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1.

図書

図書
edited by Gillian M. Davis
出版情報: Boca Raton : CRC Press, c2002  407 p. ; 25 cm
シリーズ名: The electrical engineering and applied signal processing series
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目次情報: 続きを見る
Tutorial / Section I:
Noise and Digital Signal Processing / Stephan Weiss ; Robert W. Stewart ; Gillian M. Davis1:
System Aspects / Section II:
Analog Techniques / Malcolm J. Hawksford2:
Hardware Design Considerations / Robert S. Oshana3:
Software Design Considerations for Real-Time DSP Systems / Elizabeth G. Keate4:
Evaluating the Effects of Noise on Voice Communication Systems / William D. Voiers ; Alan D. Sharpley ; Ira L. Panzer5:
Digital Algorithms and Implementation / Section III:
Single-Channel Speech Enhancement / Graham P. Eatwell6:
Microphone Arrays / Stephen J. Leese7:
Echo Cancellation / 8:
Special Applications / Section IV:
Signal and Feature Compensation Methods for Robust Speech Recognition / Rita Singh ; Richard M. Stern ; Bhiksha Raj9:
Model Compensation and Matched Condition Methods for Robust Speech Recognition / 10:
Noise and Voice Quality in VoIP Environments / Dennis Hardman11:
Noise Canceling Headsets for Speech Communication / Lars Hakansson ; Sven Johansson ; Mattias Dahl ; Per Sjosten ; Ingvar Claesson12:
Acoustic Crosstalk Reduction in Loudspeaker-Based Virtual Audio Systems / Darren B. Ward13:
Interference in Telephone Circuits / George Keratiotis ; Larry Lind ; Minesh Patel ; John W. Cook ; Pete Whelan14:
An Adaptive Beamforming Perspective on Convolutive Blind Source Separation / Lucas Parra ; Craig Fancourt15:
Use of DSP Techniques to Enhance the Performance of Hearing Aids in Noise / Douglas M. Chabries ; Victor Bray16:
Index
Tutorial / Section I:
Noise and Digital Signal Processing / Stephan Weiss ; Robert W. Stewart ; Gillian M. Davis1:
System Aspects / Section II:
2.

図書

図書
edited by Wu Chou, Biing Hwang Juang
出版情報: Boca Raton, Fla. ; London : CRC, c2003  vi, 394 p. ; 25cm
シリーズ名: The electrical engineering and applied signal processing series
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3.

図書

図書
edited by Jhing-Fa Wang, Sadaoki Furui, Biing-Hwang Juang
出版情報: Boston, Mass. : Kluwer Academic Publishers, c2004  129 p. ; 27 cm
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Guest Editorial / Jhing-Fa Wang ; Sadaoki Furui ; Biing-Hwang Juang
A Robust Bimodal Speech Section Detection / K. Murai ; S. Nakamura
Acoustic Feature Analysis and Discriminative Modeling of Filled Pauses for Spontaneous Speech Recognition / Chung-Hsien Wu ; Gwo-Lang Yan
Simultaneous Recognition of Distant-Talking Speech of Multiple Talkers Based on the 3-D N-Best Search Method / P. Heracleous ; K. Shikano
Multi-Modal Speech Recognition Using Optical-Flow Analysis for Lip Images / S. Tamura ; K. Iwano ; S. Furui
Speech Enhancement Using Perceptual Wavelet Packet Decomposition and Teager Energy Operator / Shi-Huang Chen
Use of Microphone Array and Model Adaptation for Hands-Free Speech Acquisition and Recognition / Jen-Tzung Chien ; Jain-Ray Lai
Multimedia Corpus of In-Car Speech Communication / N. Kawaguchi ; K. Takeda ; F. Itakura
Speech and Language Processing for Multimodal Human-Computer Interaction / L. Deng ; Y. Wang ; K. Wang ; A. Acero ; H. Hon ; J. Droppo ; C. Boulis ; M. Mahajan ; X.D. Huang
Blind Model Selection for Automatic Speech Recognition in Reverberant Environments / L. Couvreur ; C. Couvreur
Guest Editorial / Jhing-Fa Wang ; Sadaoki Furui ; Biing-Hwang Juang
A Robust Bimodal Speech Section Detection / K. Murai ; S. Nakamura
Acoustic Feature Analysis and Discriminative Modeling of Filled Pauses for Spontaneous Speech Recognition / Chung-Hsien Wu ; Gwo-Lang Yan
4.

図書

図書
Jacob Benesty, M. Mohan Sondhi, Yiteng Huang (eds.)
出版情報: Berlin : Springer, c2008  xxxvi, 1176 p. ; 25 cm.
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5.

図書

図書
Lajos Hanzo, F. Clare Somerville, Jason Woodard
出版情報: [New York] : IEEE press , Chichester : John Wiley, c2007  xxxv, 843 p. ; 25 cm
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About the Authors
Other Wiley and IEEE Press Books on Related Topics
Preface and Motivation
Acknowledgements
Speech Signals andWaveform Coding / I:
Predictive Coding / 2:
Analysis-by-synthesis Principles / 3:
Speech Spectral Quantization / 4:
RPE Coding / 5:
Forward-Adaptive CELP Coding / 6:
Standard CELP Codecs / 7:
Backward-Adaptive CELP Coding / 8:
Wideband Speech Coding / 9:
MPEG-4 Audio Compression and Transmission / 10:
Overview of Low-rate Speech Coding / 11:
Linear Predictive Vocoder / 12:
Wavelets and Pitch Detection / 13:
Zinc Function Excitation / 14:
Mixed-Multiband Excitation / 15:
Sinusoidal Transform Coding Below 4kbps / 16:
Conclusions on Low Rate Coding / 17:
Comparison of Speech Transceivers / 18:
Voice Over the Internet Protocol / 19:
Constructing the Quadratic Spline Wavelets / A:
Contents
Probability Density Function for Amplitudes / B:
Bibliography
Index
Author Index
About the Authors
Other Wiley and IEEE Press Books on Related Topics
Preface and Motivation
6.

図書

東工大
目次DB

図書
東工大
目次DB
Sadaoki Furui
出版情報: New York : Dekker, c2001  xx, 452 p. ; 24 cm
シリーズ名: Signal processing and communications series ; 7
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Series Introduction (K.J. Ray Liu)
Preface to the Second Edition
Acknowledgments
Preface to the First Edition
1. INTRODUCTION 1
2. PRINCIPAL CHARACTERISTICS OF SPEECH 5
   2.1 Linguistic Information 5
   2.2 Speech and Hearing 7
   2.3 Speech Production Mechanism 9
   2.4 Acoustic Characteristics of Speech 14
   2.5 Statistical Characteristics of Speech 20
   2.5.1 Distribution of amplitude level 20
   2.5.2 Long-time averaged spectrum 23
   2.5.3 Variation in fundamental frecuency 24
   2.5.4 Speech ratio 26
3. SPEECH PRODUCTION MODELS 27
   3.1 Acoustical Theory of Speech Production 27
   3.2 Linear Separable Equivalent Circuit Model 30
   3.3 Vocal Tract Transmission Model 32
   3.3.1 Progressing wave model 32
   3.3.2 Resonance model 38
   3.4 Vocal Cord Model 40
4. SPEECH ANALYSIS AND ANALYSIS-SYNTHESIS SYSTEMS 45
   4.1 Digitization 45
   4.1.1 Sampling 46
   4.1.2 Quantization and coding 47
   4.1.3 A/D and D/A conversion 51
   4.2 Spectral Analysis 52
   4.2.1 Spectral structure of speech 52
   4.2.2 Autocorrelation and Fourier transform 53
   4.2.3 Window function 57
   4.2.4 Sound spectrogram 60
   4.3 Cepstrum 62
   4.3.1 Cepstrum and its application 62
   4.3.2 Homomorphic analysis and LPC cepstrum 66
   4.4 Filter Bank and Zero-Crossing Analysis 70
   4.4.1 Digital filter bank 70
   4.4.2 Zero-crossing analysis 70
   4.5 Analysis-by-Synthesis 71
   4.6 Analysis-Synthesis Systems 73
   4.6.1 Analysis-synthesis system structure 73
   4.6.2 Example of analysis-synthesis systems 73
   4.7 Pitch Extraction 78
5. LINEAR PREDICTIVE CODING (LPC) ANALYSIS 83
   5.1 Principles of LPC Analysis 83
   5.2 LPC Analysis Procedure 86
   5.3 Maximum Likelihood Spectral Estimation 89
   5.3.1 Formulation of maximum likelihood spectral estimation 89
   5.3.2 Physical meaning of maximum likelihood spectral estimation 93
   5.4 Source Parameter Estimation from Residual Signals 98
   5.5 Speech Analysis-Synthesis System by LPC 99
   5.6 PARCOR Analysis 102
   5.6.1 Formulation of PARCOR analysis 102
   5.6.2 Relationship between PARCOR and LPC coefficients 108
   5.6.3 PARCOR synthesis filter 109
   5.6.4 Vocal tract area estimation based on PARCOR analysis 110
   5.7 Line Spectrum Pair (LSP) Analysis 116
   5.7.1 Principle of LSP analysis 116
   5.7.2 Solution of LSP analysis 119
   5.7.3 LSP synthesis filter 122
   5.7.4 Coding of LSP parameters 126
   5.7.5 Composite sinusoidal model 126
   5.7.6 Mutual relationships between LPC parameters 127
   5.8 Pole-Zero Analysis 129
6 SPEECH CODING 133
   6.1 Principal Techniques for Speech Coding 133
   6.1.1 Reversible coding 133
   6.1.2 Irreversible coding and information rate distortion theory 134
   6.1.3 Waveform coding and analysis-synthesis systems 135
   6.1.4 Basic techniques for waveform coding methods 138
   6.2 Coding in Time Domain 141
   6.2.1 Pulse code modulation (PCM) 141
   6.2.2 Adaptive quantization 143
   6.2.3 Predictive coding 143
   6.2.4 Delta modulation 149
   6.2.5 Adaptive differential PCM (ADPCM) 151
   6.2.6 Adaptive predictive coding (APC) 153
   6.2.7 Noise shaping 156
   6.3 Coding in Frequency Domain 159
   6.3.1 Subband coding (SBC) 159
   6.3.2 Adaptive transform coding (ATC) 163
   6.3.3 APC with adaptive bit allocation (APC-AB) 166
   6.3.4 Time-domain harmonic scaling (TDHS) algorithm 168
   6.4 Vector Quantization 173
   6.4.1 Multipath search coding 173
   6.4.2 Principles of vector quantization 175
   6.4.3 Tree search and multistage processing 178
   6.4.4 Vector quantization for linear predictor parameters 180
   6.4.5 Matrix quantization and finite-state vector quantization 182
   6.5 Hybrid Coding 187
   6.5.1 Residual- or speech-excited linear predictive coding 187
   6.5.2 Multipulse-excited linear predictive coding (MPC) 189
   6.5.3 Code-excited linear predictive coding (CELP) 193
   6.5.4 Coding by phase equalization and variable-rate tree coding 196
   6.6 Evaluation and Standardization of Coding Methods 199
   6.6.1 Evaluation factors of speech coding systems 199
   6.6.2 Speech coding standards 203
   6.7 Robust and Flexible Speech Coding 211
7 SPEECH SYNTHESIS 213
   7.1 Principles of Speech Synthesis 213
   7.2 Synthesis Based on Waveform Coding 217
   7.3 Synthesis Based on Analysis-Synthesis Method 221
   7.4 Synthesis Based on Speech Production Mechanism 222
   7.4.1 Vocal tract analog method 223
   7.4.2 Terminal analog method 224
   7.5 Synthesis by Rule 226
   7.5.1 Principles of synthesis by rule 226
   7.5.2 Control of prosodic features 230
   7.6 Text-to-Speech Conversion 234
   7.7 Corpus-Based Speech Synthesis 237
8 SPEECH RECOGNITION
   8.1 Principles of Speech Recognition 243
   8.1.1 Advantages of speech recognition 243
   8.1.2 Difficulties in speech recognition 245
   8.1.3 Classification of speech recognition 246
   8.2 Speech Period Detection 248
   8.3 Spectral Distance Measures 249
   8.3.1 Distance measures used in speech recognition 249
   8.3.2 Distances based on nonparametric spectral analysis 251
   8.3.3 Distances based on LPC 252
   8.3.4 Peak-weighted distances based on LPC analysis 258
   8.3.5 Weighted cepstral distance 260
   8.3.6 Transitional cepstral distance 262
   8.3.7 Prosody 264
   8.4 Structure of Word Recognition Systems 264
   8.5 Dynamic Time Warping (DTW) 266
   8.5.1 DP matching 266
   8.5.2 Variations in DP matching 270
   8.5.3 Staggered array DP matching 272
   8.6 Word Recognition Using Phoneme Units 275
   8.6.1 Principal structure 275
   8.6.2 SPLIT method 277
   8.7 Theory and Implementation of HMM 278
   8.7.1 Fundamentals of HMM 278
   8.7.2 Three basic problems for HMMs 282
   8.7.3 Solution to Problem 1-Probability evaluation 283
   8.7.4 Solution to Problem 2-optimal state sequence 286
   8.7.5 Solution to Problem 3ーparameter estimation 288
   8.7.6 Continuous observation densities in HMMs 290
   8.7.7 Tied-mixture HMM 292
   8.7.8 MMI and MCE/GPD training of HMM 292
   8.7.9 HMM system for word recognition 293
   8.8 Connected Word Recognition 295
   8.8.1 Two-level DP matching and its modifications 295
   8.8.2 Word spotting 303
   8.9 Large-Vocabulary Continuous-Speech Recognition 306
   8.9.1 Three principal structural models 306
   8.9.2 Other system constructing factors 308
   8.9.3 Statistical theory of continuous-speech recognition 311
   8.9.4 Statistical language modeling 312
   8.9.5 Typical structure of large-vocabulary continuous-speech recognition 314
   systems 318
   8.9.6 Methods for evaluating recognition systems 320
   8.10 Examples of Large-Vocabulary Continuous-Speech Recognition Systems 323
   8.10.1 DARPA speech recognition projects 323
   8.10.2 English speech recognition system at LIMSI Laboratory 324
   8.10.3 English speech recognition system at IBM Laboratory 325
   8.10.4 A Japanese speech recognition system 328
   8.11 Speaker-Independent and Adaptive Recognition 330
   8.11.1 Multi-template method 332
   8.11.2 Statistical method 333
   8.11.3 Speaker normalization method 334
   8.11.4 Speaker adaptation methods 335
   8.11.5 Unsupervised speaker adaptation method 336
   8.12 Robust Algorithms Against Noise and Channel Variations 339
   8.12.1 HMM composition/PMC 344
   8.12.2 Detection-based approach for spontaneous speech recognition 344
9 SPEAKER RECOGNIT ION 349
   9.1 Principles of Speaker Recognition 349
   9.1.1 Human and computer speaker recognition 349
   9.1.2 Individual characteristics 351
   9.2 Speaker Recognition Methods 352
   9.2.1 Classification of speaker recognition methods 352
   9.2.2 Structure of speaker recognition systems 354
   9.2.3. Relationship between error rate and number of speakers 358
   9.2.4 Intra-speaker variation and evaluation of feature parameters 360
   9.2.5 Likelihood (distance) normalization 364
   9.3 Examples of Speaker Recognition Systems 366
   9.3.1 Text-dependent speaker recognition systems 366
   9.3.2 Text-independent speaker recognition systems 368
   9.3.3 Text-prompted speaker recognition systems 373
10 FUTURE DIRECTIONS OF SPEECH INFORMATION PROCESSING 375
   10.1 Overview 375
   10.2 Analysis and Description of Dynamic Features 378
   10.3 Extraction and Normalization of Voice Individuality 379
   10.4 Adaptation to Environmental Variation 380
   10.5 Basic Units for Speech Processing 381
   10.6 Adavanced Knowledge Processing 382
   10.7 Clarification of Speech Production Mechanism 383
   10.8 Clarification of Speech Perception Mechanism 384
   10.9 Evaluation Methods fo Speech Processing Technologies 385
   10.10 LSI for Speech Processing Use 386
APPENDICES
   A Convolution and z-Transform 387
   A.1 Convolution 387
   A.2 z-Transform 388
   A.3 Stability 391
   B Vector Quantization Algorithm 393
   B.1 VQ (Vector Quantization) Technique Formulation 393
   B.2 Lloyd's Algorithm (K-Means Algorithm) 394
   B.3 LBG Algorithm 395
   C Neural Nets 399
Bibliography 405
Index 437
Series Introduction (K.J. Ray Liu)
Preface to the Second Edition
Acknowledgments
7.

図書

図書
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway, NJ : IEEE Service Center, c1995  875 p. ; 28 cm
シリーズ名: ICASSP-95 : The 1995 International Conference on Acoustics, Speech, and Signal Processing : May 9-12, 1995, Westin Hotel, Detroit, Michigan U.S.A. / sponsored by The Signal Processing Society of The Institute of Electrical and Electronics Engineers ; Vol. 1
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8.

図書

図書
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway, NJ : IEEE Service Center, c1995  p. 877-1527 ; 28 cm
シリーズ名: ICASSP-95 : The 1995 International Conference on Acoustics, Speech, and Signal Processing : May 9-12, 1995, Westin Hotel, Detroit, Michigan U.S.A. / sponsored by The Signal Processing Society of The Institute of Electrical and Electronics Engineers ; Vol. 2
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9.

図書

図書
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway, NJ : IEEE Service Center, c1995  p. 1529-2119 ; 28 cm
シリーズ名: ICASSP-95 : The 1995 International Conference on Acoustics, Speech, and Signal Processing : May 9-12, 1995, Westin Hotel, Detroit, Michigan U.S.A. / sponsored by The Signal Processing Society of The Institute of Electrical and Electronics Engineers ; Vol. 3
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10.

図書

図書
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway, NJ : IEEE Service Center, c1995  p. 2121-2762 ; 28 cm
シリーズ名: ICASSP-95 : The 1995 International Conference on Acoustics, Speech, and Signal Processing : May 9-12, 1995, Westin Hotel, Detroit, Michigan U.S.A. / sponsored by The Signal Processing Society of The Institute of Electrical and Electronics Engineers ; Vol. 4
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