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
  
• F. Liu, Y. Lee, L. 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 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, C. H. Chang, Y. H. Lin, Y. Lee, S. L. Tu, S. H. Hsieh, C. H. 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
  
• Hinrich 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
  
• P.-C. Chang, S.-H. Chen, B.-H. 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

• W. P. LeBlanc, B. Bhattacharya, S. 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
  

Last update Sat May 25 21:08:10 2013 CET by the DBLP Team — Data released under the ODC-BY 1.0 license — See also our legal information page