Outline: 1. Introduction 2. Digital processing of continuous-time signals • Retition: Sampling and sampling theorem • Quantization • AD-and DA-conversion 3. DFT and FFT • Leakage effect • Windowing • FFT structure 4. Digital filters • FIR-filters: Structures, linear phase filters, least-squares frequency domain design, Chebyshev approximation • IIR-filters: Structures, classical analog lowpass filter approximations, conversion to digital transfer functions • Finite word-length effects 5. Multirate digital signal processing • Decimation and interpolation • Filters in sampling rate alteration systems • Polyphase decomposition and efficient structures • Digital filter banks Parts of this textbook have been realized in close collaboration with Dr. Joerg Kliewer whom I warmly thank. 6. Spectral estimation • Periodogram, Bartlett's method, Welch's method, Blackman-Tukey method • ARMA modeling, Yule-Walker equation and solution Literature •
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In many practical applications of DSP, there is a problem of changing the sampling rate of a signal, either increasing it or decreasing it by some amount. For example, Telecommunication system transmits and receives different types of signals (e.g. fax, speech, video, etc), so there is a requirement to process the various signals at the different rates with corresponding bandwidth of the signals. The process of converting a signal from a given rate to a different rate is called as " sampling rate conversion " and the systems that employ multiple sampling rates in the processing of digital signals are called as " Multirate DSP systems ". ". Digital audio engineering is an area that has benefited significantly from Multirate techniques. For example, they are used in the compact disc player to simplify the D/A conversion processes, while at the same time maintaining the quality of the reproduced sound. This project is based on to design and development of highly efficient Multirate digital filter structures for filtration of noisy signal in which the high sampling rate is decreased to the desired lower sampling rate. Digital filters such as IIR filter, FIR filter and CIC filter is designed and taking their performance in case of magnitude response, step response, impulse response, pole-zero plot, filter coefficients, storage requirements, hardware requirements, number of stages and simulated waveforms for same input specifications. Finally compare them and discuss the advantages and limitations of these filters. These filter structures designed in the Simulink model in Matlab 2012a environment.
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CreateSpace Independent Publishing Platform, An Amazon.com Company, ISBN-13 : 978-1514179987
This book is a result of author's thirty-three years of experience in teaching and research in signal processing.The book will guide you from a review of continuous-time signals and systems, through the world of digital signal processing, up to some of the most advanced theory and techniques in adaptive systems, time-frequency analysis, and sparse signal processing. It provides simple examples and explanations for each, including the most complex transform, method, algorithm or approach presented in the book. The most sophisticated results in signal processing theory are illustrated on simple numerical examples. The book is written for students learning digital signal processing and for engineers and researchers refreshing their knowledge in this area. The selected topics are intended for advanced courses and for preparing the reader to solve problems in some of the state of art areas in signal processing.
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IEEE Transactions on Circuits and Systems
