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ORCIDICASSP 1994: Adelaide, South Australia, Australia
- Proceedings of ICASSP '94: IEEE International Conference on Acoustics, Speech and Signal Processing, Adelaide, South Australia, Australia, April 19-22, 1994. IEEE Computer Society 1994, ISBN 0-7803-1775-0
Volume 1
Speech Enhancement
- Srinivas Nandkumar, John H. L. Hansen:
Speech enhancement based on a new set of auditory constrained parameters. 1-4 - Gary H. Whipple:
Low residual noise speech enhancement utilizing time-frequency filtering. 5-8 - Athina P. Petropulu, Suresh Subramaniam:
Cepstrum based deconvolution for speech dereverberation. 9-12 - Hamid Sheikhzadeh, Hossein Sameti, Li Deng, Robert L. Brennan:
Comparative performance of spectral subtraction and HMM-based speech enhancement strategies with application to hearing and design. 13-16 - Anthony Teolis, John J. Benedetto:
Noise suppression using a wavelet model. 17-20 - Subrata K. Das, Arthur Nádas, David Nahamoo, Michael Picheny:
Adaptation techniques for ambience and microphone compensation in the IBM Tangora speech recognition system. 21-24 - Michael I. Savic, Huiqin Gao, Jeffrey S. Sorensen:
Co-channel speaker separation based on maximum-likelihood deconvolution. 25-28 - Michael A. Ramalho, Richard J. Mammone:
A new speech enhancement technique with application to speaker identification. 29-32
Acoustic/Phonetic Modelling
- Les T. Niles:
Acoustic modeling for speech recognition based on spotting of phonetic units. 33-36 - Tony Robinson, Mike Hochberg, Steve Renals:
IPA: improved phone modelling with recurrent neural networks. 37-40 - R. N. V. Sitaram, Thippur V. Sreenivas:
Phoneme recognition in continuous speech using large inhomogeneous hidden Markov models. 41-44 - Li Deng, Don X. Sun:
Phonetic classification and recognition using HMM representation of overlapping articulatory features for all classes of English sounds. 45-48 - S. Krishnan, P. V. S. Rao:
Segmental phoneme recognition using piecewise linear regression. 49-52 - Basavaraj I. Pawate, Eric M. Dowling:
A new method for segmenting continuous speech. 53-56 - Yifan Gong, Jean Paul Haton:
Stochastic trajectory modeling for speech recognition. 57-60 - Antonio M. Peinado
, José C. Segura, Antonio J. Rubio, M. Carmen Benítez:
Using multiple vector quantization and semicontinuous hidden Markov models for speech recognition. 61-64 - Nelly Suaudeau, Régine André-Obrecht:
An efficient combination of acoustic and supra-segmental informations in a speech recognition system. 65-68 - Takashi Yoshimura, Satoru Hayamizu, Kazuyo Tanaka:
Word accent patterns modelling by concatenation of mora hidden Markov models. 69-72 - David B. Grayden, Michael S. Scordilis:
Phonemic segmentation of fluent speech. 73-76 - Ara Samouelian:
Knowledge based approach to consonant recognition. 77-80
Telephone Based Speech Recognition and Databases
- Jared Bernstein, Kelsey Taussig, John J. Godfrey:
Macrophone: an American English telephone speech corpus for the Polyphone project. 81-84 - Mitchel Weintraub, Leonardo Neumeyer:
Constructing telephone acoustic models from a high-quality speech corpus. 85-88 - Thomas Staples, Joseph Picone, Nozomi Arai:
The voice across Japan database-the Japanese language contribution to Polyphone. 89-92 - Ronald A. Cole, David G. Novick, Daniel C. Burnett, Brian Hansen, Stephen Sutton, Mark Fant:
Towards automatic collection of the US census. 93-96 - J. Bruce Millar, Julie Vonwiller, Jonathan Harrington, Phillip Dermody:
The Australian National Database of Spoken Language. 97-100 - Kazuhiro Kondo, Joseph Picone, Barbara Wheatley:
A comparative analysis of Japanese and English digit recognition. 101-104 - Eric R. Buhrke, Régis Cardin, Yves Normandin, Mazin G. Rahim, Jay G. Wilpon:
Application of vector quantized hidden Markov modeling to telephone network based connected digit recognition. 105-108 - Pedro J. Moreno, Richard M. Stern:
Sources of degradation of speech recognition in the telephone network. 109-112 - John S. Garofolo, Tony Robinson, Jonathan G. Fiscus:
The development of file formats for very large speech corpora: SPHERE and SHORTEN. 113-116 - Yu-Hung Kao, Charles T. Hemphill, Barbara Wheatley, Raja Rajasekaran:
Toward vocabulary independent telephone speech recognition. 117-120 - Ove Andersen, Paul Dalsgaard, William J. Barry:
On the use of data-driven clustering technique for identification of poly- and mono-phonemes for four European languages. 121-124
Speaker Recognition
- Tomoko Matsui, Sadaoki Furui:
Speaker adaptation of tied-mixture-based phoneme models for text-prompted speaker recognition. 125-128 - Khaled T. Assaleh, Richard J. Mammone:
Robust cepstral features for speaker identification. 129-132 - Julian P. Eatock, John S. D. Mason:
A quantitative assessment of the relative speaker discriminating properties of phonemes. 133-136 - Shoji Hayakawa, Fumitada Itakura:
Text-dependent speaker recognition using the information in the higher frequency band. 137-140 - Michael A. Lund, Chung-Tshuy Lee, Chung-Chieh Lee, Robert W. Bossemeyer:
A distributed decision approach to speaker verification. 141-144 - Herbert Gish, Michael Schmidt, Angela Mielke:
A robust, segmental method for text independent speaker identification. 145-148 - Jean-Luc Le Floch, Claude Montacié, Marie-José Caraty:
Investigations on speaker characterization from Orphee system techniques. 149-152 - Jayant M. Naik, David M. Lubensky:
A hybrid HMM-MLP speaker verification algorithm for telephone speech. 153-156 - Jeffrey Sorensen, Michael I. Savic:
Hierarchical pattern classification for high performance text-independent speaker verification systems. 157-160 - Lynn Wilcox, Francine Chen, Don Kimber, Vijay Balasubramanian:
Segmentation of speech using speaker identification. 161-164 - Kevin R. Farrell, Richard J. Mammone:
Speaker identification using neural tree networks. 165-168 - Johan Schalkwyk, Etienne Barnard, Jeffrey R. Sachs:
Detecting an imposter in telephone speech. 169-172
Speech Coding Techniques
- Yoshinori Tanaka, Hisanari Kimura:
Low-bit-rate speech coding using a two-dimensional transform of residual signals and waveform interpolation. 173-176 - Victoria E. Sánchez, José L. Pérez-Córdoba, Juan M. López-Soler, Antonio J. Rubio:
Transform trellis coded quantization of speech using small frame sizes. 177-180 - Gao Yang, Henri Leich:
High-quality harmonic coding at very low bit rates. 181-184 - Jes Thyssen, Henrik Nielsen, Steffen Duus Hansen:
Non-linear short-term prediction in speech coding. 185-188 - Craig R. Watkins, Sam Crisafulli, Robert R. Bitmead, Robert Orsi:
Variable bit rate ADPCM via arithmetic coding. 189-192 - Roch Lefebvre, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul:
High quality coding of wideband audio signals using transform coded excitation (TCX). 193-196 - Keiichi Tokuda, Hidetoshi Matsumura, Takao Kobayashi, Satoshi Imai:
Speech coding based on adaptive mel-cepstral analysis. 197-200 - Stan A. McClellan, Jerry D. Gibson:
Spectral entropy: an alternative indicator for rate allocation? 201-204 - William R. Gardner, Bhaskar D. Rao:
Mixed-phase AR models for voiced speech and perceptual cost functions. 205-208 - Sun-Won Park:
Speech compression using ARMA model and wavelet transform. 209-212
Adaptation and Training Techniques
- Jesús Esteban Díaz Verdejo, José C. Segura, Pedro García-Teodoro, Antonio J. Rubio:
SLHMM: a continuous speech recognition system based on Alphanet-HMM. 213-216 - Ivica Rogina, Alex Waibel:
Learning state-dependent stream weights for multi-codebook HMM speech recognition systems. 217-220 - Qiang Huo, Chorkin Chan, Chin-Hui Lee:
Bayesian learning of the SCHMM parameters for speech recognition. 221-224 - Finn Tore Johansen, Magne Hallstein Johnsen:
Non-linear input transformations for discriminative HMMs. 225-228 - Yoshihiko Gotoh, Michael M. Hochberg, Harvey F. Silverman:
Using MAP estimated parameters to improve HMM speech recognition performance. 229-232 - Abdelhamid Mellouk, Patrick Gallinari:
Discriminative training for improved neural prediction systems. 233-236 - Barbara Wheatley, Kazuhiro Kondo, Wallace W. Anderson, Yeshwant K. Muthusamy:
An evaluation of cross-language adaptation for rapid HMM development in a new language. 237-240 - Shigeki Okawa, Tetsunori Kobayashi, Katsuhiko Shirai:
Automatic training of phoneme dictionary based on mutual information criterion. 241-244 - Tetsuo Kosaka, Shigeki Sagayama:
Tree-structured speaker clustering for fast speaker adaptation. 245-248 - Yasunaga Miyazawa, Jun-ichi Takami, Shigeki Sagayama, Shoichi Matsunaga:
All-phoneme ergodic hidden Markov network for unsupervised speaker adaptation. 249-252 - Yasuo Ariki, Keisuke Doi:
Phoneme recognition improvement by restricting training section in concatenated HMM training. 253-256
CELP Coding I
- Jörg-Martin Müller, Bertram Wächter:
A codec candidate for the GSM half rate speech channel. 257-260 - Michel Mauc, Geneviève Baudoin, Milan Jelinek:
Complexity reduction for FS-1016 with multistage search. 261-264 - Kumar Swaminathan, Kalyan Ganesan, Yi-Sheng Wang, Prabhat K. Gupta:
Speech and channel codec candidate for the half rate digital cellular channel. 265-268 - Kazunori Ozawa, Masahiro Serizawa, Toshiki Miyano, Toshiyuki Nomura:
M-LCELP speech coding at 4 kbps. 269-272 - Susan Yim, Dipanjan Sen, W. Harvey Holmes:
Comparison of ARMA modelling methods for low bit rate speech coding. 273-276 - Yoshiaki Asakawa, Hidetoshi Sekine, Makoto Takashima, Nobuyoshi Ishikawa, Toshiyuki Matsuda, Tom Okamoto, Ryujiro Muramatsu:
A 5.6 kb/s speech codec using a pulse codebook and improved Viterbi decoding. 277-280 - Luca Cellario, Daniele Sereno, Mario Giani, Peter Blöcher, Karl Hellwig:
A VR-CELP codec implementation for CDMA mobile communications. 281-284 - David E. Ray, Douglas J. Rahikka:
Reed-Solomon coding for CELP EDAC in land mobile radio. 285-288
Speaker and Language Recognition
- Kay M. Berkling, Takayuki Arai, Etienne Barnard:
Analysis of phoneme-based features for language identification. 289-292 - Lori Lamel, Jean-Luc Gauvain:
Language identification using phone-based acoustic likelihoods. 293-296 - Kung-Pu Li:
Automatic language identification using syllabic spectral features. 297-300 - Roger C. F. Tucker, Michael J. Carey, Eluned S. Parris:
Automatic language identification using sub-word models. 301-304 - Marc A. Zissman, Elliot Singer:
Automatic language identification of telephone speech messages using phoneme recognition and N-gram modeling. 305-308 - Chintana Griffin, Tomoko Matsui, Sakaoki Furui:
Distance measures for text-independent speaker recognition based on MAR model. 309-312 - Mark E. Forsyth, Mervyn A. Jack:
Discriminating semi-continuous HMM for speaker verification. 313-316 - Jonathan Foote, Harvey F. Silverman:
A model distance measure for talker clustering and identification. 317-320 - Lawrence G. Bahler, Jack E. Porter, Alan L. Higgins:
Improved voice identification using a nearest-neighbor distance measure. 321-324 - Chi-Shi Liu, Chin-Hui Lee, Biing-Hwang Juang, Aaron E. Rosenberg:
Speaker recognition based on minimum error discriminative training. 325-328 - Richard Ricart, Jim Cupples, Laurie Fenstermacher:
Speaker recognition in tactical communications. 329-332 - Yeshwant K. Muthusamy, Neena Jain, Ronald A. Cole:
Perceptual benchmarks for automatic language identification. 333-336
Speech Understanding and Language Modelling I
- Clifford J. Weinstein:
Demonstrations and applications of spoken language technology: highlights and perspectives from the 1993 ARPA Spoken Language Technology and Applications Day. 337-340 - Sunil Issar, Wayne H. Ward:
Unanswerable queries in a spontaneous speech task. 341-344 - Monika Woszczyna, Naomi Aoki-Waibel, Finn Dag Buø, Noah Coccaro, Keiko Horiguchi, Thomas Kemp, Alon Lavie, Arthur E. McNair, Thomas Polzin, Ivica Rogina, Carolyn P. Rosé, Tanja Schultz, Bernhard Suhm, Masaru Tomita, Alex Waibel:
JANUS 93: towards spontaneous speech translation. 345-348 - Douglas D. O'Shaughnessy:
Correcting complex false starts in spontaneous speech. 349-352 - Helmut Lucke:
Reducing the computational complexity for inferring stochastic context-free grammar rules from example text. 353-356 - Thomas Kuhn, Heinrich Niemann, Ernst Günter Schukat-Talamazzini:
Ergodic hidden Markov models and polygrams for language modeling. 357-360 - Jerry H. Wright, Gareth J. F. Jones, Harvey Lloyd-Thomas:
A robust language model incorporating a substring parser and extended n-grams. 361-364 - Finn Dag Buø, Thomas Polzin, Alex Waibel:
Learning complex output representations in connectionist parsing of spoken language. 365-368
Word Spotting and Other Topics
- Yoshiaki Itoh, Jiro Kiyama, Ryuichi Oka:
Sentence spotting applied to partial sentences and unknown words. 369-372 - Hervé Bourlard, Bart D'hoore, Jean-Marc Boite:
Optimizing recognition and rejection performance in wordspotting systems. 373-376 - David A. James, Steve J. Young:
A fast lattice-based approach to vocabulary independent wordspotting. 377-380 - Philippe Jeanrenaud, Man-Hung Siu, Jan Robin Rohlicek, Marie Meteer, Herbert Gish:
Spotting events in continuous speech. 381-384 - John W. McDonough, Kenney Ng, Philippe Jeanrenaud, Herbert Gish, Jan Robin Rohlicek:
Approaches to topic identification on the switchboard corpus. 385-388 - Richard Lippmann, Eric I. Chang, Charles R. Jankowski Jr.:
Wordspotter training using figure-of-merit back propagation. 389-392 - Rafid A. Sukkar:
Rejection for connected digit recognition based on GPD segmental discrimination. 393-396 - Takashi Otsuki, Akinori Ito, Shozo Makino, Teruhiko Otomo:
The performance prediction method on sentence recognition system using a finite state automaton. 397-400 - Kari Torkkola:
New ways to use LVQ-codebooks together with hidden Markov models. 401-404 - Jianmin Li, Ditang Fang:
Parallel distributed binary mapping models for speech recognition. 405-408
Robust Speech Recognition
- Juan Arturo Nolazco-Flores, Steve J. Young:
Continuous speech recognition in noise using spectral subtraction and HMM adaptation. 409-412 - Sahar E. Bou-Ghazale, John H. L. Hansen:
Duration and spectral based stress token generation for HMM speech recognition under stress. 413-416 - Leonardo Neumeyer, Mitchel Weintraub:
Probabilistic optimum filtering for robust speech recognition. 417-420 - Joachim Koehler, Nelson Morgan, Hynek Hermansky, Hans-Günter Hirsch, Grace Tong:
Integrating RASTA-PLP into speech recognition. 421-424 - Hidefumi Kobatake, Yousuke Matsunoo:
Degraded word recognition based on segmental signal-to-noise ratio weighting. 425-428 - Johan Smolders, Tom Claes, Gert Sablon, Dirk Van Compernolle:
On the importance of the microphone position for speech recognition in the car. 429-432 - Tasos Anastasakos, Francis Kubala, John Makhoul, Richard M. Schwartz:
Adaptation to new microphones using tied-mixture normalization. 433-436 - William C. Treurniet, Yifan Gong:
Noise independent speech recognition for a variety of noise types. 437-440 - Philip Lockwood, Patrice Alexandre:
Root adaptive homomorphic deconvolution schemes for speech recognition in noise. 441-444 - Mazin G. Rahim, Biing-Hwang Juang:
Signal bias removal for robust telephone based speech recognition in adverse environments. 445-448 - Yoichi Takebayashi, Hiroshi Kanazawa:
Adaptive noise immunity learning for word spotting. 449-452
Speech Analysis
- Gernot Kubin, W. Bastiaan Kleijn:
Time-scale modification of speech based on a nonlinear oscillator model. 453-456 - Hong Sub Choi, Seung Chan Bang, Souguil Ann:
A robust sequential parameter estimation for time-varying speech signal analysis. 457-460 - Naoto Iwahashi, Yoshinori Sagisaka:
Speech spectrum transformation by speaker interpolation. 461-464 - Suhuai Luo, Robin W. King:
A novel approach for classifying continuous speech into visible mouth-shape related classes. 465-468 - Hideyuki Mizuno, Masanobu Abe:
Voice conversion based on piecewise linear conversion rules of formant frequency and spectrum tilt. 469-472 - C. S. Ramalingam, Ramdas Kumaresan:
Voiced-speech analysis based on the residual interfering signal canceler (RISC) algorithm. 473-476