Springer Handbook of Speech Processing
Published on 25. December 2008
Book
Paperback/Softback
XXXVI, 1176 pages
978-3-662-57128-6 (ISBN)
Description
From common consumer products such as cell phones and MP3 players to more sophisticated projects such as human-machine interfaces and responsive robots, speech technologies are now everywhere. Many think that it is just a matter of time before more applications of the science of speech become inescapable in our daily life. This handbook is meant to play a fundamental role for sustainable progress in speech research and development. Springer Handbook of Speech Processing targets three categories of readers: graduate students, professors and active researchers in academia and research labs, and engineers in industry who need to understand or implement some specific algorithms for their speech-related products. The handbook could also be used as a sourcebook for one or more graduate courses on signal processing for speech and different aspects of speech processing and applications. A quickly accessible source of application-oriented, authoritative and comprehensive information about these technologies, it combines the established knowledge derived from research in such fast evolving disciplines as Signal Processing and Communications, Acoustics, Computer Science and Linguistics.
Reviews / Votes
From the reviews:"This massive volume contains 53 chapters and covers just about all aspects of the field of speech processing. . The editors are commended for producing a valuable tool in the understanding of speech and speech synthesis/recognition. The book is a valuable addition to the bookshelf of researchers, speech scientists, and engineers." (Richard J. Peppin, International Journal of Acoustics and Vibration, Vol. 13 (1), 2008)"This book is a comprehensive overview of most of the major topics associated with speech processing written by the most renowned authors in each topic. The book is well structured with a clearly organized topics. It is intended for use by the researcher . . book is organized in nine sections that cover all current speech applications. . In conclusion, I would highly recommend that anyone interested in speech processing have a copy of this encyclopaedic work." (Eduardo López Gonsalo, The Phonetician, Vol. I-II (97/98), 2008)More details
Series
Edition
Softcover reprint of the original 1st ed. 2008
Language
English
Place of publication
Berlin
Germany
Publishing group
Springer Berlin
Target group
Professional and scholarly
Illustrations
456 s/w Abbildungen, 456 farbige Tabellen
456 Tables, color; 456 Illustrations, black and white; XXXVI, 1176 p. 456 illus.
Dimensions
Height: 24.2 cm
Width: 19.3 cm
ISBN-13
978-3-662-57128-6 (9783662571286)
DOI
10.1007/978-3-540-49127-9
Schweitzer Classification
Other editions
Additional editions
Jacob Benesty | M. M. Sondhi | Yiteng Huang
Springer Handbook of Speech Processing
Book
03/2017
Springer
€640.93
Shipment within 7-9 days
Persons
Jacob BenestyJacob Benesty received the Masters degree in microwaves from Pierre & Marie Curie University, France, in 1987, and the Ph.D. degree in control and signal processing from Orsay University, France, in 1991. From January 1994 to July 1995, he worked at Telecom Paris University on multichannel adaptive filters and acoustic echo cancellation. From October 1995 to May 2003, he was first a Consultant and then a Member of the Technical Staff at Bell Laboratories, Murray Hill, NJ, USA. In May 2003, he joined the University of Quebec, INRS-EMT, in Montreal, Quebec, Canada, as a professor. His research interests are in signal processing, acoustic signal processing, and multimedia communications. Dr. Benesty received the 2001 Best Paper Award from the IEEE Signal Processing Society. He co-authored the books "Acoustic MIMO Signal Processing" (Springer-Verlag, Berlin, 2006) and "Advances in Network and Acoustic Echo Cancellation" (Springer-Verlag, Berlin, 2001). He is also a co-editor/co-author of four other books.M. M. SondhiM. Mohan Sondhi is a consultant at Avaya Research Labs, Basking Ridge, New Jersey. Prior to joining Avaya he spent 39 years at Bell Labs, from where he retired in 2001. He holds undergraduate degrees in Physics and Electrical Communication Engineering, and M.S and Ph.D. degrees in Electrical Engineering. At Bell Labs he conducted research in speech signal processing, echo cancellation, acoustical inverse problems, speech recognition, articulatory models for analysis and synthesis of speech and modeling of auditory and visual processing by humans. He has authored or co-authored several book chapters, over 120 journal articles, 10 patents, and the book Advances in Network and Acoustic Echo Cancellation. He has co-edited the book Advances in Speech Signal Processing. He has been a Distinguished Lecturer of the ASSP society, and an Associate Editor of Trans. ASSP, and has been a visiting scientist at laboratories in Sweden, France, and Japan. He has been on the editorial board of the Journal Speech Communication, and has co-edited a special issue of the Transactions of the IEEE on Speech and Audio Processing, He is a Bell Labs Fellow, a co-recipient of a best paper award of the IEEE ASSP society, and a co-recipient of IEEE's E.E. Sumner award for 1998.Y. HuangYiteng (Arden) Huang received the B.S. degree from the Tsinghua University in 1994, the M.S. and Ph.D. degrees from the Georgia Institute of Technology (Georgia Tech) in 1998 and 2001, respectively, all in electrical and computer engineering. During his doctoral studies from 1998 to 2001, he was a research assistant with the Center of Signal and Image Processing, Georgia Tech, and was a teaching assistant with the School of Electrical and Computer Engineering, Georgia Tech. In the summers from 1998 to 2000, he worked with Bell Laboratories, Murray Hill, NJ and engaged in research on passive acoustic source localization with microphone arrays. Upon graduation, he joined Bell Laboratories as a Member of Technical Staff in March 2001. His current research interests are in acoustic signal processing and multimedia communications. Dr. Huang is currently an Associated Editor of the EURASIP Journal on Applied Signal Processing. He is a member of the Signal Processing Theory and Methods and the Audio and Electroacoustics Technical Committees of the IEEE Signal Processing Society.
Content
Foreword by J. L. Flanagan Chap. 1 Introduction to Speech ProcessingPart A: Production, Perception, and Modeling of Speech (M. M. Sondhi)Part A describes the contemporary views on phonatory and articulatory mechanisms of humans to illustrate the physiological processes of speech production. It also describes the nonlinear cochlear speech processing in auditory masking, the perception of speech and sound by humans, and various methods for speech quality assessment with a focus on standardized methods. Chap. 2 Physiological Processes of Speech ProductionChap. 3 Nonlinear Cochlear Signal Processing and Masking in Speech PerceptionChap. 4 Perception of Speech and SoundChap. 5 Speech Quality Estimation Part B: Signal Processing for Speech (Y. Huang, J. Benesty)Part B gives a large number of signal processing concepts and algorithms that are widely used in speech processing and in the applications of speech. Chap. 6 Wiener and Adaptive FiltersChap. 7 Linear PredictionChap. 8 Kalman FilterChap. 9 Homomorphic Systems and Cepstrum Analysis of SpeechChap. 10 Pitch and Voicing Determination of Speech with an Extension Toward Music SignalsChap. 11 Formant Estimation and TrackingChap. 12 The STFT, Sinusoidal Models, and Speech ModificationChap. 13 Adaptive Blind Multichannel Identification Part C: Speech Coding (W. B. Kleijn)Part C discusses the attributes of speech coders as well as the underlying principles that determine their behavior and architecture. Coders for both traditional and packet networks are discussed, as well as low-bit-rate speech coding, various speech coding standards, and perceptual audio coders. Chap. 14 Principles of Speech CodingChap. 15 Voice over IP: Speech Transmission over Packet NetworksChap. 16 Low-Bit-Rate Speech CodingChap. 17 Analysis-by-Synthesis Speech CodingChap. 18 Perceptual Audio Coding of Speech Signals Part D: Text-to-Speech Synthesis (S. Narayanan)Part D presents different techniques for speech synthesis, including rule-based, corpus-based, and a combination of both. Linguistic analysis and prosodic processing, which are important parts of a text-to-speech (TTS) system, are reviewed. Other aspects of interest for TTS such as voice transformation and synthesis of expressive speech are also discussed. Chap. 19 Basic Principles of Speech SynthesisChap. 20 Rule-Based Speech SynthesisChap. 21 Corpus-Based Speech SynthesisChap. 22 Linguistic Processing for Speech SynthesisChap. 23 Prosodic ProcessingChap. 24 Voice TransformationChap. 25 Expressive/Affective Speech Synthesis Part E: Speech Recognition (L. Rabiner, B.-H. Juang)Part E describes the most important speech recognition technologies. The approach based on the powerful hidden Markov models is generously presented and some other promising approaches are outlined. The robustness issues concerning the acoustical environment are studied. Finally, several fundamental applications are also discussed. Chap. 26 Historical Perspective of the Field of ASR/NLUChap. 27 HMMs and Related Speech TechnologiesChap. 28 Speech Recognition with Weighted Finite-State TransducersChap. 29 A Machine Learning Framework for Spoken-Dialog ClassificationChap. 30 Towards Superhuman Speech RecognitionChap. 31 Natural Language UnderstandingChap. 32 Transcription and Distillation of Spontaneous SpeechChap. 33 Environmental RobustnessChap. 34 The Business of Speech TechnologiesChap. 35 Spoken Dialog Systems Part F: Speaker Recognition (S. Parthasarathy)Part F develops the field of speaker recognition. It covers text-dependent and text-independent speaker recognition and their applications. Chap. 36 Overview of Speaker RecognitionChap. 37 Text-Dependent Speaker RecognitionChap. 38 Text-Independent Speaker Recognition Part G: Language Recognition (C.-H. Lee)Part G provides an overview on principles of state-of-the-art language recognition approaches. Language characterization, identification, and modeling are addressed. Vector space characterization approaches to converting speech utterances into spoken document vectors for modeling and classification are also presented. Chap. 39 Principle of Spoken Language RecognitionChap. 40 Spoken Language CharacterizationChap. 41 Automatic Language Recognition via Spectral and Token Based ApproachesChap. 42 Vector Based Spoken Language Classification Part H: Speech Enhancement (J. Chen, S. Gannot, J. Benesty)Part H develops all classical aspects of speech enhancement: noise reduction, dereverberation, echo cancellation, feedback control, and active noise control. Chap. 43 Fundamentals of Noise ReductionChap. 44 Spectral Enhancement methodsChap. 45 Echo CancellationChap. 46 DereverberationChap. 47 Adaptive Beamforming and PostfilteringChap. 48 Feedback Control in Hearing AidsChap. 49 Active Noise Control Part I: Multichannel Speech Processing (J. Benesty, I. Cohen, Y. Huang)Part I presents modern aspects of multichannel processing, for acoustic scene analysis, speech acquisition and presentation, when a large number of microphones and loudspeakers are available. Chap. 50 Microphone ArraysChap. 51 Time Delay Estimation and Source LocalizationChap. 52 Convolutive Blind Source Separation MethodsChap. 53 Sound Field Reproduction About the Authors Subject Index