book

Hack Audio

Name: Hack Audio
Author: Eric Tarr
ISBN: 9781351018449

by Eric Tarr

June 2018

Intermediate to advanced

492 pages

9h 52m

English

Routledge

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Cover
Half Title
Series Page
Title Page
Copyright Page
Dedication Page
Table of Contents
List of Tables
List of Figures
List of Examples

Acknowledgements
Chapter 1: Introduction
1.1 Introduction: Computer Programming and Digital Signal Processing1.2 The Purpose of This Book1.3 Intended Readers1.4 Topics Covered1.5 Additional Content
Chapter 2: Basics of Programming in MATLAB
2.1 Introduction: Computer Programming in MATLAB2.2 Programming Languages2.3 Executed Commands2.3.1 Error Statements2.4 Mathematics2.4.1 Operators2.4.2 Variables2.5 Data Types2.5.1 Numbers2.5.2 Characters2.5.3 Strings2.6 Arrays2.6.1 Basic Array Creation2.6.2 Plotting Arrays2.7 Mathematical Functions2.7.1 Plotting Mathematical Functions2.8 APPENDIX: Additional Plotting Options2.8.1 Line Specification2.8.2 Axis Labels2.8.3 Figure Title2.8.4 Figure Legend2.8.5 Axis Scale2.8.6 Axis Dimensions2.8.7 Multiple Arrays
Chapter 3: Basics of Audio in MATLAB
3.1 Introduction: Digital Audio Signals3.2 MATLAB Audio Functions3.2.1 audioread3.2.2 sound3.2.3 audiowrite3.2.4 audioinfo3.3 Working with Audio Signals in Arrays3.3.1 Indexing Arrays3.3.2 Array Reversal3.3.3 Additional Methods to Create Arrays3.3.4 Array Transposition3.3.5 Determining Dimensions of Arrays3.4 Visualizing the Waveform of an Audio SignalBibliography
Chapter 4: MATLAB® Programming Environment
4.1 Introduction: MATLAB Application4.1.1 Command Window4.1.2 Workspace4.1.3 Current Folder4.2 MATLAB m-Files4.2.1 Scripts4.2.2 Commenting Code4.3 MATLAB Debugging Mode4.4 MATLAB Help Documentation
Chapter 5: Logicals and Control Structures in Programming
5.1 Introduction: Controlling the Flow of Execution5.2 Logical Data Type5.2.1 Logical Operations5.2.2 Combining Logical Operators5.2.3 String Compare5.3 Types of Control Structures5.3.1 Conditional Statements5.3.2 Loops5.3.3 Functions
Chapter 6: Signal Gain and DC Offset
6.1 Introduction: Digital Signal Processing6.1.1 Element-Wise Processing6.1.2 Element-Wise Referencing6.1.3 Block Diagrams6.2 Scalar Operations with Arrays6.2.1 Scalar Multiplication: Converting Time Units6.3 Scalar Multiplication: Signal Gain6.3.1 Signal Gain Block Diagram6.3.2 Polarity Inversion6.3.3 Decibel Scale6.4 Visualizing the Amplitude Change6.4.1 Input versus Output Characteristic Curve6.5 Scalar Addition: DC Offset6.5.1 DC Offset Block Diagram6.6 Combined Signal Gain and DC Offset6.7 Amplitude Measurements6.7.1 Signal Peak Amplitude6.7.2 Peak-to-Peak Amplitude6.7.3 Root-Mean-Square Amplitude6.7.4 Dynamic Range Crest Factor6.8 Amplitude Normalization6.8.1 Peak Normalization6.8.2 Root-Mean-Square NormalizationBibliography
Chapter 7: Introduction to Signal Synthesis
7.1 Introduction: Signal Synthesis7.2 Visualizing the Frequency Spectrum of an Audio Signal7.3 Periodic Signals7.3.1 Sine Wave7.3.2 Cosine Function7.3.3 Square Wave7.3.4 Sawtooth Wave7.3.5 Triangle Wave7.3.6 Impulse Train7.4 Aperiodic Signals7.4.1 White NoiseBibliography
Chapter 8: Digital Summing, Signal Fades, and Amplitude Modulation
8.1 Introduction: Combining Signals8.1.1 Array Operations with Arrays8.1.2 Using Array Operations to Combine Signals8.2 Array Addition: Signal Summing8.2.1 Addition Block Diagram8.2.2 Array Subtraction8.3 Array Multiplication: Ring Modulation8.4 Array Multiplication: Amplitude Fade8.4.1 Linear Fade8.4.2 Exponential Fade8.4.3 S-Curve Fade8.4.4 Equal-Amplitude and Equal-Power Fades8.5 Array Multiplication: Amplitude Modulation8.5.1 Amplitude Modulation Block Diagram8.5.2 Tremolo8.6 Appendix: Transforming Linear Scales
Chapter 9: Stereo Panning and Mid/Side Processing
9.1 Introduction: Stereo Audio Signals9.2 Stereo Panning9.2.1 Panning Functions9.2.2 Rhythmic Auto-Pan Effect9.3 Mid/Side Processing9.3.1 Mid/Side Background9.3.2 Encoding9.3.3 Decoding9.3.4 Stereo Image Widening9.4 Visualizing Stereo Width9.4.1 Polar Coordinates9.4.2 GoniometerBibliography
Chapter 10: Distortion, Saturation, and Clipping
10.1 Introduction: Linear and Nonlinear Processing10.1.1 Audio Distortion Effects10.2 Visualizing Nonlinear Processing10.2.1 Total Harmonic Distortion Plot10.3 Infinite Clipping10.3.1 Sine Wave Analysis10.4 Rectification10.4.1 Half-Wave Rectification10.4.2 Full-Wave Rectification10.5 Hard Clipping10.6 Soft Clipping10.6.1 Cubic Distortion10.6.2 Arctangent Distortion10.6.3 Additional Clipping Functions10.7 Bit reduction10.7.1 Bit Depth Background10.7.2 Bit-Reduction Algorithm10.7.3 Dither Noise10.8 Harmonic Analysis of Distortion Effects10.8.1 Even and Odd Mathematical Functions10.8.2 Asymmetrical Distortion Functions10.9 Parallel DistortionBibliography
Chapter 11: Echo Effects
11.1 Introduction: Systems with Memory11.2 Delay11.2.1 Series Delay11.2.2 Block Diagram Delay Notation11.3 Converting Delay Time to Samples11.3.1 Converting Seconds to Samples11.3.2 Converting Tempo to Samples11.4 Categorizing Echo Effects11.4.1 Perceptual Temporal Fusion11.5 Feedforward Echo11.5.1 Difference Equation11.5.2 Multi-Tap Echo11.6 Feedback Echo11.6.1 Feedback Gain Convention11.6.2 Delayed Repetitions11.6.3 System Stability11.6.4 Combined Feedforward and Feedback Echo11.6.5 Stereo Echo11.6.6 Ping-Pong Echo11.7 Impulse Response11.7.1 Finite Impulse Response Systems11.7.2 Infinite Impulse Response Systems11.8 Convolution11.8.1 MATLAB Convolution Function11.8.2 Mathematical Equation for Convolution11.8.3 Convolution Reverberation11.9 Necessary Requirements for Modeling a System with an Impulse Response and Convolution11.9.1 Linearity11.9.2 Time Invariance
Chapter 12: Finite Impulse Response Filters
12.1 Introduction: Spectral Processors12.1.1 Filter Characteristics12.1.2 Visualizing the Spectral Response of a Filter12.1.3 Units of Time Delay12.2 Basic Feedforward Filters12.2.1 Low-Pass Filter12.2.2 High-Pass Filter12.3 Additional Feedforward Filters12.3.1 Comb Filter12.3.2 Band-Pass Filter12.3.3 Feedforward Comb Filter with Gain = -112.4 Changing the Relative Gain of a Filter12.4.1 Changing the Gain of the Delay Line12.4.2 Normalizing the Overall System Gain12.5 Generalized Finite Impulse Response Filters12.5.1 MATLAB FIR Filter Functions12.6 Changing the Filter Order12.7 Processing a Signal Using an FIR Filter12.7.1 Convolution12.8 Appendix I: Combining Multiple Filters12.8.1 Series Filters12.8.2 Parallel Filters12.9 Appendix II: Using an FIR Filter to Synthesize Pink NoiseBibliography
Chapter 13: Infinite Impulse Response Filters
13.1 Introduction: Filters with Feedback13.1.1 Spectral Analysis of Filters with Feedback13.2 Basic Feedback Filters13.2.1 Negative Feedback13.2.2 Inverted Comb Filter13.3 Combined Feedforward and Feedback Filters13.3.1 All-Pass Filter13.3.2 Bi-Quadratic Filter13.4 MATLAB IIR Filter Design Functions13.4.1 MATLAB butter Function13.4.2 MATLAB ellip Function13.4.3 Filter Order, Ripple, Slope, Attenuation13.5 Series and Parallel Filter Configurations13.5.1 Equalizer13.5.2 Multi-Band Processing13.6 Processing a Signal Using an IIR Filter13.6.1 Approximating an IIR Filter as an FIR Filter13.7 Appendix I: Additional Filter Forms13.7.1 APF Direct Form II13.7.2 Nested APF13.7.3 Bi-Quad Filter Topologies13.8 Appendix II: Slew Rate Distortion13.9 Appendix III: Using an IIR Filter to Synthesize Pink Noise13.10 Appendix IV: The LKFS/LUFS Standard Measurement of LoudnessBibliography
Chapter 14: Delay Buffers and Fractional Delay Interpolation
14.1 Introduction: Fractional Delay Using Buffers14.2 Delay Buffers14.2.1 Linear Buffers14.2.2 Circular Buffers14.3 Fractional Delay14.3.1 Linear Interpolation14.3.2 Cubic InterpolationBibliography
Chapter 15: Modulated Delay Effects
15.1 Introduction: Modulated Delay Effects15.1.1 Time-Invariant and Time-Variant Systems15.1.2 Types of Modulated Delay Effects15.2 Visualizing Time-Varying Information15.2.1 Spectrogram15.3 Low-Frequency Oscillator15.4 Series Delay15.4.1 Vibrato Effect15.5 Parallel Delay15.5.1 Chorus Effect15.5.2 Flanger Effect15.6 Pitch Shifter15.6.1 Shifting by Scale Intervals15.6.2 Delay Change and Pitch Change15.6.3 Rate of Change in Delay Time15.6.4 Preserving Signal Duration15.6.5 Parallel Modulated Delay15.6.6 Harmony Pitch Shifter Effect15.7 Modulated All-Pass Filters15.7.1 Phaser Effect15.8 Modulated Low-Pass Filters15.8.1 Auto-Wah EffectBibliography
Chapter 16: Algorithmic Reverb Effects
16.1 Introduction: Algorithmic Reverb Effects16.1.1 Combining Modulated Delay Effects16.1.2 Advantages of Algorithmic Reverb16.2 Schroeder16.2.1 Feedback Comb Filters16.2.2 All-Pass Filter: Colorless Delays16.2.3 Schroeder Algorithm Implementation16.3 Moorer16.3.1 Early Reflections16.3.2 Low-Pass Filtered Feedback Delay16.3.3 Moorer Algorithm Implementation16.4 Feedback Delay Networks16.4.1 Crossover Feedback16.4.2 FDN Algorithm Implementation16.5 Appendix: Estimating Reverb TimeBibliography
Chapter 17: Amplitude Envelope Effects
17.1 Introduction: Amplitude Envelope Effects17.1.1 Defining the Amplitude Envelope17.1.2 Effects Based on the Amplitude Envelope17.2 Signal Synthesis: ADSR17.2.1 Creating the ADSR Amplitude Envelope17.2.2 Applying the ADSR Amplitude Envelope17.3 Measuring the Amplitude Envelope17.3.1 Linear Method17.3.2 Decibel Method17.3.3 RMS Method17.4 Envelope Amplitude Modulation17.5 Vocoder17.5.1 Multi-Band Envelope Modulation17.5.2 Cochlear Implant Simulation17.6 Envelope Parameter Modulation17.6.1 Envelope-Wah Effect17.7 Transient Designer17.7.1 Envelope Detection Analysis17.7.2 Controls17.7.3 Implementation of the Transient DesignerBibliography
Chapter 18: Dynamic Range Processors
18.1 Introduction: Dynamic Range Processors18.1.1 Basics of Amplitude Processing18.1.2 Compressors: Reduce Dynamic Range18.1.3 Expanders: Increase Dynamic Range18.2 Detection Path18.3 Conversion to Decibel Scale18.4 Static Characteristics18.4.1 Threshold18.4.2 Ratio18.4.3 Knee18.4.4 Gain Change Amount18.5 Response Time18.5.1 Step Response Analysis18.5.2 MATLAB stepz Function18.5.3 System Response Due to Step Input18.5.4 Rise Time of Step Response18.5.5 Using Response Time for a Compressor18.5.6 Attack and Release Time18.6 Additional Detection Path Methods18.6.1 Feedback Detection18.6.2 RMS Detection18.6.3 Side-Chain Compressor18.7 Expander18.7.1 Gate18.8 Upwards Compressor and Expander18.9 Appendix I: De-Esser Effect18.10 Appendix II: Second-Order Control Systems18.10.1 Characteristics of a Second-Order Control System18.10.2 Bi-quad Parameters and Step Response Characteristics18.10.3 Second-Order System Design FunctionBibliography
Index

Content preview from Hack Audio

Echo EffectsSystems with Memory: Audible Delay

11.1 Introduction: Systems with Memory

The systems described from Chapters 6 to 10 are all based on element-wise processing. An equivalent categorization is to describe the systems as memoryless systems. In other words, the output at any time from the system only depends on the input to the system at the same time. These systems do not require the memory of previous time samples to process a signal.

There are many other signal processing systems used in audio where the output at any time can depend on the input signal at a previous time. These systems require memory to store the input samples for later use. These are systems with memory.

11.2 Delay

Systems with memory have the ability ...

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