
Digital Signal Processing
Fundamentals, Applications, and Deep Learning
- 4th Edition - February 5, 2025
- Imprint: Academic Press
- Authors: Li Tan, Jean Jiang
- Language: English
- Paperback ISBN:9 7 8 - 0 - 4 4 3 - 2 7 3 3 5 - 3
- eBook ISBN:9 7 8 - 0 - 4 4 3 - 2 7 3 3 6 - 0
Digital Signal Processing: Fundamentals, Applications, and Deep Learning, Fourth Edition introduces students to the fundamental principles of digital signal processing (DSP) whi… Read more

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Request a sales quoteDigital Signal Processing: Fundamentals, Applications, and Deep Learning, Fourth Edition introduces students to the fundamental principles of digital signal processing (DSP) while also providing a working knowledge that they take with them into their engineering careers. Many instructive, worked examples are used to illustrate the material, and the use of mathematics is minimized for an easier grasp of concepts. As such, this title is also useful as a reference for non-engineering students and practicing engineers.
This book goes beyond DSP theory, showing the implementation of algorithms in hardware and software. Additional topics covered include DSP for artificial intelligence, adaptive filtering with noise reduction and echo cancellations, speech compression, signal sampling, digital filter realizations, filter design, multimedia applications, over-sampling, etc. More advanced topics are also covered, such as adaptive filters, speech compression such as pulse-code modulation, µ-law, adaptive differential pulse-code modulation, multi-rate DSP, oversampling analog-to-digital conversion, sub-band coding, wavelet transform, and neural networks.
- Covers DSP principles with various examples of real-world DSP applications on noise cancellation, communications, control applications, and artificial intelligence
- Includes application examples using DSP techniques for deep learning neural networks to solve real-world problems
- Provides a new chapter to cover principles of artificial neural networks and convolution neural networks with back-propagation algorithms
- Provides hands-on practice, with MATLAB code for worked examples and C programs for real-time DSP for students at https://www.elsevier.com/books-and-journals/book-companion/9780443273353
- Offers teaching support, including an image bank, full solutions manual, and MATLAB projects for qualified instructors, available for request at https://educate.elsevier.com/9780443273353
- Title of Book
- Cover image
- Title page
- Table of Contents
- Copyright
- Preface
- CHAPTER 1. Introduction to Digital Signal Processing
- 1.1 Basic Concepts of Digital Signal Processing
- 1.2 Basic Digital Signal Processing Examples in Block Diagrams
- 1.2.1 Digital Filtering
- 1.2.2 Signal Frequency (Spectrum) Analysis
- 1.3 Overview of Typical Digital Signal Processing in Real-World Applications
- 1.3.1 Digital Crossover Audio System
- 1.3.2 Interference Cancellation in Electrocardiography
- 1.3.3 Speech Coding and Compression
- 1.3.4 Vibration Signature Analysis for Defected Gear Teeth
- 1.3.5 Digital Image Enhancement
- 1.3.6 Artificial Intelligence Examples
- 1.3.6.1 Machine Fault Detection
- 1.3.6.2 Speech Feature Extraction
- 1.3.6.3 Electroencephalogram Classification
- 1.4 Digital Signal Processing Applications
- 1.5 Summary
- CHAPTER 2. Signal Sampling and Quantization
- 2.1 Sampling of Continuous Signals
- 2.2 Signal Reconstruction
- 2.2.1 Practical Considerations for Signal Sampling: Anti-Aliasing Filtering
- 2.2.2 Practical Considerations for Signal Reconstruction: Anti-Image Filter and Equalizer
- 2.3 Analog-to-Digital Conversion, Digital-to-Analog Conversion, and Quantization
- 2.4 Summary
- 2.5 MATLAB® Programs
- 2.6 Problems
- MATLAB® Projects
- Advanced Problems
- CHAPTER 3. Digital Signals and Systems
- 3.1 Digital Signals
- 3.1.1 Common Digital Sequences
- 3.1.2 Generation of Digital Signals
- 3.2 Linear Time-Invariant, Causal, and Bounded-Input and Bounded-Output Systems
- 3.2.1 Linearity
- 3.2.2 Time Invariance
- 3.2.3 Causality
- 3.2.4 Bounded-Input and Bounded-Output Stability
- 3.3 Difference Equations and Impulse Responses
- 3.3.1 Format of Difference Equations
- 3.3.2 System Representation Using Its Impulse Response
- 3.4 Digital Convolution
- 3.5 Properties of Linear Time-Invariant System
- 3.5.1 Causality of Linear Time-Invariant System
- 3.5.2 Bounded-Input and Bounded-Output Stability
- 3.6 Block Diagrams
- 3.7 Summary
- 3.8 Problems
- Advanced Problems
- CHAPTER 4. The Z-Transform
- 4.1 Definition
- 4.2 Properties of the Z-Transform
- 4.3 Inverse Z-Transform
- 4.3.1 Partial Fraction Expansion and Look-Up Table
- 4.3.2 Partial Fraction Expansion Using MATLAB®
- 4.3.3 Power Series Method
- 4.3.4 Inversion Formula Method
- 4.4 Solution of Difference Equations Using the Z-Transform
- 4.5 Two-Sided Z-Transform
- 4.6 Summary
- 4.7 Problems
- Advanced Problems
- CHAPTER 5. Discrete Fourier Transform and Signal Spectrum
- 5.1 Discrete Fourier Transform
- 5.1.1 Fourier Series Coefficients of Periodic Digital Signals
- 5.1.2 Discrete Fourier Transform Formulas
- 5.2 Amplitude Spectrum and Power Spectrum
- 5.3 Spectral Estimation Using Window Functions
- 5.4 Application to Signal Spectral Estimation
- 5.5 Fast Fourier Transform
- 5.5.1 Method of Decimation-in-Frequency
- 5.5.2 Method of Decimation-in-Time
- 5.6 Summary
- 5.7 Problems
- Computer Problems With MATLAB®
- MATLAB® Projects
- Advanced Problems
- Chapter 6. DIGITAL SIGNAL PROCESSING SYSTEMS, BASIC FILTERING TYPES, AND DIGITAL FILTER REALIZATIONS
- 6.1 Difference Equation and Digital Filtering
- 6.2 Difference Equation and Transfer Function
- 6.2.1 Impulse, Step, and System Responses
- 6.3 The Z-Plane Pole-Zero Plot and Stability
- 6.4 Digital Filter Frequency Response
- 6.5 Basic Types of Filtering
- 6.6 Realization of Digital Filters
- 6.6.1 Direct-Form I Realization
- 6.6.2 Direct-Form II Realization
- 6.6.3 Cascade (Series) Realization
- 6.6.4 Parallel Realization
- 6.7 Application: Signal Enhancement and Filtering
- 6.7.1 Preemphasis of Speech
- 6.7.2 Bandpass Filtering of Speech
- 6.7.3 Enhancement of Electrocardiogram Signals Using Notch Filtering
- 6.8 Summary
- 6.9 Problems
- MATLAB® Problems
- MATLAB® Projects
- Advanced Problems
- Chapter 7. FINITE IMPULSE RESPONSE FILTER DESIGN
- 7.1 Finite Impulse Response Filter Format
- 7.2 Fourier Transform Design
- 7.3 Window Method
- 7.4 Applications: Noise Reduction and Two-Band Digital Crossover
- 7.4.1 Noise Reduction
- 7.4.2 Speech Noise Reduction
- 7.4.3 Noise Reduction in Vibration Signals
- 7.4.4 Two-Band Digital Crossover
- 7.5 Frequency Sampling Design Method
- 7.6 Optimal Design Method
- 7.7 Design of the Finite Impulse Response Differentiator and Hilbert Transformer
- 7.8 Realization Structures of Finite Impulse Response Filters
- 7.8.1 Transversal Form
- 7.8.2 Linear Phase Form
- 7.9 Coefficient Accuracy Effects on Finite Impulse Response Filters
- 7.10 Summary of Finite Impulse Response Design Procedures and Selection of the FIR Filter Design Methods in Practice
- 7.11 Summary
- 7.12 MATLAB® Programs
- 7.13 Problems
- Computer Problems with MATLAB®
- MATLAB® Projects
- Advanced Problems:
- Chapter 8. INFINITE IMPULSE RESPONSE FILTER DESIGN
- 8.1 Infinite Impulse Response Filter Format
- 8.2 Bilinear Transformation Design Method
- 8.2.1 Analog Filters Using Lowpass Prototype Transformation
- 8.2.2 Bilinear Transformation and Frequency Warping
- 8.2.3 Bilinear Transformation Design Procedure
- 8.3 Digital Butterworth and Chebyshev Filter Designs
- 8.3.1 Lowpass Prototype Function and Its Order
- 8.3.2 Lowpass and Highpass Filter Design Examples
- 8.3.3 Bandpass and Bandstop Filter Design Examples
- 8.4 Higher-Order Infinite Impulse Response Filter Design Using the Cascade Method
- 8.5 Application: Digital Audio Equalizer
- 8.6 Impulse Invariant Design Method
- 8.7 Pole-Zero Placement Method for Simple Infinite Impulse Response Filters
- 8.7.1 Second-order Bandpass Filter Design
- 8.7.2 Second-Order Bandstop (Notch) Filter Design
- 8.7.3 First-Order Lowpass Filter Design
- 8.7.4 First-Order Highpass Filter Design
- 8.8 Realization Structures of Infinite Impulse Response Filters
- 8.8.1 Realization of Infinite Impulse Response Filters in Direct-Form I and Direct-Form II
- 8.8.2 Realization of Higher-Order Infinite Impulse Response Filters Via the Cascade Form
- 8.9 Application: 60-Hz Hum Eliminator and Heart Rate Detection Using Electrocardiography
- 8.10 Coefficient Accuracy Effects on Infinite Impulse Response Filters
- 8.11 Application: Generation and Detection of DTMF Tones Using the Goertzel Algorithm
- 8.11.1 Single-Tone Generator
- 8.11.2 Dual-Tone Multifrequency Tone Generator
- 8.11.3 Goertzel Algorithm
- 8.11.4 Dual-Tone Multifrequency Tone Detection Using the Modified Goertzel Algorithm
- 8.12 Summary of Infinite Impulse Response (IIR) Design Procedures and Selection of the IIR Filter Design Methods in Practice
- 8.13 Summary
- 8.14 Problems
- MATLAB® Projects
- Advanced Problems
- Chapter 9. ADAPTIVE FILTERS AND APPLICATIONS
- 9.1 Introduction to Least Mean Squares Adaptive Finite Impulse Response Filters
- 9.2 Basic Wiener Filter Theory and Adaptive Algorithms
- 9.2.1 Wiener Filter Theory and Linear Prediction
- 9.2.1.1 Basic Wiener Filter Theory
- 9.2.1.2 Forward Linear Prediction
- 9.2.2 Steepest Descent Algorithm
- 9.2.3 Least Mean Squares Algorithm
- 9.2.4 Recursive Least Squares Algorithm
- 9.3 Applications: Noise Cancellation, System Modeling, and Line Enhancement
- 9.3.1 Noise Cancellation
- 9.3.2 System Modeling
- 9.3.3 Line Enhancement Using Linear Prediction
- 9.3.4 Channel Equalization in Communication Systems
- 9.4 Other Application Examples
- 9.4.1 Canceling Periodic Interferences Using Linear Prediction
- 9.4.2 Electrocardiography Interference Cancellation
- 9.4.3 Echo Cancellation in Long-Distance Telephone Circuits
- 9.5 Summary
- 9.6 Problems
- Computer Problems With MATLAB®
- MATLAB® Projects
- Advanced Programs
- Chapter 10. WAVEFORM QUANTIZATION AND COMPRESSION
- 10.1 Linear Midtread Quantization
- 10.2 μ-Law Companding
- 10.2.1 Analog μ-Law Companding
- 10.2.2 Digital μ-Law Companding
- 10.3 Examples of Differential Pulse Code Modulation, Delta Modulation, and Adaptive Differential Pulse Code Modulation G.721
- 10.3.1 Examples of Differential Pulse Code Modulation and Delta Modulation
- 10.3.2 Adaptive Differential Pulse Code Modulation G.721
- Simulation Example
- 10.4 Discrete Cosine Transform, Modified Discrete Cosine Transform, and Transform Coding in MPEG Audio
- 10.4.1 Discrete Cosine Transform
- 10.4.2 Modified Discrete Cosine Transform
- 10.4.3 Transform Coding in MPEG Audio
- 10.5 Summary
- 10.6 MATLAB® Programs
- 10.7 Problems
- Computer Problems with MATLAB®
- Advanced Problems
- Chapter 11. MULTIRATE DIGITAL SIGNAL PROCESSING, OVERSAMPLING OF ANALOG-TO-DIGITAL CONVERSION, AND UNDERSAMPLING OF BANDPASS SIGNALS
- 11.1 Multirate Digital Signal Processing Basics
- 11.1.1 Sampling Rate Reduction by an Integer Factor
- 11.1.2 Sampling Rate Increase by an Integer Factor
- 11.1.3 Changing the Sampling Rate by a Noninteger Factor L/M
- 11.1.4 Application: CD Audio Player
- 11.1.5 Multistage Decimation
- 11.2 Polyphase Filter Structure and Implementation
- 11.3 Oversampling of Analog-to-Digital Conversion
- 11.3.1 Oversampling and Analog-to-Digital Conversion Resolution
- 11.3.2 Sigma-Delta Modulation Analog-to-Digital Conversion
- 11.4 Application Example: CD Player
- 11.5 Undersampling of Bandpass Signals
- 11.6 Summary
- 11.7 Problems
- MATLAB® Problems
- MATLAB® Project
- Chapter 12. SUBBAND AND WAVELET-BASED CODING
- 12.1 Subband Coding Basics
- 12.2 Subband Decomposition and Two-Channel Perfect Reconstruction-Quadrature Mirror Filter Bank
- 12.3 Subband Coding of Signals
- 12.4 Wavelet Basics and Families of Wavelets
- 12.5 Multiresolution Equations
- 12.6 Discrete Wavelet Transform
- 12.7 Wavelet Transform Coding of Signals
- 12.8 MATLAB® Programs
- 12.9 Summary
- 12.10 Problems
- MATLAB® Problems
- MATLAB® Projects
- Chapter 13. IMAGE PROCESSING BASICS
- 13.1 Image Processing Notation and Data Formats
- 13.1.1 Eight-Bit Gray-Level Images
- 13.1.2 Twenty-Four-Bit Color Images
- 13.1.3 Eight-Bit Color Images
- 13.1.4 Intensity Images
- 13.1.5 Red, Green, and Blue Components and Grayscale Conversion
- 13.1.6 MATLAB® Functions for Format Conversion
- 13.2 Image Histogram and Equalization
- 13.2.1 Grayscale Histogram and Equalization
- 13.2.2 Twenty-Four-Bit Color Image Equalization
- 13.2.3 Eight-Bit Indexed Color Image Equalization
- 13.2.4 MATLAB® Functions for Equalization
- 13.3 Image Level Adjustment and Contrast
- 13.3.1 Linear Level Adjustment
- 13.3.2 Adjusting the Level for Display
- 13.3.3 MATLAB® Functions for Image Level Adjustment
- 13.4 Image Filtering Enhancement
- 13.4.1 Lowpass Noise Filtering
- 13.4.2 Median Filtering
- 13.4.3 Edge Detection
- 13.4.4 MATLAB® Functions for Image Filtering
- 13.5 Image Pseudo–Color Generation and Detection
- 13.6 Image Spectra
- 13.7 Image Compression by Discrete Cosine Transform
- 13.7.1 Two-Dimensional Discrete Cosine Transform
- 13.7.2 Two-Dimensional JPEG Grayscale Image Compression Example
- 13.7.3 JPEG Color Image Compression
- RGB-to-YIQ transformation
- DCT on image blocks
- Quantization
- Differential pulse code modulation on direct-current coefficients:
- Run-length coding on alternating-current coefficients
- Lossless entropy coding
- Coding DC coefficients
- Coding AC coefficients
- 13.7.4 Image Compression Using Wavelet Transform Coding
- 13.8 Creating a Video Sequence by Mixing Two Images
- 13.9 Video Signal Basics
- 13.9.1 Analog Video
- Phase alternative line video
- Séquentiel couleur à mémoire video
- 13.9.2 Digital Video
- 13.10 Motion Estimation in Video
- 13.11 Summary
- 13.12 Problems
- MATLAB® Problems
- Chapter 14. SIGNAL PROCESSING FOR ARTIFICIAL INTELLIGENCE
- 14.1 Artificial Neural Network
- 14.1.1 Single-Layer Perceptron
- 14.1.2 Multilayer Perceptron
- 14.1.3 Backpropagation Algorithm for Artificial Neural Network
- 14.2 Convolutional Neural Networks
- 14.2.1 Structures of Convolutional Neural Networks
- 14.2.2 Backpropagation Algorithm for Convolutional Neural Network
- 14.3 Activation and Cost Functions
- 14.3.1 Activation Functions
- 14.3.2 Softmax Layer and Cost Function
- 14.3.3 Adam Optimizer–Based Backpropagation Algorithm
- 14.4 Applications
- 14.4.1 Signal Processing for Feature Extraction
- 14.4.2 Intelligent Systems Using Vibration, Speech, and Electroencephalogram Signals
- 14.4.2.1 Machine Fault Detection
- 14.4.2.2 Speech Recognition
- 14.4.2.3 Electroencephalogram Classification
- 14.5 Summary
- 14.6 Problems
- Projects
- 14.7 MATLAB® Programs
- Chapter 15. HARDWARE AND SOFTWARE FOR DIGITAL SIGNAL PROCESSORS
- 15.1 Digital Signal Processor Architecture
- 15.2 Digital Signal Processor Hardware Units
- 15.2.1 Multiplier and Accumulator
- 15.2.2 Shifters
- 15.2.3 Address Generators
- 15.3 Digital Signal Processors and Manufacturers
- 15.4 Fixed- and Floating-Point Formats
- 15.4.1 Fixed-Point Format
- 15.4.2 Floating-Point Format
- 15.4.3 IEEE Floating Point Formats
- 15.4.4 Fixed-Point Digital Signal Processors
- 15.4.5 Floating-Point Digital Signal Processors
- 15.5 Finite Impulse Response and Infinite Impulse Response Filter Implementations in Fixed-Point Systems
- 15.6 Digital Signal Processing Programming Examples
- 15.6.1 Overview of TMS320C67X DSK
- 15.6.2 Concept of Real-Time Processing
- 15.6.3 Linear Buffering
- 15.6.4 Sample C Programs
- 15.7 Additional Real-Time Digital Signal Processing Examples
- 15.7.1 Adaptive Filtering Using the TMS320C6713 DSK
- 15.7.2 Signal Quantization Using the TMS320C6713 DSK
- 15.7.3 Sampling Rate Conversion Using the TMS320C6713 DSK
- 15.8 Summary
- 15.9 Problems
- Appendix A. INTRODUCTION TO THE MATLAB® ENVIRONMENT
- A.1 Basic Commands and Syntax
- A.2 MATLAB® Array and Indexing
- A.3 Plot Utilities: Subplot, Plot, Stem, and Stair
- A.4 MATLAB® Script Files
- A.5 MATLAB® Functions
- Appendix B. REVIEW OF ANALOG SIGNAL PROCESSING BASICS
- B.1 Fourier Series and Fourier Transform
- B.1.1 Sine-Cosine Form
- B.1.2 Amplitude-Phase Form
- B.1.3 Complex Exponential Form
- Solution:
- B.1.4 Spectral Plots
- Solution:
- Solution:
- Solution:
- B.1.5 Fourier Transform
- Solution:
- Solution:
- Solution:
- Solution:
- B.2.1 Laplace Transform and its Table
- Solution:
- Solution:
- B.2.2 Solving Linear Differential Equations Using Laplace Transform
- Solution:
- B.2.3 Transfer Function
- Solution:
- B.3.1 Poles, Zeros, and Stability
- Solution:
- B.3.2 Convolution
- Solution:
- B.3.3 Sinusoidal Steady-State Response
- Solution:
- Appendix C. NORMALIZED BUTTERWORTH AND CHEBYSHEV FUNCTIONS
- C.1 Normalized Butterworth Function
- Solution:
- Solution:
- Solution:
- Solution:
- Solution:
- Appendix D. SINUSOIDAL STEADY-STATE RESPONSE OF DIGITAL FILTERS
- D.1 Sinusoidal Steady-State Response
- D.2 Properties of the Sinusoidal Steady-State Response
- Appendix E. FINITE IMPULSE RESPONSE FILTER DESIGN EQUATIONS BY FREQUENCY SAMPLING DESIGN METHOD
- Appendix F. WAVELET ANALYSIS AND SYNTHESIS EQUATIONS
- F.1 Basic Properties
- F.2 Analysis Equations
- F.3 Wavelet Synthesis Equations
- Appendix G. REVIEW OF DISCRETE-TIME RANDOM SIGNALS
- G.1 Random Variable Statistical Properties
- G.2 Random Signal Statistical Properties
- G.3 Wide-Sense Stationary Random Signals
- G.4 Ergodic Signals
- G.5 Statistical Properties of the Linear System Output Signal
- G.6 Z-Transform Domain Representation of Statistical Properties
- Appendix H. SOME USEFUL MATHEMATICAL FORMULAS
- L’Hospital’s rule:
- Answers to Selected Problems
- Chapter 2
- Chapter 3
- Chapter 4
- Chapter 5
- Chapter 6
- Chapter 7
- Chapter 8
- Chapter 9
- Chapter 10
- Chapter 11
- Chapter 12
- Chapter 13
- Chapter 15
- Appendix B
- Index
- Edition: 4
- Published: February 5, 2025
- Imprint: Academic Press
- No. of pages: 1032
- Language: English
- Paperback ISBN: 9780443273353
- eBook ISBN: 9780443273360
LT
Li Tan
JJ