Chapter 1 Crash Course in Digital Signal Processing
1.1 Signals and Systems
1.2 Analog-to-Digital and Digital-to-Analog Conversions
1.3 Digital Signals and Their Spectra
1.4 Digital Filtering
1.5 Speech, Music, Images, and More
Chapter Summary
Review Questions
Chapter 2 Analog-to-Digital and Digital-to-Analog Conversion
2.1 A Simple DSP System
2.2 Sampling
2.2.1 Nyquist Sampling Theory
2.2.2 The Frequency View of Sampling
2.3 Quantization
2.4 Analog-to-Digital Conversion
2.5 Digital-to-Analog Conversion
Chapter Summary
Review Questions
Chapter 3 Digital Signals
3.1 Pictures of Digital Signals
3.2 Notation for Digital Signals
3.3 Digital Functions
3.3.1 Impulse Functions
3.3.2 Step Functions
3.3.3 Power and Exponential Functions
3.3.4 Sine and Cosine Functions
3.4 Composite Functions
3.5 Two-Dimensional Digital Signals
Chapter Summary
Review Questions
Chapter 4 Difference Equations and Filtering
4.1 Filtering Basics
4.2 Analog Filters versus Digital Filters
4.3 Linear, Time-Invariant, Causal Systems
4.4 Difference Equation Structure
4.5 Superposition
4.6 Difference Equation Diagrams
4.6.1 Nonrecursive Difference Equations
4.6.2 Recursive Difference Equations
4.7 The Impulse Response
4.8 The Step Response
Chapter Summary
Review Questions
Chapter5 Convolution and Filtering
5.1 Convolution Basics
5.2 Difference Equations and Convolution
5.3 Moving Average Filters
5.4 Filtering Digital Images
Chapter Summary
Review Questions
Chapter 6 zTransforms
6.1 z Transform Basics
6.2 Transfer Functions
6.2.1 Transfer Functions and Difference Equations
6.2.2 Transfer Functions and Impulse Responses
6.2.3 Finding Filter Outputs
6.2.4 Cascade and Parallel Combinations of Transfer Functions
6.3 Back to the Time Domain
6.3.1 Standard Form
6.3.2 Simple Inverse z Transforms
6.3.3 Inverse z Transforms by Long Division
6.3.4 Inverse z Transforms by Partial Fraction Expansion
6.4 Transfer Functions and Stability
6.4.1 Poles and Zeros
6.4.2 Stability
6.4.3 First Order Systems
6.4.4 Second Order Systems
Chapter Summary
Review Questions
Chapter 7 Fourier Transforms and Filter Shape
7.1 Fourier Transform Basics
7.2 Frequency Responses and Other Forms
7.2.1 Frequency Responses and Difference Equations
7.2.2 Frequency Responses and Transfer Functions
7.2.3 Frequency Responses and Impulse Responses
7.3 Frequency Response and Filter Shape
7.3.1 Filter Effects on Sine Wave Inputs
7.3.2 Magnitude Response and Phase Response
7.3.3 Analog Frequency fend Digital Frequency
7.3.4 Filter Shape from Poles and Zeros
7.3.5 First Order Filters
7.3.6 Second Order Filters
Chapter Summary
Review Questions
Chapter 8 Digital Signal Spectra
8.1 The Meaning of the Spectrum
8.2 Nonperiodic Digital Signals
8.3 Periodic Digital Signals
Chapter Summary
Review Questions
Chapter 9 Finite Impulse Response Filters
9.1 Finite Impulse Response Filter Basics
9.2 Moving Average Filters Revisited
9.3 Phase Distortion
9.4 Approximating an Ideal Low Pass Filter
9.5 Windows
9.5.1 Rectangular Window
9.5.2 Hanning Window
9.5.3 Hamming Window
9.5.4 Blackman Window
9.5.5 Kaiser Window
9.6 Low Pass FIR Filter Design
9.6.1 Design Guidelines
9.6.2 Steps for Low Pass FIR Filter Design
9.7 Band Pass and High Pass FIR Filters
9.8 Band Stop FIR Filters
9.9 Equiripple FIR Filter Design
9.10 Hazards of Practical FIR Filters
Chapter Summary
Review Questions
Chapter 10 Infinite Impulse Response Filters
10.1 Infinite Impulse Response Filter Basics
10.2 Low Pass Analog Filters
10.3 Bilinear Transformation
10.4 Butterworth Filter Design
10.5 Chebyshev Type I Filter Design
10.6 Impulse Invariance IIR Filter Design
10.7 "Best Fit" Filter Design
10.8 Band Pass, High Pass, and Band Stop IIR Filters
10.9 Hazards of Practical IIR Filters
Chapter Summary
Review Questions
Chapter 11 DFT and FFT Processing
11.1 DFT Basics
11.2 Relationship to Fourier Transform
11.3 Relationship to Fourier Series
11.4 DFF Window Effects
11.5 Spectrograms
11.6 FFT Basics
11.7 2D DFT/FFT
Chapter Summary
Review Questions
Chapter 12 Signal Processing
12.1 Digital Audio
12.1.1 Digital Audio Basics
12.1.2 Oversampling and Decimation
12.1.3 Zero Insertion and Interpolation
12.1.4 Dithering and Companding
12.1.5 Audio Processing
12.2 Speech Recognition
12.3 Voice and Music Synthesis
Chapter Summary
References
Appendix A The Math You Need
A.I Functions
A.2 Degrees and Radians
A.3 Rational Functions
A.4 Lowest Terms
A.5 Lowest Common Multiple
A.6 Reciprocals
A.7 Logarithms
A.8 Power and Exponential Functions
A.9 Sinusoidal Function
A.10 Decibels
A. 11 Decimal, Binary, and Hexadecimal Number Systems
A.12 Area and Perimeter
A. 13 Complex Numbers
A.14 Absolute Value
A.15 Quadratic Formula
A.16 ∑ Sums
Appendix B Signal-to-Noise Ratio
Appendix C Direct Form 2 Realization of Recursive Filters
Appendix D Convolution in the Time Domain and Multiplication in the Frequency Domain
Appendix E Scaling Factor in Discrete Fourier Series and Discrete Fourier Transform
Appendix F Inverse Discrete Time Fourier Transform
Appendix G Impulse Response of Ideal Low Pass Filter
Appendix H Sampling Property
Appendix I Spectrum of Digital Cosine Signal
Appendix J Spectrum of Impulse Train
Appendix K Spectral Effects of Sampling
Appendix L Butterworth Recursive Filter Order
Appendix M Chebyshev Type I Recursive Filter Order
Appendix N Circular Convolution
N.1 Periodic Signals and Their Representations
N.2 Circular Shift
N.3 Circular Reversal
N.4 Circular Convolution
N.5 Circular Convolution Theorem
N.6 Implementation of the Linear Convolution of Two Finite-Length Sequences Using the FFT