Hack Audio

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A01=Eric Tarr
audio engineering education
audio signal processing
Author_Eric Tarr
Band Stop Filters
block
buffer
Category=UYU
comb
Comb Filter
cutoff
DC Offset
delay
Delay Blocks
Delay Buffer
Delay Lines
diagram
digital audio effects
DR Processor
eq_bestseller
eq_computing
eq_isMigrated=1
eq_isMigrated=2
eq_nobargain
eq_non-fiction
Fir Filter
Fractional Delay
frequency
IIR System
Impulse Response
Input Signal
lines
Magnitude Response
MATLAB programming techniques
Modulated Delay
Modulated Delay Effects
Modulated Time Delay
output
Parallel Delay Line
Pink Noise
practical audio DSP coding projects
RMS Amplitude
signal
Signal's Amplitude Envelope
Sine Wave LFO
Sine Wave Signal
Sine Wave Test Signal
spectral visualisation
time domain analysis
Triangle Wave
Unipolar Signal

Product details

  • ISBN 9781138497542
  • Weight: 1016g
  • Dimensions: 191 x 235mm
  • Publication Date: 20 Jun 2018
  • Publisher: Taylor & Francis Ltd
  • Publication City/Country: GB
  • Product Form: Hardback
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Computers are at the center of almost everything related to audio. Whether for synthesis in music production, recording in the studio, or mixing in live sound, the computer plays an essential part. Audio effects plug-ins and virtual instruments are implemented as software computer code. Music apps are computer programs run on a mobile device. All these tools are created by programming a computer.

Hack Audio: An Introduction to Computer Programming and Digital Signal Processing in MATLAB provides an introduction for musicians and audio engineers interested in computer programming. It is intended for a range of readers including those with years of programming experience and those ready to write their first line of code. In the book, computer programming is used to create audio effects using digital signal processing. By the end of the book, readers implement the following effects: signal gain change, digital summing, tremolo, auto-pan, mid/side processing, stereo widening, distortion, echo, filtering, equalization, multi-band processing, vibrato, chorus, flanger, phaser, pitch shifter, auto-wah, convolution and algorithmic reverb, vocoder, transient designer, compressor, expander, and de-esser.

Throughout the book, several types of test signals are synthesized, including: sine wave, square wave, sawtooth wave, triangle wave, impulse train, white noise, and pink noise. Common visualizations for signals and audio effects are created including: waveform, characteristic curve, goniometer, impulse response, step response, frequency spectrum, and spectrogram. In total, over 200 examples are provided with completed code demonstrations.

Eric Tarr is an assistant professor of Audio Engineering Technology at Belmont University in Nashville, TN. He teaches classes on digital audio, computer programming, signal processing and analysis. He received a Ph.D., M.S., and B.S. in Electrical and Computer Engineering from the Ohio State University. He received a B.A in Mathematics and a minor in Music from Capital University in Columbus, OH. His work has spanned across the topics of speech signal processing, musical robotics, sound spatialization, acoustic and electronic system modeling, hearing loss, perception and cognition. He has published articles in the Journal of the Acoustical Society of America, Journal of Speech, Language, and Hearing Research, International Journal of Audiology, and Mechanical Engineering Research. He is a member of the Audio Engineering Society. In 2015, Dr. Tarr received the Gibson Foundation Les Paul Music Innovation Award as the principal investigator of a research grant on "Blockchain Technology in the Music Industry."

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