Digital signal processing. An experimental approach

DOI10.1007/978-1-84800-119-0MaRDI QIDQ2465375zbMATH OpenOpenAlexFDO

Publication date 3 January 2008

Published in Signals and Communication Technology (Search for Journal in Brave)

Full work available at URL https://doi.org/10.1007/978-1-84800-119-0

Digital Signal Processing Experimental DSP signal theory

Numerical methods for discrete and fast Fourier transforms (65T50) Introductory exposition (textbooks, tutorial papers, etc.) pertaining to information and communication theory (94-01) Signal theory (characterization, reconstruction, filtering, etc.) (94A12) Sampling theory in information and communication theory (94A20)

Review text

Nowadays, Digital Signal Processing (DSP) is a widespread technique. For effective DSP, one must cross several steps: analyze the analog signals, transform them into discrete signals for which a beautiful theory has been developed (see the classic book of \textit{A. V. Oppenheim} and \textit{R. W. Schafer} [``Digital signal processing. (1975; Zbl 0369.94002)]). At this stage, the signals are viewed as real or complex valued sequences defined over the integers. The \(Z\) transform appears as an excellent tool to describe and handle them. But the signals are not yet prepared to enter the computer which in the end makes the calculations. So, it is necessary to know how the analog signals are transformed into digital ones by means of analog-to-digital converters. Finally, we must know how to modify the spectral content which is done by digital filters, here the Fast Fourier Transform (FFT) and various algorithms play an important role. Obviously, there must be a reverse path, by which we can restore the analog signals. All what was said is contained in many interesting books. The novelty of this book lies in providing the student the subsequent transforms in DSP in a concise manner, in an experimental way by a large use of MATLAB and SIMULINK. To make a solid connection with the technical world, available converters and controllers are used in the proposed simulations problems at the end of each chapter (e.g., ADUC841). To give a clear idea of what the book offers, we give its Content: Part I The Analysis of Discrete-time Signals: 1) Understanding Sampling; 2) Signal Reconstruction; 3) Time-limited Functions Are Not Band-limited; 4) Fourier Analysis of the Discrete Fourier Transform; 5) Windowing; 6) Signal Generation with the Help of MATLAB; 7) The Spectral Analysis of Random Signals; Part II Analog-to-Digital and Digital-to-Analog Converters; 8) The General Structure of Sampled-data Systems; 9) The Operational Amplifier: an Overview; 10) A Simple Digital-to-Analog Converter; 11) The Binary Weighted DAC; 12) The R-2R Ladder DAC; 13) The Successive Approximation Analog-to-Digital Converter; 14) The Single- and Dual-slope Analog-to-Digital Converters; 15) The Pipelined A/D; 16) Resistor-chain Converters; 17) Sigma-Delta Converters; Part III 18) Discrete-time Systems and the \(Z\)-Transform; 19) Filter Types; 20) When to use C (Rather than Assembly Language); 21) Two Simple FIR Filters; 22) Very-narrow-band Filters; 23) Design of IIR Digital Filters: The Old-Fashioned Way; 24) New Filters From Old; 25) Implementing an IIR Digital Filter; 26) IIR Filter Design Using MATLAB; 27) Group Delay and Phase Delay Filters; 28) Design of FIR Filters; 29) Implementing a Hilbert Filter; 30) The Goertzel Algorithm; References; Index. The book is accessible to students familiar with the main facts from a Signals-and-Systems course. We think the book gives a new and interesting approach for mastering DSP in an experimental way; so we conclude the present book will soon become a reference book in learning DSP.

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