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Adaptive filters : theory and applications / Behrouz Farhang-Boroujeny.

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Format:
Book
Author/Creator:
Farhang-Boroujeny, B.
Language:
English
Subjects (All):
Adaptive filters.
Adaptive signal processing.
Physical Description:
xx, 778 p. : ill.
Edition:
2nd ed.
Place of Publication:
Chichester, West Sussex, U.K. : Wiley, [2013]
Summary:
This second edition of Adaptive Filters: Theory and Applications has been updated throughout to reflect the latest developments in this field; notably an increased coverage given to the practical applications of the theory to illustrate the much broader range of adaptive filters applications developed in recent years. The book offers an easy to understand approach to the theory and application of adaptive filters by clearly illustrating how the theory explained in the early chapters of the book is modified for the various applications discussed in detail in later chapters. This integrated approach makes the book a valuable resource for graduate students; and the inclusion of more advanced applications including antenna arrays and wireless communications makes it a suitable technical reference for engineers, practitioners and researchers. Key features: Offers a thorough treatment of the theory of adaptive signal processing; incorporating new material on transform domain, frequency domain, subband adaptive filters, acoustic echo cancellation and active noise control. Provides an in-depth study of applications which now includes extensive coverage of OFDM, MIMO and smart antennas. Contains exercises and computer simulation problems at the end of each chapter. Includes a new companion website hosting MATLAB® simulation programs which complement the theoretical analyses, enabling the reader to gain an in-depth understanding of the behaviours and properties of the various adaptive algorithms.
Contents:
Cover
Title Page
Copyright
Contents
Preface
Acknowledgments
Chapter 1 Introduction
1.1 Linear Filters
1.2 Adaptive Filters
1.3 Adaptive Filter Structures
1.4 Adaptation Approaches
1.4.1 Approach Based on Wiener Filter Theory
1.4.2 Method of Least-Squares
1.5 Real and Complex Forms of Adaptive Filters
1.6 Applications
1.6.1 Modeling
1.6.2 Inverse Modeling
1.6.3 Linear Prediction
1.6.4 Interference Cancellation
Chapter 2 Discrete-Time Signals and Systems
2.1 Sequences and z-Transform
2.2 Parseval's Relation
2.3 System Function
2.4 Stochastic Processes
2.4.1 Stochastic Averages
2.4.2 z-Transform Representations
2.4.3 The Power Spectral Density
2.4.4 Response of Linear Systems to Stochastic Processes
2.4.5 Ergodicity and Time Averages
Problems
Chapter 3 Wiener Filters
3.1 Mean-Squared Error Criterion
3.2 Wiener Filter-Transversal, Real-Valued Case
3.3 Principle of Orthogonality
3.4 Normalized Performance Function
3.5 Extension to Complex-Valued Case
3.6 Unconstrained Wiener Filters
3.6.1 Performance Function
3.6.2 Optimum Transfer Function
3.6.3 Modeling
3.6.4 Inverse Modeling
3.6.5 Noise Cancellation
3.7 Summary and Discussion
Chapter 4 Eigenanalysis and Performance Surface
4.1 Eigenvalues and Eigenvectors
4.2 Properties of Eigenvalues and Eigenvectors
4.3 Performance Surface
Chapter 5 Search Methods
5.1 Method of Steepest Descent
5.2 Learning Curve
5.3 Effect of Eigenvalue Spread
5.4 Newton's Method
5.5 An Alternative Interpretation of Newton's Algorithm
Chapter 6 LMS Algorithm
6.1 Derivation of LMS Algorithm
6.2 Average Tap-Weight Behavior of the LMS Algorithm
6.3 MSE Behavior of the LMS Algorithm
6.3.1 Learning Curve.
6.3.2 Weight-Error Correlation Matrix
6.3.3 Excess MSE and Misadjustment
6.3.4 Stability
6.3.5 The Effect of Initial Values of Tap Weights on the Transient Behavior of the LMS Algorithm
6.4 Computer Simulations
6.4.1 System Modeling
6.4.2 Channel Equalization
6.4.3 Adaptive Line Enhancement
6.4.4 Beamforming
6.5 Simplified LMS Algorithms
6.6 Normalized LMS Algorithm
6.7 Affine Projection LMS Algorithm
6.8 Variable Step-Size LMS Algorithm
6.9 LMS Algorithm for Complex-Valued Signals
6.10 Beamforming (Revisited)
6.11 Linearly Constrained LMS Algorithm
6.11.1 Statement of the Problem and Its Optimal Solution
6.11.2 Update Equations
6.11.3 Extension to the Complex-Valued Case
Chapter 7 Transform Domain Adaptive Filters
7.1 Overview of Transform Domain Adaptive Filters
7.2 Band-Partitioning Property of Orthogonal Transforms
7.3 Orthogonalization Property of Orthogonal Transforms
7.4 Transform Domain LMS Algorithm
7.5 Ideal LMS-Newton Algorithm and Its Relationship with TDLMS
7.6 Selection of the Transform T
7.6.1 A Geometrical Interpretation
7.6.2 A Useful Performance Index
7.6.3 Improvement Factor and Comparisons
7.6.4 Filtering View
7.7 Transforms
7.8 Sliding Transforms
7.8.1 Frequency Sampling Filters
7.8.2 Recursive Realization of Sliding Transforms
7.8.3 Nonrecursive Realization of Sliding Transforms
7.8.4 Comparison of Recursive and Nonrecursive Sliding Transforms
7.9 Summary and Discussion
Chapter 8 Block Implementation of Adaptive Filters
8.1 Block LMS Algorithm
8.2 Mathematical Background
8.2.1 Linear Convolution Using the Discrete Fourier Transform
8.2.2 Circular Matrices
8.2.3 Window Matrices and Matrix Formulation of the Overlap-Save Method
8.3 The FBLMS Algorithm.
8.3.1 Constrained and Unconstrained FBLMS Algorithms
8.3.2 Convergence Behavior of the FBLMS Algorithm
8.3.3 Step-Normalization
8.3.4 Summary of the FBLMS Algorithm
8.3.5 FBLMS Misadjustment Equations
8.3.6 Selection of the Block Length
8.4 The Partitioned FBLMS Algorithm
8.4.1 Analysis of the PFBLMS Algorithm
8.4.2 PFBLMS Algorithm with M &gt
L
8.4.3 PFBLMS Misadjustment Equations
8.4.4 Computational Complexity and Memory Requirement
8.4.5 Modified Constrained PFBLMS Algorithm
8.5 Computer Simulations
Chapter 9 Subband Adaptive Filters
9.1 DFT Filter Banks
9.1.1 Weighted Overlap-Add Method for Realization of DFT Analysis Filter Banks
9.1.2 Weighted Overlap-Add Method for Realization of DFT Synthesis Filter Banks
9.2 Complementary Filter Banks
9.3 Subband Adaptive Filter Structures
9.4 Selection of Analysis and Synthesis Filters
9.5 Computational Complexity
9.6 Decimation Factor and Aliasing
9.7 Low-Delay Analysis and Synthesis Filter Banks
9.7.1 Design Method
9.7.2 Filters Properties
9.8 A Design Procedure for Subband Adaptive Filters
9.9 An Example
9.10 Comparison with FBLMS Algorithm
Chapter 10 IIR Adaptive Filters
10.1 Output Error Method
10.2 Equation Error Method
10.3 Case Study I: IIR Adaptive Line Enhancement
10.3.1 IIR ALE Filter, W(z)
10.3.2 Performance Functions
10.3.3 Simultaneous Adaptation of s and w
10.3.4 Robust Adaptation of w
10.3.5 Simulation Results
10.4 Case Study II: Equalizer Design for Magnetic Recording Channels
10.4.1 Channel Discretization
10.4.2 Design Steps
10.4.3 FIR Equalizer Design
10.4.4 Conversion from FIR into IIR Equalizer
10.4.5 Conversion from z Domain into s Domain
10.4.6 Numerical Results
10.5 Concluding Remarks
Chapter 11 Lattice Filters.
11.1 Forward Linear Prediction
11.2 Backward Linear Prediction
11.3 Relationship Between Forward and Backward Predictors
11.4 Prediction-Error Filters
11.5 Properties of Prediction Errors
11.6 Derivation of Lattice Structure
11.7 Lattice as an Orthogonalization Transform
11.8 Lattice Joint Process Estimator
11.9 System Functions
11.10 Conversions
11.10.1 Conversion Between Lattice and Transversal Predictors
11.10.2 Levinson-Durbin Algorithm
11.10.3 Extension of Levinson-Durbin Algorithm
11.11 All-Pole Lattice Structure
11.12 Pole-Zero Lattice Structure
11.13 Adaptive Lattice Filter
11.13.1 Discussion and Simulations
11.14 Autoregressive Modeling of Random Processes
11.15 Adaptive Algorithms Based on Autoregressive Modeling
11.15.1 Algorithms
11.15.2 Performance Analysis
11.15.3 Simulation Results and Discussion
Chapter 12 Method of Least-Squares
12.1 Formulation of Least-Squares Estimation for a Linear Combiner
12.2 Principle of Orthogonality
12.3 Projection Operator
12.4 Standard Recursive Least-Squares Algorithm
12.4.1 RLS Recursions
12.4.2 Initialization of the RLS Algorithm
12.4.3 Summary of the Standard RLS Algorithm
12.5 Convergence Behavior of the RLS Algorithm
12.5.1 Average Tap-Weight Behavior of the RLS Algorithm
12.5.2 Weight-Error Correlation Matrix
12.5.3 Learning Curve
12.5.4 Excess MSE and Misadjustment
12.5.5 Initial Transient Behavior of the RLS Algorithm
Chapter 13 Fast RLS Algorithms
13.1 Least-Squares Forward Prediction
13.2 Least-Squares Backward Prediction
13.3 Least-Squares Lattice
13.4 RLSL Algorithm
13.4.1 Notations and Preliminaries
13.4.2 Update Recursion for the Least-Squares Error Sums
13.4.3 Conversion Factor
13.4.4 Update Equation for Conversion Factor.
13.4.5 Update Equation for Cross-Correlations
13.4.6 RLSL Algorithm Using A Posteriori Errors
13.4.7 RLSL Algorithm with Error Feedback
13.5 FTRLS Algorithm
13.5.1 Derivation of the FTRLS Algorithm
13.5.2 Summary of the FTRLS Algorithm
13.5.3 Stabilized FTRLS Algorithm
Chapter 14 Tracking
14.1 Formulation of the Tracking Problem
14.2 Generalized Formulation of LMS Algorithm
14.3 MSE Analysis of the Generalized LMS Algorithm
14.4 Optimum Step-Size Parameters
14.5 Comparisons of Conventional Algorithms
14.6 Comparisons Based on Optimum Step-Size Parameters
14.7 VSLMS: An Algorithm with Optimum Tracking Behavior
14.7.1 Derivation of VSLMS Algorithm
14.7.2 Variations and Extensions
14.7.3 Normalization of the Parameter ρ
14.7.4 Computer Simulations
14.8 RLS Algorithm with Variable Forgetting Factor
14.9 Summary
Chapter 15 Echo Cancellation
15.1 The Problem Statement
15.2 Structures and Adaptive Algorithms
15.2.1 Normalized LMS (NLMS) Algorithm
15.2.2 Affine Projection LMS (APLMS) Algorithm
15.2.3 Frequency Domain Block LMS Algorithm
15.2.4 Subband LMS Algorithm
15.2.5 LMS-Newton Algorithm
15.2.6 Numerical Results
15.3 Double-Talk Detection
15.3.1 Coherence Function
15.3.2 Double-Talk Detection Using the Coherence Function
15.3.3 Numerical Evaluation of the Coherence Function
15.3.4 Power-Based Double-Talk Detectors
15.3.5 Numerical Results
15.4 Howling Suppression
15.4.1 Howling Suppression Through Notch Filtering
15.4.2 Howling Suppression by Spectral Shift
15.5 Stereophonic Acoustic Echo Cancellation
15.5.1 The Fundamental Problem
15.5.2 Reducing Coherence Between x1(n) and x2(n)
15.5.3 The LMS-Newton Algorithm for Stereophonic Systems
Chapter 16 Active Noise Control.
16.1 Broadband Feedforward Single-Channel ANC.
Notes:
Includes bibliographical references and index.
Description based on publisher supplied metadata and other sources.
ISBN:
1-118-59134-8
1-118-59135-6
1-118-59133-X
1-299-46521-8
OCLC:
830837650

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