonic distortion with ringmodulators and 2 mixers.
This is how it works:
Put 2 sines into a ringmodulator and the output frequency will be the sum and the difference of the sines. If the 2 sines are the same frequency, than the output of the ringmodulator will be 0Hz and twice the original frequency.
So what you do is to put the signal you want to distort into both inputs of a ringmodulator and you get 2th harmonics.
To get 4th, 6th etc... you feedback the output of the ringmodulator back into the input with a mixer before the input. If you put a mixer behind the ringmodulator and mix it with the original signal you have total control over the amount and contend of the even order distortion.
That sounds interesting but wouldnt it only generate octaves? The exciter process is much broader in its spectrum and works like a shelving eq across the harmonic series(somehow filtering the dissonant odd-order tones). I was looking for a full bandwidth solution which would dial out the original signal leaving the distortion (which would have to be very sweet!)
Quote from: Krunchy2 on July 08, 2008, 07:37:20 AM
That sounds interesting but wouldn't it only generate octaves?
Thats what even order harmonics are.
Of cause this circuit produces for every harmonic in the signal a series of even harmonic overtones.
Quote from: Krunchy2 on July 08, 2008, 07:37:20 AM
The exciter process is much broader in its spectrum and works like a shelving eq across the harmonic series(somehow filtering the dissonant odd-order tones).
With the circuit mentioned you can dial in the level of the second harmonic and the level of the 4th, 6th, 8th etc... More ringmodulators and mixers would allow you to mix the 8th harmonic higher in level than the 6th for example.
Quote from: Krunchy2 on July 08, 2008, 07:37:20 AM
I was looking for a full bandwidth solution which would dial out the original signal leaving the distortion (which would have to be very sweet!)
Just turn down the level of the original signal.
I'll see if I can make an example of this circuit with my Clavia micro modular.
Here can you find a picture of the circuit and a soundfile: http://www.sendspace.com/file/wxxhyl (http://www.sendspace.com/file/wxxhyl)
I used a TB-303 into the input of the Clavia, the output of the Clavia into the recorder. This circuit only produces even harmonics (and a bit of aliasing, because its digital) so it sounds quite different than normal distortion.
First you hear the direct sound, folowed by some knob tweeking. As you can hear it can be quite drastic, but also quite subtle.
I leave it up to others to make this into a stompbox.
ahh man i love the micro modular... wish i had one :'(
There is a schematic for a Ringmod type device that does something similar to what you are describing at viva analog.
http://www.lynx.bc.ca/~jc/pedalsHarmonic.html
How about an octaver?
Quote from: soggybag on July 08, 2008, 01:18:21 PM
There is a schematic for a Ringmod type device that does something similar to what you are describing at viva analog.
http://www.lynx.bc.ca/~jc/pedalsHarmonic.html
This thing only produces 2th harmonics. With a mixer at the input and feedbacking the output of the ringmodulator, you can create more even harmonics. See my description earlier. To achieve this, add a pot and a resistors to the schematic.
THX for ideas but theres a couple problems . First the even -order partials are not just octaves -they are the intervals octave,fifth,fourth,major and minor thirds and their octaves etc.. All tones generate even and odd partials-just in varying degrees of amplitude.The even partials are less dissonant and boosting them is appealing to humans.Boosting odd partials fools us into thinking its out of tune or somethings not quite right.Traditional fuzz effect boosts even and odd partials and the solution in the past was to try and eq it somehow to reduce the odd partials(not with much results from my view).
Quote from: Krunchy2 on July 08, 2008, 04:17:36 PM
THX for ideas but theres a couple problems . First the even -order partials are not just octaves -they are the intervals octave,fifth,fourth,major and minor thirds and their octaves etc..
This is not a problem, I was wrong in saying even harmonics are octaves. The circuit mentioned can produce all even harmonics including 5th, Maj3th, 4th etc... (of cause limited by the bandwith of the circuit). It of cause can't produce, for C1 as 1st harmonic, Es1, E1, F1, G1, but that must be evident.
Symmetric distortion produces uneven harmonics, asymmetric distortion produces uneven and even harmonics.
Edit: Uneven harmonics seem to put the sound more forward (in your face sound) and even harmonics seem to put the sound more backward (give air to the sound) in the soundfield.
Hey, you'll have to limit the superior order harmonics... beyond 7th, it's not musical! Even 6th sounds quite bad!
Just build a tube amp! :P
I might be wrong, but I thought that a full-wave rectifier only produced even harmonics. And it pretty much defines the octave-up sound.
The Aphex Aural Exciter uses a one-sided clipping (i.e. one side of the waveform is limited to a certain threshold, while the other side has no limitations). This will produce both even and odd harmonics.
Sean Costello
Quote from: dirk on July 08, 2008, 02:58:45 PM
Quote from: soggybag on July 08, 2008, 01:18:21 PM
There is a schematic for a Ringmod type device that does something similar to what you are describing at viva analog.
http://www.lynx.bc.ca/~jc/pedalsHarmonic.html
This thing only produces 2th harmonics. With a mixer at the input and feedbacking the output of the ringmodulator, you can create more even harmonics. See my description earlier. To achieve this, add a pot and a resistors to the schematic.
we have had many "messy" arguments about this - thanks mainly to the limitations of ascii communication and time to properly express our understanding ... a multiplier will not produce a pure octave when hit by a non-pure waveform, what comes out of a string instrument ... what the HarmonicMultiplier puts out is principally a second harmonic (octave) and a bunch of side terms depending on how impure the waveform is (there is no way around that unless you go artificial-digital reconstruction) why some people like to roll off the treble on their signal before hitting an analogue octaver circuit ...
having said that a multiplier will in general produce a cleaner second harmonic on a pure single tone than other types of "squaring" circuits ... my circuit ends up sounding like a dry Bassman "sort-of" when the mixer is set between 5-20% wet to dry mixing - but mine's not an easy build and the nulling also requires work to get right ... there are simpler pre-nulled multiplying chips (AD533 ?? I forget) that you can pair to a linear mixer to achieve a similar result ... the circuit is a neat tool to have when doing doubled parts on a recording, adding a pleasing difference to a basic sound ...
best ...
Harmonic Percolator IIRC
mac
Quote from: Eb7+9 on July 08, 2008, 07:52:13 PM
Quote from: dirk on July 08, 2008, 02:58:45 PM
Quote from: soggybag on July 08, 2008, 01:18:21 PM
There is a schematic for a Ringmod type device that does something similar to what you are describing at viva analog.
http://www.lynx.bc.ca/~jc/pedalsHarmonic.html
This thing only produces 2th harmonics. With a mixer at the input and feedbacking the output of the ringmodulator, you can create more even harmonics. See my description earlier. To achieve this, add a pot and a resistors to the schematic.
we have had many "messy" arguments about this - thanks mainly to the limitations of ascii communication and time to properly express our understanding ... a multiplier will not produce a pure octave when hit by a non-pure waveform, what comes out of a string instrument ... what the HarmonicMultiplier puts out is principally a second harmonic (octave) and a bunch of side terms depending on how impure the waveform is (there is no way around that unless you go artificial-digital reconstruction) why some people like to roll off the treble on their signal before hitting an analogue octaver circuit ...
I would really like to hear these arguments.
As I understand it every signal is made up from sines (Fourier) and therefore multiplication of a signal with itself will produce for every harmonic in the original signal an octave of that harmonic.
Actually, one of the Aphex Exciters (I forget which model) used this exact approach, feeding in a copy of the input signal to one half of an LM13600, and a second copy to a high-pass filter which then fed the Iabc pin of the 13600. In other words the modulation of the original signal by a copy of itself was restricted to mid-harmonics and above such that one would get doubling and augmentation of harmonic content but not added lower-order harmonics from the fundamental.
Quote from: Mark Hammer on July 09, 2008, 12:42:47 PM
Actually, one of the Aphex Exciters (I forget which model) used this exact approach, feeding in a copy of the input signal to one half of an LM13600, and a second copy to a high-pass filter which then fed the Iabc pin of the 13600. In other words the modulation of the original signal by a copy of itself was restricted to mid-harmonics and above such that one would get doubling and augmentation of harmonic content but not added lower-order harmonics from the fundamental.
This would also act as a "transient exciter," as you are squaring not only the harmonics, but also the amplitude envelope of the signal, which results in a much faster decay.
Sean Costello
Which is precisely the desired result.
Quote from: Mark Hammer on July 09, 2008, 01:37:29 PM
Which is precisely the desired result.
I guess. When I have programmed "harmonic enhancers" (the Aphex patent has lapsed, but the Aural Exciter name is still a trademark), I found that the asymmetric clipping sounded much better than squaring the input signal. You would hear much more "sheen" on cymbals, and no weird staccato transients.
I know that some bass boost algorithms square up the low-frequency filtered signal, but precede the squaring with a compressor, to undo the squaring of the amplitude envelope. A full wave rectifier is nice in that it preserves the signal envelope, but the even order harmonics produce too much of an octave artifact, which is not what you want for psychoacoustic bass boost.
Sean Costello
Quote from: dirk on July 09, 2008, 12:32:29 PM
I would really like to hear these arguments.
As I understand it every signal is made up from sines (Fourier) and therefore multiplication of a signal with itself will produce for every harmonic in the original signal an octave of that harmonic.
this is how I see things ... Fourier analysis strictly applies to periodic signals first of all so the idea of a signal made up of harmonics is a touchy concept as it applies to string derived signals - one should be talking of harmonic bursting, the question is can we really be talking about an infinite sum of sine waves in the context of rapidly changing harmonic bursts ?? ...that's where the discussion almost gets stupid and many people opt for a simplified understanding of "harmonics" ... the modified Fourier theory as it applies to non-periodic signals gets really messy as well and only yields similar results "in the limit" ...
in music signals only the decay portion comes close to being quasi-periodic - so Fourier analysis doesn't apply "idealy" the same way that periodic theory predicts when applied to plucked string signals, esp. near the front end where the periodicity can be conceived as "quite" non-existant depending on how one wants to look at things ... ignoring that part a non-pure signal (ie carrying harmonics) will produce sums and differences of frequencies when going through any non-linear transfer in addition to (self-product) doubled terms ... amongst other things this is why/how you get this one frequency term heading towards DC when bending one note into another (as we do on the guitar) when going through certain non-linear processing blocks like octavers ...
Quote from: dirk on July 09, 2008, 12:32:29 PM
I would really like to hear these arguments.
As I understand it every signal is made up from sines (Fourier) and therefore multiplication of a signal with itself will produce for every harmonic in the original signal an octave of that harmonic.
It's algebra. Imagine that you have an input signal consisting of two sine waves, a and b, with two different frequencies, freq(a) and freq(b). The signal can be represented as
a+b
Multiplying the signal by itself will yield
(a+b)(a+b) = a^2 + 2ab + b^2
a^2 will result in a sine wave of freq(a)+freq(a), which is 2*freq(a), which is an octave above freq(a). It will also create a frequency at freq(a)-freq(a), which is 0 Hz, or DC.
b^2 will result in a sine wave of freq(b)+freq(b), which is 2*freq(ab), which is an octave above freq(b). It will also create a frequency at freq(b)-freq(b), which is 0 Hz, or DC.
2ab will result in a sine wave at freq(a)+freq(b), as well as a sine wave at freq(a)-freq(b). These will not be simple octaves of a or b.
So, it is true that multiplication of a signal by itself will result in an octave of every harmonic in that signal. But, you get a whole mess of other frequencies, as every frequency in the signal is multiplied by every other frequency in the signal. This is why a squaring circuit is not a pitch shifter.
In addition, the amplitude envelope is squared by itself. Imagine that you just have a single sine wave, of amplitude A. When the amplitude is 1.0, the squared amplitude is also 1.0. If the input amplitude is 0.5, the squared output amplitude is 0.25. If the input amplitude is -0.5, assuming a bipolar signal, the output amplitude will be 0.25. This ends up resulting in transients that decay away much quicker, as well as a unipolar output signal (i.e. >0 for all input signals). Such behaviour is useful in envelope detection, and is the S in RMS (root mean squared) which is used in compressors, VU meters, and other places.
Sean Costello
Thanks Sean Costello, that makes perfect sense.
Of topic: As I understand Fourier, it will apply to every real world signal. Because every real world signal is bandlimited, you have a finite amount of sines that would make up the signal. But I'm not a mathematician, so I could be wrong.
Quote from: dirk on July 10, 2008, 03:11:50 AM
Thanks Sean Costello, that makes perfect sense.
Thanks! I am not that good at math, but I was able to dig out the old high-school algebra for that one.
Quote
Of topic: As I understand Fourier, it will apply to every real world signal. Because every real world signal is bandlimited, you have a finite amount of sines that would make up the signal. But I'm not a mathematician, so I could be wrong.
It is actually the opposite: real world signals are NOT bandlimited. In many of the signals, the higher order harmonics end up having insignificant amplitudes, but they are still there. In many cases, the signals above our high frequency thresholds (20+ KHz for younguns, 17 Khz and lower as you reach adulthood) have significant energy. That is why A/D convertors have brick-wall lowpass filters, so the frequencies outside of the audible range do not alias down into the area where we can here them.
Digital signals are bandlimited - they have to be - so they can be represented by a finite number of sines. Psychoacoustic data compression works by throwing out those frequencies that will not be perceived, due to masking and other perceptual pheonomena.
With regards to multiplying signals by themselves, if a harmonic has insignificant amplitude, its products will also be insignificant. It is also worth noting that multiplying a signal by itself will double the bandwidth of the output (i.e. the highest frequency of the output will be twice the highest frequency of the input). No big deal in the analog domain, but a big problem in the digital domain, as this will cause aliasing.
Sean Costello
Quote from: dirk on July 08, 2008, 09:16:56 AM
Quote from: Krunchy2 on July 08, 2008, 07:37:20 AM
That sounds interesting but wouldn't it only generate octaves?
Thats what even order harmonics are.
Of cause this circuit produces for every harmonic in the signal a series of even harmonic overtones.
Quote from: Krunchy2 on July 08, 2008, 07:37:20 AM
The exciter process is much broader in its spectrum and works like a shelving eq across the harmonic series(somehow filtering the dissonant odd-order tones).
With the circuit mentioned you can dial in the level of the second harmonic and the level of the 4th, 6th, 8th etc... More ringmodulators and mixers would allow you to mix the 8th harmonic higher in level than the 6th for example.
Quote from: Krunchy2 on July 08, 2008, 07:37:20 AM
I was looking for a full bandwidth solution which would dial out the original signal leaving the distortion (which would have to be very sweet!)
Just turn down the level of the original signal.
I'll see if I can make an example of this circuit with my Clavia micro modular.
I'm pretty sure that only perfect squares are octaves. 6th harmonic is an octave of the 3rd.
edit- nevermind, someone already caught this.
Quote from: SeanCostello on July 10, 2008, 02:19:11 PM
Quote from: dirk on July 10, 2008, 03:11:50 AM
Of topic: As I understand Fourier, it will apply to every real world signal. Because every real world signal is bandlimited, you have a finite amount of sines that would make up the signal. But I'm not a mathematician, so I could be wrong.
It is actually the opposite: real world signals are NOT bandlimited. In many of the signals, the higher order harmonics end up having insignificant amplitudes, but they are still there.
I don't agree with this.
Every real world signal is bandlimited and has a signal to noise ratio, I was not complete before. If real world signals are not bandlimited, then they would have infinite energy (an endless amount of frequencies with insignificant amplitudes, still adds up to infinity). That is simply not possible.
And therefore Fourier applies to every real world signal.
Here you can find a table that shows what type of Fourier transform applies to what type of signal.
http://en.wikipedia.org/wiki/Fourier_analysis#Discrete_Fourier_transform_.28DFT.29 (http://en.wikipedia.org/wiki/Fourier_analysis#Discrete_Fourier_transform_.28DFT.29)
Quote from: Krunchy2 on July 08, 2008, 07:37:20 AM
That sounds interesting but wouldnt it only generate octaves? The exciter process is much broader in its spectrum and works like a shelving eq across the harmonic series(somehow filtering the dissonant odd-order tones). I was looking for a full bandwidth solution which would dial out the original signal leaving the distortion (which would have to be very sweet!)
Actually, there are three of those. They are the JFET Doubler, the MOS Doubler and the Mu Doubler, both at GEO. And they work exactly as you describe. They take a signal, use a phase inverter to generate a second, out of phase signal, then run the two split signals into a differential amplifier.
But the differential amplifier also adds the two amplified, out of phase signals back together at the output by tying the drains of a diffamp pair of FETs together. This cancels out the original signal and leaves only even order harmonics, which all reinforce. The original signal and odd-order harmonics cancel. It worked the first time I tried it. JFETs and MOSFETs have a square-law nature, so the result is primarily second harmonic only.
So why don't you see this more? Because the linear characteristics of JFETs and MOSFETs are quirky, and it's difficult to get off-the-shelf devices to work. There is nearly always some tinkering needed to get it to cancel properly and have enough output level. I've been working with a guy on one of these recently.
You need a well-matched pair of FETs for the diffamp, which is where the magic happens. That is what generates the second-order distortion. And you need a "dirty" pair, ones that generate a lot of square-law. Nicely behaved linear FETs don't distort much, so they generate only a tiny amount of distortion to use, and then you have to really work at amplifying it back up.
The closest to repeatable is the MOS Doubler, which uses a CD4007 logic chip for its accessible MOSFET devices. They are monolithic (and so well matched inherently) and logic chips (so nobody slaved over a hot display trying to make them linear). There's even enough devices there to make a post amplifier to get more output and a little grit. The complaints I've received, other than it being quirky to set up, are largely that it's either too quiet or not distorted enough, and a little post-clipping enhances things.
So if you want to tinker with an embodiment of your idea, check out the Doublers at GEO.
Quote from: Krunchy2 on July 07, 2008, 02:40:16 PM
I was wondering if anyone knows of a fuzz or distortion pedal that that produces only EVEN ORDER HARMONICS. Like the process used on Aphex Exciters but carried even further?IF not then any ideas on its development would be cool! ::)
The Ampeg Scrambler does a pretty good job at this IMHO. With careful adjustment of the BLEND control, you can get a shimmery, Fender-ampish even-order distortion. Match the differential xstrs and use three germanium diodes in series instead of the stock silicons for a softer sound and better touch sensitivity.
Regards,
Joe
PS: Just found this:
http://electronicdesign.com/Articles/Index.cfm?AD=1&ArticleID=6379
Might work for a FWR-type distortion. Haven't tried it yet, but looks promising!