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ORCIDICASSP 1992: San Francisco, California, USA
- 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '92, San Francisco, California, USA, March 23-26, 1992. IEEE Computer Society 1992, ISBN 0-7803-0532-9
Volume 1
- Richard M. Schwartz, Steve Austin, Francis Kubala, John Makhoul, Long Nguyen, Paul Placeway, George Zavaliagkos:
New uses for the N-Best sentence hypotheses within the BYBLOS speech recognition system. 1-4 - Ronald Rosenfeld
, Xuedong Huang, Merrick L. Furst:
Exploiting correlations among competing models with application to large vocabulary speech recognition. 5-8 - Hermann Ney, Reinhold Häb-Umbach, Bach-Hiep Tran, Martin Oerder:
Improvements in beam search for 10000-word continuous speech recognition. 9-12 - Reinhold Häb-Umbach, Hermann Ney:
Linear discriminant analysis for improved large vocabulary continuous speech recognition. 13-16 - Lalit R. Bahl, Peter V. de Souza, P. S. Gopalakrishnan, David Nahamoo, Michael A. Picheny:
A fast match for continuous speech recognition using allophonic models. 17-20 - Katunobu Itou, Satoru Hayamizu, Hozumi Tanaka:
Continuous speech recognition by context-dependent phonetic HMM and an efficient algorithm for finding N-Best sentence hypotheses. 21-24 - Douglas B. Paul:
An efficient A* stack decoder algorithm for continuous speech recognition with a stochastic language model. 25-28 - Kiyoshi Asai, Satoru Hayamizu, Ken'ichi Handa:
Dividing the distributions of HMM and linear interpolation in speech recognition. 29-32 - Mei-Yuh Hwang, Xuedong Huang:
Subphonetic modeling with Markov states-Senone. 33-36 - Tomokazu Yamada, Shoichi Matsunaga, Kiyohiro Shikano:
Japanese dictation system using character source modeling. 37-40 - R. Soheili, Ahmet M. Kondoz, Barry G. Evans:
Techniques for improving the quality of LD-CELP coders at 8 kb/s. 41-44 - Jey-Hsin Yao, John J. Shynk, Allen Gersho:
Low-delay VXC at 8 kbit/s with interframe coding. 45-48 - Y. J. Liu:
On reducing the bit rate of a CELP-based speech coder. 49-52 - Michel Mauc, Geneviève Baudoin:
Reduced complexity CELP coder. 53-56 - Per Hedelin:
A multi-stage perspective on CELP speech coding. 57-60 - Nicolas Moreau, Przemyslaw Dymarski:
Successive orthogonalizations in the multistage CELP coder. 61-64 - Xiongwei Zhang, Chen Xianzhi:
A new excitation model for LPC vocoder at 2.4 kb/s. 65-68 - Juin-Hwey Chen, Nikil Jayant, Richard V. Cox:
Improving the performance of the 16 kb/s LD-CELP speech coder. 69-72 - Marco Fratti, Gian Antonio Mian, Giuseppe Riccardi:
On the effectiveness of parameter reoptimization in multipulse based coders. 73-76 - Sam Crisafulli, James D. Mills, Robert R. Bitmead:
Kalman filtering techniques in speech coding. 77-80 - Benoit Sylvestre, Peter Kabal:
Time-scale modification of speech using an incremental time-frequency approach with waveform structure compensation. 81-84 - Toshio Irino, Hideki Kawahara:
Signal reconstruction from modified wavelet transform-An application to auditory signal processing. 85-88 - V. Ramasubramanian, Kuldip K. Paliwal:
An efficient approximation-elimination algorithm for fast-nearest-neighbour search (speech coding). 89-92 - V. Ralph Algazi, David H. Irvine, C. Caldwell, Michael J. Ready, Kathy L. Brown, Sang Chung:
Speech coding by the efficient transformation of the spectral envelope of subwords. 93-96 - Chih-Chung Kuo, Fu-Rong Jean, Hsiao-Chuan Wang:
Low bit-rate quantization of LSP parameters using two-dimensional differential coding. 97-100 - Tzung Kwang Wang, John Foster, Shahryar S. Ardalan:
Adaptive vector quantization for waveform coding. 101-104 - Bhaskar Bhattacharya, Wilf P. LeBlanc, Samy A. Mahmoud, Vladimir Cuperman:
Tree searched multi-stage vector quantization of LPC parameters for 4 kb/s speech coding. 105-108 - Mitsuru Nakai, Hiroshi Shimodaira, Masayuki Kimura:
A fast VQ codebook design algorithm for a large number of data. 109-112 - Deepen Sinha, Ahmed H. Tewfik:
Synthesis/coding of audio signals using optimized wavelets. 113-116 - Cheung-Fat Chan, Kwok-Wah Law:
Thinned lattice filter for LPC analysis. 117-120 - Hynek Hermansky, Nelson Morgan, Aruna Bayya, Phil Kohn:
RASTA-PLP speech analysis technique. 121-124 - Nancy Hubing, Kyung Y. Yoo:
Exploiting recursive parameter trajectories in speech analysis. 125-128 - Mithat C. Dogan, Jerry M. Mendel:
Real-time robust pitch detector. 129-132 - Asunción Moreno, José A. R. Fonollosa:
Pitch determination of noisy speech using higher order statistics. 133-136 - Toshiaki Fukada, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai:
An adaptive algorithm for mel-cepstral analysis of speech. 137-140 - Shie Qian, Dapang Chen, Kebo Chen:
Signal approximation via data-adaptive normalized Gaussian functions and its applications for speech processing. 141-144 - Hélène Valbret, Éric Moulines, Jean-Pierre Tubach:
Voice transformation using PSOLA technique. 145-148 - Keikichi Hirose, Hiroya Fujisaki, Shigenobu Seto:
A scheme for pitch extraction of speech using autocorrelation function with frame length proportional to the time lag. 149-152 - Claude Montacié, Paul Deléglise, Frédéric Bimbot, Marie-José Caraty:
Cinematic techniques for speech processing: temporal decomposition and multivariate linear prediction. 153-156 - Giulio Maltese, Federico Mancini:
An automatic technique to include grammatical and morphological information in a trigram-based statistical language model. 157-160 - Ute Essen, Volker Steinbiss:
Cooccurrence smoothing for stochastic language modeling. 161-164 - Shoichi Matsunaga, Tomokazu Yamada, Kiyohiro Shikano:
Task adaptation in stochastic language models for continuous speech recognition. 165-168 - Jeremy H. Wright, Gareth J. F. Jones, E. N. Wrigley:
Hybrid grammar-bigram speech recognition system with first-order dependence model. 169-172 - Egidio P. Giachin:
Automatic training of stochastic finite-state language models for speech understanding. 173-176 - Julian Kupiec:
Hidden Markov estimation for unrestricted stochastic context-free grammars. 177-180 - David Goddeau, Victor Zue:
Integrating probabilistic LR parsing into speech understanding systems. 181-184 - Keh-Yih Su, Tung-Hui Chiang, Yi-Chung Lin:
A unified framework to incorporate speech and language information in spoken language processing. 185-188 - Stephanie Seneff:
Robust parsing for spoken language systems. 189-192 - Roberto Pieraccini, Evelyne Tzoukermann, Zakhar Gorelov, Jean-Luc Gauvain, Esther Levin, Chin-Hui Lee, Jay G. Wilpon:
A speech understanding system based on statistical representation of semantics. 193-196 - Hiroyuki Tsuboi, Yoichi Takebayashi:
A real-time task-oriented speech understanding system using keyword-spotting. 197-200 - Laura G. Miller, Allen L. Gorin:
A structured network architecture for adaptive language acquisition. 201-204 - Ajay N. Jain, Alex Waibel, David S. Touretzky:
PARSEC: a structured connectionist parsing system for spoken language. 205-208 - Louise Osterholtz, Charles Augustine, Arthur E. McNair, Ivica Rogina, Hiroaki Saito, Tilo Sloboda, Joe Tebelskis, Alex Waibel:
Testing generality in JANUS: a multi-lingual speech translation system. 209-212 - David B. Roe, Fernando C. N. Pereira, Richard Sproat, Michael D. Riley, Pedro J. Moreno, Alejandro Macarrón:
Efficient grammar processing for a spoken language translation system. 213-216 - Hiroshi Shimodaira, Masayuki Kimura:
Accent phrase segmentation using pitch pattern clustering. 217-220 - Colin W. Wightman, Mari Ostendorf:
Automatic recognition of intonational features. 221-224 - Jim L. Hieronymus, David McKelvie, Fergus R. McInnes:
Use of acoustic sentence level and lexical stress in HSMM speech recognition. 225-227 - Francine R. Chen, Margaret Withgott:
The use of emphasis to automatically summarize a spoken discourse. 229-232 - Mark J. F. Gales, Steve J. Young:
An improved approach to the hidden Markov model decomposition of speech and noise. 233-236 - Beth A. Carlson, Mark A. Clements:
Speech recognition in noise using a projection-based likelihood measure for mixture density HMM's. 237-240 - D. Charles Bateman, David Bye, M. J. Hunt:
Spectral contrast normalization and other techniques for speech recognition in noise. 241-244 - Stefan Dobler, Peter Meyer, Hans-Wilhelm Rühl:
A robust connected-words recognizer. 245-248 - Lorenzo Fissore, Pietro Laface, P. Ruscitti:
HMM modeling for speaker independent voice dialing in car environment. 249-252 - Michelle Q. Wang, Steve J. Young:
Speech recognition using hidden Markov model decomposition and a general background speech model. 253-256 - Fu-Hua Liu, Alejandro Acero, Richard M. Stern:
Efficient joint compensation of speech for the effects of additive noise and linear filtering. 257-260 - Solomon Lerner, Baruch Mazor:
Telephone channel normalization for automatic speech recognition. 261-264 - Philip Lockwood, Jérôme Boudy, Marc Blanchet:
Non-linear spectral subtraction (NSS) and hidden Markov models for robust speech recognition in car noise environments. 265-268 - Brian Mak, Jean-Claude Junqua, Ben Reaves:
A robust speech/non-speech detection algorithm using time and frequency-based features. 269-272 - Harald Singer, Shigeki Sagayama:
Pitch dependent phone modelling for HMM based speech recognition. 273-276 - Alan V. Oppenheim, Ehud Weinstein, Kambiz C. Zangi, Meir Feder, D. Gauger:
Single sensor active noise cancellation based on the EM algorithm. 277-280 - Stephen Oh, Vishu Viswanathan, Panos E. Papamichalis:
Hands-free voice communication in an automobile with a microphone array. 281-284 - Kevin R. Farrell, Richard J. Mammone, James L. Flanagan:
Beamforming microphone arrays for speech enhancement. 285-288 - Yariv Ephraim:
Speech enhancement using state dependent dynamical system model. 289-292 - Victor C. Soon, Lang Tong, Y. F. Huang, R. Liu:
A wideband blind identification approach to speech acquisition using a microphone array. 293-296 - Srinivas Nandkumar, John H. L. Hansen:
Dual-channel speech enhancement with auditory spectrum based constraints. 297-300 - Biing-Hwang Juang, Kuldip K. Paliwal:
Vector equalization in hidden Markov models for noisy speech recognition. 301-304 - Yves Grenier:
A microphone array for car environments. 305-308 - Ki Yong Lee, Byung-Gook Lee, Iickho Song, Souguil Ann:
Robust estimation of AR parameters and its application for speech enhancement. 309-312 - Richard L. Zinser, Steven R. Koch:
CELP coding at 4.0 kb/sec and below: improvements to FS-1016. 313-316 - Daniel Lin:
Ultra-fast CELP coding using deterministic multi-codebook innovations. 317-320 - Toshiki Miyano, Masahiro Serizawa, Junichi Takizawa, Shigeji Ikeda, Kazunori Ozawa:
Improved 4.8 kb/s CELP coding using two-stage vector quantization with multiple candidates (LCELP). 321-324 - Tomohiko Taniguchi, Yoshinori Tanaka, Yasuji Ohta:
Tree-structured delta codebook for an efficient implementation of CELP. 325-328 - M. Delprat, C. Gruet, F. Dervaux, C. Baroux:
Fractional excitation and other efficient transformed codebooks for CELP coding of speech. 329-332 - Peter Lupini, Vladimir Cuperman:
Excitation modeling based on speech residual information. 333-336 - W. Bastiaan Kleijn, Ravi Prakash Ramachandran, Peter Kroon:
Generalized analysis-by-synthesis coding and its application to pitch prediction. 337-340 - Domingo Docampo, Victoria Abreu-Sernández, Fernando Pérez-Cruz, Francisco González:
A deconvolution-based efficient method for generating the excitation in linear predictive speech coding. 341-344 - Adil Benyassine, Hüseyin Abut:
Mixture excitations and finite-state CELP speech coders. 345-348 - Shihua Wang, Allen Gersho:
Improved phonetically-segmented vector excitation coding at 3.4 kb/s. 349-352 - Takeshi Kawabata:
Predictor codebooks for speaker-independent speech recognition. 353-356 - Jun Mo Koo, Hwang Soo Lee, C. K. Un:
An improved VQ codebook design algorithm for HMM. 357-360 - Christian Wellekens:
Mixture density estimators in Viterbi training. 361-364 - Tatsuya Kawahara, Shuji Doshita:
HMM based on pair-wise Bayes classifiers. 365-368 - M. Taylor, Frédéric Bimbot:
Temporal decomposition for the initialization of a HMM isolated word-recognizer. 369-372 - Zhi-ping Hu, Satoshi Imai:
Modeling improvement of the continuous hidden Markov model for speech recognition. 373-376 - Fabio Brugnara, Renato De Mori, Diego Giuliani, Maurizio Omologo:
A family of parallel hidden Markov models. 377-380 - Padma Ramesh, Jay G. Wilpon:
Modeling state durations in hidden Markov models for automatic speech recognition. 381-384 - Tsuneo Nitta, Jun'ichi Iwasaki, Yasuyuki Masai, Hiroshi Matsu'ura:
Representing dynamic features of phonetic segment in an orthogonalized codebook of HMM based speech recognition system. 385-388 - Mari Ostendorf, Ibrahim Bechwati, Owen Kimball:
Context modeling with the stochastic segment model. 389-392 - Gerhard Rigoll:
Unsupervised information theory-based training algorithms for multilayer neural networks. 393-396 - Bojan Petek, Joe Tebelskis:
Context-dependent hidden control neutral network architecture for continuous speech recognition. 397-400 - Eric R. Buhrke, Joseph L. LoCicero:
Fast learning for multi-layer perceptrons using statistical techniques. 401-404 - Yasuhiro Komori:
A neural fuzzy training approach for continuous speech recognition improvement. 405-408 - Satoru Nakamura, Hidefumi Sawai, Masahide Sugiyama:
Speaker-independent phoneme recognition using large-scale neural networks. 409-412 - E. L. Richards:
A multi-task neural network approach to speech recognition. 413-416 - Erik McDermott, Shigeru Katagiri:
Prototype-based discriminative training for various speech units. 417-420 - Christian Dugast, Laurence Devillers:
Incorporating acoustic-phonetic knowledge in hybrid TDNN/HMM frameworks. 421-424 - Les T. Niles, Lynn D. Wilcox, Marcia A. Bush:
Error-correcting training for phoneme spotting. 425-428 - Huaiyu Zeng, Tiecheng Yu:
Parallel sequential running neural network and its application to automatic speech recognition. 429-432 - Keiji Fukuzawa, Yasuhiro Komori, Hidefumi Sawai, Masahide Sugiyama:
A segment-based speaker adaptation neural network applied to continuous speech recognition. 433-436 - Burhan F. Necioglu, Mari Ostendorf, Jan Robin Rohlicek:
A Bayesian approach to speaker adaptation for the stochastic segment model. 437-440 - Otto Schmidbauer, Joe Tebelskis:
An LVQ based reference model for speaker-adaptive speech recognition. 441-444 - Jerome R. Bellegarda, Peter V. de Souza, Arthur Nádas, David Nahamoo, Michael A. Picheny, Lalit R. Bahl:
Robust speaker adaptation using a piecewise linear acoustic mapping. 445-448 - Hiroshi Matsukoto, Hirowo Inoue:
A piecewise linear spectral mapping for supervised speaker adaptation. 449-452 - Michael Witbrock, Patrick Haffner:
Rapid connectionist speaker adaptation. 453-456 - Tetsunori Kobayashi, Y. Uchiyama, J. Osada, Katsuhiko Shirai:
Speaker adaptive phoneme recognition based on feature mapping from spectral domain to probabilistic domain. 457-460 - Fritz Class, Alfred Kaltenmeier, Peter Regel-Brietzmann, Karl Trottler:
Fast speaker adaptation combined with soft vector quantization in an HMM speech recognition system. 461-464 - Xuedong Huang:
Speaker normalization for speech recognition. 465-468 - Masakatsu Hoshimi, Maki Miyata, Shoji Hiraoka, Katsuyuki Niyada:
Speaker independent speech recognition method using training speech from a small number of speakers. 469-472 - Wu Chou, Biing-Hwang Juang, Chin-Hui Lee:
Segmental GPD training of HMM based speech recognizer. 473-476 - Lalit R. Bahl, Peter V. de Souza, David Nahamoo, Michael A. Picheny, Salim Roukos:
Adaptation of large vocabulary recognition system parameters. 477-480 - Jean-Luc Gauvain, Chin-Hui Lee:
Improved acoustic modeling with Bayesian learning. 481-484 - Hsiao-Wuen Hon, Kai-Fu Lee:
Vocabulary learning and environment normalization in vocabulary-independent speech recognition. 485-488