Digital Filters And Signal Processing

Digital Filters and Signal Processing, Third Edition ... with MATLAB Exercises presents a general survey of digital signal processing concepts, design methods, and implementation considerations, with an emphasis on digital filters. It is suitable as a textbook for senior undergraduate or first-year graduate courses in digital signal processing. While mathematically rigorous, the book stresses an intuitive understanding of digital filters and signal processing systems, with numerous realistic and relevant examples. Hence, practicing engineers and scientists will also find the book to be a most useful reference. The Third Edition contains a substantial amount of new material including, in particular, the addition of MATLAB exercises to deepen the students' understanding of basic DSP principles and increase their proficiency in the application of these principles. The use of the exercises is not mandatory, but is highly recommended. Other new features include: normalized frequency utilized in the DTFT, e.g., X(ejomega); new computer generated drawings and MATLAB plots throughout the book; Chapter 6 on sampling the DTFT has been completely rewritten; expanded coverage of Types I-IV linear-phase FIR filters; new material on power and doubly-complementary filters; new section on quadrature-mirror filters and their application in filter banks; new section on the design of maximally-flat FIR filters; new section on roundoff-noise reduction using error feedback; and many new problems added throughout.

Digital filters, together with signal processing, are being employed in the new technologies and information systems, and are implemented in different areas and applications. Digital filters and signal processing are used with no costs and they can be adapted to different cases with great flexibility and reliability. This book presents advanced developments in digital filters and signal process methods covering different cases studies. They present the main essence of the subject, with the principal approaches to the most recent mathematical models that are being employed worldwide.

A complete up-to-date reference for advanced analog and digital IIR filter design rooted in elliptic functions. "Revolutionary" in approach, this book opens up completely new vistas in basic analog and digital IIR filter design--regardless of the technology. By introducing exceptionally elegant and creative mathematical stratagems (e.g., accurate replacement of Jacobi elliptic functions by functions comprising polynomials, square roots, and logarithms), optimization routines carried out with symbolic analysis by "Mathematica," and the advance filter design software of MATLAB, it shows readers how to design many types of filters that cannot be designed using conventional techniques. The filter design algorithms can be directly programed in any language or environment such as Visual BASIC, Visual C, Maple, DERIVE, or MathCAD. Signals; Systems; Transforms; Classical Analog Filter Design; Advanced Analog Filter Design Case Studies; Advanced Analog Filter Design Algorithms; Multi-criteria Optimization of Analog Filter Designs; Classical Digital Filter Design; Advanced Digital Filter Design Case Studies; Advanced Digital Filter Design Algorithms; Multi-criteria Optimization of Digital Filter Designs; Elliptic Functions; Elliptic Rational Function.

From industrial and teaching experience the authors provide a blend of theory and practice of digital signal processing (DSP) for advanced undergraduate and post-graduate engineers reading electronics. This fast-moving, developing area is driven by the information technology revolution. It is a source book in research and development for embedded system design engineers, designers in real-time computing, and applied mathematicians who apph DSP techniques in telecommunications, aerospace (control systems), satellite communications, instrumentation, and medical technology (ultrasound and magnetic resonance imaging).The book is particularly useful at the hardware end of DSP, with its emphasis on practical I)SP devices and the integration of basic processes with appropriate software. It is unique to find in one volume the implementation of the equations as algorithms, not only in \IATLAB but right up to a working DSP-based scheme. Other relevant architectural features include number representations, multiply-accumulate, special addressing modes, zero overhead iteration schemes. and single and multiple nlicroprocessors which will allow the readers to compare and understand both current processors and future DSP developments.Fundamental signal processing procedures are introduced and developed: also convolution. correlation, the Discrete Fourier Transform and its fast computation algorithms. Then follo finite impulse response (FIR) filters, infinite impulse response (IlR) filters, multirate filters, adaptive filters, and topics from communication and control. I)esign examples are given in all of these cases, taken through an algorithm testing stage using MATLAB. The design of the latter. using C language models, is explained together with the experimental results of real time integer implementations.Academic prerequisites are first and second year university mathematics, an introductor knowledge of circuit theor ‘and microprocessors. and C Language. Provides an unusual blend of theory and practice of digital signal processing (DSP) Discusses fundamental signal processing procedures, convolution, correlation, the Discrete Fourier Transform and its fast computation algorithms Includes number representations, multiply-accumulate, special addressing modes, zero overhead iteration schemes, and single and multiple instructions

Dealing with digital filtering methods for 1-D and 2-D signals, this book provides the theoretical background in signal processing, covering topics such as the z-transform, Shannon sampling theorem and fast Fourier transform. An entire chapter is devoted to the design of time-continuous filters which provides a useful preliminary step for analog-to-digital filter conversion. Attention is also given to the main methods of designing finite impulse response (FIR) and infinite impulse response (IIR) filters. Bi-dimensional digital filtering (image filtering) is investigated and a study on stability analysis, a very useful tool when implementing IIR filters, is also carried out. As such, it will provide a practical and useful guide to those engaged in signal processing.

This text provides a broad introduction to the field of digital signal processing and contains sufficient material for a two-semester sequence in this multifaceted subject. It is also written with the practicing engineer or scientist in mind, having many observations and examples of practical significance drawn from the author's industrial experience. The first semester, at the junior, senior, or first-year graduate level, could cover chapters 2 through 7 with topics perhaps from chapters 8 and 9, depending upon the background of the students. The only requisite background is linear systems theory for continuous-time systems, including Fourier and Laplace trans forms. Many students will also have had some previous exposure to discrete-time systems, in which case chapters 2 through 4 may serve to review and expand that preparation. Note, in particular, that knowledge of probability theory and random processes is not required until chapters 10 and 11, except for section 7. 6 on the periodogram. A second, advanced course could utilize material from chapters 8 through 13. A comprehensive one-semester course for suitably prepared graduate students might cover chapters 4 through 9 and additional topics from chapters 10 through 13. Sections marked with a dagger Ct) cover advanced or specialized topics and may be skipped without loss of continuity. Notable features of the book include the following: 1. Numerous useful filter examples early in the text in chapters 4 and 5. 2. State-space representation and structures in chapters 4 and 11.

A digital filter can be pictured as a "black box" that accepts a sequence of numbers and emits a new sequence of numbers. In digital audio signal processing applications, such number sequences usually represent sounds. For example, digital filters are used to implement graphic equalizers and other digital audio effects. This book is a gentle introduction to digital filters, including mathematical theory, illustrative examples, some audio applications, and useful software starting points. The theory treatment begins at the high-school level, and covers fundamental concepts in linear systems theory and digital filter analysis. Various "small" digital filters are analyzed as examples, particularly those commonly used in audio applications. Matlab programming examples are emphasized for illustrating the use and development of digital filters in practice.