Now published as a second edition 380 page paperback, ISBN 0-333-96356-3 24.99 UK pounds, Palgrave, Houndmills, Basingstoke, RG21 6XS, United Kingdom. fax +44 1256 302945.
If you have a postscript reader you can get this page with added figures here
This is a revised and updated text which covers the basic principles and operation of digital signal processing. It is prepared to give the student the essentials of this mathematical subject so that it can be easily understood and assimilated. The text concentrates on discrete-time sampled data systems covering initially digital filters and discrete Fourier transforms. These are then extended into adaptive filters and spectrum analysers with the minimum of mathematical derivation, concentrating on demonstrating the performance which is achievable from these processors in communications and radar system applications.
This book is aimed at readers who are completing a graduate level B.Eng./M.Eng. first degree course in Electronics or Electrical Engineering. It is assumed that these readers will have competence in the mathematical concepts explored in earlier courses in basic mathematical techniques, to handle comfortably the material.
This new textbook is also appropriate to M.Sc. courses in signals and systems and signal processing and for professional engineers who wish to have a simple easy to read reference book on DSP techniques.
The book contains self-assessment questions with associated solutions and numerous worked examples on processor design and performance simulation. Many of these examples are augmented by animated simulations available to students via world wide web access.
We have deliberately extended our coverage of signal processing to include the practical aspects of systems. With this balance between theory, applications and systems implementation we hope that this text will be readily used both in academia and in the rapidly growing communications industry.
For assessment we have mounted two specimen chapters in postscript below:
Chapter 1 SIGNAL REPRESENTATION AND SYSTEM RESPONSE
Chapter 2 TIME DOMAIN DESCRIPTION AND CONVOLUTION
Chapter 3 TRANSFER FUNCTIONS AND SYSTEM CHARACTERISATION
Chapter 4 SAMPLED DATA SYSTEMS AND THE z-TRANSFORM
Chapter 5 INFINITE IMPULSE RESPONSE DIGITAL FILTERS
Chapter 6 FINITE IMPULSE RESPONSE DIGITAL FILTERS
Chapter 7 RANDOM SIGNAL ANALYSIS
Chapter 8 ADAPTIVE FILTERS
Chapter 9 THE FOURIER TRANSFORM AND SPECTRAL ANALYSIS
Chapter 10 THE FAST FOURIER TRANSFORM
Chapter 11 MULTIRATE SIGNAL PROCESSING
Appendix A MATRIX THEORY
Appendix B TABLES OF COMMONLY USED TRANSFORMS
BIBLIOGRAPHY and REFERENCES
MATLABTM source code is provided on an open access basis to assist the instructor with presentation of the material and the student in understanding of the material. In general the source code provides a computer animation of some of the figures in the book. For example the m-file "fig1_4.m" contains MATLAB code which produces an animation of the complex phasor of Figure 1.4. These are identified within the text by the square symbol.
Further JAVA DSP demonstrations are available from Dr David Laurenson and Mike Jackson here in Edinburgh.
Solutions to the tutorials and slide copies are available to authorised instructors, in postcript form, by entering the username and password at the prompt.
Edinburgh | Bernard Mulgrew, Peter Grant and John Thompson |
October 2002 |