The best-selling A First Course in Fuzzy Logic provides an introduction to the theory of fuzzy sets and to its various areas of application in engineering and decision making. This third edition has been completely revised and expanded with new information on copulas and t-norms and features a new section on type-2 fuzzy sets. It also includes expanded information on generalized modus ponens, the compositional rule of inference, possibility theory, partial knowledge, and fuzzy integrals. With exercises at the end of each chapter, this book remains an essential text for students from such fields as artificial intelligence, computer science, engineering, cognitive science, and probability and statistics.

This text provides a self-contained, "no frills," mathematically rigorous derivation from first principles of all basic Kalman filter recursions. This approach relies on a pared-down version of more general state-space models found most often in the literature. Such simplification saves notational complexity without sacrificing conceptual understanding. The rigor found in the book ensures a fundamental understanding of how the Kalman filter actually works, which builds confidence for those employing the filter in their research and writing code to implement it in practice. The author provides implementations of the Kalman filter in Java available for download from his Web site.

A Practical Handbook of Speech Coders offers in-depth treatment of the basics of speech coding plus the innovations to the basic methods that make the coders useful and efficient. It describes the fundamentals of auditory information processing and how they relate to speech coding, and shows readers how to evaluate the strengths and weaknesses of all publicly available codes and choose the right one. It explains how to measure the quality of speech coders with objective, subjective, and perceptual measures. The book also shows engineers how to tailor existing speech coders and provides the building blocks to create new coders.

Aims to bridge the gap between parallel computer architectures and the creation of parallel digital signal processing (DSP) algorithms. This work offers an approach to digital signal processing utilizing the unified signal algebra environment to develop naturally occurring parallel DSP algorithms.;College or university book shops may order five or more copies at a special student price. Price is available on request.

Telecommunication systems and human-machine interfaces have begun using multiple microphones and loudspeakers to make conversation and interaction more lifelike, and hence more efficient. This development raises a variety of acoustic signal processing problems under multiple-input multiple-output (MIMO) scenarios, encompassing distant speech acquisition, sound source localization and tracking, echo and noise control, source separation and speech dereverberation, and many others. Acoustic MIMO Signal Processing presents the theoretical and the practical in separate sections. The authors open by introducing an acoustic MIMO paradigm, establishing the fundamental of the field, and linking acoustic MIMO signal processing with the concepts of classical signal processing and communication theories in terms of system identification, equalization, and adaptive algorithms. The second part of the book presents a novel and penetrating analysis of aforementioned acoustic applications that is carried out in the paradigm to reinforce the fundamental concepts of acoustic MIMO signal processing.