Stop the war!Остановите войну!
for scientists:
default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCIDICASSP 1991: Toronto, Ontario, Canada
- 1991 International Conference on Acoustics, Speech, and Signal Processing, ICASSP '91, Toronto, Ontario, Canada, May 14-17, 1991. IEEE Computer Society 1991, ISBN 0-7803-0003-3
- Schuyler Quackenbush:
A 7 kHz bandwidth, 32 kbps speech coder for ISDN. 1-4 - Yair Shoham:
On the use of direct vector quantization in LPC-based analysis-by-synthesis coding systems. 5-8 - Erik Ordentlich, Yair Shoham:
Low-delay code-excited linear-predictive coding of wideband speech at 32 kbps. 9-12 - Claude Laflamme, Jean-Pierre Adoul, Redwan Salami, Sarto Morissette, Philippe Mabilleau:
16 kbps wideband speech coding technique based on algebraic CELP. 13-16 - Guylain Roy, Peter Kabal:
Wideband CELP speech coding at 16 kbits/sec. 17-20 - Juin-Hwey Chen, Yen-Chun Lin, Richard V. Cox:
A fixed-point 16 kb/s LD-CELP algorithm. 21-24 - Majid Foodeei, Peter Kabal:
Low-delay CELP and tree coders: comparison and performance improvements. 25-28 - Rong Peng, Vladimir Cuperman:
Variable-rate low-delay analysis-by-synthesis speech coding at 8-16 kb/s. 29-32 - Alv I. Aarskog, Arild Nilsen, Ole Berg, Hans C. Guren:
A long-term predictive ADPCM coder with short-term prediction and vector quantization. 37-40 - Carl Rudolph Nassar, Mohammad Reza Soleymani:
Globally optimal trellis quantizers. 41-44 - Hidefumi Sawai:
Frequency-time-shift-invariant time-delay neural networks for robust continuous speech recognition. 45-48 - Nelson Morgan, Hynek Hermansky, Hervé Bourlard, Phil Kohn, Chuck Wooters
:
Continuous speech recognition using PLP analysis with multilayer perceptrons. 49-52 - Hidefumi Sawai:
TDNN-LR continuous speech recognition system using adaptive incremental TDNN training. 53-56 - Ken-ichi Iso, Takao Watanabe:
Large vocabulary speech recognition using neural prediction model. 57-60 - Joe Tebelskis, Alex Waibel, Bojan Petek, Otto Schmidbauer:
Continuous speech recognition using linked predictive neural networks. 61-64 - Gerhard Rigoll:
Information theory-based supervised learning methods for self-organizing maps in combination with hidden Markov modeling. 65-68 - Jhing-Fa Wang, Chung-Hsien Wu, Chaug-Ching Haung, Jau-Yien Lee:
Integrating neural nets and one-stage dynamic programming for speaker independent continuous Mandarin digit recognition. 69-72 - Frédéric Bimbot, Gérard Chollet, Jean-Pierre Tubach:
TDNNs for phonetic features extraction: a visual exploration. 73-76 - Sung Jun Lee, Ki Chul Kim, Hyunsoo Yoon, Jung Wan Cho:
Application of fully recurrent neural networks for speech recognition. 77-80 - Fabio Greco, Andrea Paoloni, Giacomo Ravaioli:
A recurrent time-delay neural network for improved phoneme recognition. 81-84 - Masami Nakamura, Shinichi Tamura, Shigeki Sagayama:
Phoneme recognition by phoneme filter neural networks. 85-88 - Jun-ichi Takami, Shigeki Sagayama:
A pairwise discriminant approach to robust phoneme recognition by time-delay neural networks. 89-92 - Luc Mathan, Laurent Miclet:
Rejection of extraneous input in speech recognition applications, using multi-layer perceptrons and the trace of HMMs. 93-96 - Younès Bennani, Nasser Chaourar, Patrick Gallinari, Abdelhamid Mellouk:
Validation of neural net architectures on speech recognition tasks. 97-100 - Jari Kangas:
Phoneme recognition using time-dependent versions of self-organizing maps. 101-104 - Patrick Haffner, Michael A. Franzini, Alex Waibel:
Integrating time alignment and neural networks for high performance continuous speech recognition. 105-108 - Michael M. Hochberg, Les T. Niles, J. T. Foote, Harvey F. Silverman:
Hidden Markov model/neural network training techniques for connected alphadigit speech recognition. 109-112 - Padma Ramesh, Shigeru Katagiri, Chin-Hui Lee:
A new connected word recognition algorithm based on HMM/LVQ segmentation and LVQ classification. 113-116 - Ulrich Bodenhausen, Alex Waibel:
Learning the architecture of neural networks for speech recognition. 117-120 - Yifan Gong, Jean-Paul Haton:
Non-linear vector interpolation by neural network for phoneme identification in continuous speech. 121-124 - Yasuhiro Komori:
Time-state neural networks (TSNN) for phoneme identification by considering temporal structure of phonemic features. 125-128 - Raymond L. Watrous:
Source decomposition of acoustic variability in a modular connectionist network. 129-131 - Peter Brauer, Per Hedelin, Dieter Huber, Petter Knagenhjelm:
Probability based optimization for network classifiers. 133-136 - Jan-Erik Strömberg, Jalel Zrida, Alf J. Isaksson:
Neural trees-using neural nets in a tree classifier structure. 137-140 - David M. Lubensky:
Word recognition using neural nets, multi-state Gaussian and k -nearest neighbor classifiers. 141-144 - Laurent Barbier, Gérard Chollet:
Robust speech parameters extraction for word recognition in noise using neural networks. 145-148 - Timothy R. Anderson:
Speaker independent phoneme recognition with an auditory model and a neural network: a comparison with traditional techniques. 149-152 - Yifan Gong, Ying Cheng, Jean-Paul Haton:
Neural network coupled with IIR sequential adapter for phoneme recognition in continuous speech. 153-156 - Eric R. Buhrke, Joseph L. LoCicero:
Speech recognition with neural networks and network fusion. 157-160 - Chin-Hui Lee, Egidio P. Giachin, Lawrence R. Rabiner, Roberto Pieraccini, Aaron E. Rosenberg:
Improved acoustic modeling for speaker independent large vocabulary continuous speech recognition. 161-164 - Yves Laprie:
Phonetic triplets in acoustic-phonetic decoding of continuous speech. 165-168 - Tomokazu Yamada, Toshiyuki Hanazawa, Takeshi Kawabata, Shoichi Matsunaga, Kiyohiro Shikano:
Phonetic typewriter based on phoneme source modeling. 169-172 - Lalit R. Bahl, Subrata K. Das, Peter DeSouza, M. Epstein, Robert L. Mercer, Bernard Mérialdo, David Nahamoo, Michael A. Picheny, J. Powell:
Automatic phonetic baseform determination. 173-176 - Lalit R. Bahl, Jerome R. Bellegarda, Peter V. de Souza, P. S. Gopalakrishnan, David Nahamoo, Michael A. Picheny:
A new class of fenonic Markov word models for large vocabulary continuous speech recognition. 177-180 - Lynn C. Wood, David J. B. Pearce, Frederic Novello:
Improved vocabulary-independent sub-word HMM modelling. 181-184 - Lalit R. Bahl, Peter V. de Souza, P. S. Gopalakrishnan, David Nahamoo, Michael A. Picheny:
Decision trees for phonological rules in continuous speech. 185-188 - Emilio Sanchis Arnal, Francisco Casacuberta, Isabel Galiano, Encarna Segarra:
Learning structural models of subword units through grammatical inference techniques. 189-192 - Li Deng, Kevin Erler:
Microstructural speech units and their HMM representation for discrete utterance speech recognition. 193-196 - Paul Dalsgaard, Ove Andersen, William J. Barry:
Multi-lingual label alignment using acoustic-phonetic features derived by neural-network technique. 197-200 - W. Bastiaan Kleijn:
Continuous representations in linear predictive coding. 201-204 - Ira A. Gerson, Mark A. Jasiuk:
Techniques for improving the performance of CELP type speech coders. 205-208 - Alain Le Guyader, Renaud J. Di Francesco, Claude Lamblin:
Derivation of efficient CELP coding algorithms using the Z-transform approach. 209-212 - Kazunori Ozawa, Toshiki Miyano:
4 kb/s improved CELP coder with efficient vector quantization. 213-216 - W. Granzow, Bishnu S. Atal, Kuldip K. Paliwal, Juergen Schroeter:
Speech coding at 4 kb/s and lower using single-pulse and stochastic models of LPC excitation. 217-220 - Mark Johnson, Tomohiko Taniguchi:
Low-complexity multi-mode VXC using multi-stage optimization and mode selection [speech coding]. 221-224 - Per Hedelin, Anders Bergström:
Amplitude quantization for CELP excitation signals. 225-228 - Kimio Miseki, Masami Akamine:
Adaptive bit-allocation between the pole-zero synthesis filter and excitation in CELP. 229-232 - Maurizio Copperi:
Efficient excitation modeling in a low bit-rate CELP coder. 233-236 - U. Kipper, Herbert Reininger, Dietrich Wolf:
Improved CELP coding using adaptive excitation codebooks. 237-240 - Tomohiko Taniguchi, Mark Johnson, Yasuji Ohta:
Pitch sharpening for perceptually improved CELP, and the sparse-delta codebook for reduced computation. 241-244 - Philipos C. Loizou, Andreas S. Spanias:
Vector quantization of transform components for speech coding at 1200 bps. 245-248 - John C. Hardwick, Jae S. Lim:
The application of the IMBE speech coder to mobile communications. 249-252 - Lorenzo Fissore, Pietro Laface, Giorgio Micca:
Comparison of discrete and continuous HMMs in a CSR task over the telephone. 253-256 - Yuqing Gao, Taiyi Huang, Zhiwei Lin, Bo Xu, Dongxin Xu:
A real-time Chinese speech recognition system with unlimited vocabulary. 257-260 - Kari Torkkola, Mikko Kokkonen:
Using the topology-preserving properties of SOFMs in speech recognition. 261-264 - Tatsuo Matsuoka, Kiyohiro Shikano:
Robust HMM phoneme modeling for different speaking styles. 265-268 - Kenji Kita, Wayne H. Ward:
Incorporating LR parsing into SPHINX. 269-272 - Shozo Makino, Akinori Ito, Mitsuru Endo, Ken'iti Kido:
A Japanese text dictation system based on phoneme recognition and a dependency grammar. 273-276 - John S. Bridle, L. Dodd:
An Alphanet approach to optimising input transformations for continuous speech recognition. 277-280 - Jeffrey N. Marcus, Victor W. Zue:
A variable duration acoustic segment HMM for hard-to-recognize words and phrases. 281-284 - Helen M. Meng, Victor W. Zue:
Signal representation comparison for phonetic classification. 285-288 - Vassilios Digalakis, Jan Robin Rohlicek, Mari Ostendorf:
A dynamical system approach to continuous speech recognition. 289-292 - Kathy L. Brown, V. Ralph Algazi:
Speech recognition using dynamic features of acoustic subword spectra. 293-296 - Helene Cerf-Danon, Marc El-Bèze:
Three different probabilistic language models: comparison and combination. 297-300 - Stephan Euler, Joachim Zinke:
Extending the vocabulary of a speaker independent recognition system. 301-304 - Ayman Asadi, Richard M. Schwartz, John Makhoul:
Automatic modeling for adding new words to a large-vocabulary continuous speech recognition system. 305-308 - Jay G. Wilpon, Laura G. Miller, P. Modi:
Improvements and applications for key word recognition using hidden Markov modeling techniques. 309-312 - David P. Morgan, Christopher L. Scofield, John E. Adcock:
Multiple neural network topologies applied to keyword spotting. 313-316 - Richard C. Rose, Eric I. Chang, Richard Lippmann:
Techniques for information retrieval from voice messages. 317-320 - Colin W. Wightman, Mari Ostendorf:
Automatic recognition of prosodic phrases. 321-324 - Ronald A. Cole, Mark A. Fanty, M. Gopalakrishnan, Rik D. T. Janssen:
Speaker-independent name retrieval from spellings using a database of 50000 names. 325-328 - Douglas B. Paul:
The Lincoln tied-mixture HMM continuous speech recognizer. 329-332 - Yunxin Zhao, Hisashi Wakita, Xinhua Zhuang:
An HMM based speaker-independent continuous speech recognition system with experiments on the TIMIT database. 333-336 - William S. Meisel, Mark T. Anikst, S. S. Pirzadeh, J. E. Schumacher, Matthew C. Soares, David J. Trawick:
The SSI large-vocabulary speaker-independent continuous speech recognition system. 337-340 - Vishwa Gupta, Matthew Lennig, Paul Mermelstein, Patrick Kenny, Franz Seitz, Douglas D. O'Shaughnessy:
Using phoneme duration and energy contour information to improve large vocabulary isolated-word recognition. 341-344 - Xuedong Huang, Kai-Fu Lee, Hsiao-Wuen Hon, Mei-Yuh Hwang:
Improved acoustic modeling with the SPHINX speech recognition system. 345-348 - Jay G. Wilpon, Chin-Hui Lee, Lawrence R. Rabiner:
Improvements in connected digit recognition using higher order spectral and energy features. 349-352 - Yeshwant K. Muthusamy, Ronald A. Cole, M. Gopalakrishnan:
A segment-based approach to automatic language identification. 353-356 - Horacio Franco, António Joaquim Serralheiro:
Training HMMs using a minimum recognition error approach. 357-360 - Jean-Claude Junqua:
The influence of psychoacoustic and psycholinguistic factors on listener judgments of intelligibility of normal and Lombard speech. 361-364 - Wayne H. Ward:
Understanding spontaneous speech: the Phoenix system. 365-367 - Steve Renals, David McKelvie, Fergus McInnes:
A comparative study of continuous speech recognition using neural networks and hidden Markov models. 369-372 - Dimitrios A. Gaganelis, Eleftherios D. Frangoulis:
A novel approach to speaker verification. 373-376 - Tomoko Matsui, Sadaoki Furui:
A text-independent speaker recognition method robust against utterance variations. 377-380 - Aaron E. Rosenberg, Chin-Hui Lee, Sedat Gokcen:
Connected word talker verification using whole word hidden Markov models. 381-384 - Younès Bennani, Patrick Gallinari:
On the use of TDNN-extracted features information in talker identification. 385-388 - Laszlo Rudasi, Stephen A. Zahorian:
Text-independent talker identification with neural networks. 389-392 - John Oglesby, John S. D. Mason:
Radial basis function networks for speaker recognition. 393-396 - Michael J. Carey, Eluned S. Parris, John S. Bridle:
A speaker verification system using alpha-nets. 397-400 - Richard C. Rose, J. Fitzmaurice, Edward M. Hofstetter, Douglas A. Reynolds:
Robust speaker identification in noisy environments using noise adaptive speaker models. 401-404 - Alan L. Higgins, Lawrence G. Bahler:
Text-independent speaker verification by discriminator counting. 405-408 - Jesse W. Fussell:
Automatic sex identification from short segments of speech. 409-412 - Eleftherios D. Frangoulis:
Isolated word recognition in noisy environment by vector quantization of the HMM and noise distributions. 413-416 - Petros Maragos:
Fractal aspects of speech signals: dimension and interpolation. 417-420 - Petros Maragos, Thomas F. Quatieri, James F. Kaiser:
Speech nonlinearities, modulations, and energy operators. 421-424 - Brent Townshend:
Nonlinear prediction of speech. 425-428 - Kuldip K. Paliwal, Man Mohan Sondhi:
Recognition of noisy speech using cumulant-based linear prediction analysis. 429-432 - Les E. Atlas, Patrick J. Loughlin, James W. Pitton:
Truly nonstationary techniques for the analysis and display of voiced speech. 433-436 - Ronald P. Cohn:
Robust voiced/unvoiced speech classification using a neural net. 437-440 - Thea B. Ghiselli-Crippa, Amro El-Jaroudi:
A fast neural net training algorithm and its application to voiced-unvoiced-silence classification of speech. 441-444 - Krishna S. Nathan, Harvey F. Silverman:
Classification of unvoiced stops based on formant transitions prior to release. 445-448 - Shubha Kadambe, Gloria Faye Boudreaux-Bartels:
A comparison of a wavelet functions for pitch detection of speech signals. 449-452 - Alex I. C. Monaghan, D. Robert Ladd:
Manipulating synthetic intonation for speaker characterisation. 453-456 - Shiufun Cheung, Jae S. Lim:
Combined multi-resolution (wideband/narrowband) spectrogram. 457-460 - David Rainton:
Speech recognition-a time-frequency subspace filtering based approach. 461-464 - David J. Pepper, Mark A. Clements:
On the phonetic structure of a large hidden Markov model. 465-468 - Neri Merhav, Yariv Ephraim:
Hidden Markov modeling using the most likely state sequence. 469-472 - Andrej Ljolje, Michael Riley:
Automatic segmentation and labeling of speech. 473-476 - Gabriele C. Hegerl, Harald Höge:
Numerical simulation of the glottal flow by a model based on the compressible Navier-Stokes equations. 477-480 - Sunil K. Gupta, Juergen Schroeter:
Low update rate articulatory analysis/synthesis of speech. 481-484 - Mazin G. Rahim, Willem B. Kleijn, Juergen Schroeter, Colin C. Goodyear:
Acoustic to articulatory parameter mapping using an assembly of neural networks. 485-488 - Tetsunori Kobayashi, Masayuki Yagyu, Katsuhiko Shirai:
Application of neural networks to articulatory motion estimation. 489-492