Fundamentals of digital signal processing

　　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