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해외주문 [Book] Springer Handbook of Speech Processing [With Dvdrom]

Springer Handbook of | HARDCOVER/DVD | Hardcover
Benesty, Jacob , Sondhi, M. Mohan , Huang, Yiteng 지음 | Springer | 2008년 01월 01일
Springer Handbook of Speech Processing [With Dvdrom]
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상품상세정보
ISBN 9783540491255(3540491252)
쪽수 1176쪽
언어 English
크기 203(W) X 245(H) X 57(T) (mm)
HARDCOVER/DVD
제본형태 Hardcover
삽화유무 삽화있음
총권수 1권
Textual Format Computer Applications
리딩지수 Level Scholarly/Graduate

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목차


List of Abbreviations xxxi
Introduction to Speech Processing 1(6)
J. Benesty
M. M. Sondhi
Y. Huang
A Brief History of Speech Processing 1(1)
Applications of Speech Processing 2(2)
Organization of the Handbook 4(3)
References 4(3)
Part A Production, Perception, and Modeling of Speech
Physiological Processes of Speech Production 7(20)
K. Honda
Overview of Speech Apparatus 7(1)
Voice Production Mechanisms 8(6)
Articulatory Mechanisms 14(10)
Summary 24(3)
References 25(2)
Nonlinear Cochlear Signal Processing and Masking in Speech Perception 27(34)
J. B. Allen
Basics 27(8)
The Nonlinear Cochlea 35(10)
Neural Masking 45(10)
Discussion and Summary 55(6)
References 56(5)
Perception of Speech and Sound 61(22)
B. Kollmeier
T. Brand
B. Meyer
Basic Psychoacoustic Quantities 62(8)
Acoustical Information Required for Speech Perception 70(4)
Speech Feature Perception 74(9)
References 81(2)
Speech Quality Assessment 83(20)
V. Grancharov
W. B. Kleijn
Degradation Factors Affecting Speech Quality 84(1)
Subjective Tests 85(5)
Objective Measures 90(5)
Conclusions 95(8)
References 96(7)
Part B Signal Processing for Speech
Wiener and Adaptive Filters 103(18)
J. Benesty
Y. Huang
J. Chen
Overview 103(1)
Signal Models 104(2)
Derivation of the Wiener Filter 106(1)
Impulse Response Tail Effect 107(1)
Condition Number 108(2)
Adaptive Algorithms 110(6)
MIMO Wiener Filter 116(3)
Conclusions 119(2)
References 120(1)
Linear Prediction 121(14)
J. Benesty
J. Chen
Y. Huang
Fundamentals 121(1)
Forward Linear Prediction 122(1)
Backward Linear Prediction 123(1)
Levinson--Durbin Algorithm 124(2)
Lattice Predictor 126(1)
Spectral Representation 127(1)
Linear Interpolation 128(1)
Line Spectrum Pair Representation 129(1)
Multichannel Linear Prediction 130(3)
Conclusions 133(2)
References 133(2)
The Kalman Filter 135(26)
S. Gannot
A. Yeredor
Derivation of the Kalman Filter 136(5)
Examples: Estimation of Parametric Stochastic Process from Noisy Observations 141(3)
Extensions of the Kalman Filter 144(5)
The Application of the Kalman Filter to Speech Processing 149(8)
Summary 157(4)
References 157(4)
Homomorphic Systems and Cepstrum Analysis of Speech 161(20)
R. W. Schafer
Definitions 161(3)
Z-Transform Analysis 164(1)
Discrete-Time Model for Speech Production 165(1)
The Cepstrum of Speech 166(3)
Relation to LPC 169(2)
Application to Pitch Detection 171(1)
Applications to Analysis/Synthesis Coding 172(4)
Applications to Speech Pattern Recognition 176(4)
Summary 180(1)
References 180(1)
Pitch and Voicing Determination of Speech with an Extension Toward Music Signals 181(32)
W. J. Hess
Pitch in Time--Variant Quasiperiodic Acoustic Signals 182(3)
Short-Term Analysis PDAs 185(7)
Selected Time--Domain Methods 192(3)
A Short Look into Voicing Determination 195(2)
Evaluation and Postprocessing 197(4)
Applications in Speech and Music 201(2)
Some New Challenges and Developments 203(4)
Concluding Remarks 207(6)
References 208(5)
Formant Estimation and Tracking 213(16)
D. O'Shaughnessy
Historical 213(2)
Vocal Tract Resonances 215(1)
Speech Production 216(2)
Acoustics of the Vocal Tract 218(3)
Short-Time Speech Analysis 221(2)
Formant Estimation 223(3)
Summary 226(3)
References 226(3)
The STFT, Sinusoidal Models, and Speech Modification 229(30)
M. M. Goodwin
The Short-Time Fourier Transform 230(12)
Sinusoidal Models 242(11)
Speech Modification 253(6)
References 256(3)
Adaptive Blind Multichannel Identification 259(24)
Y. Huang
J. Benesty
J. Chen
Overview 259(1)
Signal Model and Problem Formulation 260(1)
Identifiability and Principle 261(1)
Constrained Time--Domain Multichannel LMS and Newton Algorithms 262(4)
Unconstrained Multichannel LMS Algorithm with Optimal Step-Size Control 266(2)
Frequency-Domain Blind Multichannel Identification Algorithms 268(8)
Adaptive Multichannel Exponentiated Gradient Algorithm 276(3)
Summary 279(4)
References 279(4)
Part C Speech Coding
Principles of Speech Coding 283(24)
W. B. Kleijn
The Objective of Speech Coding 283(1)
Speech Coder Attributes 284(2)
A Universal Coder for Speech 286(7)
Coding with Autoregressive Models 293(3)
Distortion Measures and Coding Architecture 296(6)
Summary 302(5)
References 303(4)
Voice over IP: Speech Transmission over Packet Networks 307(24)
J. Skoglund
E. Kozica
J. Linden
R. Hagen
W. B. Kleijn
Voice Communication 307(1)
Properties of the Network 308(5)
Outline of a VoIP System 313(4)
Robust Encoding 317(9)
Packet Loss Concealment 326(1)
Conclusion 327(4)
References 328(3)
Low-Bit-Rate Speech Coding 331(20)
A. V. McCree
Speech Coding 331(1)
Fundamentals: Parametric Modeling of Speech Signals 332(5)
Flexible Parametric Models 337(7)
Efficient Quantization of Model Parameters 344(1)
Low-Rate Speech Coding Standards 345(2)
Summary 347(4)
References 347(4)
Analysis-by-Synthesis Speech Coding 351(42)
J.-H. Chen
J. Thyssen
Overview 352(1)
Basic Concepts of Analysis-by-Synthesis Coding 353(4)
Overview of Prominent Analysis-by-Synthesis Speech Coders 357(3)
Multipulse Linear Predictive Coding (MPLPC) 360(2)
Regular-Pulse Excitation with Long-Term Prediction (RPE-LTP) 362(1)
The Original Code Excited Linear Prediction (CELP) Coder 363(4)
US Federal Standard FS1016 CELP 367(1)
Vector Sum Excited Linear Prediction (VSELP) 368(2)
Low-Delay CELP (LD-CELP) 370(1)
Pitch Synchronous Innovation CELP (PSI-CELP) 371(1)
Algebraic CELP (ACELP) 371(6)
Conjugate Structure CELP (CS-CELP) and CS-ACELP 377(1)
Relaxed CELP (RCELP) -- Generalized Analysis by Synthesis 378(3)
eX-CELP 381(1)
iLBC 382(1)
TSNFC 383(3)
Embedded CELP 386(2)
Summary of Analysis-by-Synthesis Speech Coders 388(2)
Conclusion 390(3)
References 390(3)
Perceptual Audio Coding of Speech Signals 393(20)
J. Herre
M. Lutzky
History of Audio Coding 393(1)
Fundamentals of Perceptual Audio Coding 394(2)
Some Successful Standardized Audio Coders 396(2)
Perceptual Audio Coding for Real-Time Communication 398(5)
Hybrid/Crossover Coders 403(6)
Summary 409(4)
References 409(4)
Part D Text-to-Speech Synthesis
Basic Principles of Speech Synthesis 413(16)
J. Schroeter
The Basic Components of a TTS System 413(8)
Speech Representations and Signal Processing for Concatenative Synthesis 421(2)
Speech Signal Transformation Principles 423(2)
Speech Synthesis Evaluation 425(1)
Conclusions 426(3)
References 426(3)
Rule-Based Speech Synthesis 429(8)
R. Carlson
B. Granstrom
Background 429(1)
Terminal Analog 429(3)
Controlling the Synthesizer 432(2)
Special Applications of Rule-Based Parametric Synthesis 434(1)
Concluding Remarks 434(3)
References 434(3)
Corpus-Based Speech Synthesis 437(20)
T. Dutoit
Basics 437(1)
Concatenative Synthesis with a Fixed Inventory 438(9)
Unit-Selection-Based Synthesis 447(3)
Statistical Parametric Synthesis 450(3)
Conclusion 453(4)
References 453(4)
Linguistic Processing for Speech Synthesis 457(14)
R. Sproat
Why Linguistic Processing is Hard 457(1)
Fundamentals: Writing Systems and the Graphical Representation of Language 457(1)
Problems to be Solved and Methods to Solve Them 458(7)
Architectures for Multilingual Linguistic Processing 465(1)
Document-Level Processing 465(1)
Future Prospects 466(5)
References 467(4)
Prosodic Processing 471(18)
J. van Santen
T. Mishra
E. Klabbers
Overview 471(4)
Historical Overview 475(1)
Fundamental Challenges 476(1)
A Survey of Current Approaches 477(7)
Future Approaches 484(1)
Conclusions 485(4)
References 485(4)
Voice Transformation 489(16)
Y. Stylianou
Background 489(1)
Source-Filter Theory and Harmonic Models 490(2)
Definitions 492(2)
Source Modifications 494(4)
Filter Modifications 498(1)
Conversion Functions 499(1)
Voice Conversion 500(1)
Quality Issues in Voice Transformations 501(1)
Summary 502(3)
References 502(3)
Expressive/Affective Speech Synthesis 505(16)
N. Campbell
Overview 505(1)
Characteristics of Affective Speech 506(2)
The Communicative Functionality of Speech 508(2)
Approaches to Synthesizing Expressive Speech 510(2)
Modeling Human Speech 512(3)
Conclusion 515(6)
References 515(6)
Part E Speech Recognition
Historical Perspective of the Field of ASR/NLU 521(18)
L. Rabiner
B.-H. Juang
ASR Methodologies 521(2)
Important Milestones in Speech Recognition History 523(1)
Generation 1 -- The Early History of Speech Recognition 524(1)
Generation 2 -- The First Working Systems for Speech Recognition 524(1)
Generation 3 -- The Pattern Recognition Approach to Speech Recognition 525(5)
Generation 4 -- The Era of the Statistical Model 530(4)
Generation 5 -- The Future 534(1)
Summary 534(5)
References 535(4)
HMMs and Related Speech Recognition Technologies 539(20)
S. Young
Basic Framework 539(1)
Architecture of an HMM-Based Recognizer 540(7)
HMM-Based Acoustic Modeling 547(3)
Normalization 550(1)
Adaptation 551(3)
Multipass Recognition Architectures 554(1)
Conclusions 554(5)
References 555(4)
Speech Recognition with Weighted Finite-State Transducers 559(26)
M. Mohri
F. Pereira
M. Riley
Definitions 559(1)
Overview 560(7)
Algorithms 567(7)
Applications to Speech Recognition 574(8)
Conclusion 582(3)
References 582(3)
A Machine Learning Framework for Spoken-Dialog Classification 585(12)
C. Cortes
P. Haffner
M. Mohri
Motivation 585(1)
Introduction to Kernel Methods 586(1)
Rational Kernels 587(2)
Algorithms 589(2)
Experiments 591(2)
Theoretical Results for Rational Kernels 593(1)
Conclusion 594(3)
References 595(2)
Towards Superhuman Speech Recognition 597(20)
M. Picheny
D. Nahamoo
Current Status 597(1)
A Multidomain Conversational Test Set 598(1)
Listening Experiments 599(2)
Recognition Experiments 601(6)
Speculation 607(10)
References 614(3)
Natural Language Understanding 617(10)
S. Roukos
Overview of NLU Applications 618(2)
Natural Language Parsing 620(3)
Practical Implementation 623(1)
Speech Mining 623(2)
Conclusion 625(2)
References 626(1)
Transcription and Distillation of Spontaneous Speech 627(26)
S. Furui
T. Kawahara
Background 627(1)
Overview of Research Activities on Spontaneous Speech 628(4)
Analysis for Spontaneous Speech Recognition 632(3)
Approaches to Spontaneous Speech Recognition 635(5)
Metadata and Structure Extraction of Spontaneous Speech 640(4)
Speech Summarization 644(3)
Conclusions 647(6)
References 647(6)
Environmental Robustness 653(28)
J. Droppo
A. Acero
Noise Robust Speech Recognition 653(3)
Model Retraining and Adaptation 656(1)
Feature Transformation and Normalization 657(7)
A Model of the Environment 664(3)
Structured Model Adaptation 667(4)
Structured Feature Enhancement 671(4)
Unifying Model and Feature Techniques 675(2)
Conclusion 677(4)
References 677(4)
The Business of Speech Technologies 681(24)
J. Wilpon
M. E. Gilbert
J. Cohen
Introduction 682(4)
Network-Based Speech Services 686(6)
Device-Based Speech Applications 692(5)
Vision/Predications of Future Services -- Fueling the Trends 697(4)
Conclusion 701(4)
References 702(3)
Spoken Dialogue Systems 705(20)
V. Zue
S. Seneff
Technology Components and System Development 707(5)
Development Issues 712(2)
Historical Perspectives 714(1)
New Directions 715(3)
Concluding Remarks 718(7)
References 718(7)
Part F Speaker Recognition
Overview of Speaker Recognition 725(18)
A. E. Rosenberg
F. Bimbot
S. Parthasarathy
Speaker Recognition 725(4)
Measuring Speaker Features 729(2)
Constructing Speaker Models 731(4)
Adaptation 735(1)
Decision and Performance 735(2)
Selected Applications for Automatic Speaker Recognition 737(2)
Summary 739(4)
References 739(4)
Text-Dependent Speaker Recognition 743(20)
M. Hebert
Brief Overview 743(4)
Text-Dependent Challenges 747(3)
Selected Results 750(10)
Concluding Remarks 760(3)
References 760(3)
Text-Independent Speaker Recognition 763(22)
D. A. Reynolds
W. M. Campbell
Introduction 763(1)
Likelihood Ratio Detector 764(2)
Features 766(1)
Classifiers 767(9)
Performance Assessment 776(2)
Summary 778(7)
References 779(6)
Part G Language Recognition
Principles of Spoken Language Recognition 785(12)
C.-H. Lee
Spoken Language 785(1)
Language Recognition Principles 786(2)
Phone Recognition Followed by Language Modeling (PRLM) 788(1)
Vector-Space Characterization (VSC) 789(1)
Spoken Language Verification 790(1)
Discriminative Classifier Design 791(2)
Summary 793(4)
References 793(4)
Spoken Language Characterization 797(14)
M. P. Harper
M. Maxwell
Language versus Dialect 798(2)
Spoken Language Collections 800(1)
Spoken Language Characteristics 800(4)
Human Language Identification 804(2)
Text as a Source of Information on Spoken Languages 806(1)
Summary 807(4)
References 807(4)
Automatic Language Recognition Via Spectral and Token Based Approaches 811(14)
D.A. Reynolds
W. M. Campbell
W. Shen
E. Singer
Automatic Language Recognition 811(1)
Spectral Based Methods 812(3)
Token-Based Methods 815(3)
System Fusion 818(2)
Performance Assessment 820(3)
Summary 823(2)
References 823(2)
Vector-Based Spoken Language Classification 825(18)
H. Li
B. Ma
C.-H. Lee
Vector Space Characterization 826(1)
Unit Selection and Modeling 827(3)
Front-End: Voice Tokenization and Spoken Document Vectorization 830(1)
Back-End: Vector-Based Classifier Design 831(4)
Language Classification Experiments and Discussion 835(3)
Summary 838(5)
References 839(4)
Part H Speech Enhancement
Fundamentals of Noise Reduction 843(30)
J. Chen
J. Benesty
Y. Huang
E. J. Diethorn
Noise 843(2)
Signal Model and Problem Formulation 845(1)
Evaluation of Noise Reduction 846(1)
Noise Reduction via Filtering Techniques 847(10)
Noise Reduction via Spectral Restoration 857(6)
Speech-Model-Based Noise Reduction 863(5)
Summary 868(5)
References 869(4)
Spectral Enhancement Methods 873(30)
I. Cohen
S. Gannot
Spectral Enhancement 874(1)
Problem Formulation 875(1)
Statistical Models 876(3)
Signal Estimation 879(2)
Signal Presence Probability Estimation 881(1)
A Priori SNR Estimation 882(6)
Noise Spectrum Estimation 888(3)
Summary of a Spectral Enhancement Algorithm 891(5)
Selection of Spectral Enhancement Algorithms 896(2)
Conclusions 898(5)
References 899(4)
Adaptive Echo Cancelation for Voice Signals 903(26)
M. M. Sondhi
Network Echoes 904(11)
Single-Channel Acoustic Echo Cancelation 915(6)
Multichannel Acoustic Echo Cancelation 921(4)
Summary 925(4)
References 926(3)
Dereverberation 929(16)
Y. Huang
J. Benesty
J. Chen
Background and Overview 929(2)
Signal Model and Problem Formulation 931(1)
Source Model-Based Speech Dereverberation 932(4)
Separation of Speech and Reverberation via Homomorphic Transformation 936(1)
Channel Inversion and Equalization 937(4)
Summary 941(4)
References 942(3)
Adaptive Beamforming and Postfiltering 945(34)
S. Gannot
I. Cohen
Problem Formulation 947(1)
Adaptive Beamforming 948(5)
Fixed Beamformer and Blocking Matrix 953(2)
Identification of the Acoustical Transfer Function 955(5)
Robustness and Distortion Weighting 960(2)
Multichannel Postfiltering 962(5)
Performance Analysis 967(5)
Experimental Results 972(1)
Summary 972(1)
Appendix: Derivation of the Expected Noise Reduction for a Coherent Noise Field 973(1)
Appendix: Equivalence Between Maximum SNR and LCMV Beamformers 974(5)
References 975(4)
Feedback Control in Hearing Aids 979(22)
A. Spriet
S. Doclo
M. Moonen
J. Wouters
Problem Statement 980(2)
Standard Adaptive Feedback Canceller 982(4)
Feedback Cancellation Based on Prior Knowledge of the Acoustic Feedback Path 986(4)
Feedback Cancellation Based on Closed-Loop System Identification 990(5)
Comparison 995(2)
Conclusions 997(4)
References 997(4)
Active Noise Control 1001(20)
S. M. Kuo
D. R. Morgan
Broadband Feedforward Active Noise Control 1002(4)
Narrowband Feedforward Active Noise Control 1006(4)
Feedback Active Noise Control 1010(1)
Multichannel ANC 1011(4)
Summary 1015(6)
References 1015(6)
Part I Multichannel Speech Processing
Microphone Arrays 1021(22)
G. W. Elko
J. Meyer
Microphone Array Beamforming 1021(8)
Constant-Beamwidth Microphone Array System 1029(1)
Constrained Optimization of the Directional Gain 1030(1)
Differential Microphone Arrays 1031(3)
Eigenbeamforming Arrays 1034(3)
Adaptive Array Systems 1037(3)
Conclusions 1040(3)
References 1040(3)
Time Delay Estimation and Source Localization 1043(22)
Y. Huang
J. Benesty
J. Chen
Technology Taxonomy 1043(1)
Time Delay Estimation 1044(10)
Source Localization 1054(7)
Summary 1061(4)
References 1062(3)
Convolutive Blind Source Separation Methods 1065(30)
M. S. Pedersen
J. Larsen
U. Kjems
L. C. Parra
The Mixing Model 1066(2)
The Separation Model 1068(3)
Identification 1071(1)
Separation Principle 1071(5)
Time Versus Frequency Domain 1076(2)
The Permutation Ambiguity 1078(6)
Results 1084(1)
Conclusion 1084(11)
References 1084(11)
Sound Field Reproduction 1095(20)
R. Rabenstein
S. Spors
Sound Field Synthesis 1095(1)
Mathematical Representation of Sound Fields 1096(4)
Stereophony 1100(3)
Vector-Based Amplitude Panning 1103(1)
Ambisonics 1104(5)
Wave Field Synthesis 1109(6)
References 1113(2)
Acknowledgements 1115(2)
About the Authors 1117(16)
Detailed Contents 1133(28)
Subject Index 1161

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