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ORCIDEUROSPEECH 1991: Genova, Italy
- Second European Conference on Speech Communication and Technology, EUROSPEECH 1991, Genova, Italy, September 24-26, 1991. ISCA 1991
Plenary
- Sadaoki Furui:
Recent advances in speech recognition. 3-12 - Frank Fallside:
On the acquisition of speech by machines, ASM. 13-14
Continuous Speech Recognition
- 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 - P. S. Gopalakrishnan, David Nahamoo:
Immediate recognition of embedded command words. 21-24 - Lynn Wilcox, Marcia A. Bush:
HMM-based wordspotting for voice editing and indexing. 25-28 - Janet M. Baker:
Large vocabulary speaker-adaptive continuous speech recognition research overview at dragon systems. 29-32 - 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
- Katsuhiko Shirai, Kazuo Hashimoto, Tetsunori Kobayashi:
Text-to-speech synthesizer using superposition of sinusoidal waves generated by synchronized oscillators. 39-42 - M. Guerti, Gérard Bailly:
Synthesis-by-rule using compost: modelling resonance trajectories. 43-46 - Yasushi Ishikawa, Kunio Nakajima:
Neural network based spectral interpolation method for speech synthesis by rule. 47-50 - Martine Garnier-Rizet:
A rule-based segmental synthesis module for French. 51-54
Human Factors
- Norman M. Fraser, G. Nigel Gilbert:
Effects of system voice quality on user utterances in speech dialogue systems. 57-60 - P. Day, Andreas Grünupp, Klaus-Peter Muthig:
A human factors study of speech-to-text technology: consequences of discrete speech. 61-64 - Iain R. Murray, John L. Arnott, Alan F. Newell:
A comparison of document composition using a listening typewriter and conventional office systems. 65-68 - Paulus H. Vossen:
Evaluating speech input and output in a CAD-system using the hidden-operator method. 69-72 - 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
- 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 - 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 - Klaus Fellbaum, Dieter Becker:
Isolated word recognition with integrated noise reduction. 87-90 - 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
- Jorma Laaksonen:
A new reliability-based phoneme segmentation method for the "neural" phonetic typewriter. 97-100 - Bruno Apolloni, Francesco Pazienti, Vincenzo Trotta:
Isolated word adaptive recognizer based on neural networks. 101-104 - Nobuo Hataoka, Alex Waibel:
Evaluation of speaker-independent phoneme recognition on TIMIT database using TDNNs. 105-108 - Nelson Morgan, Hervé Bourlard, Chuck Wooters, Phil Kohn, Michael Cohen:
Phonetic context in hybrid HMM/MLP continuous speech recognition. 109-112 - E. C. Andrews, John S. Mason:
Neural network classification of complex-valued speech features. 113-116 - Dennis Norris:
Rewiring lexical networks on the fly. 117-120 - Kjell Elenius, G. Takacs:
Phoneme recognition with an artificial neural network. 121-124 - Jianxin Jiang, Kechu Yi, Zheng Hu:
A new self-organization algorithm of forming a phoneme map. 125-128 - Shuping Ran, J. Bruce Millar:
Phoneme classification using neural networks based on acoustic-phonetic structure. 129-132 - Nigel Dodd, Donald MacFarlane, Chris Marland:
Networks for speech recognition structurally optimised by genetic techniques implemented on parallel hardware. 133-136
Phonetics I, II
- Jeff Pittam, John Ingram:
Influence of vietnamese tone and prosody on the acquisition of English stress patterns. 139-142 - Walter F. Sendlmeier:
The voiced/unvoiced distinction of initial stops by normal and hearing impaired listeners. 143-146 - Krishna S. Nathan:
Comparison of formant transition based stop classifiers: time-varying and time-invariant signal models. 147-150 - Christian Benoît, Christian Abry, L. J. Roe:
The effect of context on labiality in French. 151-156 - A. K. Datta, N. R. Ganguli, B. Mukherjee:
Nasalisation in bengali speech sounds acoustic-phonetic study. 157-160 - N. R. Ganguli:
Vowel formant frequency distribution of a major indian language. 161-164 - Bernard Harmegnies, Marielle Bruyninckx, Joaquim Llisterri, Dolors Poch:
Effects of language change on voice quality in bilingual speakers, corpus content effect. 165-168 - T. I. Shevchenko, T. S. Skopintseva:
Effects of social and regional backgrounds on LTAS in british English. 169-172 - Henk van den Heuvel, Bert Cranen, Toni C. M. Rietveld:
Speaker related variability in the durations of dutch speech segments. 251-254 - Johan Liljencrants:
Numerical simulations of glottal flow. 255-258 - Joop Jansen, Bert Cranen, Louis Boves:
Modelling of source characteristics of speech sounds by means of the LF-model. 259-262 - Hanspeter Herzel, J. Wendler:
Evidence of chaos in phonatory samples. 263-266 - 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)
- 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 - 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 - Hermann Ney, Roberto Billi:
Prototype systems for large-vocabulary speech recognition: polyglot and spicos. 193-200
Spoken Language Parsing
- J. H. Wright:
Adaptation of grammar-based language models for continuous speech recognition. 203-206 - Keh-Yih Su, Tung-Hui Chiang, Yi-Chung Lin:
A robustness and discrimination oriented score function for integrating speech and language processing. 207-210 - Paolo Baggia, Lorenzo Fissore, Elisabetta Gerbino, Egidio P. Giachin, Claudio Rullent:
Improving speech understanding performance through feedback verification. 211-214 - Anna Corazza, Renato de Mori, Roberto Gretter, Giorgio Satta:
Computation of upper-bounds for island-driven stochastic parsers. 215-218 - François Andry, J. H. Simon Thornton:
A parser for speech lattices using a UCG grammar. 219-222 - Sheryl Young, Michael Matessa:
Using pragmatic and semantic knowledge to correct parsing of spoken language utterances. 223-227
Speech Coding I-IV
- Arnaldo J. Abrantes, Jorge S. Marques, Isabel Trancoso:
Hybrid sinusoidal modeling of speech without voicing decision. 231-234 - Jorge S. Marques, Isabel Trancoso, Arnaldo J. Abrantes:
Harmonic coding of speech: an experimental study. 235-238 - David Rowe, William G. Cowley, Andrew Perkis:
A multiband excitation linear predictive speech coder. 239-242 - Shu Hung Leung, K. L. Lai, O. Y. Wong, Andrew Luk:
A new coded excitation model using multifrequency decomposition. 245-248 - Daniele Sereno:
Frame substitution and adaptive post-filtering in speech coding. 595-598 - S. A. Atungsiri, R. Soheili, Ahmet M. Kondoz, Barry G. Evans:
Effective lost speech frame reconstruction for CELP coders. 599-602 - Hiromi Nagabuchi, Nobuhiko Kitawaki:
Evaluation and improvement of coded speech quality degraded by cell loss in ATM networks. 603-606 - Alain J. Vigier:
Combined source-channel coding for a very noisy channed. 607-610 - G. Rosina, Marcello Sant' Agostino, E. Turco, Luigi Vetrano:
Testing and quality enhancement of the GSM full rate voice channel. 611-614 - U. Kipper, Herbert Reininger, Dietrich Wolf:
Low bit rate speech coding using CELP with adaptive excitation codebook. 893-896 - Arild Fuldseth, Erik Harborg, Finn Tore Johansen, Jan E. Knudsen:
A real-time implementable 7 khz speech coder at 16 kbit/s. 897-900 - D. J. Zarkadis:
Adaptive spectral weighting for vector predictive coding of the LPC-spectra. 901-904 - Samir Saoudi, Jean-Marc Boucher, Alain Le Guyader:
Medium band speech coding using optimal scalar quantization of LSP. 905-908 - Philip Secker, Andrew Perkis:
Joint source and channel coding of line spectrum pairs. 909-912 - C. F. Chan, K. W. Law:
An algorithm for computing LSP frequencies directly from the reflection coefficients. 913-916 - Peter Meyer, W. Peters, Jürgen Paulus:
Variable rate speech coding using perceptive thresholds and adaptive VUS detection. 809-812 - 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 - Carlos M. Ribeiro, Isabel Trancoso:
A 4.8 kbps celp coder with post-processing. 817-820 - K. W. Law, O. Y. Wong, C. F. Chan:
A real-time high quality joint-excitation linear predictive coder at 8 kbps. 821-824 - Rosario Drogo de Iacovo, Roberto Montagna:
Some experiments in perceptual masking of quantizing noise in analysis-by-synthesis speech coders. 825-828 - Gao Yang, Henri Leich, René Boite:
A very high-quality CELP coder at the rate of 2400 bps. 829-832 - Z. Yong Liu:
An effective pulse adaptive code-excited linear predictive coder at 4kb/S. 835-838 - 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
- 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 - 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 - P. A. Howard-Jones:
'SOAP' - a speech output assessment package for controlled multilingual evaluation of synthetic speech. 281-284 - Tammo Houtgast, Jan A. Verhave:
A physical approach to speech quality assessment: correlation patterns in the speech spectrogram. 285-288 - Hiroyuki Miyata, Tammo Houtgast:
Weighted MTF for predicting speech intelligibility in reverberant sound fields. 289-292 - Ute Jekosch:
Speech intelligibility studies for the european hermes spaceplane. 293-296 - 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 - 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 - Yolande Anglade, Jean-Marie Pierrel, Jean-Claude Junqua:
A spoken language interface for a telephone switchboard operator center. 307-310 - 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
- Thomas Portele, Birgit Steffan, Rainer Preuß, Wolfgang Hess:
German speech synthesis by concatenation of non-parametric units. 317-320 - Giuseppe Abbattista, Antonello Riccio, Enzo Mumolo:
Automatic document reader with speech output capabilities. 321-324 - Robin W. King:
Tools and processes for developing low-cost and high-quality text-to-speech synthesis for communication aids. 325-329 - Hynek Hermansky, Louis Anthony Cox Jr.:
Perceptual linear predictive (PLP) analysis-resynthesis technique. 329-332 - Reinhold Greisbach, Bernd J. Kröger, O. Esser, G. Plaßmann:
A display technique for measurements of natural and synthetic articulatory dynamics. 333-336 - Yueh-Chin Chang, Yi-Fan Lee, Bang-Er Shia, Hsiao-Chuan Wang:
Statistical models for the Chinese text-to-speech system. 337-340 - P. A. Taylor, I. A. Nairn, Andrew M. Sutherland, Mervyn A. Jack:
A realtime speech synthesis system. 341-344 - Hélène Valbret, Eric Moulines, Jean-Pierre Tubach:
Voice tranformation using PSOLA technique. 345-348 - Massimo Giustiniani, Piero Pierucci:
Phonetic ergodic HMM for speech synthesis. 349-352 - 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 - H. Zingte, Cl. Hennebois:
Helping young children to associate sounds and letters through speech synthesis. 357-360 - Hervé Bourlard:
Neural nets and hidden Markov models: review and generalizations. 363-369 - Nikil S. Jayant, James D. Johnston, Yair Shoham:
Coding of wideband speech. 373-379
Probabilistic Language Models for Speech Recognition
- Roberto Pieraccini, Esther Levin:
Stochastic representation of semantic structure for speech understanding. 383-386 - Colin Matheson, Fergus R. McInnes:
Incorporating probabilities into the dualgram language model. 387-390 - Egidio P. Giachin:
A dynamic programming based framework for stochastic spoken language understanding. 391-394 - Natividad Prieto, Enrique Vidal:
Learning language models through the ECGI method. 395-398 - 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
- Katsuhiko Shirai, Eiichiro Kitagawa, T. Endo:
Optimal construction of context sensitive quantizer for phoneme recognition in continuous speech. 405-408 - Mary O'Kane, P. E. Kenne, D. Landy, S. Atkins:
Generalising from single-speaker recognition in a feature-based recogniser. 409-412 - Hans-Günter Hirsch, Peter Meyer, Hans-Wilhelm Rühl:
Improved speech recognition using high-pass filtering of subband envelopes. 413-416 - Yifan Gong, Jean Paul Haton:
Comparing two phoneme identification methods using a continuous speech recognizer. 417-420 - D. Ederveen, Louis Boves:
Knowledge-based phoneme recognition. 421-424
Speaker Identification and Verification
- J. Kraayeveld, A. C. M. Rietveld, Vincent J. van Heuven:
Speaker characterization in dutch using prosodic parameters. 427-430 - Alan K. Hunt:
New commercial applications of telephone-network-based speech recognition and speaker verification. 431-434 - Jean-François Bonastre, Henri Meloni, Philippe Langlais:
Analytical strategy for speaker identification. 435-438 - L. Xu, John S. Mason:
Optimization of perceptually-based spectral transforms in speaker identification. 439-442
Pitch Determination and Voice Separation
- Alain de Cheveigné:
A mixed speech F0 estimation algorithm. 445-448 - Edward Jones, Eliathamby Ambikairajah:
A perceptually-based pitch extractor for band-limited speech. 449-452 - Yu-Hua Gu:
A robust pseudo perceptual pitch estimator. 453-456 - Neviano Dal Degan, Marco Fratti:
Pitch estimation based on a "narrowed" autocorrelation function. 457-460
Speech Recognition: Understanding Systems
- Seiichi Nakagawa, Yoshimitsu Hirata, Isao Murase:
The syntax-oriented spoken Japanese understanding system SPOJOS-SYNO II. 463-466 - Henning Bergmann, Hans-Hermann Hamer, Andreas Noll, Annedore Paeseler, Horst Tomaschewski:
An adaptable man-machine interface using connected-word recognition. 467-470 - 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 - I. Lee Hetherington, Hong C. Leung, Victor W. Zue:
Toward vocabulary-independent recognition of telephone speech. 475-478 - Ronald A. Cole, Krist Roginski, Mark A. Fanty:
English alphabet recognition with telephone speech. 479-482 - Jean-Yves Fiset, Jean-Marc Robert, Raymond Descout:
Evolutionary language models in air traffic control training. 483-486 - 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 - Mitchell Hood:
Lexical access in a speech understanding and dialogue system. 490-493 - Reinhold Haeb-Umbach, Hermann Ney:
A look-ahead search technique for large vocabulary continuous speech recognition. 495-498 - Carlos Teixeira, Isabel Trancoso:
Spectral subtraction for front-end noise reduction in a speech recognizer. 499-502
Speech Databases, Analysis And Assessment
- Lori F. Larnel, Jean-Luc Gauvain, Maxine Eskénazi:
BREF, a large vocabulary spoken corpus for French. 505-508 - Luc Mathan, Dominique Morin:
Speech field databases: development and analysis. 509-512 - Shuichi Itahashi:
Large scale Japanese dialect speech corpora. 513-516 - Paulus H. Vossen:
Outline of a design-oriented evaluation framework for speech-driven applications. 517-520 - Richard Winski, Kamran Kordi:
Assessment of continuous speech recognisers using recogniser sensitivity analysis. 521-524 - Christine Bourjot, Anne Boyer, Dominique Fohr:
A tool for assessment of acoustic phonetic lattices. 525-528 - 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 - C. Ma, L. F. Willems:
The audibility of narrow band noise in fiat spectral complex sounds. 1125-1128 - 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 - Herman J. M. Steeneken, Tammo Houtgast:
On the mutual dependency of octave-band-specific contributions to speech intelligibility. 1133-1136 - Brit van Ooyen, Anne Cutler, Dennis Norris:
Detection times for vowels versus consonants. 1451-1454 - Dick R. van Bergem:
The influence of sentence accent, word stress, and word class on the quality of vowels. 1455-1458 - 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 - John Ingram, Jeff Pittam:
Connected speech processes in second language learning. 1463-1466
Speech Synthesis and Prosody
- Rodmonga K. Potapova:
Modification of acoustic features in Russian connected speech. 1141-1144 - Helmer Strik, Louis Boves:
On the relation between voice source characteristics and prosody. 1145-1148 - Sverre Stensby:
Prosody in a rule-based norwegian text-to-speech system. 1149-1152 - A. S. Madhukumar, S. Rajendran, C. Chandra Sekhar, B. Yegnanarayana:
Synthesizing intonation for speech in hindi. 1153-1156 - James Hieronymus, Briony J. Williams:
An investigation of the relation between perceived pitch accent and automatically-located accent in british English. 1157-1160 - Silvia Quazza:
Modelling Italian intonation in a text-to-speech system. 1161-1164 - Michael H. O'Malley, Howard Resnick, Michelle Caisse:
An analysis of strategies for finding prosodic clues in text. 1165-1168 - Marcello Balestri:
A coded dictionary for stress assignment rules in Italian. 1169-1172
Text-to-Speech Synthesis Systems
- Enrico te Lindert, Hugo C. van Leeuwen:
Speech maker: text-to-speech conversion based on a multi-level, synchronized data structure. 1231-1234 - Eric Lewis, Mark Tatham:
A new text-to-speech synthesis system. 1235-1238 - Luís C. Oliveira, Céu Viana, Isabel Trancoso:
DIXI - portuguese text-to-speech system. 1239-1242 - P. Molbaek Hansen, N. Reinholt Petersen, Jørgen Rischel, Carsten Henriksen:
Higher-level linguistic information in a text-to-speech system for danish. 1243-1246 - Gábor Olaszy:
Adaptation of the multivox text-to-speech system to Italian. 1247-1250
Phonetic Modelling
- Partha Niyogi, Victor W. Zue:
Correlation analysis of vowels and their application to speech recognition. 1253-1256 - John N. Holmes:
Use of phonetic knowledge when designing and training stochastic models for speech recognition. 1257-1260 - Bernhard Kaspar, Karlheinz Schuhmacher:
Modelling phones by microsegments in a phonetically oriented recognition system. 1261-1264 - Il K. Kim, H. S. Lee:
An extended LVQ2 algorithm and its application to phoneme classification. 1265-1268 - Paul J. Dix, Gert-Jan Vernooij, Gerrit Bloothooft:
A hierarchical broad phonetic classification scheme. 1269-1272
Generation of Prosody
- Julia Hirschberg:
Using text analysis to predict intonational boundaries. 1275-1278 - Merle Horne
:
Why do speakers accent 'given' information ? 1279-1282 - Julie Vonwiller, Robin W. King, R. W. T. Lloyd:
Automatic prosody assignment for interactive synthesized dialogue systems. 1283-1286 - NickYoud NickYoud, Jill House:
Generating intonation in a voice dialogue system. 1287-1290 - Rodolfo Delmonte, Roberto Dolci:
Computing linguistic knowledge for text-to-speech systems with PROSO. 1291-1294
Speech Processing and Analysis
- C. Acker, Peter Vary, H. Ostendarp:
Acoustic echo cancellation using prediction residual signals. 1297-1300 - H. S. Dabis, Alan Wrench:
An evaluation of adaptive noise cancelling for speech recognition. 1301-1304 - Enzo Mumolo, Antonello Riccio, Giuseppe Abbattista:
An efficient algorithm for real-time voiced/unvoiced decision. 1305-1308 - Tim Aarset, Ben Gold:
Models of pitch perception. 1309-1312 - P. Corney, John S. Mason:
A new perspective on LPC excitation using singular value decomposition. 1315-1318 - Werner Verhelst, Marcel Borger:
Intra-speaker transplantation of speech characteristics an application of waveform vocoding techniques and DTW. 1319-1322 - S. H. Leung, O. Y. Wong, K. L. Lai:
Decomposition of the LPC excitation using wavelet functions. 1327-1330 - Eliathamby Ambikairajah, Liam Kilmartin:
An adaptive cochlear model for speech recognition. 1331-1334 - Giovanni Jacovitti, Piero Pierucci, Alessandro Falaschi:
Speech segmentation and classification using higher order moments. 1335-1338
Automatic Speech Recognition: Hardware and Noise Reduction
- Alberto Ciaramella, Davide Clementino, Roberto Pacifici:
A PC-housed speaker independent large vocabulary continuous telephonic speech recognizer. 1341-1344 - Abdulmesih Aktas, Klaus Zünkler:
Speaker independent continuous HMM-based recognition of isolated words on a real-time multi-DSP system. 1345-1348 - Anastasios Tsopanoglou, Efstathios D. Kyriakis-Bitzaros, John Mourjopoulos, George K. Kokkinakis:
A real time speech decoder using instantaneous frequency and energy. 1349-1352 - Martin Schultheiß, Arild Lacroix:
Fast hardware for efficient parallel processing of speech signals. 1353-1356 - Jan Sedivý, Jiff Filcev, Jan Uhlír, Tomas Vanek, Václav Hanzl, Zdenek Oliva, Petr Kotek:
The one chip speech recognition system. 1357-1631 - Luis Villarrubia, M. J. Poza, Carlos Crespo-Casas:
Influence of the telephone line on automatic speech recognition. 1363-1366 - 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 - 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 - Susanne Dvorak, Thomas Hormann:
High-performance speech recognition in noise by continuously updated reference templates. 1375-1378 - Klára Vicsi:
Speech enhancement in the case of speech recognizers. 1379-1381 - 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
- Lorenzo Fissore, Egidio P. Giachin, Pietro Laface, Giorgio Micca:
Selection of speech units for a speaker-independent CSR task. 1389-1392 - 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 - Akito Nagai, Shigeki Sagayama, Kenji Kita:
Phoneme-context-dependent LR parsing algorithms for HMM-based continuous speech recognition. 1397-1400 - 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
- Tore Fjällbrant, Fisseha Mekuria:
Signal processing using an auditory filter bank with side-lobes and phase-jumps. 1429-1431 - J. S. C. van Dijk:
Notes on auditive coding of sophisticated signals. 1433-1436 - Manfred Beham:
An auditorily based spectral transformation of speech signals. 1437-1440 - Andrew C. Morris, Pierre Escudier, Jean-Luc Schwartz:
On and off units detect information bottle-necks for speech recognition. 1441-1444 - Jose A. Pozas-Alvarez:
A new logic operator-based auditory system model. 1445-1448
Speech Interfaces: Dialogue and Human Factors
- Jeremy Peckham:
Speech understanding and dialogue over the telephone: an overview of progress in the sundial project. 1469-1472 - Jean-Pierre Tubach, P. Doignon:
A system for natural spoken language queries design, implementation and assessment. 1473-1476 - Guy Deville, Pierre Mousel:
Operational validation of syntactic-semantic models in a spoken man-machine dialogue system. 1477-1480 - Bertrand Gaiffe, Laurent Romary, Jean-Marie Pierrel:
References in a multimodal dialogue: towards a unified processing. 1481-1485 - Pierre Lefebvre, G. Duncan, Frank Poirier:
The user-unix dialogue: a novel integrated approach to enhancing the operating system interface. 1487-1490 - Bodo Arndt:
Adoption op verbal and visual dialogue behaviour in document handling systems. 1491-1494 - Paula M. T. Smeele, Anne C. Sittig:
The contribution of vision to speech perception. 1495-1497 - Robin J. Lickley, R. C. Shillcock, Ellen Gurman Bard:
Processing disfluent speech: how and when are disfluencies found? 1499-1502 - A. Chointere, Jean-Marc Robert, Raymond Descout:
Building a user interface for a speech recognition-based telephone application system. 1503-1506 - A. C. Murray, Clive Frankish, Dylan M. Jones:
System design and human factors in auditory interfaces. 1507-1510
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