Guitar one octave Up

In response to all:

Thanks for the ideas. Some of them are much too complicated or expensive for the scope I was aiming at.

The simple techniques with spurious frequencys are already implemented, and the circuits are available on the web, so I don't think I would try reinventing the wheel.

I was just trying to implement my original idea, but it doesn't seem to be an easy task and I don't think it would be reliable in the end.

So, could I arrive to the conclusion that getting a 2nd order harmonic from an audio source is not a trivial task (with as little distortion as possible)?

In conventional 12 string tuning, the bottom 4 pairs are an octave apart, but the G is then pretty much at the tension limit for current string technology, so the higher B and E are instead tuned in unison.

But thanks for the 12 string reference - that's the ONLY way to get the OP's original request for an analog solution. All the other analog ideas presented here are reasonable until you factor in the OP's "guitar" reference, which absolutely demands either a 12 string guitar or a DSP-based pitch-shifter (there are lots around, but the technology is way beyond an amateur DIY project).

Cheers, Tony

"Jim Thompson" wrote in message news: snipped-for-privacy@4ax.com...

See alt.binaries.schematics.electronic. I just posted a schematic for a guitar pedal called an "Octave" made by Boss. I had this one. It works pretty well.

Bob

I think the best solution would be to do what another poster suggested, use an A2D on a processor, sample into a buffer and play the buffer back through a D2A at twice the sampled rate. This would of cource last 1/2 the time, so you would need to 'repeat' sections. To get the distortion down you'd choose zero crossing points in the waveform and repeat small sections. It's going to add some distortion but that's the best I can come up with.

Mark.

That sounds a neat way of doing it, I suggested repeating blocks between zero crossing points but your odd/even sample approach may be better to reduce overall distortion. Choosing the sample repeating block size could be a parameterisable thing, I think that will be the difficult bit...

Mark.

On a sunny day (Sun, 28 Mar 2010 19:05:29 +0100) it happened "markp" wrote in :

I think: audio -> ADC -> n byte audio sample -> fft -> modify spectral components ->

reverse fft -> DAC -> audio

A bit like I do here for video:

Interesting. However it's the time taken to do this that might kill it. There can't be too much delay from signal in to signal out, so rules out a micro. I like your approach though...

Mark.

I did try this on my own, and achieved results that a guitarist friend of mine told me resembled those of some commercial products:

Precision full wave rectifier. Use a DC blocking capacitor and the following resistor to ground afterwards.

A precision fullwave rectifier is an op-amp circuit that I expect to be easy enough to find on the web.

The main faults here:

(I base this on when I was "most-into" guitar effects in the early

- Don Klipstein ( snipped-for-privacy@misty.com)

ALL the (many) solutions that work use a micro of some sort. You play back short segments of recorded signal at 2x speed, each about as long as the wavelength of the common fundamental (so they overlap), choosing the transitions between segments where the signal value and at least the first few derivatives match. A few ms of average delay is never a problem (same as standing a few feet further from the speaker). Most modern multi-fx boxes have decent pitch-shift algorithms along these lines, and the specialist ones like Digitech, TC Helicon etc go to far greater lengths to make it work better.

Tony

Indeed, the A2D and D2A approach only adds a few ms, but taking the fft, changing co-efficients and taking the inverse fft is going to take somewhat longer than that with a micro...

Mark.

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I just had an interesting idea. Since we are trying to double the frequencies, the Walsh waveforms would also work.

Basically, this makes all the constants in the FFT -1 or +1 no sin() and cos() are needed.

You also need to avoid jumps where one FFT ends and the next starts.

I have an idea that is all analog and just reachable for DIY. I need to=20 set up a simulation and see if i can get it to do what i think it will.=20 If it works the way i think it might, it will do a real neat octave = trick. I'll be back when i get my SPICE simulation done.

You can certainly double the frequency of a single sinusoid with a multiplier, but you'll get a real mess with music and its chords. ...Jim Thompson

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I have something rather different in mind. I need (multiply tuned) tank=20 circuits in the megahertz region and other kinds of circuits.

OK :-)

I've done sub-bass stuff for boom boxes, by using 1/2 octave filters at the low end, digital DIV2, then reconstruct the envelope. Sounds good even on Souza, and classical stuff, particularly Beethoven :-)

If you've ever done Bobby McGee's in Phoenix, that's my stuff.) ...Jim Thompson

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