dblp: IEEE Transactions on Speech and Audio Processing, Volume 5

William R. Gardner, Bhaskar D. Rao: Noncausal all-pole modeling of voiced speech. 1-10

Saeed Vaseghi, Ben P. Milner: Noise compensation methods for hidden Markov model speech recognition in adverse environments. 11-21

Jean-François Mari, Jean Paul Haton, Abdelaziz Kriouile: Automatic word recognition based on second-order hidden Markov models. 22-25

Yaxin Zhang, Roberto Togneri, Michael D. Alder: Phoneme-based vector quantization in a discrete HMM speech recognizer. 26-32

Yifan Gong: Stochastic trajectory modeling and sentence searching for continuous speech recognition. 33-44

Michael S. Brandstein, John E. Adcock, Harvey F. Silverman: A closed-form location estimator for use with room environment microphone arrays. 45-50

Davide Rocchesso, Julius O. Smith: Circulant and elliptic feedback delay networks for artificial reverberation. 51-63

Nathalie Delprat: Global frequency modulation laws extraction from the Gabor transform of a signal: a first study of the interacting components case. 64-71

Adoram Erell, David Burshtein: Noise adaptation of HMM speech recognition systems using tied-mixtures in the spectral domain. 72-74

Yumin Lee, Lin-Shan Lee, Chiu-yu Tseng: Isolated Mandarin syllable recognition with limited training data specially considering the effect of tones. 75-80

Javier Hernando, Climent Nadeu: Linear prediction of the one-sided autocorrelation sequence for noisy speech recognition. 80-84

Mihailo S. Zilovic, Ravi P. Ramachandran, Richard J. Mammone: A fast algorithm for finding the adaptive component weighted cepstrum for speaker recognition. 84-86

Ning Bi, Yingyong Qi: Application of speech conversion to alaryngeal speech enhancement. 97-105

Chung-Hsien Wu, Jau-Hung Chen: A novel two-level method for the computation of the LSP frequencies using a decimation-in-degree algorithm. 106-115

Jan S. Erkelens, Piet M. T. Broersen: Bias propagation in the autocorrelation method of linear prediction. 116-119

Stan McClellan, Jerry D. Gibson: Variable-rate CELP based on subband flatness. 120-130

Benjamim Tang, Albert Shen, Abeer Alwan, Gregory J. Pottie: A perceptually based embedded subband speech coder. 131-140

Yuval Cohen, Adoram Erell, Yuval Bistritz: Enhancement of connected words in an extremely noisy environment. 141-148

Nam Soo Kim, Chong Kwan Un: Frame-correlated hidden Markov model based on extended logarithmic pool. 149-160

Qiang Huo, Chin-Hui Lee: On-line adaptive learning of the continuous density hidden Markov model based on approximate recursive Bayes estimate. 161-172

Jean-Claude Junqua: SmarTspelL^TM: a multipass recognition system for name retrieval over the telephone. 173-182

Ben Shahshahani: A Markov random field approach to Bayesian speaker adaptation. 183-191

Garrison W. Greenwood: Training partially recurrent neural networks using evolutionary strategies. 192-194

Hsin-Min Wang, Tai-Hsuan Ho, Rung-Chiung Yang, Jia-Lin Shen, Bo-Ren Bai, Jenn-Chau Hong, Wei-Peng Chen, Tong-Lo Yu, Lin-Shan Lee: Complete recognition of continuous Mandarin speech for Chinese language with very large vocabulary using limited training data. 195-200

Andrew Choi: Real-time fundamental frequency estimation by least-square fitting. 201-205

Yuchang Cao, Sridha Sridharan, Miles Moody: Multichannel speech separation by eigendecomposition and its application to co-talker interference removal. 209-219

You-Li Chen, Bor-Sen Chen: Model-based multirate representation of speech signals and its application to recovery of missing speech packets. 220-231

Rathinavelu Chengalvarayan, Li Deng: Use of generalized dynamic feature parameters for speech recognition. 232-242

Rathinavelu Chengalvarayan, Li Deng: HMM-based speech recognition using state-dependent, discriminatively derived transforms on mel-warped DFT features. 243-256

Biing-Hwang Juang, Wu Hou, Chin-Hui Lee: Minimum classification error rate methods for speech recognition. 257-265

Mazin G. Rahim, Chin-Hui Lee, Biing-Hwang Juang: Discriminative utterance verification for connected digits recognition. 266-277

Michael A. Blommer, Gregory H. Wakefield: Pole-zero approximations for head-related transfer functions using a logarithmic error criterion. 278-287

Maurizio Omologo, Piergiorgio Svaizer: Use of the crosspower-spectrum phase in acoustic event location. 288-292

Nam Soo Kim, Chong Kwan Un: Statistically reliable deleted interpolation. 292-295

Michael A. Cohen, Stephen Grossberg: Parallel auditory filtering by sustained and transient channels separates coarticulated vowels and consonants. 301-318

Li Deng, Michael Aksmanovic: Speaker-independent phonetic classification using hidden Markov models with mixtures of trend functions. 319-324

Yoichi Haneda, Shoji Makino, Yutaka Kaneda: Multiple-point equalization of room transfer functions by using common acoustical poles. 325-333

Michael P. Nowak, Barry D. Van Veen: A constrained transform domain adaptive IIR filter structure for active noise control. 334-347

Sen M. Kuo, Mansour Tahernezhadi, Li Ji: Frequency-domain periodic active noise control and equalization. 348-358

Seymour Shlien: The modulated lapped transform, its time-varying forms, and its applications to audio coding standards. 359-366

Shude Zhang, Gordon Lockhart: Embedded RPE based on multistage coding. 367-371

Carlos Avendaño, Hynek Hermansky: On the effects of short-term spectrum smoothing in channel normalization. 372-374

Sen M. Kuo, Dipa Vijayan: A secondary path modeling technique for active noise control systems. 374-377

Scott D. Snyder, Nobuo Tanaka: Algorithm adaptation rate in active control: is faster necessarily better? 378-381

E. Bryan George, Mark J. T. Smith: Speech analysis/synthesis and modification using an analysis-by-synthesis/overlap-add sinusoidal model. 389-406

David P. Morgan, E. Bryan George, Leonard T. Lee, Steven M. Kay: Cochannel speaker separation by harmonic enhancement and suppression. 407-424

Sofiène Affes, Yves Grenier: A signal subspace tracking algorithm for microphone array processing of speech. 425-437

Ramesh R. Sarukkai, Dana H. Ballard: Word set probability boosting for improved spontaneous dialog recognition. 438-450

Brian Strope, Abeer Alwan: A model of dynamic auditory perception and its application to robust word recognition. 451-464

Thomas F. Quatieri, Thomas E. Hanna, Gerald C. O'Leary: AM-FM separation using auditory-motivated filters. 465-480

Pierre A. Laurent: Expression of spectral distortion using line spectrum frequencies. 481-484

Régine Le Bouquin-Jeannès, Ahmad Akbari Azirani, Gérard Faucon: Enhancement of speech degraded by coherent and incoherent noise using a cross-spectral estimator. 484-487

Dionysis E. Tsoukalas, J. Mourjopoulos, George Kokkinakis: Speech enhancement based on audible noise suppression. 497-514

Sanyogita Shamsunder, Georgios B. Giannakis: Multichannel blind signal separation and reconstruction. 515-528

Joseph G. Desloge, William M. Rabinowitz, Patrick M. Zurek: Microphone-array hearing aids with binaural output .I. Fixed-processing systems. 529-542

Daniel P. Welker, Julie E. Greenberg, Joseph G. Desloge, Patrick M. Zurek: Microphone-array hearing aids with binaural output. II. A two-microphone adaptive system. 543-551

Jozsef Vass, Yunxin Zhao, Xinhua Zhuang: Adaptive forward-backward quantizer for low bit rate high-quality speech coding. 552-557

Michael W. Macon, Mark A. Clements: Sinusoidal modeling and modification of unvoiced speech. 557-560

Larry P. Heck, Koorosh Naghshineh: Optimal interpolation of structural responses for active noise control. 560-565

Sen M. Kuo, Mansour Tahernezhadi, Li Ji: Correction To "Frequency-domain Periodic Active Noise Control And Equalization". 565

IEEE Transactions on Speech and Audio Processing, Volume 5

Volume 5, Number 1, January 1997

Volume 5, Number 2, March 1997

Volume 5, Number 3, May 1997

Volume 5, Number 4, July 1997

Volume 5, Number 5, September 1997

Volume 5, Number 6, November 1997