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This book offers an introduction to digital signal processing (DSP) with an emphasis on audio signals and computer music. It covers the mathematical foundations of DSP, important DSP theories including sampling, LTI systems, the z-transform, FIR/IIR filters, classic sound synthesis algorithms, various digital effects, topics in time and frequency-domain analysis/synthesis, and associated musical/sound examples. Whenever possible, pictures and graphics are included when presenting DSP concepts of various abstractions. To further facilitate understanding of ideas, a plethora of MATLAB® code examples are provided, allowing the reader tangible means to “connect dots” via mathematics, visuals, as well as aural feedback through synthesis and modulation of sound. This book is designed for both technically and musically inclined readers alike–folks with a common goal of exploring digital signal processing.
Sample Chapter(s)
Chapter 1: Acoustics, Hearing Limitations, and Sampling (5,310 KB)
Contents:
- Acoustics, Hearing Limitations, and Sampling
- Time-Domain Signal Processing I
- Time-Domain Processes II
- Sine Waves
- Linear Time-Invariant Systems
- Frequency Response
- Filters
- Frequency-Domain and the Fourier Transform
- Spectral Analysis, Vocoders, and Other Goodies
Readership: Undergraduate and graduate students, researchers and academics in digital audio and multimedia as well as music.
