Journal: IEEE Transactions on Audio, Speech & Language Processing

Volume 3, Issue 6

433 -- 438Wolfgang G. Knecht, Markus E. Schenkel, George S. Moschytz. Neural network filters for speech enhancement
439 -- 448Søren Holdt Jensen, Per Christian Hansen, Steffen Duus Hansen, John Aasted Sorensen. Reduction of broad-band noise in speech by truncated QSVD
449 -- 457Qiguang Lin. A fast algorithm for computing the vocal-tract impulse response from the transfer function
458 -- 463Jianing Dai. Isolated word recognition using Markov chain models
464 -- 472Andrei Popescu 0004, Nicolas Moreau, Claude Lamblin. CELP coding using trellis-coded vector quantization of the excitation
473 -- 480Jinho Choi 0001. A fast determination of stochastic excitation without codebook search in CELP coder
481 -- 489Keiichi Tokuda, Takao Kobayashi, Satoshi Imai. Adaptive cepstral analysis of speech
490 -- 503Michael Paraskevas, John Mourjopoulos. A differential perceptual audio coding method with reduced bitrate requirements
504 -- 514Elias Bjarnason. Analysis of the filtered-X LMS algorithm
515 -- 519Thomas F. Quatieri, Robert B. Dunn, Thomas E. Hanna. A subband approach to time-scale expansion of complex acoustic signals

Volume 3, Issue 5

325 -- 333R. Smits, B. Yegnanarayana. Determination of instants of significant excitation in speech using group delay function
334 -- 345Qiang Huo, Chorkin Chan, Chin-Hui Lee. Bayesian adaptive learning of the parameters of hidden Markov model for speech recognition
346 -- 356Chafic Mokbel, Gérard Chollet. Automatic word recognition in cars
357 -- 366Vassilios Digalakis, Dimitry Rtischev, Leonardo Neumeyer. Speaker adaptation using constrained estimation of Gaussian mixtures
367 -- 381William R. Gardner, Bhaskar D. Rao. Theoretical analysis of the high-rate vector quantization of LPC parameters
382 -- 395Kuansan Wang, Shihab A. Shamma. Spectral shape analysis in the central auditory system
396 -- 406James M. Kates. Two-tone suppression in a cochlear model
407 -- 415John H. L. Hansen, Mark A. Clements. Source generator equalization and enhancement of spectral properties for robust speech recognition in noise and stress
415 -- 421John H. L. Hansen, Sahar E. Bou-Ghazale. Robust speech recognition training via duration and spectral-based stress token generation
421 -- 424Spiros Dimolitsas, Franklin L. Corcoran, Channasandra Ravishankar. Dependence of opinion scores on listening sets used in degradation category rating assessments
424 -- 428Futoshi Asano, Yôiti Suzuki, Toshio Sone. Weighted RLS adaptive beamformer with initial directivity

Volume 3, Issue 4

229 -- 241Sven Anderson, Diane Kewley-Port. Evaluation of speech recognizers for speech training applications
242 -- 250Alan V. McCree, Thomas P. Barnwell III. A mixed excitation LPC vocoder model for low bit rate speech coding
251 -- 266Yariv Ephraim, Harry L. van Trees. A signal subspace approach for speech enhancement
267 -- 278Simon J. Godsill, Peter J. W. Rayner. A Bayesian approach to the restoration of degraded audio signals
279 -- 285Nam Soo Kim, Chong Kwan Un. On estimating robust probability distribution in HMM-based speech recognition
286 -- 293Charles R. Jankowski Jr., Hoang-Doan H. Vo, Richard P. Lippmann. A comparison of signal processing front ends for automatic word recognition
294 -- 303Saeed Gazor, Yves Grenier. Criteria for positioning of sensors for a microphone array
304 -- 313Joseph A. Maxwell, Patrick M. Zurek. Reducing acoustic feedback in hearing aids
314 -- 317Miguel Angel Ferrer-Ballester, Aníbal R. Figueiras-Vidal. Efficient adaptive vector quantization of LPC parameters

Volume 3, Issue 3

157 -- 168Ravi P. Ramachandran, Man Mohan Sondhi, Nambi Seshadri, Bishnu S. Atal. A two codebook format for robust quantization of line spectral frequencies
169 -- 184John H. L. Hansen, Levent M. Arslan. Robust feature-estimation and objective quality assessment for noisy speech recognition using the Credit Card corpus
185 -- 192Philip A. Nelson, Felipe Orduna-Bustamante, Hareo Hamada. Inverse filter design and equalization zones in multichannel sound reproduction
193 -- 203Michael W. Hoffman, Kevin M. Buckley. Robust time-domain processing of broadband microphone array data
204 -- 209Tan Lee, P. C. Ching, Lai-Wan Chan, Y. H. Cheng, Brian Mak. Tone recognition of isolated Cantonese syllables
209 -- 213Donald G. Childers, Jose C. Principe, Y. T. Ting. Adaptive WRLS-VFF for speech analysis
213 -- 217Carl D. Mitchell, Mary P. Harper, Leah H. Jamieson. On the complexity of explicit duration HMM's
217 -- 222Sen M. Kuo, Min J. Ji. Development and analysis of an adaptive noise equalizer

Volume 3, Issue 2

117 -- 125Ravi P. Ramachandran, Mihailo S. Zilovic, Richard J. Mammone. A comparative study of robust linear predictive analysis methods with applications to speaker identification
126 -- 136Dennis R. Morgan. Slow asymptotic convergence of LMS acoustic echo cancelers
137 -- 141Craig R. Watkins, Robert R. Bitmead, Sam Crisafulli. Destabilization effects of adaptive quantization in ADPCM
141 -- 145Sin-Horng Chen, Wen-Yuan Chen. Generalized minimal distortion segmentation for ANN-based speech recognition
146 -- 150Sin-Horng Chen, Yih-Ru Wang. Tone recognition of continuous Mandarin speech based on neural networks

Volume 3, Issue 1

1 -- 21Ronald A. Cole, Lynette Hirschman, Les A. Atlas, Mary E. Beckman, Alan Biermann, Marcia Bush, Mark Clements, Jordan Cohen, Oscar Garcia, Brian A. Hanson, Hynek Hermansky, Steve Levinson, Kathy McKeown, Nelson Morgan, David G. Novick, Mari Ostendorf, Sharon Oviatt, Patti Price, Harvey Silverman, Judy Spitz, Alex Waibel, Clifford J. Weinstein, Steve Zahorian, Victor Zue. The challenge of spoken language systems: research directions for the nineties
22 -- 34Srinivas Nandkumar, John H. L. Hansen. Dual-channel iterative speech enhancement with constraints on an auditory-based spectrum
35 -- 39Venkatesh R. Chari, Carol Y. Espy-Wilson. Adaptive enhancement of Fourier spectra
40 -- 47Gao Yang 0002, Henri Leich, René Boite. Voiced speech coding at very low bit rates based on forward-backward waveform prediction
48 -- 58Ajay Ingle, Vhay A. Vaishampayan. DPCM system design for diversity systems with applications to packetized speech
59 -- 71Juin-Hwey Chen, Allen Gersho. Adaptive postfiltering for quality enhancement of coded speech
72 -- 83Douglas A. Reynolds, Richard C. Rose. Robust text-independent speaker identification using Gaussian mixture speaker models
84 -- 93Olivier Cappé, Jean Laroche. Evaluation of short-time spectral attenuation techniques for the restoration of musical recordings
94 -- 98Chih-Chung Kuo, Fu-Rong Jean, Hsiao-Chuan Wang. Speech classification embedded in adaptive codebook search for low bit-rate CELP coding
98 -- 104John H. L. Hansen, Levent M. Arslan. Markov model-based phoneme class partitioning for improved constrained iterative speech enhancement