DSP Functions
Introduction to Algorithm Programming
16-1
Chapter 16
DSP Functions
This chapter explains the DSP functions that can be inserted into the algorithms in the Program
Editor. As you configure each algorithm, the DSP functions you select determine the type of
synthesis you apply to your sounds. Deciding which algorithm to use depends on what you
want to do; there’s no hard and fast rule. If you want to create a classic analog sound, for
example, you’ll choose one of the algorithms containing one or more blocks that can have filter
functions assigned to them. If you want real-time panning effects, choose an algorithm that
includes the PANNER function in the F3 block. Your best approach is to study the algorithm
charts in the Musician’s Reference, and choose the algorithm that includes the functions you want
to work with.
Note that Triple Mode offers even more algorithmic possibilities than those described here; see
Chapter 12 of the Musician’s Reference for details.
Before we get to the explanations of the DSP functions, we’ve included a brief discussion of a
few general concepts of sound synthesis. This should help you understand the workings of the
DSP functions. We’ll refer to these concepts repeatedly as we go along.
Any single sound waveform is composed of numerous sine wave components, each at a
different frequency. These components are called partials. The lowest frequency is perceived by
the ear as the pitch of the sound, and is called the fundamental. The other components are called
harmonics. The relative amplitudes (volume) of each of the partials in a sound determine its
timbre, its most recognizable characteristic. When you think of the difference between the sound
of a piano and a saxophone, you’re thinking about their different timbres. A dull sound has a
strong fundamental and weak harmonics, while a bright sound has strong harmonics.
Sound synthesis can be most simply described as the manipulation of either the amplitude or
phase of one or more of the partials constituting a sound. The K2661’s various DSP functions
give you a variety of methods for manipulating those partials. We’ve grouped our explanations
of the DSP functions according to the types of specialized manipulation they enable you to
perform on a given sound. The categories are as follows:
Introduction to Algorithm Programming
Programming the algorithms is a multi-step process. The first step is selecting an algorithm.
Changing the algorithm of an existing program’s layer is likely to alter the sound of the layer
dramatically. As a rule, then, you won’t want to change a layer’s algorithm unless you’re
building a sound from scratch. Furthermore, when you change a layer’s algorithm, the values
for each of the DSP functions within the algorithm may be set at nonmusical values; you should
lower the K2661’s volume slider before changing algorithms.
Filters Added Waveforms
Equalization (EQ) Nonlinear Functions
Pitch / Amplitude / Pan Position Waveforms with Nonlinear Inputs
Mixers MIxers with Nonlinear Inputs
Waveforms Synchronizing (Hard Sync) Functions
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