I've long been in the process of setting up a live environment to play guitar through using pd. I've got my audio interface, the presonus firebox (still saving up for a nice laptop- anyone have experience with adkproaudio laptops?), I've made table-based loopers that work well and conform to a given beat, and I've got a decent distortion using hardoff's tangent-based saturator idea, and I've done a few other effects I'm pretty happy with. But the one effect I've been swooning over, and that I cannot wrap my head around, is that of an octave shift that works well in real-time.

I've tried using hilbert~, which works well for vibrato, but nothing close to an octave down, and I've tried numerous times to create a pitch shifter using multiple vd~'s but those, and the versions I've tried made by others, always sort of sound mediocre, and have a sort of phasing to them, which results in a muffled sound with none of the sort of "pluck" you'd want in emulating a bass guitar.

As far as I can tell the last option to try is using rfft and such, but I can't quite understand it enough to do anything. I can follow the resynthesis example pretty well, as far as I can tell it uses a table lookup to modify amplitudes of certain frequencies. What I don't quite understand is what is actively indexing that table. Is the sample # for the fft window considered the index? And then, only half-way understanding that, I'm completely baffled by the phase-vocoder example (which is designed for time-stretching rather than pitch shifting, so I'd have to know how to adapt it) which is what I assume I'd need to use to do something like this.

So I guess what I'm curious about is whether the phase-vocoder is what I'm looking for, if there are any other options to consider, and whether there are any pd-based tutorials for making phase-vocoder pitch-shifters, or if anyone has made patches for such a thing with good results. I've got msp's book but the fourier analysis chapter is for the most part above my head; I cannot for the life of me figure out how to apply any of it in making a functional pitch-shifter for live input. Thanks.

Hi. I'm still kind of a novice at pd, so what I'm trying to might not be a good idea. It doesn't seem to be working any how.

Anyways, I was trying to make a distortion type deal where you select a window in the signal, say .5 to .75 of the absolute value of the signal, or whatever you like, and have it so the signal below that window (say 0 to .5) isn't reduced in amplitude, the signal above that window (.75 to 1) is reduced by a factor of your choice, I'll call it a, a value between 1 and 0, and the signal within the window ramps steadily from no reduction to reduction by a factor of a, so it's smooth.

What I've been doing for this is multiplying the whole signal by a clipped version of the absolute value of the signal that's multiplied and added to such that instead of going from .5 to .75 it goes from 1 to a. So, anything below .5 is multiplied by 1, anything above .75 is multiplied by a, and everything in between is multiplied by something between 1 and a. I thought this would produce a smooth sort of distortion, depending on how much the amplitude is reduced by and the width of the window, but no matter what I do I get a harsher distortion, and there are little rectangular cuts in the peaks that shouldn't be there, unless my logic about all of this is wrong.

This is roughly what's going on.

Signal
|
abs~
|
clip~ .5 .75
|
-~ .5 ( 0 to .25)
|
/~ .25 (0 to 1)
|
*~ -(1 - a) (given a reduction factor of .6 this would be 0 to -.4)
|
+~ 1

And that would result in a range of 1 to .6. And I multiply that by the original signal. I'm just kinda messing around with pd trying to make stuff, so maybe this is just a bad way to do this. Is it just too many operations on the clip to multiply it at the right time?

I can put a patch on this later if you need.

Hi, I've been messing around with making electronic music on the computer for about a year and a half. I started out with project 5, and still use project 5. At first I just used the preset instruments and fooled around for a bit, but I didn't like the cheesey factory settings, which gave everything a out-of-the-box fruityloops feel. Somehow I found out about absynth, and have since put a lot of time into understanding it and learning how to use it to make decent music.

I've grown fairly competent with it, and have a pretty good grasp of how to sculpt the sounds I want with oscillators, envelopes, filters, and lfo's. But I feel somewhat limited, regardless of the fact that I haven't come close to pushing absynth as far as it can go, and I want to open up opportunities for making more complex sounds in different ways. pd seemed the logical choice as it's free, while reaktor isn't.

I'm not a bad mathematician and have a bit of programming experience, so using pd for the basics hasn't been too difficult. I completed most of the tutorial/documentation that was included without too much trouble. I still need to learn a lot more about all the elements I'll need to create to make good synths and whatnot, but pd makes enough sense that I'm pretty sure I'll be able to figure out how to do what I want to do with some study and practice.

My problems are with how to implement pd as a tool to create music. How does one do this? Is it easy to make pd patches into vst's for use with project 5, or say, minihost? If so, how? I don't have any midi controllers and so I want to use pd to create instruments that I can use with the sequencers I'm already familiar with. Or with minihost for when I get a midi controller, so I can quickly switch between a pd patch and other vsts. I understand the basics of programming, but first and foremost I'm just a guy that likes to make music on the computer. Not a programming guru, so I don't really plan on making sequencers or environments, as it seems it would be a lot more difficult than just making synths.

Are there any pd guides out there for people that are predominantly electronic musicians?

I'd appreciate advice from anyone, whether you've been in my shoes or not.

Hey thanks a lot both of you for your replies. Hardoff, I see what you're saying about the signal dropping below the rest. I guess I didn't give the math enough thought.

Obiwannabe I understand most of what you're saying but it might be a bit before I'll be able to apply it. I've just started reading puckette's book, before I mostly just experimented with things. Would the purpose of low-pass filtering be to smooth the signal and yield a less sporadic amplitude? Is that what you mean by averaging? Also your site's a great resource I plan on going through it when I'm done with the book.