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EUROSPEECH 1991: Genova, Italy

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  • Plenary
  • Robust Isolated Word Recognition
  • Spoken Language Parsing
  • Probabilistic Language Models for Speech Recognition
  • Speech Recognition: Understanding Systems
  • Modelling Duration in Speech
  • Natural Language Processing
  • Assessment
  • Dialogue and Translation
  • Speech Synthesis and Prosody
  • Speech Processing and Analysis
  • Speech Interfaces: Dialogue and Human Factors

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    Second European Conference on Speech Communication and Technology, EUROSPEECH 1991, Genova, Italy, September 24-26, 1991. ISCA 1991

Plenary

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    Sadaoki Furui:
    Recent advances in speech recognition. 3-12
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  Frank Fallside:
  On the acquisition of speech by machines, ASM. 13-14

Continuous Speech Recognition

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    Padma Ramesh, Jay G. Wilpon, Maureen A. McGee, David B. Roe, Chin-Hui Lee, Lawrence R. Rabiner:
    Speaker independent recognition of spontaneously spoken connected digits. 17-20
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    P. S. Gopalakrishnan, David Nahamoo:
    Immediate recognition of embedded command words. 21-24
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    Lynn Wilcox, Marcia A. Bush:
    HMM-based wordspotting for voice editing and indexing. 25-28
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    Janet M. Baker:
    Large vocabulary speaker-adaptive continuous speech recognition research overview at dragon systems. 29-32
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    Victoria Sgardoni, Dimitrios A. Gaganelis, Eleftherios D. Frangoulis:
    Continuous density HMM context dependent phones for speech recognition over the telephone. 33-36

Segmental Speech Synthesis

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    Katsuhiko Shirai, Kazuo Hashimoto, Tetsunori Kobayashi:
    Text-to-speech synthesizer using superposition of sinusoidal waves generated by synchronized oscillators. 39-42
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    M. Guerti, Gérard Bailly:
    Synthesis-by-rule using compost: modelling resonance trajectories. 43-46
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    Yasushi Ishikawa, Kunio Nakajima:
    Neural network based spectral interpolation method for speech synthesis by rule. 47-50
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    Martine Garnier-Rizet:
    A rule-based segmental synthesis module for French. 51-54

Human Factors

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    Norman M. Fraser, G. Nigel Gilbert:
    Effects of system voice quality on user utterances in speech dialogue systems. 57-60
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    P. Day, Andreas Grünupp, Klaus-Peter Muthig:
    A human factors study of speech-to-text technology: consequences of discrete speech. 61-64
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    Iain R. Murray, John L. Arnott, Alan F. Newell:
    A comparison of document composition using a listening typewriter and conventional office systems. 65-68
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    Paulus H. Vossen:
    Evaluating speech input and output in a CAD-system using the hidden-operator method. 69-72
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    Mary Zajicek, Jill Hewitt:
    Mixed mode input for a standard wordprocessor. investigating links between input mode, speech and keyboard, and specific task areas. 73-76

Robust Isolated Word Recognition

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    Philip Lockwood, Jérôme Boudy:
    Experiments with a non-linear spectral subtractor (NSS), hidden Markov models and the projection, for robust speech recognition in cars. 79-82
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    Philip Lockwood, C. Baillargeat, J. M. Gillot, Jérôme Boudy, Gérard Faucon:
    Noise reduction for speech enhancement in cars: non-linear spectral subtraction / kalman filtering. 83-86
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    Klaus Fellbaum, Dieter Becker:
    Isolated word recognition with integrated noise reduction. 87-90
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    Javier Hernando, Climent Nadeu:
    A comparative study of parameters and distances for noisy speech recognition. 91-94

Neural Nets: Phonetic Features, Phoneme Recognition, and Time Alignment

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    Jorma Laaksonen:
    A new reliability-based phoneme segmentation method for the "neural" phonetic typewriter. 97-100
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    Bruno Apolloni, Francesco Pazienti, Vincenzo Trotta:
    Isolated word adaptive recognizer based on neural networks. 101-104
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    Nobuo Hataoka, Alex Waibel:
    Evaluation of speaker-independent phoneme recognition on TIMIT database using TDNNs. 105-108
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    Nelson Morgan, Hervé Bourlard, Chuck Wooters, Phil Kohn, Michael Cohen:
    Phonetic context in hybrid HMM/MLP continuous speech recognition. 109-112
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    E. C. Andrews, John S. Mason:
    Neural network classification of complex-valued speech features. 113-116
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    Dennis Norris:
    Rewiring lexical networks on the fly. 117-120
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    Kjell Elenius, G. Takacs:
    Phoneme recognition with an artificial neural network. 121-124
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    Jianxin Jiang, Kechu Yi, Zheng Hu:
    A new self-organization algorithm of forming a phoneme map. 125-128
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    Shuping Ran, J. Bruce Millar:
    Phoneme classification using neural networks based on acoustic-phonetic structure. 129-132
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    Nigel Dodd, Donald MacFarlane, Chris Marland:
    Networks for speech recognition structurally optimised by genetic techniques implemented on parallel hardware. 133-136

Phonetics I, II

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    Jeff Pittam, John Ingram:
    Influence of vietnamese tone and prosody on the acquisition of English stress patterns. 139-142
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    Walter F. Sendlmeier:
    The voiced/unvoiced distinction of initial stops by normal and hearing impaired listeners. 143-146
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    Krishna S. Nathan:
    Comparison of formant transition based stop classifiers: time-varying and time-invariant signal models. 147-150
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    Christian Benoît, Christian Abry, L. J. Roe:
    The effect of context on labiality in French. 151-156
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    A. K. Datta, N. R. Ganguli, B. Mukherjee:
    Nasalisation in bengali speech sounds acoustic-phonetic study. 157-160
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    N. R. Ganguli:
    Vowel formant frequency distribution of a major indian language. 161-164
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    Bernard Harmegnies, Marielle Bruyninckx, Joaquim Llisterri, Dolors Poch:
    Effects of language change on voice quality in bilingual speakers, corpus content effect. 165-168
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    T. I. Shevchenko, T. S. Skopintseva:
    Effects of social and regional backgrounds on LTAS in british English. 169-172
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    Henk van den Heuvel, Bert Cranen, Toni C. M. Rietveld:
    Speaker related variability in the durations of dutch speech segments. 251-254
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    Johan Liljencrants:
    Numerical simulations of glottal flow. 255-258
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    Joop Jansen, Bert Cranen, Louis Boves:
    Modelling of source characteristics of speech sounds by means of the LF-model. 259-262
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    Hanspeter Herzel, J. Wendler:
    Evidence of chaos in phonatory samples. 263-266
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    Van Loan Trinh, Bernard Guérin, Eric Castelli:
    Source-tract coupling and the subglottal system in an articulatory synthesizer. 267-270

Multilingual Speech Recognition Systems (Special Session)

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    Paul G. Bamberg, Anne Demedts, John Elder, Caroline B. Huang, Charles Ingold, Mark A. Mandel, Linda Manganaro, Stijn Van Even:
    Phoneme-based training for large-vocabulary recognition in six european languages. 175-182
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    Helene Cerf-Danon, Steven DeGennaro, Marco Ferretti, Jorge Gonzalez, Eric Keppel:
    1.0 TANGORA - a large vocabulary speech recognition system for five languages. 183-192
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    Hermann Ney, Roberto Billi:
    Prototype systems for large-vocabulary speech recognition: polyglot and spicos. 193-200

Spoken Language Parsing

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    J. H. Wright:
    Adaptation of grammar-based language models for continuous speech recognition. 203-206
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    Keh-Yih Su, Tung-Hui Chiang, Yi-Chung Lin:
    A robustness and discrimination oriented score function for integrating speech and language processing. 207-210
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    Paolo Baggia, Lorenzo Fissore, Elisabetta Gerbino, Egidio P. Giachin, Claudio Rullent:
    Improving speech understanding performance through feedback verification. 211-214
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    Anna Corazza, Renato de Mori, Roberto Gretter, Giorgio Satta:
    Computation of upper-bounds for island-driven stochastic parsers. 215-218
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    François Andry, J. H. Simon Thornton:
    A parser for speech lattices using a UCG grammar. 219-222
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    Sheryl Young, Michael Matessa:
    Using pragmatic and semantic knowledge to correct parsing of spoken language utterances. 223-227

Speech Coding I-IV

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    Arnaldo J. Abrantes, Jorge S. Marques, Isabel Trancoso:
    Hybrid sinusoidal modeling of speech without voicing decision. 231-234
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    Jorge S. Marques, Isabel Trancoso, Arnaldo J. Abrantes:
    Harmonic coding of speech: an experimental study. 235-238
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    David Rowe, William G. Cowley, Andrew Perkis:
    A multiband excitation linear predictive speech coder. 239-242
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    Shu Hung Leung, K. L. Lai, O. Y. Wong, Andrew Luk:
    A new coded excitation model using multifrequency decomposition. 245-248
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    Daniele Sereno:
    Frame substitution and adaptive post-filtering in speech coding. 595-598
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    S. A. Atungsiri, R. Soheili, Ahmet M. Kondoz, Barry G. Evans:
    Effective lost speech frame reconstruction for CELP coders. 599-602
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    Hiromi Nagabuchi, Nobuhiko Kitawaki:
    Evaluation and improvement of coded speech quality degraded by cell loss in ATM networks. 603-606
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    Alain J. Vigier:
    Combined source-channel coding for a very noisy channed. 607-610
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    G. Rosina, Marcello Sant' Agostino, E. Turco, Luigi Vetrano:
    Testing and quality enhancement of the GSM full rate voice channel. 611-614
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    U. Kipper, Herbert Reininger, Dietrich Wolf:
    Low bit rate speech coding using CELP with adaptive excitation codebook. 893-896
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    Arild Fuldseth, Erik Harborg, Finn Tore Johansen, Jan E. Knudsen:
    A real-time implementable 7 khz speech coder at 16 kbit/s. 897-900
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    D. J. Zarkadis:
    Adaptive spectral weighting for vector predictive coding of the LPC-spectra. 901-904
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    Samir Saoudi, Jean-Marc Boucher, Alain Le Guyader:
    Medium band speech coding using optimal scalar quantization of LSP. 905-908
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    Philip Secker, Andrew Perkis:
    Joint source and channel coding of line spectrum pairs. 909-912
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    C. F. Chan, K. W. Law:
    An algorithm for computing LSP frequencies directly from the reflection coefficients. 913-916
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    Peter Meyer, W. Peters, Jürgen Paulus:
    Variable rate speech coding using perceptive thresholds and adaptive VUS detection. 809-812
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    M. R. Suddle, S. A. Atungsiri, Ahmet M. Kondoz, Barry G. Evans:
    A secure and robust CELP coder for land and satellite mobile systems. 813-816
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    Carlos M. Ribeiro, Isabel Trancoso:
    A 4.8 kbps celp coder with post-processing. 817-820
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    K. W. Law, O. Y. Wong, C. F. Chan:
    A real-time high quality joint-excitation linear predictive coder at 8 kbps. 821-824
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    Rosario Drogo de Iacovo, Roberto Montagna:
    Some experiments in perceptual masking of quantizing noise in analysis-by-synthesis speech coders. 825-828
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    Gao Yang, Henri Leich, René Boite:
    A very high-quality CELP coder at the rate of 2400 bps. 829-832
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    Z. Yong Liu:
    An effective pulse adaptive code-excited linear predictive coder at 4kb/S. 835-838
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    C. F. Chan, S. H. Leung:
    A vocoder using high-order LPC filter with very few non-zero coefficients. 839-842

Assessment, Intelligibility and Aids for Disabled

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    Mario Rossi, Robert Espesser, Chaslav Pavlovic:
    The effects of in internal reference system and cross-modality matching on the subjective rating of speech synthesisers. 273-276
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    H. A. Sydeserff, R. J. Caley, Stephen D. Isard, Mervyn A. Jack, Alex I. C. Monaghan, Jo Verhoeven:
    Evaluation of speech synthesis techniques in a comprehension task. 277-280
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    P. A. Howard-Jones:
    'SOAP' - a speech output assessment package for controlled multilingual evaluation of synthetic speech. 281-284
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    Tammo Houtgast, Jan A. Verhave:
    A physical approach to speech quality assessment: correlation patterns in the speech spectrogram. 285-288
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    Hiroyuki Miyata, Tammo Houtgast:
    Weighted MTF for predicting speech intelligibility in reverberant sound fields. 289-292
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    Ute Jekosch:
    Speech intelligibility studies for the european hermes spaceplane. 293-296
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    Jianing Wei, Andrew Faulkner, Adrian Fourcin:
    An application of speech processing and encoding scheme for Chinese lexical tone and consonant perception by hearing impaired listeners. 299-302
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    Dimitri Kanevsky, P. Gopalakrishan, Catalina Danis, Gregg Daggett, Edward A. Epstein, David Nahamoo:
    On the development of a phone communication aid for the hearing impaired. 303-306
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    Yolande Anglade, Jean-Marie Pierrel, Jean-Claude Junqua:
    A spoken language interface for a telephone switchboard operator center. 307-310
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    Iain R. Murray, John L. Arnott, Norman Alm, Alan F. Newell:
    A communication system for the disabled with emotional synthetic speech produced by rule. 311-314

Speech Synthesis: Techniques and Applications

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    Thomas Portele, Birgit Steffan, Rainer Preuß, Wolfgang Hess:
    German speech synthesis by concatenation of non-parametric units. 317-320
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    Giuseppe Abbattista, Antonello Riccio, Enzo Mumolo:
    Automatic document reader with speech output capabilities. 321-324
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    Robin W. King:
    Tools and processes for developing low-cost and high-quality text-to-speech synthesis for communication aids. 325-329
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    Hynek Hermansky, Louis Anthony Cox Jr.:
    Perceptual linear predictive (PLP) analysis-resynthesis technique. 329-332
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    Reinhold Greisbach, Bernd J. Kröger, O. Esser, G. Plaßmann:
    A display technique for measurements of natural and synthetic articulatory dynamics. 333-336
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    Yueh-Chin Chang, Yi-Fan Lee, Bang-Er Shia, Hsiao-Chuan Wang:
    Statistical models for the Chinese text-to-speech system. 337-340
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    P. A. Taylor, I. A. Nairn, Andrew M. Sutherland, Mervyn A. Jack:
    A realtime speech synthesis system. 341-344
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    Hélène Valbret, Eric Moulines, Jean-Pierre Tubach:
    Voice tranformation using PSOLA technique. 345-348
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    Massimo Giustiniani, Piero Pierucci:
    Phonetic ergodic HMM for speech synthesis. 349-352
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    Cristina Delogu, P. Paoloni, Paolo Pocci, Ciro Sementina:
    Quality evaluation of text-to-speech synthesizers using magnitude estimation, categorical estimation, pair comparison and reaction time methods. 353-356
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    H. Zingte, Cl. Hennebois:
    Helping young children to associate sounds and letters through speech synthesis. 357-360
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    Hervé Bourlard:
    Neural nets and hidden Markov models: review and generalizations. 363-369
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    Nikil S. Jayant, James D. Johnston, Yair Shoham:
    Coding of wideband speech. 373-379

Probabilistic Language Models for Speech Recognition

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    Roberto Pieraccini, Esther Levin:
    Stochastic representation of semantic structure for speech understanding. 383-386
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    Colin Matheson, Fergus R. McInnes:
    Incorporating probabilities into the dualgram language model. 387-390
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    Egidio P. Giachin:
    A dynamic programming based framework for stochastic spoken language understanding. 391-394
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    Natividad Prieto, Enrique Vidal:
    Learning language models through the ECGI method. 395-398
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    Roberto Cremonini, Marco Ferretti, M. C. Galimberti, Giulio Maltese, Federico Mancini:
    Using a generative grammar to train a probabilistic language model for speaker-independent speech recognition. 399-402

Speech Recognition and Phonetic Modelling

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    Katsuhiko Shirai, Eiichiro Kitagawa, T. Endo:
    Optimal construction of context sensitive quantizer for phoneme recognition in continuous speech. 405-408
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    Mary O'Kane, P. E. Kenne, D. Landy, S. Atkins:
    Generalising from single-speaker recognition in a feature-based recogniser. 409-412
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    Hans-Günter Hirsch, Peter Meyer, Hans-Wilhelm Rühl:
    Improved speech recognition using high-pass filtering of subband envelopes. 413-416
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    Yifan Gong, Jean Paul Haton:
    Comparing two phoneme identification methods using a continuous speech recognizer. 417-420
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    D. Ederveen, Louis Boves:
    Knowledge-based phoneme recognition. 421-424

Speaker Identification and Verification

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    J. Kraayeveld, A. C. M. Rietveld, Vincent J. van Heuven:
    Speaker characterization in dutch using prosodic parameters. 427-430
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    Alan K. Hunt:
    New commercial applications of telephone-network-based speech recognition and speaker verification. 431-434
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    Jean-François Bonastre, Henri Meloni, Philippe Langlais:
    Analytical strategy for speaker identification. 435-438
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    L. Xu, John S. Mason:
    Optimization of perceptually-based spectral transforms in speaker identification. 439-442

Pitch Determination and Voice Separation

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    Alain de Cheveigné:
    A mixed speech F0 estimation algorithm. 445-448
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    Edward Jones, Eliathamby Ambikairajah:
    A perceptually-based pitch extractor for band-limited speech. 449-452
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    Yu-Hua Gu:
    A robust pseudo perceptual pitch estimator. 453-456
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    Neviano Dal Degan, Marco Fratti:
    Pitch estimation based on a "narrowed" autocorrelation function. 457-460

Speech Recognition: Understanding Systems

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    Seiichi Nakagawa, Yoshimitsu Hirata, Isao Murase:
    The syntax-oriented spoken Japanese understanding system SPOJOS-SYNO II. 463-466
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    Henning Bergmann, Hans-Hermann Hamer, Andreas Noll, Annedore Paeseler, Horst Tomaschewski:
    An adaptable man-machine interface using connected-word recognition. 467-470
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    M. J. Poza, Celinda de la Torre, Daniel Tapias, Luis Villarrubia:
    An approach to automatic recognition of keywords in unconstrained speech using parametric models. 471-474
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    I. Lee Hetherington, Hong C. Leung, Victor W. Zue:
    Toward vocabulary-independent recognition of telephone speech. 475-478
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    Ronald A. Cole, Krist Roginski, Mark A. Fanty:
    English alphabet recognition with telephone speech. 479-482
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    Jean-Yves Fiset, Jean-Marc Robert, Raymond Descout:
    Evolutionary language models in air traffic control training. 483-486
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    Gareth J. F. Jones, Jeremy H. Wright, E. N. Wrigley, Michael J. Carey, Eluned S. Parris:
    Isolated-word sentence recognition using probabilistic context-free grammar. 487-489
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    Mitchell Hood:
    Lexical access in a speech understanding and dialogue system. 490-493
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    Reinhold Haeb-Umbach, Hermann Ney:
    A look-ahead search technique for large vocabulary continuous speech recognition. 495-498
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    Carlos Teixeira, Isabel Trancoso:
    Spectral subtraction for front-end noise reduction in a speech recognizer. 499-502

Speech Databases, Analysis And Assessment

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    Lori F. Larnel, Jean-Luc Gauvain, Maxine Eskénazi:
    BREF, a large vocabulary spoken corpus for French. 505-508
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    Luc Mathan, Dominique Morin:
    Speech field databases: development and analysis. 509-512
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    Shuichi Itahashi:
    Large scale Japanese dialect speech corpora. 513-516
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    Paulus H. Vossen:
    Outline of a design-oriented evaluation framework for speech-driven applications. 517-520
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    Richard Winski, Kamran Kordi:
    Assessment of continuous speech recognisers using recogniser sensitivity analysis. 521-524
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    Christine Bourjot, Anne Boyer, Dominique Fohr:
    A tool for assessment of acoustic phonetic lattices. 525-528
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    Herman J. M. Steeneken, Jeroen G. van Velden:
    Ramos - recognizer assessment by means of manipulation of speech applied to connected speech recognition. 529-532
• 
  • 
  • 
  • 
  • 
    Paul van Alphen, Louis C. W. Pols:
    Comparing various feature vectors in automatic speech recognition. 533-536
• 
  • 
  • 
  • 
  • 
    Victor W. Zue, James R. Glass, David Goodine, Lynette Hirschman, Hong C. Leung, Michael S. Phillips, Joseph Polifroni, Stephanie Seneff:
    The MIT ATIS system; preliminary development, spontaneous speech data collection, and performance evaluation. 537-540
• 
  • 
  • 
  • 
  • 
    S. Benaouicha, A. Rajouani, M. Zyoute:
    Construction of an Arabic speech data base - duration model of Arabic vowels. 541-544
• 
  • 
  • 
  • 
  • 
    P. N. Denbigh, J. Zhao:
    Pitch extraction and separation of overlapping speech. 545-548

Neural Nets I, II

• 
  • 
  • 
  • 
  • 
    Yoshua Bengio, Renato de Mori, Giovanni Flammia, Ralf Kompe:
    Phonetically motivated acoustic parameters for continuous speech recognition using artificial neural networks. 551-554
• 
  • 
  • 
  • 
  • 
    Michael J. Carey, Eluned S. Parris:
    Adapting input transformations using alpha-nets for whole word speech recognition. 555-558
• 
  • 
  • 
  • 
  • 
    Les T. Niles:
    TIMIT phoneme recognition using an HMM-derived recurrent neural network. 559-562
• 
  • 
  • 
  • 
  • 
    P. O. Husoy, Torbjørn Svendsen:
    ANN-based speech recognition using a preprocessor for non-linear time compression. 563-566
• 
  • 
  • 
  • 
  • 
    Helge B. D. Sørensen, Uwe Hartmann:
    A self-structuring neural noise reduction model. 567-570
• 
  • 
  • 
  • 
  • 
    Bojan Petek, Alex Waibel, Joseph M. Tebelskis:
    Integrated phoneme-function word architecture of hidden control neural networks for continuous speech recognition. 1407-1410
• 
  • 
  • 
  • 
  • 
    X. Zhang, John S. Mason, E. C. Andrews:
    Multiple dynamic features to enhance neural net based speaker verification. 1411-1414
• 
  • 
  • 
  • 
  • 
    Patrick Haffner, Alex Waibel:
    Time-delay neural networks embedding time alignment: a performance analysis. 1415-1418
• 
  • 
  • 
  • 
  • 
    Yohji Fukuda, Haruya Matsumoto:
    Phoneme recognition using recurrent neural networks. 1419-1423
• 
  • 
  • 
  • 
  • 
    Yasuhiro Komori, Kaichiro Hatazaki:
    An integration of knowledge and neural networks toward a phoneme typewriter without a language model. 1423-1426

Parsing and Lexical Access

• 
  • 
  • 
  • 
  • 
    Junko Hosaka, Toshiyuki Takezawa, Terumasa Ehara:
    Utilizing empirical data for postposition classification toward spoken Japanese speech recognition. 573-576
• 
  • 
  • 
  • 
  • 
    Michael S. Phillips, James R. Glass, Victor W. Zue:
    Automatic learning of lexical representations for sub-word unit based speech recognition systems. 577-580
• 
  • 
  • 
  • 
  • 
    Roxane Lacouture, Renato de Mori:
    Lexical tree compression. 581-584
• 
  • 
  • 
  • 
  • 
    Michael D. Riley, Andrej Ljolje:
    Lexical access with a statistically-derived phonetic network. 585-588
• 
  • 
  • 
  • 
  • 
    Giuliano Antoniol, Fabio Brugnara, Diego Giuliani:
    Admissible strategies for acoustic matching with a large vocabulary. 589-592

Modelling Duration in Speech

• 
  • 
  • 
  • 
  • 
    Alejandro Macarrón, J. Gregorio Escalada, Miguel Ángel Rodríguez Crespo:
    Generation of duration rules for a Spanish text-to-speech synthesizer. 617-620
• 
  • 
  • 
  • 
  • 
    L. Mortamet:
    Implementing duration expert rules into a text-to-speech synthesis system. 621-624
• 
  • 
  • 
  • 
  • 
    Nobuyoshi Kaiki, Katsuhiko Mimura, Yoshinori Sagisaka:
    Statistical modeling of segmental duration and power control for Japanese. 625-628
• 
  • 
  • 
  • 
  • 
    W. Nick Campbell:
    Phrase-level factors affecting timing in speech. 629-632
• 
  • 
  • 
  • 
  • 
    Matti Karjalainen, Toomas Altosaar:
    Phoneme duration rules for speech synthesis by neural networks. 633-636

Automatic Speech Recognition: Algorithms I-III

• 
  • 
  • 
  • 
  • 
    Fergus R. McInnes:
    Context-sensitive phoneme lattice generation using interpolated demi-diphone and triphone models. 639-642
• 
  • 
  • 
  • 
  • 
    J. M. Song, T. Thomas, M. Patel:
    Experiments of 991-word speaker independent continuous speech recognition on DARPA RM task. 643-646
• 
  • 
  • 
  • 
  • 
    Henri Meloni, Frédéric Béchet, Philippe Gilles:
    Bottom-up acoustic-phonetic decoding for the selection of word cohorts from a large vocabulary. 647-650
• 
  • 
  • 
  • 
  • 
    Antonio M. Peinado, Ramon Román, José C. Segura, Antonio J. Rubio, Pedro García-Teodoro, Jesús Esteban Díaz Verdejo:
    Entropic training for HMM speech recognition. 651-654
• 
  • 
  • 
  • 
  • 
    Patrick Kenny, Sarangarajan Parthasarathy, Vishwa Gupta, Matthew Lennig, Paul Mermelstein, Douglas D. O'Shaughnessy:
    Energy, duration and Markov models. 655-658
• 
  • 
  • 
  • 
  • 
    J. J. Nijtmans:
    A new recursive Markov model with a new state pruning approach for large vocabulary continuous speech recognition. 659-663
• 
  • 
  • 
  • 
  • 
    Fergus R. McInnes:
    Context-sensitive phoneme lattice generation using interpolated demi-diphone and triphone models. 663-666
• 
  • 
  • 
  • 
  • 
    Peter Nowell, Henry S. Thompson:
    An efficient implementation of the n-best algorithm for lexical access. 667-670
• 
  • 
  • 
  • 
  • 
    Alessandro Falaschi, Massimo Pucci:
    Automatic derivation of HMM alternative pronunciation network topologies. 671-674
• 
  • 
  • 
  • 
  • 
    Isabel Galiano, Francisco Casacuberta, Emilio Sanchis:
    On the structure of subword units for a speaker independent continuous speech task. 675-678
• 
  • 
  • 
  • 
  • 
    Yunxin Zhao, Hisashi Wakita, Xinhua Zhuang:
    Generate word transcription dictionary from sentence utterances and evaluate its effect on speaker-independent continuous speech recognition. 679-682
• 
  • 
  • 
  • 
  • 
    Andrew Varga, Roger K. Moore:
    Simultaneous recognition of concurrent speech signals using hidden Markov model decomposition. 1175-1178
• 
  • 
  • 
  • 
  • 
    I. A. Ballantyne, Andrew M. Sutherland, J. M. Hannah, Mervyn A. Jack:
    A large vocabulary parallel processing continuous speech recognition system. 1179-1182
• 
  • 
  • 
  • 
  • 
    Richard C. Rose, Edward M. Hofstetter:
    Techniques for robust word spotting in continuous speech messages. 1183-1186
• 
  • 
  • 
  • 
  • 
    Alessandro Falaschi, Alfredo Micozzi:
    Word spotting by CSR through vector quantized background models. 1187-1190
• 
  • 
  • 
  • 
  • 
    Jean-Claude Junqua, Hisashi Wakita:
    Towards an artificial laboratory for the design and simulation of cooperative speech processing algorithms. 1191-1194
• 
  • 
  • 
  • 
  • 
    Keith Edwards, Fergus R. McInnes, Mervyn A. Jack:
    Accent specific modifications for continuous speech recognition based on a sub-word lattice approach. 1195-1198
• 
  • 
  • 
  • 
  • 
    Eduardo Lleida, José B. Mariño, Climent Nadeu, Albert Oliveras:
    Two level continuous speech recognition using demisyllable-based HMM word spotting. 1199-1202
• 
  • 
  • 
  • 
  • 
    Ted H. Applebaum, Brian A. Hanson:
    Tradeoffs in the design of regression features for word recognition. 1203-1206
• 
  • 
  • 
  • 
  • 
    Lalit R. Bahl, Peter F. Brown, Peter V. de Souza, Robert L. Mercer, David Nahamoo:
    A fast algorithm for deleted interpolation. 1209-1212
• 
  • 
  • 
  • 
  • 
    Michael A. Franzini, Alex Waibel, Kai-Fu Lee:
    Recent work in continuous speech recognition using the connectionist viterbi training procedure. 1213-1216
• 
  • 
  • 
  • 
  • 
    Volker Steinbiss:
    A search organization for large-vocabulary recognition based on n-best decoding. 1217-1220
• 
  • 
  • 
  • 
  • 
    Yifan Gong, Jean Paul Haton:
    VINICS: a continuous speech recognizer based on a new robust formulation. 1221-1224
• 
  • 
  • 
  • 
  • 
    Shigeki Sagayama:
    A matrix representation of HMM-based speech recognition algorithms. 1225-1228

Segmentation

• 
  • 
  • 
  • 
  • 
    Paul Dalsgaard, Ove Andersen, William J. Barry:
    Multi-lingual acoustic-phonetic features for a number of european languages. 685-688
• 
  • 
  • 
  • 
  • 
    Harouna Kabré, Guy Perennou, Nadine Vigouroux:
    A non-linear filtering method applied to automatic segmentation of multilingual speech corpora. 689-692
• 
  • 
  • 
  • 
  • 
    Piero Cosi, Daniele Falavigna, Maurizio Omologo:
    A preliminary statistical evaluation of manual and automatic segmentation discrepancies. 693-696
• 
  • 
  • 
  • 
  • 
    James M. McQueen, Edward John Briscoe:
    A computational tool for examining lexical segmentation in continuous speech. 697-700
• 
  • 
  • 
  • 
  • 
    M. S. Schmidt, G. S. Watson:
    The evaluation and optimization of automatic speech segmentation. 701-704
• 
  • 
  • 
  • 
  • 
    G. Feng, N. Achab, R. Combescure:
    On-line speech segmentation using adaptive models: application to variable rate speech coding. 705-708
• 
  • 
  • 
  • 
  • 
    P. A. Taylor, Stephen D. Isard:
    Automatic diphone segmentation. 709-711
• 
  • 
  • 
  • 
  • 
    Georg Ottesen:
    An automatic diphone segmentation system. 713-716
• 
  • 
  • 
  • 
  • 
    Richard Brierton, Barry M. G. Cheetham:
    An evaluation oof spectral transitivity functions for speech segmentation in variable frame-rate speech vocoding. 717-720

Automatic Speech Recognition: Applications

• 
  • 
  • 
  • 
  • 
    Dirk Van Compernolle, Johan Smolders, P. Jaspers, T. Hellemans:
    Speaker clustering for dialectic robustness in speaker independent recognition. 723-726
• 
  • 
  • 
  • 
  • 
    Dina Yashchin, William C. G. Ortel:
    Experience with speech recognition in automating telephone operator functions. 727-730
• 
  • 
  • 
  • 
  • 
    Franco Canavesio, Lorenzo Fissore, Mario Oreglia, P. Ruscitti:
    HMM modeling in the public telephone network environment: experiments and results. 731-734
• 
  • 
  • 
  • 
  • 
    Dominique Morin:
    Influence of field data in HMM training for a vocal server. 735-738
• 
  • 
  • 
  • 
  • 
    Alberto Ciaramella, Lorenzo Fissore, Alberto Pacchiotti, Roberto Pacifici:
    An isolated word speech recognizer prototype for mobile-radio applications. 739-742

Natural Language Processing

• 
  • 
  • 
  • 
  • 
    James Monaghan, Christine Cheepen:
    Linguistic modelling for a speech interface in the office context. 745-748
• 
  • 
  • 
  • 
  • 
    Andrea Di Carlo, Rino Falcone:
    Ill-formedness problem in the spoken language processing. 749-752
• 
  • 
  • 
  • 
  • 
    Giulio Maltese, Federico Mancini:
    A technique to automatically assign parts-of-speech to words taking into account word-ending information through a probabilistic model. 753-756
• 
  • 
  • 
  • 
  • 
    Marcello Pelillo, Mario Refice:
    Syntactic category disambiguation through relaxation processes. 757-760
• 
  • 
  • 
  • 
  • 
    E. N. Wrigley, Jeremy H. Wright:
    Computational requirements of probabilistic LR parsing for speech recognition using a natural language grammar. 761-764

Symbolic Processing in Speech Synthesis

• 
  • 
  • 
  • 
  • 
    J. Tihoni, G. Pérennon:
    Phonotypical transcription through the GEPH expert system. 767-770
• 
  • 
  • 
  • 
  • 
    Briony Williams, Franziska Maier:
    A spelling corrector for use in text-to-speech synthesis for English. 771-774
• 
  • 
  • 
  • 
  • 
    Thomas Russi:
    Robust and efficient parsing for applications such as text-to-speech conversion. 775-778
• 
  • 
  • 
  • 
  • 
    Robert W. P. Luk, Robert I. Damper:
    Stochastic transduction for English text-to-phoneme conversion. 779-782

Sub-Lexical Unit Modelling

• 
  • 
  • 
  • 
  • 
    Mei-Yuh Hwang, Xuedong Huang:
    Acoustic distribution clustering in phonetic hidden Markov models. 785-788
• 
  • 
  • 
  • 
  • 
    Mats Blomberg:
    Modelling articulatory inter-timing variation in a speech recognition system based on synthetic references. 789-792
• 
  • 
  • 
  • 
  • 
    Kari Torkkola, Mikko Kokkonen, Mikko Kurimo, Pekka Utela:
    Improving short-time speech frame recognition results by using context. 793-796
• 
  • 
  • 
  • 
  • 
    Panagiotis A. Rentzepopoulos, George K. Kokkinakis:
    Phoneme to grapheme conversion using HMM. 797-800
• 
  • 
  • 
  • 
  • 
    S. H. Parfitt, Richard A. Sharman:
    A bi-directional model of English pronunciation. 801-804

Speech Understanding and Dialogue

• 
  • 
  • 
  • 
  • 
    David Goodine, Stephanie Seneff, Lynette Hirschman, Michael S. Phillips:
    Full integration of speech and language understanding in the MIT spoken language system. 845-848
• 
  • 
  • 
  • 
  • 
    Takayuki Yamaoka, Hitoshi Iida:
    Dialogue interpretation model and its application to next utterance prediction for spoken language processing. 849-852
• 
  • 
  • 
  • 
  • 
    W. Boogers:
    Dialogue construction by compilation. 853-856
• 
  • 
  • 
  • 
  • 
    Izuru Nogaito, Masahiko Takahashi, Shingo Kuroiwa, Fumihiro Yato:
    Dialogue management in an extension number guidance system. 857-860
• 
  • 
  • 
  • 
  • 
    Encarna Segarra, Pedro García:
    Automatic learning of acoustic and syntactic-semantic levels in continuous speech understanding. 861-864
• 
  • 
  • 
  • 
  • 
    Paolo Baggia, Alberto Ciaramella, Davide Clementino, Lorenzo Fissore, Elisabetta Gerbino, Egidio P. Giachin, Giorgio Micca, Luciano Nebbia, Roberto Pacifici, G. Pirani, Claudio Rullent:
    A man-machine dialogue system for speech access to e-mail information using the telephone: implementation and first results. 865-868

Assessment

• 
  • 
  • 
  • 
  • 
    Renée van Bezooijen, Louis C. W. Pols:
    Performance of text-to-speech conversion for dutch: a comparative evaluation of allophone and diphone based synthesis at the level of the segment, the word, and the paragraph. 871-874
• 
  • 
  • 
  • 
  • 
    Christian Benoît, Françoise Emerard, Betina Schnabel, A. Tseva:
    Quality comparisons of prosodic and of acoustic components of various synthesisers. 875-878
• 
  • 
  • 
  • 
  • 
    Martine Griee, Kiki Vagges, Daniel Hirst:
    Assessment of intonation in text-to-speech synthesis systems - a pilot test in English and Italian. 879-882
• 
  • 
  • 
  • 
  • 
    Alex I. C. Monaghan:
    Evaluation of the naturalness of prosody generated by the CSTR TTS system. 883-886
• 
  • 
  • 
  • 
  • 
    Ulrich Halka:
    Speech-model processes for objective quality measurements of speech-coding systems. 887-890

Speech Recognition: Stochastic Modelling

• 
  • 
  • 
  • 
  • 
    Stephan Euler:
    Adaptation techniques in tied density hidden Markov models. 919-922
• 
  • 
  • 
  • 
  • 
    Denis Jouvet, Katarina Bartkova, Jean Monné:
    On the modelization of allophones in an HMM based speech recognition system. 923-926
• 
  • 
  • 
  • 
  • 
    Denis Jouvet, Laurent Mauuary, Jean Monné:
    Automatic adjustments of the structure of Markov models for speech recognition applications. 927-930
• 
  • 
  • 
  • 
  • 
    Hong C. Leung, I. Lee Hetherington, Victor W. Zue:
    Speech recognition using stochastic explicit-segment modeling. 931-934
• 
  • 
  • 
  • 
  • 
    Dominique Dubois:
    Comparison of time-dependent acoustic features for a speaker-independent speech recognition system. 935-938
• 
  • 
  • 
  • 
  • 
    Jean-Luc Gauvain, Chin-Hui Lee:
    Bayesian learning for hidden Markov model with Gaussian mixture state observation densities. 939-942

Speech Interfaces: Systems and Applications

• 
  • 
  • 
  • 
  • 
    Hans-Wilhelm Rühl:
    Voice controlled mail ordering via telephone using SPREIN. 945-948
• 
  • 
  • 
  • 
  • 
    Stefan Dobler, Werner Armbruester, Peter Meyer, Hans-Wilhelm Rühl:
    A voice dialling device for mobile radio. 949-952
• 
  • 
  • 
  • 
  • 
    Kamel Smaïli, François Charpillet, Jean-Marie Pierrel, Jean Paul Haton:
    A continuous speech recognition approach for the design of a dictation machine. 953-956
• 
  • 
  • 
  • 
  • 
    David L. Thomson, Jay G. Wilpon, Rafid A. Sukkar, Dimitrios P. Prezas:
    Automatic speech recognition in the Spanish telephone network. 957-960
• 
  • 
  • 
  • 
  • 
    Roberto Billi, P. Buttafava, P. De Stefani, M. Gamba, D. Voltolini:
    Computer-aided, voice-based, medical report preparation: an application to radiology. 961-964
• 
  • 
  • 
  • 
  • 
    Filipe N. Carlos, Jose P. Carmona, Pedro M. Chagas, Luís C. Oliveira, António Joaquim Serralheiro, Isabel Trancoso:
    A recognition / synthesis system applied to database access through the telephone network. 965-968
• 
  • 
  • 
  • 
  • 
    Seppo Helle:
    An experiment in using a hypertext system in phonetics and speech processing education. 969-972
• 
  • 
  • 
  • 
  • 
    Giuliano Antoniol, Fabio Brugnara, F. Dalla Palma, Gianni Lazzari, E. Moser:
    A. RE. s. : an interface for automatic reporting by speech. 973-976
• 
  • 
  • 
  • 
  • 
    U. Schultheiß, Bernd Lochschmidt:
    COGNITO - an experimental voice-controlled telecommunication system. 977-980
• 
  • 
  • 
  • 
  • 
    Jared Bernstein, Dimitry Rtischev:
    A voice interactive language instruction system. 981-984
• 
  • 
  • 
  • 
  • 
    Edmund Rooney, Steven M. Hiller, John Laver, Maria-Gabriella Di Benedetto:
    Macro and micro features for automated pronunciation improvement in the spell system. 985-988

Neural Nets: Comparative Studies, Lexical Recognition

• 
  • 
  • 
  • 
  • 
    Laurence Devillers, Christian Dugast:
    Comparison of continuous mixture densities and TDNN in a viterbi-framework: experiments on speaker dependent DARPA RM1+. 991-994
• 
  • 
  • 
  • 
  • 
    Peter Thurston, Dennis Norris:
    A comparison of two compression functions used for noisy vowel detection with back-propagation networks. 995-998
• 
  • 
  • 
  • 
  • 
    Javier Ferreiros, A. Castro, José M. Pardo:
    Comparison between two different approaches in speaker - independent isolated digit recognition. 999-1002
• 
  • 
  • 
  • 
  • 
    Franck Poirier:
    DVQ: dynamic vector quantization application to speech processing. 1003-1006
• 
  • 
  • 
  • 
  • 
    Yoshua Bengio, Renato de Mori, Giovanni Flammia, Ralf Kompe:
    A comparative study on hybrid acoustic phonetic decoders based on artificial neural networks. 1007-1010
• 
  • 
  • 
  • 
  • 
    Hidefumi Sawai, Satoru Nakamura:
    Time-delay neural network architectures for high-performance speaker-independent recognition. 1011-1014
• 
  • 
  • 
  • 
  • 
    Peter Wittenburg, R. Couwenberg:
    Recurrent neural nets as building blocks for human word recognition. 1015-1018
• 
  • 
  • 
  • 
  • 
    Fisseha Mekuria, Tore Fjällbrant:
    A neural net model for vector quantization. 1019-1022
• 
  • 
  • 
  • 
  • 
    N. H. Russell, Frank Fallside, Anthony J. Robinson, Richard W. Prager:
    Lexical access using a recurrent error propagation network. 1023-1026
• 
  • 
  • 
  • 
  • 
    Peter Brauer, Per Hedelin, Dieter Huber, Petter Knagenhjelm, Johan Molnö:
    Model or non-model based classifiers. 1027-1030
• 
  • 
  • 
  • 
  • 
    Toomas Altosaar, Matti Karjalainen:
    Event-based recognition and analysis of speech by neural networks. 1031-1034
• 
  • 
  • 
  • 
  • 
    Frederick Jelinek:
    Up from trigrams! - the struggle for improved language models. 1037-1040
• 
  • 
  • 
  • 
  • 
    Rolf Carlson:
    Synthesis: modelling variability and constraints. 1043-1048

Dialogue and Translation

• 
  • 
  • 
  • 
  • 
    Marc Guyomard, Jacques Siroux, Alain Cozannet:
    The role of dialogue in speech recognition the case of the yellow. 1051-1054
• 
  • 
  • 
  • 
  • 
    Elisabetta Gerbino, Paolo Baggia:
    Interpretation of context-dependent utterances in man-machine dialogue. 1055-1058
• 
  • 
  • 
  • 
  • 
    S. Eggins, Julie Vonwiller, Christian Matthiessen, P. Sefton:
    The description of minor clauses in information-seeking telephone dialogues. 1059-1062
• 
  • 
  • 
  • 
  • 
    David B. Roe, Fernando Pereira, Richard Sproat, Michael D. Riley, Pedro J. Moreno, Alejandro Macarrón:
    Toward a spoken language translator for restricted-domain context-free languages. 1063-1066
• 
  • 
  • 
  • 
  • 
    N. Venkata Subramaniam, Narayanan Alwar, G. Mallikarjuna, P. Prabhakar Rao, Subramanian Raman:
    Bidirectional machine translation in indian languages. 1067-1070

Speech Analysis and Signal Representation

• 
  • 
  • 
  • 
  • 
    Constantin Papaodysseus, Elias Koukoutsis, C. Triantafillou, C. Vasilatos:
    Exact monitoring of the numerical error in various speech algorithms. 1073-1076
• 
  • 
  • 
  • 
  • 
    Jacques C. Koreman, Bert Cranen, Louis Boves:
    Automatic computation and comparison of dynamically varying voice source parameters. 1077-1080
• 
  • 
  • 
  • 
  • 
    Paavo Alku:
    Glottal wave analysis with pitch synchronous iterative adaptive inverse filtering. 1081-1084
• 
  • 
  • 
  • 
  • 
    Thierry Galas, Xavier Rodet:
    Generalized functional approximation for source-filter system modeling. 1085-1088
• 
  • 
  • 
  • 
  • 
    Frédéric Bimbot, Bishnu S. Atal:
    An evaluation of temporal decomposition. 1089-1092

Discriminant Training and Speaker Adaptation

• 
  • 
  • 
  • 
  • 
    Klaus Zünkler:
    A discriminative recognizer for isolated and continuous speech using statistical separability measures. 1095-1098
• 
  • 
  • 
  • 
  • 
    Otto Schmidbauer, Harald Höge:
    Speaker adaptation based on articulatory features. 1099-1102
• 
  • 
  • 
  • 
  • 
    Fabio Brugnara, Renato de Mori, Diego Giuliani, Maurizio Omologo:
    A parallel HMM approach to speech recognition. 1103-1106
• 
  • 
  • 
  • 
  • 
    Tsuneo Nitta, Jun'ichi Iwasaki, Hiroshi Matsu'ura:
    Speaker independent word recognition using HMMs with an orthogonalized phonetic segment codebook. 1107-1110
• 
  • 
  • 
  • 
  • 
    Pascale Fung, Tatsuya Kawahara, Shuji Doshita:
    Unsupervised speaker normalization by speaker Markov model converter for speaker-independent speech recognition. 1111-1114

Perception I

• 
  • 
  • 
  • 
  • 
    R. J. J. H. van Son, Louis C. W. Pols:
    The influence of formant track shape on the perception of synthetic vowels. 1117-1120
• 
  • 
  • 
  • 
  • 
    P. A. Howard-Jones:
    Fluctuation of noise background: measurement and significance in relation to speech masking. 1121-1124
• 
  • 
  • 
  • 
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    C. Ma, L. F. Willems:
    The audibility of narrow band noise in fiat spectral complex sounds. 1125-1128
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    Gitta P. M. Laan, Dick R. van Bergem, Florien J. Koopmans-van Beinum:
    The importance of spectral quality of vowels for the intelligibility of sentences. 1129-1132
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    Herman J. M. Steeneken, Tammo Houtgast:
    On the mutual dependency of octave-band-specific contributions to speech intelligibility. 1133-1136
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    Brit van Ooyen, Anne Cutler, Dennis Norris:
    Detection times for vowels versus consonants. 1451-1454
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    Dick R. van Bergem:
    The influence of sentence accent, word stress, and word class on the quality of vowels. 1455-1458
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    Florien J. Koopmans-van Beinum:
    A peak-and-level model for focus words in read and spontaneous natural speech and in synthetic speech. 1459-1462
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    John Ingram, Jeff Pittam:
    Connected speech processes in second language learning. 1463-1466

Speech Synthesis and Prosody

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    Rodmonga K. Potapova:
    Modification of acoustic features in Russian connected speech. 1141-1144
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    Helmer Strik, Louis Boves:
    On the relation between voice source characteristics and prosody. 1145-1148
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    Sverre Stensby:
    Prosody in a rule-based norwegian text-to-speech system. 1149-1152
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    A. S. Madhukumar, S. Rajendran, C. Chandra Sekhar, B. Yegnanarayana:
    Synthesizing intonation for speech in hindi. 1153-1156
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    James Hieronymus, Briony J. Williams:
    An investigation of the relation between perceived pitch accent and automatically-located accent in british English. 1157-1160
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    Silvia Quazza:
    Modelling Italian intonation in a text-to-speech system. 1161-1164
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    Michael H. O'Malley, Howard Resnick, Michelle Caisse:
    An analysis of strategies for finding prosodic clues in text. 1165-1168
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    Marcello Balestri:
    A coded dictionary for stress assignment rules in Italian. 1169-1172

Text-to-Speech Synthesis Systems

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    Enrico te Lindert, Hugo C. van Leeuwen:
    Speech maker: text-to-speech conversion based on a multi-level, synchronized data structure. 1231-1234
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    Eric Lewis, Mark Tatham:
    A new text-to-speech synthesis system. 1235-1238
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    Luís C. Oliveira, Céu Viana, Isabel Trancoso:
    DIXI - portuguese text-to-speech system. 1239-1242
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    P. Molbaek Hansen, N. Reinholt Petersen, Jørgen Rischel, Carsten Henriksen:
    Higher-level linguistic information in a text-to-speech system for danish. 1243-1246
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    Gábor Olaszy:
    Adaptation of the multivox text-to-speech system to Italian. 1247-1250

Phonetic Modelling

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    Partha Niyogi, Victor W. Zue:
    Correlation analysis of vowels and their application to speech recognition. 1253-1256
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    John N. Holmes:
    Use of phonetic knowledge when designing and training stochastic models for speech recognition. 1257-1260
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    Bernhard Kaspar, Karlheinz Schuhmacher:
    Modelling phones by microsegments in a phonetically oriented recognition system. 1261-1264
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    Il K. Kim, H. S. Lee:
    An extended LVQ2 algorithm and its application to phoneme classification. 1265-1268
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    Paul J. Dix, Gert-Jan Vernooij, Gerrit Bloothooft:
    A hierarchical broad phonetic classification scheme. 1269-1272

Generation of Prosody

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    Julia Hirschberg:
    Using text analysis to predict intonational boundaries. 1275-1278
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    Merle Horne:
    Why do speakers accent 'given' information ? 1279-1282
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    Julie Vonwiller, Robin W. King, R. W. T. Lloyd:
    Automatic prosody assignment for interactive synthesized dialogue systems. 1283-1286
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    NickYoud NickYoud, Jill House:
    Generating intonation in a voice dialogue system. 1287-1290
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    Rodolfo Delmonte, Roberto Dolci:
    Computing linguistic knowledge for text-to-speech systems with PROSO. 1291-1294

Speech Processing and Analysis

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    C. Acker, Peter Vary, H. Ostendarp:
    Acoustic echo cancellation using prediction residual signals. 1297-1300
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    H. S. Dabis, Alan Wrench:
    An evaluation of adaptive noise cancelling for speech recognition. 1301-1304
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    Enzo Mumolo, Antonello Riccio, Giuseppe Abbattista:
    An efficient algorithm for real-time voiced/unvoiced decision. 1305-1308
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    Tim Aarset, Ben Gold:
    Models of pitch perception. 1309-1312
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    P. Corney, John S. Mason:
    A new perspective on LPC excitation using singular value decomposition. 1315-1318
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    Werner Verhelst, Marcel Borger:
    Intra-speaker transplantation of speech characteristics an application of waveform vocoding techniques and DTW. 1319-1322
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    S. H. Leung, O. Y. Wong, K. L. Lai:
    Decomposition of the LPC excitation using wavelet functions. 1327-1330
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    Eliathamby Ambikairajah, Liam Kilmartin:
    An adaptive cochlear model for speech recognition. 1331-1334
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    Giovanni Jacovitti, Piero Pierucci, Alessandro Falaschi:
    Speech segmentation and classification using higher order moments. 1335-1338

Automatic Speech Recognition: Hardware and Noise Reduction

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    Alberto Ciaramella, Davide Clementino, Roberto Pacifici:
    A PC-housed speaker independent large vocabulary continuous telephonic speech recognizer. 1341-1344
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    Abdulmesih Aktas, Klaus Zünkler:
    Speaker independent continuous HMM-based recognition of isolated words on a real-time multi-DSP system. 1345-1348
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    Anastasios Tsopanoglou, Efstathios D. Kyriakis-Bitzaros, John Mourjopoulos, George K. Kokkinakis:
    A real time speech decoder using instantaneous frequency and energy. 1349-1352
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    Martin Schultheiß, Arild Lacroix:
    Fast hardware for efficient parallel processing of speech signals. 1353-1356
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    Jan Sedivý, Jiff Filcev, Jan Uhlír, Tomas Vanek, Václav Hanzl, Zdenek Oliva, Petr Kotek:
    The one chip speech recognition system. 1357-1631
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    Luis Villarrubia, M. J. Poza, Carlos Crespo-Casas:
    Influence of the telephone line on automatic speech recognition. 1363-1366
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    Hynek Hermansky, Nelson Morgan, Aruna Bayya, Phil Kohn:
    Compensation for the effect of the communication channel in auditory-like analysis of speech (RASTA-PLP). 1367-1370
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    Jean-Claude Junqua, Ben Reaves, Brian Mak:
    A study of endpoint detection algorithms in adverse conditions: incidence on a DTW and HMM recognizer. 1371-1374
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    Susanne Dvorak, Thomas Hormann:
    High-performance speech recognition in noise by continuously updated reference templates. 1375-1378
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    Klára Vicsi:
    Speech enhancement in the case of speech recognizers. 1379-1381
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    Juan Gómez-Mena, J. Santos-Suarez, Ramón García-Gómez:
    A robust feature extraction method for automatic speech recognition in noisy environments. 1383-1386

Sub-Word Units for Automatic Speech Recognition

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    Lorenzo Fissore, Egidio P. Giachin, Pietro Laface, Giorgio Micca:
    Selection of speech units for a speaker-independent CSR task. 1389-1392
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    Egidio P. Giachin, Chin-Hui Lee, Lawrence R. Rabiner, Aaron E. Rosenberg, Roberto Pieraccini:
    Word juncture modeling using inter-word context-dependent phone-like units. 1393-1396
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    Akito Nagai, Shigeki Sagayama, Kenji Kita:
    Phoneme-context-dependent LR parsing algorithms for HMM-based continuous speech recognition. 1397-1400
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    H. Drexler, R. Roddeman, Louis Boves, Helmer Strik:
    Optimizing lexical fast search in a large vocabulary isolated word speech recognition system. 1401-1404

Auditory Modelling

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    Tore Fjällbrant, Fisseha Mekuria:
    Signal processing using an auditory filter bank with side-lobes and phase-jumps. 1429-1431
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    J. S. C. van Dijk:
    Notes on auditive coding of sophisticated signals. 1433-1436
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    Manfred Beham:
    An auditorily based spectral transformation of speech signals. 1437-1440
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    Andrew C. Morris, Pierre Escudier, Jean-Luc Schwartz:
    On and off units detect information bottle-necks for speech recognition. 1441-1444
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    Jose A. Pozas-Alvarez:
    A new logic operator-based auditory system model. 1445-1448

Speech Interfaces: Dialogue and Human Factors

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    Jeremy Peckham:
    Speech understanding and dialogue over the telephone: an overview of progress in the sundial project. 1469-1472
• 
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  • 
    Jean-Pierre Tubach, P. Doignon:
    A system for natural spoken language queries design, implementation and assessment. 1473-1476
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  • 
    Guy Deville, Pierre Mousel:
    Operational validation of syntactic-semantic models in a spoken man-machine dialogue system. 1477-1480
• 
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  • 
    Bertrand Gaiffe, Laurent Romary, Jean-Marie Pierrel:
    References in a multimodal dialogue: towards a unified processing. 1481-1485
• 
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  • 
    Pierre Lefebvre, G. Duncan, Frank Poirier:
    The user-unix dialogue: a novel integrated approach to enhancing the operating system interface. 1487-1490
• 
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  • 
  • 
    Bodo Arndt:
    Adoption op verbal and visual dialogue behaviour in document handling systems. 1491-1494
• 
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    Paula M. T. Smeele, Anne C. Sittig:
    The contribution of vision to speech perception. 1495-1497
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    Robin J. Lickley, R. C. Shillcock, Ellen Gurman Bard:
    Processing disfluent speech: how and when are disfluencies found? 1499-1502
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    A. Chointere, Jean-Marc Robert, Raymond Descout:
    Building a user interface for a speech recognition-based telephone application system. 1503-1506
• 
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  • 
    A. C. Murray, Clive Frankish, Dylan M. Jones:
    System design and human factors in auditory interfaces. 1507-1510

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