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DSP for MATLAB and LabVIEW. Volume I, Fundamentals of discrete signal processing [electronic resource] / Forester W. Isen

Author:
Isen, Forester W. (Forester William), 1949-
Additional Titles:
Fundamentals of discrete signal processing
Published:
San Rafael, Calif. (1537 Fourth Street, San Rafael, CA 94901 USA) : Morgan & Claypool Publishers, [2008]
Copyright Date:
©2008
Physical Description:
1 electronic text (xvi, 199 pages : illustrations) : digital file
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Restrictions on Access:
Abstract freely available; full-text restricted to subscribers or individual document purchasers.
Contents:
An overview of DSP -- Signals,waves, and digital processing -- Advantages of digital processing -- DSP nomenclature and topics -- Time domain processing -- Frequency transforms -- Frequency domain processing -- Organization of this volume of the series -- Contents of volume II in the series -- Contents of volume III in the series -- Contents of volume IV in the series -- Discrete signals and concepts -- Overview -- Software for use with this book -- Discrete sequence notation -- Useful signals, sequences, and concepts -- Sine and cosine -- Sequence and time position vector -- The unit impulse (Delta) function -- The unit step function -- Real exponential sequence -- Periodic sequences -- Harmonic sequences -- Folded sequence -- Even and odd decomposition -- Geometric sequence -- Random or noise sequences -- Chirp -- Complex power sequence -- Specific frequency generation -- Energy of a signal -- Power of a signal -- Discrete time systems -- LTI systems -- Method of analysis of LTI systems -- Graphic method -- A few properties of convolution -- Stability and causality -- LTI system as a filter -- Difference equations -- References -- Exercises -- Sampling and binary representation -- Overview -- Software for use with this book -- Aliasing -- Folding diagram -- Normalized frequency -- Conversion from decimal to binary format -- Quantization error -- Binary-to-decimal via algorithm -- Decimal-to-binary via algorithm -- Offset to input to reduce error -- Clipping -- Offset and sign-plus-magnitude -- DAC with variable LSB -- ADC with variable LSB -- Zero-order hold conversion -- Changing sample rate -- Interpolation -- Decimation -- Combining interpolation with decimation -- Bandlimited interpolation using the sinc function -- Efficient methods for changing sample rate -- Frequency generation -- Variable SR -- Constant SR, integral decimation -- Constant SR, non-integral decimation -- Compression -- References -- Exercises -- Transform and filtering principles -- Overview -- Software for use with this book -- Correlation at the Zeroth lag (CZL) -- CZL equal-frequency sine/cosine orthogonality -- CZL of sinusoid pairs, arbitrary frequencies -- Orthogonality of complex exponentials -- Sum of samples of single complex exponential -- Identifying specific sinusoids in a signal -- Single frequency correlation and reconstruction -- Multiple frequency correlation and reconstruction -- Using orthogonality in signal transmission -- The correlation sequence -- Correlation via convolution -- Matched filtering -- Estimating frequency response -- Frequency selectivity -- Single correlator filters of arbitrary frequency -- Multiple correlator filters -- Deficiencies of simple filters -- Sinusoidal fidelity -- Determination of time delay using correlation -- The single-pole IIR -- Physical arrangement -- Recursive computation -- M-code implementation -- Impulse response, unit step response, and stability -- Leaky integrator -- Frequency response -- Complex conjugate poles -- 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 -- Complex numbers -- Definition -- Rectangular v. polar -- Addition and subtraction -- Multiplication -- Rectangular coordinates --Polar coordinates -- Division and complex conjugate -- Using rectangular coordinates -- Using polar coordinates -- Polar notation using cosine and sine -- The complex exponential -- Uses for signal processing.
Summary:
This book is Volume I of the series DSP for MATLAB and LabVIEW. 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. Volume I consists of four chapters. The first chapter gives a brief overview of the field of digital signal processing. This is followed by a chapter detailing many useful signals and concepts, including convolution, recursion, difference equations, LTI systems, etc. The third chapter covers 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, conversion from one sample rate to another, waveform generation at various sample rates from stored wave data, and Mu-law compression. The fourth and final chapter of the present volume introduces the reader to many important principles of signal processing, including correlation, the correlation sequence, the Real DFT, correlation by convolution, matched filtering, simple FIR filters, and simple IIR filters. Chapter 4, in particular, provides an intuitive or "first principle" understanding of how digital filtering and frequency transforms work, preparing the reader for Volumes II and III, which provide, respectively, detailed coverage of discrete frequency transforms (including the Discrete Time Fourier Transform, the Discrete Fourier Transform, and the z-Transform) and digital filter design (FIR design using Windowing, Frequency Sampling, and Optimum Equiripple techniques, and Classical IIR design). Volume IV, the culmination of the series, is an introductory treatment of LMS Adaptive Filtering and applications.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.
Subject(s):
ISBN:
9781598298918 (electronic bk.)
9781598298901 (pbk.)
Related Titles:
Synthesis digital library of engineering and computer science
Note:
Part of: Synthesis digital library of engineering and computer science.
Title from PDF t.p. (viewed on December 10, 2008).
Series from website.
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Also available in print.
Technical Details:
Mode of access: World Wide Web.
System requirements: Adobe Acrobat reader.
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View MARC record | catkey: 7058443