Trent's Guitar Effects Processor
Wicked-awesome guitar effects by Trent Rolf
I started graduate school at the University of Utah in January of '08. After going back and forth on what I want to study, I finally settled on a focus in Digital Signal Processing. Specifically, audio DSP is what I am most interested in.
At the time I worked for Harman Music Group in Sandy, UT, and I was surrounded by some hard-core DSP developers. They create algorithms for reverberation, guitar effects, live sound management, etc. This is something that really interested me and might be something I want to do in the future.
For a final project in my DSP class (ECE 5580) I chose to write my own version of some of these algorithms. I didn't have time to create as many as I wanted, but it's not a bad start.
The target processor for this project was the Texas Instruments TMS320C6713 floating-point digital signal processor. I carried out the project on the TMS3206713B DSK development board using TI's Code Composer Studio and MATLAB.
This web page is not part of my project, but since I couldn't keep the development kit I decided to capture some audio so I can remember what I accomplished. The page is mostly just audio clips, but I will give a brief explanation before each clip.
Also, do me a favor and listen to these on a good set of speakers or headphones... no crappy laptop speakers allowed!
Original Track - Steve playing Pearl Jam
Since I am not a guitar dude, I had my friend Steve Noall record some dry guitar tracks for me (thanks, Steve!). This first clip is Steve playing some Pearl Jam with no audio processing.
Distortion - Hard Clipping
This first algorithm is the simplest. It simulates classic distortion where the guitar amplifier gain is turned up so high that the signal gets "clipped" at the power supply rails. Imagine a sine wave with the peaks sliced off and you'll get the picture.
FIR Delay
In this clip I apply a finite impulse response (FIR) comb filter, or a delay. There is no feedback path, only a single delay that is attenuated by some constant. So you hear the orignal track shadowed by a quieter version of the same signal. The delay and attenuation are adjustable in real time, but I can't really show you that here.
IIR Echo
The infinite impulse response (IIR) comb filter is more like a real-world echo. This time there is a feedback path that infinitely adds old attenuated copies of the audio sample to itself. So instead of hearing just one delayed copy of the track, you hear an infinite number of copies.
Original Track 2 - Steve playing Steve
Here is another dry track. This is Steve's own composition; I chose it because it has sustained chords that allow you to hear these next effects easier.
Tremolo
Tremolo is amplitude modulation usually driven by a sine wave, meaning that the signal is multiplied by some gain that is constantly changing. In this case, the gain is changing at 7 Hz which according to my DSP book is the most "pleasing to the ear."
Flanger
Okay, here is where it gets a little more complicated. A flanger is much like the FIR delay, only the delay time is constantly changing. What does changing delay sound like? Pitch shifting. This is kind of like the doppler effect when you hear the pitch change on a car horn as it drives by. With delay changing slowly (known as low frequency oscillation or LFO) and only by 20-30 milliseconds, you get this sweet effect.
Chorus
A chorus simulates multiple voices. For instance, if you have a group of three singers singing the exact same tune, they will always sing with slight variation to each other. To accomplish this in my algorithm, I used three flangers summed together each with independent LFO frequencies.