11. Sample-based Synthesis#
This chapter begins the second part of this book. In the first part (Chapters 1 through 10), we studied the fundamental principles of computer music in depth: how sound becomes digital audio, how to synthesize it, how to analyze it in the frequency domain, and how to process it with filters and frames. Now we shift gears. In the second part, we will survey a breadth of computer music topics that build on these principles. We won’t study each topic in as much depth as the fundamentals. Instead, the goal is to expose you to many topics, so that you can find the ones that interest you most and go deeper through self-study.
For our first topic, we will examine sampling and sample-based synthesis.
“Sampling” is overloaded
So far in this book, “sampling” has been shorthand for digital sampling, the step of analog-to-digital conversion that measures a signal \(f_s\) times per second (Chapter 7). Likewise, a “sample” has referred to a single audio sample \(x[n]\).
In this chapter, “sampling” takes on its colloquial meaning: lifting a clip of audio from one source and reusing it in another context, often with some manipulation. Similarly, a “sample” now refers to a clip of audio, not an individual audio sample. When the distinction matters, we will write “audio sample” for \(x[n]\).
We will cover two use cases for sampling:
Sampling and sequencing: extracting samples from recordings and playing them back in new orders and with new timing.
Sample-based synthesis: manipulating pre-recorded samples on the fly, for example by repitching them, to create instruments that can be played with different expressive parameters.