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1.

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図書
1. Speech analysis, synthesis, and perception. 2nd, expanded ed (g.w. ; u.s.)
by James L. Flanagan
出版情報: Berlin ; New York : Springer-Verlag, 1972  x, 444 p. ; 24 cm
シリーズ名: Kommunikation und Kybernetik in Einzeldarstellungen ; 3
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2.

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図書
2. Speech synthesis and recognition systems (: U.K ; : U.S)
E.J. Yannakoudakis and P.J. Hutton
出版情報: Chichester [West Sussex] : E. Horwood , New York : Halsted Press, 1987  184 p. ; 25 cm
シリーズ名: Ellis Horwood series in computers and their applications
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3.

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図書
3. Computer speech processing
edited by Frank Fallside and William A. Woods
出版情報: Englewood Cliffs, NJ : Prentice-Hall International, c1985  xxi, 506 p. ; 24 cm
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4.

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4. Speech processing and synthesis toolboxes
D. G. Childers
出版情報: New York : Wiley, c2000  ix, 483 p. ; 26 cm
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目次情報: 続きを見る
Speech Analysis Toolbox
Speech Production, Labeling, and Characteristics
Data and Measurements
Linear Prediction
Speech Synthesis and a Formant Speech Synthesis Toolbox
Vocos - A Voice Conversion Toolbox
Time Modification of Speech Toolbox
Animated Vocal Fold Model Toolbox
Articulatory Speech Synthesis Toolbox
Appendices
Index
Speech Analysis Toolbox
Speech Production, Labeling, and Characteristics
Data and Measurements
5.

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5. Applications of digital signal processing to audio and acoustics (: alk. paper)
edited by Mark Kahrs, Karlheinz Brandenburg
出版情報: Boston ; Dordrecht ; London : Kluwer, c1998  xxxi, 545 p. ; 25 cm
シリーズ名: The Kluwer international series in engineering and computer science ; SECS 437
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6.

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6. Speech technology for telecommunications
edited by F. A. Westall, R. D. Johnston and A. V. Lewis
出版情報: London : Chapman & Hall, 1998  xii, 564 p. ; 24cm
シリーズ名: BT telecommunications series ; 11
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7.

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7. Modern methods of speech processing
edited by Ravi P. Ramachandran, Richard J. Mammone
出版情報: Boston : Kluwer Academic Publishers, c1995  xvii, 470 p. ; 25 cm
シリーズ名: The Kluwer international series in engineering and computer science ; SECS 327 . VLSI, computer architecture, and digital signal processing
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8.

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8. Speech coding and synthesis
edited by W.B. Kleijn, K.K. Paliwal
出版情報: Amsterdam ; New York : Elsevier, 1995  xvii, 755 p. ; 25 cm
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目次情報: 続きを見る
Chapter Headings. An Introduction to Speech Coding / W.B. Kleijn ; K.K. Paliwal
Speech Coding Standards / R.V. Cox
Linear-Prediction based Analysis-by-Synthesis Coding / P. Kroon
Sinusoidal Coding / R.J. McAulay ; T.F. Quatieri
Waveform Interpolation for Coding and Synthesis / J. Haagen
Low-Delay Coding of Speech / J.-H. Chen
Multimode and Variable-Rate Coding of Speech / A. Das et al.
Wideband Speech Coding / J-P. Adoul ; R. Lefebvre
Vector Quantization for Speech Transmission / P. Hedelin et al.
Theory for Transmission of Vector Quantization Data
Waveform Coding and Auditory Masking / R. Veldhuis ; A. Kohlrausch
Quantization of LPC Parameters
Evaluation of Speech Coders
A Robust Algorithm for Pitch Tracking (RAPT) / D. Talkin
Time-Domain and Frequency-Domain Techniques for Prosodic Modification of Speech / E. Moulines ; W. Verhelst
Nonlinear Processing of Speech / G. Kubin
An Approach to Text-to-Speech Synthesis / R. Sp roat ; J. Olive
The Generation of Prosodic Structure and Intonation in Speech Synthesis / J. Terken ; R. Collier
Computation of Timing in Text-to-Speech Synthesis / J.P.H. van Santen
Objective Optimization in Algorithms for Text-to-Speech Synthesis / Y. Sagisaka ; N. Iwahashi
Quality Evaluation of Synthesized Speech / V.J. van Heuven ; R. van Bezooijen
Subject Index
Chapter Headings. An Introduction to Speech Coding / W.B. Kleijn ; K.K. Paliwal
Speech Coding Standards / R.V. Cox
Linear-Prediction based Analysis-by-Synthesis Coding / P. Kroon
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9. Speech (: softbound ; : casebound)
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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10.

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10. Digital signal processing (: softbound ; : casebound)
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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11.

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11. Statistical signal and array processing (: softbound ; : casebound)
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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12.

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図書
12. Image and multi-dimensional signal processing ; Signal processing applications development (: softbound ; : casebound)
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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13.

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13. Special sessions ; Audio and electroacoustics ; Underwater acoustic signal processing ; Very large scale integration for signal processing ; Neural networks for signal processing ; Joint sessions (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway, NJ : IEEE Service Center, c1995  p. 2763-3662 ; 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. 5
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14. Subspace methods of pattern recognition (: Research Studies Press ; : John Wiley & Sons)
E. Oja
出版情報: Letchworth, Hertfordshire, England : Research Studies Press , New York : Wiley, c1983  xii, 187 p. ; 24 cm
シリーズ名: Electronic & electrical engineering research studies ; . Pattern recognition & image processing series ; 6
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15.

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15. Computational models of discourse
edited by Michael Brady and Robert C. Berwick ; contributors, James Allen ... [et al.]
出版情報: Cambridge, Mass. : MIT Press, c1983  xxiii, 403 p. ; 24 cm
シリーズ名: The MIT Press series in artificial intelligence
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目次情報: 続きを見る
Foreword / Michael Brady
Preface / David Israel
Computational aspects of discourse / Robert C. Berwick
Recognizing intentions from natural language utterances / James Allen
Cooperative responses from a portable natural language database query system / Jerrold Kaplan
Natural language generation as a computational problem: an Introduction / David D. McDonald
Focusing in the comprehension of definite anaphora / Candace L. Sidner
So what can we talk about now? / Bonnie L. Webber
Bibliography
Index
Foreword / Michael Brady
Preface / David Israel
Computational aspects of discourse / Robert C. Berwick
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16. Pitch determination of speech signals : algorithms and devices (: Germany ; : U.S.)
Wolfgang Hess
出版情報: Berlin ; Tokyo : Springer-Verlag, 1983  xiv, 698 p. ; 23 cm
シリーズ名: Springer series in information sciences ; 3
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17.

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17. ICASSP 91 : 1991 International Conference on Acoustics, Speech, and Signal Processing, May 14-17, 1991, the Sheraton Centre Hotel & Towers, Toronto, Ontario, Canada (v. 1 : casebound - v. 5 : softbound)
sponsored by The Institute of Electrical and Electronics Engineers, Acoustics, Speech and Signal Processing Society
出版情報: New York : IEEE, c1991  5 v. ; 29 cm
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18.

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18. Advances in speech signal processing
edited by Sadaoki Furui, M. Mohan Sondhi
出版情報: New York : M. Dekker, c1992  xv, 871 p. ; 24 cm
シリーズ名: Electrical engineering and electronics
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19.

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19. Neural networks and speech processing
by David P. Morgan, Christopher L. Scofield ; foreword by Leon N. Cooper
出版情報: Boston : Kluwer Academic Publishers, c1991  xvi, 391 p. ; 25 cm
シリーズ名: The Kluwer international series in engineering and computer science ; . VLSI, computer architecture, and digital signal processing
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20.

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20. ICASSP-92 : 1992 International Conference on Acoustics, Speech, and Signal Processing, March 23-26, 1992, the San Francisco Marriott, San Francisco, California (v. 1 : casebound - v. 5 : softbound)
sponsored by The Institute of Electrical and Electronics Engineers, Signal Processing Society
出版情報: Piscataway : Additional copies may be ordered from: IEEE Service Center, c1992  5 v. ; 29 cm
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21.

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21. Cognitive models of speech processing : psycholinguistic and computational perspectives
edited by Gerry T.M. Altmann
出版情報: Cambridge, Mass. : MIT Press, c1990  x, 540 p. ; 24 cm
シリーズ名: ACL-MIT Press series in natural language processing / Aravind K. Joshi, Mark Liberman, and Karen Sparck Jones, editors
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22.

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22. Speech enhancement
edited by Jae S. Lim
出版情報: Englewood Cliffs, N.J. : Prentice-Hall, c1983  xv, 363 p. ; 29 cm
シリーズ名: Prentice Hall signal processing series
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23.

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23. ICASSP 90 : 1990 International Conference on Acoustics, Speech, and Signal Processing, April 3-6, 1990, Albuquerque Convention Centre, Albuquerque, New Mexico, USA (v. 1 - v. 5)
sponsored by The Institute of Electrical and Electronics Engineers, Acoustics, Speech and Signal Processing Society
出版情報: New York : IEEE, c1990  5 v. ; 29 cm
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24.

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24. Speech intelligibility and speaker recognition (: Dowden, Hutchinson & Ross ; : Halsted Press)
edited by Mones E. Hawley
出版情報: Stroudsburg, Pa. : Dowden, Hutchinson & Ross , [New York] : exclusive distributor, Halsted Press, c1977  xvi, 451 p. ; 26 cm
シリーズ名: Benchmark papers in acoustics series ; 11
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25.

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25. Recent advances in speech understanding and dialog systems (: Germany ; : U.S.)
edited by H. Niemann, M. Lang, G. Sagerer
出版情報: Berlin ; Tokyo : Springer-Verlag, c1988  x, 521 p. ; 25 cm
シリーズ名: NATO ASI series ; ser. F . Computer and systems sciences ; v. 46
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26.

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26. Digital processing of speech signals
Lawrence R. Rabiner, Ronald W. Schafer
出版情報: Englewood Cliffs, N.J. : Prentice-Hall, c1978  xvi, 512 p. ; 25 cm
シリーズ名: Prentice Hall signal processing series
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目次情報: 続きを見る
Introduction / 1：
Fundamentals of Digital Speech Processing / 2：
Digital Models for the Speech Signal / 3：
Time-Domain Models for Speech Processing / 4：
Digital Representation of the Speech Waveform / 5：
Short-Time Fourier Analysis / 6：
Homomorphic Speech Processing / 7：
Linear Predictive Coding of Speech / 8：
Digital Speech Processing for Man-Machine Communication by Voice / 9：
Introduction / 1：
Fundamentals of Digital Speech Processing / 2：
Digital Models for the Speech Signal / 3：
27.

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27. Human communication : a unified view
Edited by Edward E. David, Jr. [and] Peter B. Denes
出版情報: New York : McGraw-Hill, 1972  xvii, 458 p ; 24 cm
シリーズ名: Inter-university electronics series ; no. 15
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28.

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28. A production system version of the Hearsay-II speech understanding system
by Donald L. McCracken
出版情報: Ann Arbor, Mich. : UMI Research Press, c1981  x, 139 p. ; 24 cm
シリーズ名: Computer science ; Artificial intelligence ; no. 2
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29.

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29. Linear prediction of speech (:gw ; :us)
J.D. Markel, A.H. Gray, Jr
出版情報: Berlin ; New York : Springer-Verlag, 1976  xii, 288 p. ; 25 cm
シリーズ名: Communication and cybernetics ; 12
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30.

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30. International Conference on Speech Input/Output, Techniques and Applications, 24-26 March 1986 (pbk.)
organised by the Science, Education, and Technology Division of the Institution of Electrical Engineers ; in association with the British Society of Audiology ... [et al.]
出版情報: London : The Institution, c1986  xii, 323 p. ; 30 cm
シリーズ名: IEE conference publication ; no. 258
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31.

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31. ICASSP 89 : 1989 International Conference on Acoustics, Speech, and Signal Processing, 23-26 May 1989, Scottish Exhibition+Conference Centre, Glasgow, Scotland (v. 1 - v. 4)
sponsored by The Institute of Electrical and Electronics Engineers, Acoustics, Speech and Signal Processing Society
出版情報: New York : IEEE, c1989  4 v. (2833 p.) ; 29 cm
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32.

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32. Plenary, special, audio, underwater acoustics, VLSI, neural networks (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : Additional copies may be ordered from: IEEE Service Center, c1993  652 p. ; 28 cm
シリーズ名: ICASSP-93 : 1993 International Conference on Acoustics, Speech, and Signal Processing, April 27-30, 1993, Minneapolis Convention Center, Minneapolis, Minnesota, USA / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 1
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33.

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33. Speech processing (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : Additional copies may be ordered from: IEEE Service Center, c1993  735 p. ; 28 cm
シリーズ名: ICASSP-93 : 1993 International Conference on Acoustics, Speech, and Signal Processing, April 27-30, 1993, Minneapolis Convention Center, Minneapolis, Minnesota, USA / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 2
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34.

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34. Digital speech processing (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : Additional copies may be ordered from: IEEE Service Center, c1993  606 p. ; 28 cm
シリーズ名: ICASSP-93 : 1993 International Conference on Acoustics, Speech, and Signal Processing, April 27-30, 1993, Minneapolis Convention Center, Minneapolis, Minnesota, USA / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 3
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35.

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35. Statistical signal and array processing (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : Additional copies may be ordered from: IEEE Service Center, c1993  559 p. ; 28 cm
シリーズ名: ICASSP-93 : 1993 International Conference on Acoustics, Speech, and Signal Processing, April 27-30, 1993, Minneapolis Convention Center, Minneapolis, Minnesota, USA / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 4
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36.

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36. Image and multidimensional signal processing (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : Additional copies may be ordered from: IEEE Service Center, c1993  681 p. ; 28 cm
シリーズ名: ICASSP-93 : 1993 International Conference on Acoustics, Speech, and Signal Processing, April 27-30, 1993, Minneapolis Convention Center, Minneapolis, Minnesota, USA / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 5
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37.

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37. Speech and audio coding for wireless and network applications
edited by Bishnu S. Atal, Vladimir Cuperman, Allen Gersho
出版情報: Boston : Kluwer Academic, c1993  viii, 283 p. ; 25 cm
シリーズ名: The Kluwer international series in engineering and computer science ; Communications and information theory
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38.

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38. Digital speech : coding for low bit rate communication systems (: pbk)
A.M. Kondoz
出版情報: Chichester ; New York : J. Wiley, c1994  xii, 442 p. ; 25 cm
シリーズ名: Wiley series in communication and distributed systems
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目次情報: 続きを見る
Preface
Acknowledgements
Introduction / 1：
Coding Strategies and Standards / 2：
Speech Coding Techniques / 2.1：
Algorithm Objectives and Requirements / 2.3：
Standard Speech Coders / 2.4：
Summary / 2.5：
Bibliography
Sampling and Quantization / 3：
Sampling / 3.1：
Scalar Quantization / 3.3：
Vector Quantization / 3.4：
Speech Signal Analysis and Modelling / 3.5：
Short-Time Spectral Analysis / 4.1：
Linear Predictive Modelling of Speech Signals / 4.3：
Pitch Prediction / 4.4：
Efficient LPC Quantization Methods / 4.5：
Alternative Representation of LPC / 5.1：
LPC to LSF Transformation / 5.3：
LSF to LPC Transformation / 5.4：
Properties of LSFs / 5.5：
LSF Quantization / 5.6：
Codebook Structures / 5.7：
MSVQ Performance Analysis / 5.8：
Inter-frame Correlation / 5.9：
Improved LSF Estimation Through Anti-Aliasing Filtering / 5.10：
Pitch Estimation and Voiced-Unvoiced Classification of Speech / 5.11：
Pitch Estimation Methods / 6.1：
Voiced-Unvoiced Classification / 6.3：
Analysis by Synthesis LPC Coding / 6.4：
Generalized AbS Coding / 7.1：
Code-Excited Linear Predictive Coding / 7.3：
Harmonic Speech Coding / 7.4：
Sinusoidal Analysis and Synthesis / 8.1：
Parameter Estimation / 8.3：
Common Harmonic Coders / 8.4：
Multimode Speech Coding / 8.5：
Design Challenges of a Hybrid Coder / 9.1：
Summary of Hybrid Coders / 9.3：
Synchronized Waveform-Matched Phase Model / 9.4：
Hybrid Encoder / 9.5：
Speech Classification / 9.6：
Hybrid Decoder / 9.7：
Performance Evaluation / 9.8：
Quantization Issues of Hybrid Coder Parameters / 9.9：
Variable Bit Rate Coding / 9.10：
Acoustic Noise and Channel Error Performance / 9.11：
Voice Activity Detection / 9.12：
Standard VAD Methods / 10.1：
Likelihood-Ratio-Based VAD / 10.3：
Speech Enhancement / 10.4：
Review of STSA-based Speech Enhancement / 11.1：
Noise Adaptation / 11.3：
Echo Cancellation / 11.4：
Index / 11.5：
Preface
Acknowledgements
Introduction / 1：
39.

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39. Advanced speech applications : European research on speech technology (: gw ; : us)
K. Varghese, S. Pfleger, J.-P. Lefèvre (eds.)
出版情報: Berlin ; New York : Springer-Verlag, c1994  x, 319 p. ; 24 cm
シリーズ名: Research reports ESPRIT ; Project group speech technology ; v. 1
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40.

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40. Fundamentals of speech recognition (: pbk)
Lawrence Rabiner, Biing-Hwang Juang
出版情報: Englewood Cliffs, NJ : Prentice Hall, c1993  xxxv, 507 p. ; 24 cm
シリーズ名: Prentice Hall signal processing series
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目次情報: 続きを見る
Fundamentals of Speech Recognition / 1：
The Speech Signal: Production, Perception, and Acoustic-Phonetic Characterization / 2：
Signal Processing and Analysis Methods for Speech Recognition / 3：
Pattern Comparison Techniques / 4：
Speech Recognition System Design and Implementation Issues / 5：
Theory and Implementation of Hidden Markov Models / 6：
Speech Recognition Based on Connected Word Models / 7：
Large Vocabulary Continuous Speech Recognition / 8：
Task-Oriented Applications of Automatic Speech Recognition / 9：
Fundamentals of Speech Recognition / 1：
The Speech Signal: Production, Perception, and Acoustic-Phonetic Characterization / 2：
Signal Processing and Analysis Methods for Speech Recognition / 3：
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41. Discrete-time processing of speech signals
John R. Deller, John G. Proakis, John H.L. Hansen
出版情報: New York : Macmillan Pub. Co. , Toronto : Maxwell Maxmillan Canada , New York : Maxwell Macmillan International, c1993  xx, 908 p. ; 25 cm
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42.

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42. ICASSP-93 : 1993 International Conference on Acoustics, Speech, and Signal Processing, April 27-30, 1993, Minneapolis Convention Center, Minneapolis, Minnesota, USA (v. 1 - v. 5)
sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society
出版情報: Piscataway : Additional copies may be ordered from: IEEE Service Center, c1993  5 v. ; 29 cm
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43.

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43. Speech processing 1 (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : IEEE Service Center, c1994  640 p. ; 30 cm
シリーズ名: ICASSP-94 : 1994 International Conference on Acoustics, Speech, and Signal Processing, April 19-22, 1994, Adelaide Convention Centre, Adelaide, South Australia / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 1
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44.

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44. Speech processing 2, audio, underwater acoustics, VLSI & neural networks (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : IEEE Service Center, c1994  688 p. ; 30 cm
シリーズ名: ICASSP-94 : 1994 International Conference on Acoustics, Speech, and Signal Processing, April 19-22, 1994, Adelaide Convention Centre, Adelaide, South Australia / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 2
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45.

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図書
45. Digital signal processing (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : IEEE Service Center, c1994  628 p. ; 30 cm
シリーズ名: ICASSP-94 : 1994 International Conference on Acoustics, Speech, and Signal Processing, April 19-22, 1994, Adelaide Convention Centre, Adelaide, South Australia / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 3
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46.

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図書
46. Statistical signal and array processing (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : IEEE Service Center, c1994  596 p. ; 30 cm
シリーズ名: ICASSP-94 : 1994 International Conference on Acoustics, Speech, and Signal Processing, April 19-22, 1994, Adelaide Convention Centre, Adelaide, South Australia / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 4
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47.

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図書
47. Image and multidimensional signal processing (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : IEEE Service Center, c1994  632 p. ; 30 cm
シリーズ名: ICASSP-94 : 1994 International Conference on Acoustics, Speech, and Signal Processing, April 19-22, 1994, Adelaide Convention Centre, Adelaide, South Australia / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 5
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48.

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48. Plenary and special sessions (: softbound ; : casebound)
IEEE International Conference on Acoustics, Speech, and Signal Processing ; IEEE Signal Processing Society
出版情報: Piscataway : IEEE Service Center, c1994  198 p. ; 30 cm
シリーズ名: ICASSP-94 : 1994 International Conference on Acoustics, Speech, and Signal Processing, April 19-22, 1994, Adelaide Convention Centre, Adelaide, South Australia / sponsored by the Institute of Electrical and Electronics Engineers, Signal Processing Society ; Vol. 6
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49.

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49. Noise reduction in speech applications
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：
50.

図書
図書
50. Speech coding algorithms : foundation and evolution of standardized coders
Wai C. Chu
出版情報: New York : John Wiley & Sons, Inc., c2003  xxiv, 558 p. ; 25 cm
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Preface
Acronyms
Notation
Introduction / 1：
Overview of Speech Coding / 1.1：
Classification of Speech Coders / 1.2：
Speech Production and Modeling / 1.3：
Some Properties of the Human Auditory System / 1.4：
Speech Coding Standards / 1.5：
About Algorithms / 1.6：
Summary and References / 1.7：
Signal Processing Techniques / 2：
Pitch Period Estimation / 2.1：
All-Pole and All-Zero Filters / 2.2：
Convolution / 2.3：
Exercises / 2.4：
Stochastic Processes and Models / 3：
Power Spectral Density / 3.1：
Periodogram / 3.2：
Autoregressive Model / 3.3：
Autocorrelation Estimation / 3.4：
Other Signal Models / 3.5：
Linear Prediction / 3.6：
The Problem of Linear Prediction / 4.1：
Linear Prediction Analysis of Nonstationary Signals / 4.2：
Examples of Linear Prediction Analysis of Speech / 4.3：
The Levinson-Durbin Algorithm / 4.4：
The Leroux-Gueguen Algorithm / 4.5：
Long-Term Linear Prediction / 4.6：
Synthesis Filters / 4.7：
Practical Implementation / 4.8：
Moving Average Prediction / 4.9：
Scalar Quantization / 4.10：
Uniform Quantizer / 5.1：
Optimal Quantizer / 5.3：
Quantizer Design Algorithms / 5.4：
Algorithmic Implementation / 5.5：
Pulse Code Modulation and its Variants / 5.6：
Uniform Quantization / 6.1：
Nonuniform Quantization / 6.2：
Differential Pulse Code Modulation / 6.3：
Adaptive Schemes / 6.4：
Vector Quantization / 6.5：
Multistage VQ / 7.1：
Predictive VQ / 7.5：
Other Structured Schemes / 7.6：
Scalar Quantization of Linear Prediction Coefficient / 7.7：
Spectral Distortion / 8.1：
Quantization Based on Reflection Coefficient and Log Area Ratio / 8.2：
Line Spectral Frequency / 8.3：
Quantization Based on Line Spectral Frequency / 8.4：
Interpolation of LPC / 8.5：
Linear Prediction Coding / 8.6：
Speech Production Model / 9.1：
Structure of the Algorithm / 9.2：
Voicing Detector / 9.3：
The FS1015 LPC Coder / 9.4：
Limitations of the LPC Model / 9.5：
Regular-Pulse Excitation Coders / 9.6：
Multipulse Excitation Model / 10.1：
Regular-Pulse-Excited-Long-Term Prediction / 10.2：
Code-Excited Linear Prediction / 10.3：
The CELP Speech Production Model / 11.1：
The Principle of Analysis-by-Synthesis / 11.2：
Encoding and Decoding / 11.3：
Excitation Codebook Search / 11.4：
Postfilter / 11.5：
The Federal Standard Version of Celp / 11.6：
Improving the Long-Term Predictor / 12.1：
The Concept of the Adaptive Codebook / 12.2：
Incorporation of the Adaptive Codebook to the CELP Framework / 12.3：
Stochastic Codebook Structure / 12.4：
Adaptive Codebook Search / 12.5：
Stochastic Codebook Search / 12.6：
Encoder and Decoder / 12.7：
Vector Sum Excited Linear Prediction / 12.8：
The Core Encoding Structure / 13.1：
Search Strategies for Excitation Codebooks / 13.2：
Excitation Codebook Searches / 13.3：
Gain Related Procedures / 13.4：
Low-Delay Celp / 13.5：
Strategies to Achieve Low Delay / 14.1：
Basic Operational Principles / 14.2：
Linear Prediction Analysis / 14.3：
Backward Gain Adaptation / 14.4：
Codebook Training / 14.6：
Vector Quantization of Linear Prediction Coefficient / 14.8：
Correlation Among the LSFs / 15.1：
Split VQ / 15.2：
Algebraic Celp / 15.3：
Algebraic Codebook Structure / 16.1：
Adaptive Codebook / 16.2：
Algebraic Codebook Search / 16.3：
Gain Quantization Using Conjugate VQ / 16.5：
Other ACELP Standards / 16.6：
Mixed Excitation Linear Prediction / 16.7：
The MELP Speech Production Model / 17.1：
Fourier Magnitudes / 17.2：
Shaping Filters / 17.3：
Pitch Period and Voicing Strength Estimation / 17.4：
Encoder Operations / 17.5：
Decoder Operations / 17.6：
Source-Controlled Variable Bit-Rate Celp / 17.7：
Adaptive Rate Decision / 18.1：
LP Analysis and LSF-Related Operations / 18.2：
Decoding and Encoding / 18.3：
Speech Quality Assessment / 18.4：
The Scope of Quality and Measuring Conditions / 19.1：
Objective Quality Measurements for Waveform Coders / 19.2：
Subjective Quality Measures / 19.3：
Improvements on Objective Quality Measures / 19.4：
Minimum-Phase Property of the Forward Prediction-Error Filter / Appendix A：
Some Properties of Line Spectral Frequency / Appendix B：
Research Directions in Speech Coding / Appendix C：
Linear Combiner for Pattern Classification / Appendix D：
Celp: Optimal Long-Term Predictor to Minimize the Weighted Difference / Appendix E：
Review of Linear Algebra: Orthogonality, Basis, Linear Independence, and the Gram-Schmidt Algorithm / Appendix F：
Bibliography
Index
Preface
Acronyms
Notation
51.

図書
図書
51. Robustness in language and speech technology
edited by Jean-Claude Junqua and Gertjan van Noord
出版情報: Dordrecht ; Boston : Kluwer Academic Publishers, c2001  x, 269 p. ; 25 cm
シリーズ名: Text, speech, and language technology ; v. 17
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52.

図書

東工大
目次DB
図書
東工大
目次DB
52. Digital speech processing, synthesis, and recognition. 2nd ed., rev. and expanded
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
53.

図書
図書
53. Voice and audio compression for wireless communications. 2nd ed
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
54.

図書
図書
54. Speech enhancement : theory and practice
Philipos C. Loizou
出版情報: Boca Raton, Fla. : CRC Press, c2007  608 p. ; 24 cm
シリーズ名: Signal processing and communications series ; 30
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Preface
The Author
Introduction / Chapter 1：
Understanding the Enemy: Noise / 1.1：
Noise Sources / 1.1.1：
Noise and Speech Levels in Various Environments / 1.1.2：
Classes of Speech Enhancement Algorithms / 1.2：
Book Organization / 1.3：
References
Fundamentals / Part 1：
Discrete-Time Signal Processing and Short-Time Fourier Analysis / Chapter 2：
Discrete-Time Signals / 2.1：
Linear Time-Invariant Discrete-Time Systems / 2.2：
Difference Equations / 2.2.1：
Linear Convolution / 2.2.2：
The z-Transform / 2.3：
Properties / 2.3.1：
The z-Domain Transfer Function / 2.3.2：
Discrete-Time Fourier Transform / 2.4：
DTFT Properties / 2.4.1：
Discrete Fourier Transform / 2.4.2：
Windowing / 2.4.3：
Short-Time Fourier Transform / 2.5：
Definition / 2.5.1：
Interpretations of the STFT / 2.5.2：
Sampling the STFT in Time and Frequency / 2.5.3：
Short-Time Synthesis of Speech / 2.5.4：
Filterbank Summation for Short-Time Synthesis of Speech / 2.5.4.1：
Overlap-and-Add Method for Short-Time Synthesis / 2.5.4.2：
Spectrographic Analysis of Speech Signals / 2.6：
Summary / 2.7：
Speech Production and Perception / Chapter 3：
The Speech Signal / 3.1：
The Speech Production Process / 3.2：
Lungs / 3.2.1：
Larynx and Vocal Folds / 3.2.2：
Vocal Tract / 3.2.3：
Engineering Model of Speech Production / 3.3：
Classes of Speech Sounds / 3.4：
Acoustic Cues in Speech Perception / 3.5：
Vowels and Diphthongs / 3.5.1：
Semivowels / 3.5.2：
Nasals / 3.5.3：
Stops / 3.5.4：
Fricatives
Noise Compensation by Human Listeners / 3.6：
Intelligibility of Speech in Multiple-Talker Conditions / 4.1：
Effect of Masker's Spectral/Temporal Characteristics and Number of Talkers: Monaural Hearing / 4.1.1：
Effect of Source Spatial Location: Binaural Hearing / 4.1.2：
Acoustic Properties of Speech Contributing to Robustness / 4.2：
Shape of the Speech Spectrum / 4.2.1：
Spectral Peaks / 4.2.2：
Periodicity / 4.2.3：
Rapid Spectral Changes Signaling Consonants / 4.2.4：
Perceptual Strategies for Listening in Noise / 4.3：
Auditory Streaming / 4.3.1：
Listening in the Gaps and Glimpsing / 4.3.2：
Use of F0 Differences / 4.3.3：
Use of Linguistic Knowledge / 4.3.4：
Use of Spatial and Visual Cues / 4.3.5：
Algorithms / 4.4：
Spectral-Subtractive Algorithms / Chapter 5：
Basic Principles of Spectral Subtraction / 5.1：
A Geometric View of Spectral Subtraction / 5.2：
Upper Bounds on the Difference Between the Noisy and Clean Signals' Phases / 5.2.1：
Alternate Spectral-Subtractive Rules and Theoretical Limits / 5.2.2：
Shortcomings of the Spectral Subtraction Method / 5.3：
Spectral Subtraction Using Oversubtraction / 5.4：
Nonlinear Spectral Subtraction / 5.5：
Multiband Spectral Subtraction / 5.6：
MMSE Spectral Subtraction Algorithm / 5.7：
Extended Spectral Subtraction / 5.8：
Spectral Subtraction Using Adaptive Gain Averaging / 5.9：
Selective Spectral Subtraction / 5.10：
Spectral Subtraction Based on Perceptual Properties / 5.11：
Performance of Spectral Subtraction Algorithms / 5.12：
Wiener Filtering / 5.13：
Introduction to Wiener Filter Theory / 6.1：
Wiener Filters in the Time Domain / 6.2：
Wiener Filters in the Frequency Domain / 6.3：
Wiener Filters and Linear Prediction / 6.4：
Wiener Filters for Noise Reduction / 6.5：
Square-Root Wiener Filter / 6.5.1：
Parametric Wiener Filters / 6.5.2：
Iterative Wiener Filtering / 6.6：
Mathematical Speech Production Model / 6.6.1：
Statistical Parameter Estimation of the All-Pole Model in Noise / 6.6.2：
Imposing Constraints on Iterative Wiener Filtering / 6.7：
Across-Time Spectral Constraints / 6.7.1：
Across-Iterations Constraints / 6.7.2：
Constrained Iterative Wiener Filtering / 6.8：
Constrained Wiener Filtering / 6.9：
Mathematical Definitions of Speech and Noise Distortions / 6.9.1：
Limiting the Noise Distortion Level / 6.9.2：
Estimating the Wiener Gain Function / 6.10：
Incorporating Psychoacoustic Constraints in Wiener Filtering / 6.11：
Shaping the Noise Distortion in the Frequency Domain / 6.11.1：
Using Masking Thresholds as Constraints / 6.11.2：
Codebook-Driven Wiener Filtering / 6.12：
Audible Noise Suppression Algorithm / 6.13：
Statistical-Model-Based Methods / 6.14：
Maximum-Likelihood Estimators / 7.1：
Bayesian Estimators / 7.2：
MMSE Estimator / 7.3：
MMSE Magnitude Estimator / 7.3.1：
MMSE Complex Exponential Estimator / 7.3.2：
Estimating the A Priori SNR / 7.3.3：
Maximum-Likelihood Method / 7.3.3.1：
Decision-Directed Approach / 7.3.3.2：
Improvements to the Decision-Directed Approach / 7.4：
Reducing the Bias / 7.4.1：
Improving the Adaptation Speed / 7.4.2：
Implementation and Evaluation of the MMSE Estimator / 7.5：
Elimination of Musical Noise / 7.6：
Log-MMSE Estimator / 7.7：
MMSE Estimation of the pth-Power Spectrum / 7.8：
MMSE Estimators Based on Non-Gaussian Distributions / 7.9：
Maximum A Posteriori (MAP) Estimators / 7.10：
General Bayesian Estimators / 7.11：
Perceptually Motivated Bayesian Estimators / 7.12：
Psychoacoustically Motivated Distortion Measure / 7.12.1：
Weighted Euclidean Distortion Measure / 7.12.2：
Itakura-Saito Measure / 7.12.3：
Cosh Measure / 7.12.4：
Weighted Likelihood Ratio / 7.12.5：
Modified IS Distortion Measure / 7.12.6：
Incorporating Speech Absence Probability in Speech Enhancement / 7.13：
Incorporating Speech-Presence Uncertainty in Maximum-Likelihood Estimators / 7.13.1：
Incorporating Speech-Presence Uncertainty in MMSE Estimators / 7.13.2：
Incorporating Speech-Presence Uncertainty in Log-MMSE Estimators / 7.13.3：
Implementation Issues Regarding A Priori SNR Estimation / 7.13.4：
Methods for Estimating the A Priori Probability of Speech Absence / 7.14：
Subspace Algorithms / 7.15：
Definitions / 8.1：
Projections / 8.1.2：
Low-Rank Modeling / 8.1.3：
Using SVD for Noise Reduction: Theory / 8.2：
SVD Analysis of "Noisy" Matrices / 8.2.1：
Least-Squares and Minimum-Variance Estimates of the Signal Matrix / 8.2.2：
SVD-Based Algorithms: White Noise / 8.3：
SVD Synthesis of Speech / 8.3.1：
Determining the Effective Rank / 8.3.2：
Noise Reduction Algorithm / 8.3.4：
SVD-Based Algorithms: Colored Noise / 8.4：
SVD-Based Methods: A Unified View / 8.5：
EVD-Based Methods: White Noise / 8.6：
Eigenvalue Analysis of "Noisy" Matrices / 8.6.1：
Subspace Methods Based on Linear Estimators / 8.6.2：
Linear Minimum Mean-Square Estimator (LMMSE) / 8.6.2.1：
Time-Domain-Constrained Estimator / 8.6.2.2：
Spectral-Domain-Constrained Estimator / 8.6.2.3：
Implementation / 8.6.3：
Covariance Estimation / 8.6.3.1：
Estimating the Lagrange Multiplier / 8.6.3.2：
Estimating the Signal Subspace Dimension / 8.6.3.3：
EVD-Based Methods: Colored Noise / 8.7：
Prewhitening Approach / 8.7.1：
Signal/Noise KLT-Based Method / 8.7.2：
Adaptive KLT Approach / 8.7.3：
Subspace Approach with Embedded Prewhitening / 8.7.4：
Spectrum-Domain-Constrained Estimator / 8.7.4.1：
Relationship Between Subspace Estimators and Prewhitening / 8.7.4.3：
EVD-Based Methods: A Unified View / 8.8：
Perceptually Motivated Subspace Algorithms / 8.9：
Fourier to Eigen-Domain Relationship / 8.9.1：
Incorporating Psychoacoustic Model Constraints / 8.9.2：
Incorporating Auditory Masking Constraints / 8.9.3：
Subspace-Tracking Algorithms / 8.10：
Block Algorithms / 8.10.1：
Recursive Algorithms / 8.10.2：
Modified Eigenvalue Problem Algorithms / 8.10.2.1：
Adaptive Algorithms / 8.10.2.2：
Using Subspace-Tracking Algorithms in Speech Enhancement / 8.10.3：
Noise Estimation Algorithms / 8.11：
Voice Activity Detection Vs. Noise Estimation / 9.1：
Introduction to Noise Estimation Algorithms / 9.2：
Minimal-Tracking Algorithms / 9.3：
Minimum Statistics (MS) Noise Estimation / 9.3.1：
Principles / 9.3.1.1：
Derivation of the Bias Factor / 9.3.1.2：
Derivation of Optimal Time- and Frequency-Dependent Smoothing Factor / 9.3.1.3：
Searching for the Minimum / 9.3.1.4：
Minimum Statistics Algorithm / 9.3.1.5：
Continuous Spectral Minimum Tracking / 9.3.2：
Time-Recursive Averaging Algorithms for Noise Estimation / 9.4：
SNR-Dependent Recursive Averaging / 9.4.1：
Weighted Spectral Averaging / 9.4.2：
Recursive Averaging Algorithms Based on Signal-Presence Uncertainty / 9.4.3：
Likelihood Ratio Approach / 9.4.3.1：
Minima-Controlled Recursive Averaging (MCRA) Algorithms / 9.4.3.2：
Histogram-Based Techniques / 9.5：
Other Noise Estimation Algorithms / 9.6：
Objective Comparison of Noise Estimation Algorithms / 9.7：
Evaluation / 9.8：
Evaluating Performance of Speech Enhancement Algorithms / Chapter 10：
Quality vs. Intelligibility / 10.1：
Evaluating Intelligibility of Processed Speech / 10.2：
Nonsense Syllable Tests / 10.2.1：
Word Tests / 10.2.2：
Phonetically Balanced Word Tests / 10.2.2.1：
Rhyming Word Tests / 10.2.2.2：
Sentence Tests / 10.2.3：
Measuring Speech Intelligibility / 10.2.4：
Speech Reception Threshold / 10.2.4.1：
Using Statistical Tests to Assess Significant Differences: Recommended Practice / 10.2.4.2：
Evaluating Quality of Processed Speech / 10.3：
Relative Preference Methods / 10.3.1：
Absolute Category Rating Methods / 10.3.2：
Mean Opinion Scores / 10.3.2.1：
Diagnostic Acceptability Measure / 10.3.2.2：
The ITU-T P.835 Standard / 10.3.2.3：
Evaluating Reliability of Quality Judgments: Recommended Practice / 10.4：
Intrarater Reliability Measures / 10.4.1：
Interrater Reliability Measures / 10.4.2：
Objective Quality Measures / 10.5：
Segmental SNR Measures: Time and Frequency / 10.5.1：
Spectral Distance Measures Based on LPC / 10.5.2：
Perceptually Motivated Measures / 10.5.3：
Weighted Spectral Slope (WSS) Distance Measure / 10.5.3.1：
Bark Distortion Measures / 10.5.3.2：
Perceptual Evaluation of Speech Quality (PESQ) Measure / 10.5.3.3：
Composite Measures / 10.5.4：
Nonintrusive Objective Quality Measures / 10.6：
Figures of Merit of Objective Quality Measures / 10.7：
Challenges and Future Directions in Objective Quality Evaluation / 10.8：
Comparison of Speech Enhancement Algorithms / 10.9：
NOIZEUS: A Noisy Speech Corpus for Quality Evaluation of Speech Enhancement Algorithms / 11.1：
Comparison of Speech Enhancement Algorithms: Quality / 11.2：
Quality Evaluation: Procedure / 11.2.1：
Subjective Quality Evaluation: Results / 11.2.2：
Within-Class Algorithm Comparisons / 11.2.3：
Across-Class Algorithm Comparisons / 11.2.4：
Comparisons in Reference to Noisy Speech / 11.2.5：
Contribution of Speech and Noise Distortion to Judgment of Overall Quality / 11.2.6：
Summary of Findings / 11.2.7：
Comparison of Speech Enhancement Algorithms: Intelligibility / 11.3：
Listening Tests: Procedure / 11.3.1：
Intelligibility Evaluation: Results / 11.3.2：
Intelligibility Comparison Among Algorithms / 11.3.3：
Intelligibility Comparison Against Noisy Speech / 11.3.4：
Comparison of Objective Measures for Quality Evaluation / 11.3.5：
Objective Measures / 11.4.1：
Correlations of Objective Measures with Quality / 11.4.2：
Special Functions and Integrals / 11.4.3：
Derivation of the MMSE Estimator / Appendix B：
Speech Databases and MATLAB Code / Appendix C：
Index
Preface
The Author
Introduction / Chapter 1：
55.

図書
図書
55. Applied speech technology
edited by A. Syrdal, R. Bennett, S. Greenspan
出版情報: Boca Raton : CRC Press, c1995  viii, 641 p. ; 25 cm
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56.

図書
図書
56. Real world speech processing
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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目次情報: 続きを見る
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
57.

図書
図書
57. Springer handbook of speech processing
Jacob Benesty, M. Mohan Sondhi, Yiteng Huang (eds.)
出版情報: Berlin : Springer, c2008  xxxvi, 1176 p. ; 25 cm.
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58.

図書
図書
58. Pattern recognition in speech and language processing
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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59.

電子ブック
ＥＢ
59. Advances in Chinese spoken language processing (: electronic bk)
Chin-Hui Lee ... [et al.]
出版情報: World Scientific  1 online resource (xvi, 545 p.)
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目次情報: 続きを見る
Preface
List of Contributors
Principles of CSLP / Part I：
Speech Analysis: The Production-Perception Perspective / L. Deng ; J. DangChapter 1：
Phonetic and Phonological Background of Chinese Spoken Languages / C.-C. KuoChapter 2：
Prosody Analysis / C. TsengChapter 3：
Tone Modeling for Speech Synthesis / S.-H. Chen ; H.-m. WangChapter 4：
Mandarin Text-To-Speech Synthesis / R.-H. Wang ; J. Tao ; M. ChuChapter 5：
Large Vocabulary Continuous Speech Recognition for Mandarin Chinese: Principles, Application Tasks and Prototype Examples / L.-s. LeeChapter 6：
Acoustic Modeling for Mandarin Large Vocabulary Continuous Speech Recognition / M.-Y. HwangChapter 7：
Tone Modeling for Speech Recognition / T. Lee ; Y. QianChapter 8：
Some Advances in Language Modeling / C.-H. Chueh ; M.-S. Wu ; J.-T. ChienChapter 9：
Spontaneous Mandarin Speech Pronunciation Modeling / P. Fung ; Y. LiuChapter 10：
Corpus Design and Annotation for Speech Synthesis and Recognition / A. Li ; Y. ZuChapter 11：
CSLP Technology Integration / Part II：
Speech-to-Speech Translation / Y. Gao ; L. Gu ; B. ZhouChapter 12：
Spoken Document Retrieval and Summarization / B. ChenChapter 13：
Speech Act Modeling and Verification in Spoken Dialogue Systems / C.-H. Wu ; J.-F. Yeh ; G.-L. YanChapter 14：
Transliteration / H. Li ; S. Bai ; J.-S. KuoChapter 15：
Cantonese Speech Recognition and Synthesis / P. C. Ching ; W. K. Lo ; H. M. MengChapter 16：
Taiwanese Min-nan Speech Recognition and Synthesis / R.-y. Lyu ; M.-s. Liang ; D.-c. Lyu ; Y.-c. ChiangChapter 17：
Putonghua Proficiency Test and Evaluation / Q. Liu ; S. WeiChapter 18：
Systems, Applications and Resources / Part III：
Audio-Based Digital Content Management and Retrieval / B. Xu ; S. Zhang ; T. HuangChapter 19：
Multilingual Dialog Systems / Chapter 20：
Directory Assistance System / J.-K. Chen ; C.-C. YangChapter 21：
Robust Car Navigation System / J.-F. Wang ; H.-C. Wang ; J.-C. WangChapter 22：
CSLP Corpora and Language Resources / T. F. ZhengChapter 23：
Index
Preface
List of Contributors
Principles of CSLP / Part I：
60.

電子ブック
ＥＢ
60. Speech and audio signal processing : processing and perception of speech and music. 2nd ed (: electronic bk)
Ben Gold, Nelson Morgan, Dan Ellis ; with contributions from Hervé Bourlard ... [et al.]
出版情報: Wiley Online Library, 2011  1 online resource (xxii, 661p.)
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Preface To The 2011 Edition
Introduction / Chapter 1：
Historical Background / Part I：
Synthetic A Udio: A Brief History / Chapter 2：
Speech Analysis And Synthesis Overview / Chapter 3：
Brief History Of Automatic Speech Recognition / Chapter 4：
Speech-Recognition Overview / Chapter 5：
Mathematical Background / Part II：
Digital Signal Processing / Chapter 6：
Digital Filtersand Discrete Fourier Transform / Chapter 7：
Pattern Classification / Chapter 8：
Statistical Pattern Classification / Chapter 9：
Acoustics / Part III：
Wave Basics / Chapter 10：
Acoustic Tube Modeling Of Speech Production / Chapter 11：
Musical Instrument Acoustics / Chapter 12：
Room Acoustics / Chapter 13：
Auditory Perception / Part IV：
Ear Physiology / Chapter 14：
Psychoacoustics / Chapter 15：
Models Of Pitch Perception / Chapter 16：
Speech Perception / Chapter 17：
Human Speech Recognition / Chapter 18：
Speech Features / Part V：
The Auditory System As A Filter Bank / Chapter 19：
The Cepstrum As A Spectral Analyzer / Chapter 20：
Linear Prediction / Chapter 21：
A Utomatic Speech Recognition / Part VI：
Feature Extraction For Asr / Chapter 22：
Linguistic Categories For Speech Recognition / Chapter 23：
Deterministic Sequence Recognition For Asr / Chapter 24：
Statistical Sequence Recognition / Chapter 25：
Statistical Model Training / Chapter 26：
Discriminant Acoustic Probability Estimation / Chapter 27：
Acoustic Model Training: Further Topics / Chapter 28：
Speech Recognition And Understanding / Chapter 29：
Synthesis And Coding / Part VII：
Speech Synthesis / Chapter 30：
Pitch Detection / Chapter 31：
Vocoders / Chapter 32：
Low-Rate Vocoders / Chapter 33：
Medium-Rate And High-Rate Vocoders / Chapter 34：
Perceptual A Udio Coding / Chapter 35：
Other Applications / Part VIII：
Some Aspects Of Computer Music Synthesis / Chapter 36：
Music Signal Analysis / Chapter 37：
Music Retrieval / Chapter 38：
Source Separation / Chapter 39：
Speech Transformations / Chapter 40：
Speaker Verification / Chapter 41：
Speaker Diarization / Chapter 42：
Preface To The 2011 Edition
Introduction / Chapter 1：
Historical Background / Part I：
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