Introduction to Digital Signal Processing Using MATLAB with Application to Digital Communications

✍ Scribed by K. S. Thyagarajan

• Publisher: Springer
• Year: 2018
• Tongue: English
• Leaves: 510
• Category: Library

✦ Synopsis

This textbook provides engineering students with instruction on processing signals encountered in speech, music, and wireless communications using software or hardware by employing basic mathematical methods. The book starts with an overview of signal processing, introducing readers to the field. It goes on to give instruction in converting continuous time signals into digital signals and discusses various methods to process the digital signals, such as filtering. The author uses MATLAB throughout as a user-friendly software tool to perform various digital signal processing algorithms and to simulate real-time systems. Readers learn how to convert analog signals into digital signals; how to process these signals using software or hardware; and how to write algorithms to perform useful operations on the acquired signals such as filtering, detecting digitally modulated signals, correcting channel distortions, etc. Students are also shown how to convert MATLAB codes into firmware codes. Further, students will be able to apply the basic digital signal processing techniques in their workplace. The book is based on the author's popular online course at University of California, San Diego.

✦ Table of Contents

Preface
Contents
Chapter 1: Introduction
1.1 What Is Digital Signal Processing
1.2 A Few Applications of Digital Signal Processing
1.2.1 Audio/Speech Processing
1.2.2 Digital Communications
1.2.3 Digital Image Processing
1.2.4 Digital Signal/Image Processing in Medicine
1.3 A Typical Digital Signal Processing System
1.4 Continuous-Time Signals and Systems
1.4.1 Continuous-Time Signals in the Time Domain
1.4.2 Continuous-Time Systems
1.4.3 Frequency Domain Representation of Signals and Systems
1.5 Summary
1.6 Problems
References
Chapter 2: Discrete-Time Signals and Systems
2.1 Introduction
2.2 Typical Discrete-Time Signals
2.3 Discrete-Time Systems
2.4 Convolution Sum
2.5 Linear Difference Equation
2.6 Sampling a Continuous-Time Signal
2.7 Conversion of Continuous-Time Signals to Digital Signals
2.8 Performance of A/D Converters
2.9 Summary
2.10 Problems
References
Chapter 3: Z-Transform
3.1 Z-Transform Definition
3.2 Properties of Z-Transform
3.3 Z-Transform and Difference Equation
3.4 Poles and Zeros
3.5 Inverse Z-Transform
3.6 MATLAB Examples
3.7 Summary
3.8 Problems
References
Chapter 4: Frequency Domain Representation of Discrete-Time Signals and Systems
4.1 Introduction
4.2 Discrete-Time Fourier Transform
4.2.1 DTFT and Z-Transform
4.2.2 DTFT of Some Typical Sequences
4.3 Inverse Discrete-Time Fourier Transform
4.4 Properties of DTFT
4.5 Frequency Domain Representation of LTI Discrete-Time Systems
4.5.1 Steady State Response of LTI Discrete-Time Systems
4.5.2 Concept of Filtering
4.5.3 Calculation of DTFT Using MATLAB
4.6 Summary
4.7 Problems
References
Chapter 5: Discrete Fourier Transform
5.1 Introduction
5.2 Definition of DFT
5.3 Relationship Between DTFT and DFT
5.4 Inverse DFT
5.5 Effect of Sampling the DTFT on the Reconstructed Sequence
5.6 Circular Convolution
5.7 Properties of the DFT
5.8 Linear Convolution Using Circular Convolution
5.9 Linear Convolution of a Finite-Length Sequence with an Infinite-Length Sequence
5.9.1 Overlap and Add
5.9.2 Overlap and Save
5.9.3 DFT Leakage
5.10 Discrete Transforms
5.10.1 Unitary Transform
5.10.2 Orthogonal Transform
5.10.3 Discrete Cosine Transform
5.10.4 Hadamard Transform
5.11 Summary
5.12 Problems
References
Chapter 6: IIR Digital Filters
6.1 Introduction
6.2 Impulse Invariance Technique
6.3 Design of IIR Digital Filters in the Frequency Domain
6.3.1 Digital Filter Frequency Specifications
6.3.2 Design Using Bilinear Transformation
6.3.3 Butterworth Lowpass IIR Digital Filter
6.3.4 Chebyshev Type I Lowpass IIR Digital Filter
6.3.5 Chebyshev Type II Lowpass IIR Digital Filter
6.3.6 Elliptic Lowpass IIR Digital Filter
6.4 Design of IIR Digital Filters Using Frequency Transformation
6.4.1 Lowpass-to-Lowpass Conversion
6.4.2 Lowpass-to-Highpass Conversion
6.4.3 Lowpass-to-Bandpass and Bandstop Conversion
6.5 Computer-Aided Design of IIR Digital Filters
6.6 Group Delay
6.6.1 Group Delay Equalization
6.7 Simulation Using Simulink
6.8 Summary
6.9 Problems
References
Chapter 7: FIR Digital Filters
7.1 Types of Linear-Phase FIR Filters
7.2 Linear-Phase FIR Filter Design
7.2.1 Lowpass FIR Filter Design
7.2.2 Gibbs Phenomenon
7.2.3 Windowed Lowpass Linear-Phase FIR Filter Design
7.2.3.1 Bartlett Window
7.2.3.2 Haan Window
7.2.3.3 Hamming Window
7.2.3.4 Blackman Window
7.2.3.5 Determination of FIR Filter Order
7.2.3.6 Adjustable Window Functions
7.2.3.6.1 Dolph-Chebyshev (DC) Window
7.2.3.6.2 Kaiser Window
7.2.4 Design of a Highpass Linear-Phase FIR Filter
7.2.5 Design of a Bandpass Linear-Phase FIR Filter
7.2.6 Design of a Bandstop Linear-Phase FIR Filter
7.3 Computer-Aided Design of Linear-Phase FIR Filters
7.4 Discrete-Time Hilbert Transformer
7.5 Summary
7.6 Problems
References
Chapter 8: Digital Filter Structures
8.1 Signal Flow Graph
8.2 IIR Digital Filter Structures
8.2.1 Direct Form I and II Structures
8.2.2 Parallel Structure
8.2.3 Cascade Structure
8.3 FIR Filter Structures
8.3.1 Direct Form Structure of an FIR Filter
8.3.2 Cascade Structure of an FIR Digital Filter
8.3.3 Linear-Phase FIR Filter Structure
8.3.4 Polyphase FIR Filter Structure
8.4 Finite Word Length Effect
8.4.1 Fixed-Point Binary Representation
8.4.2 Floating-Point Binary Representation
8.4.3 Filter Coefficient Sensitivity
8.4.4 Error Due to Finite Word Length Arithmetic
8.4.5 Limit Cycles in IIR Digital Filters
8.5 FIR Lattice Structure
8.6 Summary
8.7 Problems
References
Chapter 9: Fast Fourier Transform
9.1 Brute-Force Computation of DFT
9.2 Fast Fourier Transform
9.2.1 Decimation-in-Time FFT
9.2.2 Decimation-in-Frequency FFT
9.2.3 Inverse FFT
9.3 Spectral Analysis of Discrete-Time Sequences
9.3.1 Autocorrelation of a Discrete-Time Sequence
9.3.2 Relation Between Autocorrelation and DTFT of a Discrete-Time Sequence
9.3.3 Autocorrelation of Periodic Sequences
9.3.4 Short-Time Fourier Transform
9.4 Fixed-Point Implementation of FFT
9.5 Sliding Discrete Fourier Transform
9.6 Energy Compaction Property Revisited
9.7 Zoom FFT
9.8 Chirp Fourier Transform
9.9 Summary
9.10 Problems
References
Chapter 10: DSP in Communications
10.1 Introduction
10.2 Sampling Rate Conversion
10.2.1 Upsampling
10.2.2 Downsampling
10.3 Oversampled ADC
10.3.1 Transition Bandwidth Reduction
10.3.2 Analysis of Oversampled ADC
10.4 Oversampled DAC
10.5 Cancelation of Inter-Symbol Interference
10.5.1 Pulse Shaping
10.5.2 Equalization
10.5.3 Matched Filter
10.5.4 Phase-Locked Loop
10.5.5 OFDM
10.5.6 Software-Defined Radio
10.6 Summary
10.7 Problems
References
Index

✦ Subjects

Technology & Engineering, Electronics, General, Telecommunications, Computers, Networking, Hardware, Online Services, Imaging Systems, Electrical