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The Resource DSP for MATLAB and LabVIEW, Volume IV, LMS adaptive filtering, Forester W. Isen, (electronic resource)
DSP for MATLAB and LabVIEW, Volume IV, LMS adaptive filtering, Forester W. Isen, (electronic resource)
Resource Information
The item DSP for MATLAB and LabVIEW, Volume IV, LMS adaptive filtering, Forester W. Isen, (electronic resource) represents a specific, individual, material embodiment of a distinct intellectual or artistic creation found in University of Oklahoma Libraries.This item is available to borrow from all library branches.
Resource Information
The item DSP for MATLAB and LabVIEW, Volume IV, LMS adaptive filtering, Forester W. Isen, (electronic resource) represents a specific, individual, material embodiment of a distinct intellectual or artistic creation found in University of Oklahoma Libraries.
This item is available to borrow from all library branches.
 Summary
 This book is Volume IV of the series DSP for MATLAB and LabVIEW. Volume IV is an introductory treatment of LMS Adaptive Filtering and applications, and covers cost functions, performance surfaces, coefficient perturbation to estimate the gradient, the LMS algorithm, response of the LMS algorithm to narrowband signals, and various topologies such as ANC (Active Noise Cancelling) or system modeling, Noise Cancellation, Interference Cancellation, Echo Cancellation (with single and dualH topologies), and Inverse Filtering/Deconvolution. The entire series consists of four volumes that collectively cover basic digital signal processing in a practical and accessible manner, but which nonetheless include all essential foundation mathematics. As the series title implies, the scripts (of which there are more than 200) described in the text and supplied in code form (available via the internet at http://www.morganclaypool.com/page/isen) will run on both MATLAB and LabVIEW. The text for all volumes contains many examples, and many useful computational scripts, augmented by demonstration scripts and LabVIEW Virtual Instruments (VIs) that can be run to illustrate various signal processing concepts graphically on the user's computer screen. Volume I consists of four chapters that collectively set forth a brief overview of the field of digital signal processing, useful signals and concepts (including convolution, recursion, difference equations, LTI systems, etc), conversion from the continuous to discrete domain and back (i.e., analogtodigital and digitaltoanalog conversion), aliasing, the Nyquist rate, normalized frequency, sample rate conversion, and Mulaw compression, and signal processing principles including correlation, the correlation sequence, the Real DFT, correlation by convolution, matched filtering, simple FIR filters, and simple IIR filters. Chapter 4 of Volume I, in particular, provides an intuitive or "first principle" understanding of how digital filtering and frequency transforms work. Volume II provides detailed coverage of discrete frequency transforms, including a brief overview of common frequency transforms, both discrete and continuous, followed by detailed treatments of the Discrete Time Fourier Transform (DTFT), the zTransform (including definition and properties, the inverse ztransform, frequency response via ztransform, and alternate filter realization topologies including Direct Form, Direct Form Transposed, Cascade Form, Parallel Form, and Lattice Form), and the Discrete Fourier transform (DFT) (including Discrete Fourier Series, the DFTIDFT pair, DFT of common signals, bin width, sampling duration, and sample rate, the FFT, the Goertzel Algorithm, Linear, Periodic, and Circular convolution, DFT Leakage, and computation of the Inverse DFT). Volume III covers digital filter design, including the specific topics of FIR design via windowedideallowpass filter, FIR highpass, bandpass, and bandstop filter design from windowedideal lowpass filters, FIR design using the transitionbandoptimized Frequency Sampling technique (implemented by InverseDFT or Cosine/Sine Summation Formulas), design of equiripple FIRs of all standard types including Hilbert transformers and Differentiators via the Remez Exchange Algorithm, design of Butterworth, Chebyshev (Types I and II), and Elliptic analog prototype lowpass filters, conversion of analog lowpass prototype filters to highpass, bandpass, and bandstop filters, and conversion of analog filters to digital filters using the Impulse Invariance and Bilinear Transform techniques. Certain filter topologies specific to FIRs are also discussed, as are two simple FIR types, the Comb and Moving Average filters
 Language
 eng
 Extent
 1 electronic text (xvii, 109 p. : ill.)
 Note

 Part of: Synthesis digital library of engineering and computer science
 Title from PDF t.p. (viewed on January 8, 2009)
 Series from website
 Contents

 Introduction to LMS adaptive filtering
 Overview
 In previous volumes
 In this volume
 In this chapter
 Software for use with this book
 Cost function
 Performance surface
 Coefficient perturbation
 Method of steepest descent
 Two variable performance surface
 An improved gradient search method
 LMS used in an FIR
 Typical arrangements
 Derivation
 Limitation on Mu
 NLMS algorithm
 Contrasttrue MSE
 LMS adaptive FIR summary
 References
 Exercises
 Applied adaptive filtering
 Overview
 Active noise cancellation
 System modeling
 Echo cancellation
 SingleH
 DualH
 Sparse computation
 Periodic component elimination or enhancement
 Interference cancellation
 Equalization/deconvolution
 Deconvolution of a reverberative signal
 Simulation
 Spectral effect of reverberation
 Estimating delay
 Estimating decay rate
 Deconvolution
 References
 Exercises
 Software for use with this book
 File types and naming conventions
 Downloading the software
 Using the software
 Singleline function calls
 Multiline Mcode examples How to successfully copyandpaste Mcode
 Learning to use Mcode
 What you need with MATLAB and LabVIEW
 Vector/matrix operations in Mcode
 Row and column vectors
 Vector products
 Inner product
 Outer product
 Product of corresponding values
 Matrix multiplied by a vector or matrix
 Matrix inverse and pseudoinverse
 Biography
 Isbn
 9781598299007
 Label
 DSP for MATLAB and LabVIEW, Volume IV, LMS adaptive filtering
 Title
 DSP for MATLAB and LabVIEW
 Title number
 Volume IV
 Title part
 LMS adaptive filtering
 Statement of responsibility
 Forester W. Isen
 Title variation
 LMS adaptive filtering
 Language
 eng
 Summary
 This book is Volume IV of the series DSP for MATLAB and LabVIEW. Volume IV is an introductory treatment of LMS Adaptive Filtering and applications, and covers cost functions, performance surfaces, coefficient perturbation to estimate the gradient, the LMS algorithm, response of the LMS algorithm to narrowband signals, and various topologies such as ANC (Active Noise Cancelling) or system modeling, Noise Cancellation, Interference Cancellation, Echo Cancellation (with single and dualH topologies), and Inverse Filtering/Deconvolution. The entire series consists of four volumes that collectively cover basic digital signal processing in a practical and accessible manner, but which nonetheless include all essential foundation mathematics. As the series title implies, the scripts (of which there are more than 200) described in the text and supplied in code form (available via the internet at http://www.morganclaypool.com/page/isen) will run on both MATLAB and LabVIEW. The text for all volumes contains many examples, and many useful computational scripts, augmented by demonstration scripts and LabVIEW Virtual Instruments (VIs) that can be run to illustrate various signal processing concepts graphically on the user's computer screen. Volume I consists of four chapters that collectively set forth a brief overview of the field of digital signal processing, useful signals and concepts (including convolution, recursion, difference equations, LTI systems, etc), conversion from the continuous to discrete domain and back (i.e., analogtodigital and digitaltoanalog conversion), aliasing, the Nyquist rate, normalized frequency, sample rate conversion, and Mulaw compression, and signal processing principles including correlation, the correlation sequence, the Real DFT, correlation by convolution, matched filtering, simple FIR filters, and simple IIR filters. Chapter 4 of Volume I, in particular, provides an intuitive or "first principle" understanding of how digital filtering and frequency transforms work. Volume II provides detailed coverage of discrete frequency transforms, including a brief overview of common frequency transforms, both discrete and continuous, followed by detailed treatments of the Discrete Time Fourier Transform (DTFT), the zTransform (including definition and properties, the inverse ztransform, frequency response via ztransform, and alternate filter realization topologies including Direct Form, Direct Form Transposed, Cascade Form, Parallel Form, and Lattice Form), and the Discrete Fourier transform (DFT) (including Discrete Fourier Series, the DFTIDFT pair, DFT of common signals, bin width, sampling duration, and sample rate, the FFT, the Goertzel Algorithm, Linear, Periodic, and Circular convolution, DFT Leakage, and computation of the Inverse DFT). Volume III covers digital filter design, including the specific topics of FIR design via windowedideallowpass filter, FIR highpass, bandpass, and bandstop filter design from windowedideal lowpass filters, FIR design using the transitionbandoptimized Frequency Sampling technique (implemented by InverseDFT or Cosine/Sine Summation Formulas), design of equiripple FIRs of all standard types including Hilbert transformers and Differentiators via the Remez Exchange Algorithm, design of Butterworth, Chebyshev (Types I and II), and Elliptic analog prototype lowpass filters, conversion of analog lowpass prototype filters to highpass, bandpass, and bandstop filters, and conversion of analog filters to digital filters using the Impulse Invariance and Bilinear Transform techniques. Certain filter topologies specific to FIRs are also discussed, as are two simple FIR types, the Comb and Moving Average filters
 Additional physical form
 Also available in print.
 Cataloging source
 CaBNvSL
 Citation source

 Compendex
 INSPEC
 Google scholar
 Google book search
 http://bibfra.me/vocab/lite/collectionName
 Synthesis digital library of engineering and computer science
 http://library.link/vocab/creatorDate
 1949
 http://library.link/vocab/creatorName
 Isen, Forester W.
 Dewey number
 621.3822
 Illustrations
 illustrations
 Index
 no index present
 LC call number
 TK5102.9
 LC item number
 .I8334 2009
 Literary form
 non fiction
 Nature of contents

 dictionaries
 abstracts summaries
 Series statement
 Synthesis lectures on signal processing,
 Series volume
 # 7
 http://library.link/vocab/subjectName

 Signal processing
 Adaptive filters
 Target audience

 adult
 specialized
 Label
 DSP for MATLAB and LabVIEW, Volume IV, LMS adaptive filtering, Forester W. Isen, (electronic resource)
 Note

 Part of: Synthesis digital library of engineering and computer science
 Title from PDF t.p. (viewed on January 8, 2009)
 Series from website
 Color
 multicolored
 Contents
 Introduction to LMS adaptive filtering  Overview  In previous volumes  In this volume  In this chapter  Software for use with this book  Cost function  Performance surface  Coefficient perturbation  Method of steepest descent  Two variable performance surface  An improved gradient search method  LMS used in an FIR  Typical arrangements  Derivation  Limitation on Mu  NLMS algorithm  Contrasttrue MSE  LMS adaptive FIR summary  References  Exercises  Applied adaptive filtering  Overview  Active noise cancellation  System modeling  Echo cancellation  SingleH  DualH  Sparse computation  Periodic component elimination or enhancement  Interference cancellation  Equalization/deconvolution  Deconvolution of a reverberative signal  Simulation  Spectral effect of reverberation  Estimating delay  Estimating decay rate  Deconvolution  References  Exercises  Software for use with this book  File types and naming conventions  Downloading the software  Using the software  Singleline function calls  Multiline Mcode examples How to successfully copyandpaste Mcode  Learning to use Mcode  What you need with MATLAB and LabVIEW  Vector/matrix operations in Mcode  Row and column vectors  Vector products  Inner product  Outer product  Product of corresponding values  Matrix multiplied by a vector or matrix  Matrix inverse and pseudoinverse  Biography
 Dimensions
 unknown
 Extent
 1 electronic text (xvii, 109 p. : ill.)
 File format
 multiple file formats
 Form of item
 electronic
 Governing access note
 Abstract freely available; fulltext restricted to subscribers or individual document purchasers
 Isbn
 9781598299007
 Isbn Type
 (electronic bk.)
 Other control number
 10.2200/S00164ED1V01Y200811SPR007
 Other physical details
 digital file.
 Reformatting quality
 access
 Specific material designation
 remote
 System control number

 317996201okla_normanlaw
 (SIRSI)3179962
 (Sirsi) i9781598299007
 (CaBNvSL)gtp00532332
 (OCoLC)299608625
 System details

 Mode of access: World Wide Web
 System requirements: Adobe Acrobat reader
 Label
 DSP for MATLAB and LabVIEW, Volume IV, LMS adaptive filtering, Forester W. Isen, (electronic resource)
 Note

 Part of: Synthesis digital library of engineering and computer science
 Title from PDF t.p. (viewed on January 8, 2009)
 Series from website
 Color
 multicolored
 Contents
 Introduction to LMS adaptive filtering  Overview  In previous volumes  In this volume  In this chapter  Software for use with this book  Cost function  Performance surface  Coefficient perturbation  Method of steepest descent  Two variable performance surface  An improved gradient search method  LMS used in an FIR  Typical arrangements  Derivation  Limitation on Mu  NLMS algorithm  Contrasttrue MSE  LMS adaptive FIR summary  References  Exercises  Applied adaptive filtering  Overview  Active noise cancellation  System modeling  Echo cancellation  SingleH  DualH  Sparse computation  Periodic component elimination or enhancement  Interference cancellation  Equalization/deconvolution  Deconvolution of a reverberative signal  Simulation  Spectral effect of reverberation  Estimating delay  Estimating decay rate  Deconvolution  References  Exercises  Software for use with this book  File types and naming conventions  Downloading the software  Using the software  Singleline function calls  Multiline Mcode examples How to successfully copyandpaste Mcode  Learning to use Mcode  What you need with MATLAB and LabVIEW  Vector/matrix operations in Mcode  Row and column vectors  Vector products  Inner product  Outer product  Product of corresponding values  Matrix multiplied by a vector or matrix  Matrix inverse and pseudoinverse  Biography
 Dimensions
 unknown
 Extent
 1 electronic text (xvii, 109 p. : ill.)
 File format
 multiple file formats
 Form of item
 electronic
 Governing access note
 Abstract freely available; fulltext restricted to subscribers or individual document purchasers
 Isbn
 9781598299007
 Isbn Type
 (electronic bk.)
 Other control number
 10.2200/S00164ED1V01Y200811SPR007
 Other physical details
 digital file.
 Reformatting quality
 access
 Specific material designation
 remote
 System control number

 317996201okla_normanlaw
 (SIRSI)3179962
 (Sirsi) i9781598299007
 (CaBNvSL)gtp00532332
 (OCoLC)299608625
 System details

 Mode of access: World Wide Web
 System requirements: Adobe Acrobat reader
Subject
 Active noise cancellation (ANC)
 Adaptive Line Enhancer (ALE)
 Adaptive filters
 Deconvolution
 Dereverberation
 Echo cancellation
 Equalization
 Interference cancellation
 Inverse filtering
 LMS adaptive filter
 LabVIEW
 Least mean square
 MATLAB
 Signal processing  Digital techniques
Member of
 Synthesis digital library of engineering and computer science
 Synthesis lectures on signal processing (Online), # 7
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