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IEEE Transactions on Speech and Audio Processing, Volume 1
Volume 1, Number 1, January 1993
- Kuldip K. Paliwal

, Bishnu S. Atal
:
Efficient vector quantization of LPC parameters at 24 bits/frame. 3-14 - Frank K. Soong, Biing-Hwang Juang:

Optimal quantization of LSP parameters. 15-24 - Yunus Hussain, Nariman Farvardin:

Variable-rate finite-state vector quantization and applications to speech and image coding. 25-38 - Jean-Claude Junqua, Hisashi Wakita, Hynek Hermansky:

Evaluation and optimization of perceptually-based ASR front-end. 39-48 - Patrick Kenny, Rene Hollan, Vishwa Gupta, Matthew Lennig, Paul Mermelstein, Douglas D. O'Shaughnessy:

A*-admissible heuristics for rapid lexical access. 49-58 - Lalit R. Bahl, Steven V. De Gennaro, P. S. Gopalakrishnan, Robert L. Mercer:

A fast approximate acoustic match for large vocabulary speech recognition. 59-67 - Adoram Erell, Mitchel Weintraub:

Filterbank-energy estimation using mixture and Markov models for recognition of noisy speech. 68-76 - Lalit R. Bahl, Peter F. Brown, Peter V. de Souza, Robert L. Mercer:

Estimating hidden Markov model parameters so as to maximize speech recognition accuracy. 77-83 - Adoram Erell, Mitchel Weintraub:

Energy conditioned spectral estimation for recognition of noisy speech. 84-89 - Neri Merhav, Chin-Hui Lee:

A minimax classification approach with application to robust speech recognition. 90-100 - Shoji Makino

, Yutaka Kaneda, Nobuo Koizumi:
Exponentially weighted stepsize NLMS adaptive filter based on the statistics of a room impulse response. 101-108 - G. T. H. Wright, F. J. Owens:

An optimized multirate sampling technique for the dynamic variation of vocal tract length in the Kelly-Lochbaum speech synthesis model. 109-113 - Fu-hua Liu, Yumin Lee, Lin-Shan Lee:

A direct-concatenation approach to train hidden Markov models to recognize the highly confusing Mandarin syllables with very limited training data. 113-119
Volume 1, Number 2, April 1993
- Roy C. Snell, Fausto Milinazzo:

Formant location from LPC analysis data. 129-134 - Pao-Chung Chang, Biing-Hwang Juang:

Discriminative training of dynamic programming based speech recognizers. 135-143 - Bryan Beresford-Smith, Jens Breckling, Heiko Schröder, Peter F. Summons:

Systolic codebook generation [speech recognition]. 144-149 - Xuedong Huang, Kai-Fu Lee:

On speaker-independent, speaker-dependent, and speaker-adaptive speech recognition. 150-157 - Lin-Shan Lee, Chiu-yu Tseng, Hung-Yan Gu, Fu-hua Liu, Robert Chen-Hao Chang

, Yueh-hong Lin, Yumin Lee, Shih-Lung Tu, Shew-Heng Hsieh, Chian-hung Chen:
Golden Mandarin (I)-A real-time Mandarin speech dictation machine for Chinese language with very large vocabulary. 158-179 - V. Ralph Algazi, Kathy L. Brown, Michael J. Ready, David H. Irvine, Christie L. Cadwell, Sang Chung:

Transform representation of the spectra of acoustic speech segments with applications. I. General approach and application to speech recognition. 180-195 - Kevin T. Malone, Thomas R. Fischer:

Trellis-searched adaptive predictive coding of speech. 196-206 - Yan Ming Cheng, Douglas D. O'Shaughnessy:

On 450-600 b/s natural sounding speech coding. 207-220 - Lee-Feng Chien, Keh-Jiann Chen, Lin-Shan Lee:

A best-first language processing model integrating the unification grammar and Markov language model for speech recognition applications. 221-240 - Søren Laugesen, Stephen J. Elliott:

Multichannel active control of random noise in a small reverberant room. 241-249 - Yingyong Qi, Bobby R. Hunt:

Voiced-unvoiced-silence classifications of speech using hybrid features and a network classifier. 250-255 - Bernard Mérialdo:

On the locality of the forward-backward algorithm [speech recognition]. 255-257 - H. Schütze:

Convergence of acoustic echo cancellers for hands-free telephones operating under feedback conditions. 257-260
Volume 1, Number 3, July 1993
- Il-Taek Lim, Byeong Gi Lee:

Lossless pole-zero modeling of speech signals. 269-276 - V. Ralph Algazi, Kathy L. Brown, Michael J. Ready, David H. Irvine, Christie L. Cadwell, Sang Chung:

Transform representation of the spectra of acoustic speech segments with applications. II. Speech analysis, synthesis, and coding. 277-286 - Lin-Shan Lee, Chiu-yu Tseng, Ching Jiang Hsieh:

Improved tone concatenation rules in a formant-based Chinese text-to-speech system. 287-294 - Nurgun Erdol, Claude Castelluccia, Ali Zilouchian:

Recovery of missing speech packets using the short-time energy and zero-crossing measurements. 295-303 - Kevin T. Malone, Thomas R. Fischer:

Enumeration and trellis-searched coding schemes for speech LSP parameters. 304-314 - M. Elshafei-Ahmed, M. I. Al-Suwaiyel:

Fast methods for code search in CELP. 315-325 - Pao-Chung Chang, S.-H. Chen, Biing-Hwang Juang:

Discriminative analysis of distortion sequences in speech recognition. 326-333 - Lalit R. Bahl, Jerome R. Bellegarda, Peter V. de Souza, P. S. Gopalakrishnan, David Nahamoo, Michael A. Picheny:

Multonic Markov word models for large vocabulary continuous speech recognition. 334-344 - Yunxin Zhao:

A speaker-independent continuous speech recognition system using continuous mixture Gaussian density HMM of phoneme-sized units. 345-361 - Ben Pinkowski:

LPC spectral moments for clustering acoustic transients. 362-368
Volume 1, Number 4, October 1993
- Wilf P. LeBlanc, B. Bhattacharya, Samy A. Mahmoud, Vladimir Cuperman:

Efficient search and design procedures for robust multi-stage VQ of LPC parameters for 4 kb/s speech coding. 373-385 - W. Bastiaan Kleijn

:
Encoding speech using prototype waveforms. 386-399 - Carl R. Nassar, M. Reza Soleymani:

Codebook design for trellis quantization using simulated annealing. 400-404 - Ehud Weinstein, Meir Feder, Alan V. Oppenheim:

Multi-channel signal separation by decorrelation. 405-413 - Mei-Yuh Hwang, Xuedong Huang:

Shared-distribution hidden Markov models for speech recognition. 414-420 - Shigeru Katagiri, Chin-Hui Lee:

A new hybrid algorithm for speech recognition based on HMM segmentation and learning vector quantization. 421-430 - Vassilios Digalakis

, Jan Robin Rohlicek, Mari Ostendorf:
ML estimation of a stochastic linear system with the EM algorithm and its application to speech recognition. 431-442 - Lalit R. Bahl, Peter F. Brown, Peter V. de Souza, Robert L. Mercer, Michael A. Picheny:

A method for the construction of acoustic Markov models for words. 443-452 - James M. Kates:

Accurate tuning curves in a cochlear model. 453-462 - Marco Fratti, Gian Antonio Mian, Giuseppe Riccardi:

An approach to parameter reoptimization in multipulse-based coders. 463-465 - John R. Deller Jr., R. K. Snider:

Reducing redundant computation in HMM evaluation. 465-471 - Li Deng:

A stochastic model of speech incorporating hierarchical nonstationarity. 471-474

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