# DSP for MATLAB and LabVIEW. Volume IV, LMS adaptive filtering [electronic resource] / Forester W. Isen.

Material type: TextSeries: Synthesis lectures on signal processing ; # 7.Publication details: San Rafael, Calif. (1537 Fourth Street, San Rafael, CA 94901 USA) :: Morgan & Claypool Publishers,, c2009Description: 1 electronic text (xvii, 109 p. : ill.) : digital fileISBN: 9781598299007 (electronic bk.); 9781598298994 (pbk.)Other title: LMS adaptive filteringUniform titles: Synthesis digital library of engineering and computer science. Subject(s): MATLAB | LabVIEW | Signal processing -- Digital techniques | Adaptive filters | LMS adaptive filter | Least mean square | Active noise cancellation (ANC) | Deconvolution | Equalization | Inverse filtering | Interference cancellation | Echo cancellation | Dereverberation | Adaptive Line Enhancer (ALE)DDC classification: 621.3822 LOC classification: TK5102.9 | .I8334 2009Online resources: Abstract with links to resource Also available in print.Item type | Current library | Call number | Status | Date due | Barcode |
---|---|---|---|---|---|

Ebooks | Indian Institute of Technology Delhi - Central Library | Available |

Mode of access: World Wide Web.

System requirements: Adobe Acrobat reader.

Part of: Synthesis digital library of engineering and computer science.

Series from website.

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 -- Contrast-true MSE -- LMS adaptive FIR summary -- References -- Exercises -- Applied adaptive filtering -- Overview -- Active noise cancellation -- System modeling -- Echo cancellation -- Single-H -- Dual-H -- 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 -- Single-line function calls -- Multi-line M-code examples --How to successfully copy-and-paste M-code -- Learning to use M-code -- What you need with MATLAB and LabVIEW -- Vector/matrix operations in M-code -- Row and column vectors -- Vector products -- Inner product -- Outer product -- Product of corresponding values -- Matrix multiplied by a vector or matrix -- Matrix inverse and pseudo-inverse -- Biography.

Abstract freely available; full-text restricted to subscribers or individual document purchasers.

Compendex

INSPEC

Google scholar

Google book search

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 narrow-band signals, and various topologies such as ANC (Active Noise Cancelling) or system modeling, Noise Cancellation, Interference Cancellation, Echo Cancellation (with single- and dual-H 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., analog-to-digital and digital-to-analog conversion), aliasing, the Nyquist rate, normalized frequency, sample rate conversion, and Mu-law 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 z-Transform (including definition and properties, the inverse z-transform, frequency response via z-transform, 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 windowed-ideal-lowpass filter, FIR highpass, bandpass, and bandstop filter design from windowed-ideal lowpass filters, FIR design using the transition-band-optimized Frequency Sampling technique (implemented by Inverse-DFT 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.

Also available in print.

Title from PDF t.p. (viewed on January 8, 2009).

There are no comments on this title.