Digital Signal Processing Using MATLAB, 2/e (IE)

Table of Contents

Chapter 1 - Introduction
1.1 Overview of Digital Signal Processing
1.2 A Few Words about MATLAB

Chapter 2 - Discrete-time Signals and Systems
2.1 Discrete-time Signals
2.2 Discrete Systems
2.3 Convolution
2.4 Difference Equations
2.5 Problems

Chapter 3 - The Discrete-time Fourier Analysis
3.1 The discrete-time Fourier Transform
3.2 The Properties of the DTFT
3.3 The Frequency Domain Representation of LSI Systems
3.4 Sampling and Reconstruction of Analog Signals
3.5 Problems

Chapter 4 - The z-Transform
4.1 The Bilateral z-Transform
4.2 Important Properties of the z-Transform
4.3 Inversion of the z-Transform
4.4 System Representation in the z-Domain
4.5 Solutions of the Difference Equations
4.6 Problems

Chapter 5 - The Discrete Fourier Transform
5.1 The Discrete Fourier Series
5.2 Sampling and Reconstruction in the z-Domain
5.3 The Discrete Fourier Transform
5.4 Properties of the Discrete Fourier Transform
5.5 Linear Convolution using the DFT
5.6 Circulant matrices
5.7 The Fast Fourier Transform
5.8 Problems

Chapter 6 - Digital Filter Structures
6.1 Basic Elements
6.2 IIR Filter Structures
6.3 FIR Filter Structures
6.4 Lattice Structures
6.5 Problems

Chapter 7 - FIR Filter Design
7.1 Preliminaries
7.2 Properties of Linear Phase FIR Filters
7.3 Window Design Techniques
7.4 Frequency Sampling Design Techniques
7.5 Optimal Equiripple Design Technique
7.6 Problems

Chapter 8 - IIR Filter Design
8.1 Some preliminaries
8.2 Characteristics of Prototype Analog Filters
8.3 Analog-to-Digital Filter Transformations
8.4 Lowpass Filter Design using MATLAB
8.5 Frequency-band Transformations
8.6 Comparison of FIR vs. IIR Filters
8.7 Problems

Chapter 9 - Finite Word-Length Effects
9.1 Overview
9.2 Representation of Numbers
9.3 The Process of Quantization and Error Characterization
9.4 Quantization of Filter Coefficients
9.5 Analysis of A/D Quantization Noise
9.6 Round-Off Effects in IIR Digital Filters
9.7 Round-Off Noise in FIR Filter Realizations
9.8 Summary
9.9 Problems

Chapter 10 - Sampling Rate Conversion
10.1 Introduction
10.2 Decimation by a Factor D
10.3 Interpolation by a Factor
10.4 Sampling Rate Conversion by a Rational Factor I/D
10.5 FIR Filter Designs for Sample Rate Conversion
10.6 FIR Filter Structures for Sampling-Rate Conversion
10.7 Summary
10.8 Problems

Chapter 11 - Applications in Adaptive Filtering
11.1 LMS Algorithm for Coefficient Adjustment
11.2 System Identification or System Modeling
11.3 Suppression of Narrowband Interference in a Wideband Signal
11.4 Adaptive Line Enhancement
11.5 Adaptive Channel Equalization
11.6 Summary

Chapter 12 - Applications in Communications
12.1 Pulse Code Modulation
12.2 Differential PCM (DPCM)
12.3 Adaptive PCM and DPCM (ADPCM)
12.4 Delta Modulation (DM)
12.5 Linear Predictive Coding (LPC) of Speech
12.6 Dual-Tone Multifrequency (DTMF) Signals
12.7 Binary Digital Communications
12.8 Spread Spectrum Communications
12.9 Summary