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IEEE Transactions on Speech and Audio Processing, Volume 3
Volume 3, Number 1, January 1995
- Ronald A. Cole, Lynette Hirschman, Les E. Atlas, Mary E. Beckman, Alan Biermann, Marcia A. Bush, Mark Clements, Jordan Cohen, Oscar Garcia, Brian A. Hanson, Hynek Hermansky
, Steve Levinson, Kathy McKeown, Nelson Morgan, David G. Novick, Mari Ostendorf, Sharon L. Oviatt, Patti Price, Harvey F. Silverman, Judy Spitz, Alex Waibel, Clifford J. Weinstein, Stephen A. Zahorian, Victor Zue:
The challenge of spoken language systems: research directions for the nineties. 1-21 - Srinivas Nandkumar, John H. L. Hansen:
Dual-channel iterative speech enhancement with constraints on an auditory-based spectrum. 22-34 - Venkatesh R. Chari, Carol Y. Espy-Wilson:
Adaptive enhancement of Fourier spectra. 35-39 - Gao Yang, Henri Leich, René Boite:
Voiced speech coding at very low bit rates based on forward-backward waveform prediction. 40-47 - Ajay Ingle, Vinay A. Vaishampayan
DPCM system design for diversity systems with applications to packetized speech. 48-58 - Juin-Hwey Chen, Allen Gersho:
Adaptive postfiltering for quality enhancement of coded speech. 59-71 - Douglas A. Reynolds, Richard C. Rose:
Robust text-independent speaker identification using Gaussian mixture speaker models. 72-83 - Olivier Cappé, Jean Laroche:
Evaluation of short-time spectral attenuation techniques for the restoration of musical recordings. 84-93 - Chih-Chung Kuo, Fu-Rong Jean, Hsiao-Chuan Wang:
Speech classification embedded in adaptive codebook search for low bit-rate CELP coding. 94-98 - John H. L. Hansen, Levent M. Arslan
Markov model-based phoneme class partitioning for improved constrained iterative speech enhancement. 98-104
Volume 3, Number 2, March 1995
- Ravi Prakash Ramachandran, Mihailo S. Zilovic, Richard J. Mammone:
A comparative study of robust linear predictive analysis methods with applications to speaker identification. 117-125 - Dennis R. Morgan:
Slow asymptotic convergence of LMS acoustic echo cancelers. 126-136 - Craig R. Watkins, Robert R. Bitmead, Sam Crisafulli:
Destabilization effects of adaptive quantization in ADPCM. 137-141 - Sin-Horng Chen, Wen-Yuan Chen:
Generalized minimal distortion segmentation for ANN-based speech recognition. 141-145 - Sin-Horng Chen, Yih-Ru Wang:
Tone recognition of continuous Mandarin speech based on neural networks. 146-150
Volume 3, Number 3, May 1995
- Ravi Prakash Ramachandran, Man Mohan Sondhi, Nambi Seshadri, Bishnu S. Atal
A two codebook format for robust quantization of line spectral frequencies. 157-168 - John H. L. Hansen, Levent M. Arslan
Robust feature-estimation and objective quality assessment for noisy speech recognition using the Credit Card corpus. 169-184 - Philip Arthur Nelson
Inverse filter design and equalization zones in multichannel sound reproduction. 185-192 - Michael W. Hoffman, Kevin M. Buckley:
Robust time-domain processing of broadband microphone array data. 193-203 - Tan Lee
Tone recognition of isolated Cantonese syllables. 204-209 - Donald G. Childers, José C. Príncipe, Y. T. Ting:
Adaptive WRLS-VFF for speech analysis. 209-213 - Carl D. Mitchell, Mary P. Harper, Leah H. Jamieson:
On the complexity of explicit duration HMM's. 213-217 - Sen M. Kuo, Min J. Ji:
Development and analysis of an adaptive noise equalizer. 217-222
Volume 3, Number 4, July 1995
- Sven Anderson, Diane Kewley-Port:
Evaluation of speech recognizers for speech training applications. 229-241 - Alan McCree, Thomas P. Barnwell III:
A mixed excitation LPC vocoder model for low bit rate speech coding. 242-250 - Yariv Ephraim, Harry L. Van Trees:
A signal subspace approach for speech enhancement. 251-266 - Simon J. Godsill, Peter J. W. Rayner:
A Bayesian approach to the restoration of degraded audio signals. 267-278 - Nam Soo Kim, Chong Kwan Un:
On estimating robust probability distribution in HMM-based speech recognition. 279-285 - Charles R. Jankowski Jr., Hoang-Doan H. Vo, Richard P. Lippmann:
A comparison of signal processing front ends for automatic word recognition. 286-293 - Saeed Gazor
Criteria for positioning of sensors for a microphone array. 294-303 - Joseph A. Maxwell, Patrick M. Zurek:
Reducing acoustic feedback in hearing aids. 304-313 - Miguel Angel Ferrer-Ballester
Efficient adaptive vector quantization of LPC parameters. 314-317
Volume 3, Number 5, September 1995
- R. Smits, B. Yegnanarayana:
Determination of instants of significant excitation in speech using group delay function. 325-333 - Qiang Huo, Chorkin Chan, Chin-Hui Lee:
Bayesian adaptive learning of the parameters of hidden Markov model for speech recognition. 334-345 - Chafic Mokbel, Gérard Chollet
Automatic word recognition in cars. 346-356 - Vassilios Digalakis
Speaker adaptation using constrained estimation of Gaussian mixtures. 357-366 - William R. Gardner, Bhaskar D. Rao:
Theoretical analysis of the high-rate vector quantization of LPC parameters. 367-381 - Kuansan Wang, Shihab A. Shamma:
Spectral shape analysis in the central auditory system. 382-395 - James M. Kates:
Two-tone suppression in a cochlear model. 396-406 - John H. L. Hansen, Mark A. Clements:
Source generator equalization and enhancement of spectral properties for robust speech recognition in noise and stress. 407-415 - John H. L. Hansen, Sahar E. Bou-Ghazale:
Robust speech recognition training via duration and spectral-based stress token generation. 415-421 - Spiros Dimolitsas, Franklin L. Corcoran, Channasandra Ravishankar:
Dependence of opinion scores on listening sets used in degradation category rating assessments. 421-424 - Futoshi Asano, Yôiti Suzuki, Toshio Sone:
Weighted RLS adaptive beamformer with initial directivity. 424-428
Volume 3, Number 6, November 1995
- Wolfgang G. Knecht
Neural network filters for speech enhancement. 433-438 - Søren Holdt Jensen, Per Christian Hansen
Reduction of broad-band noise in speech by truncated QSVD. 439-448 - Qiguang Lin:
A fast algorithm for computing the vocal-tract impulse response from the transfer function. 449-457 - Jianing Dai:
Isolated word recognition using Markov chain models. 458-463 - Andrei Popescu, Nicolas Moreau, Claude Lamblin:
CELP coding using trellis-coded vector quantization of the excitation. 464-472 - Jinho Choi
A fast determination of stochastic excitation without codebook search in CELP coder. 473-480 - Keiichi Tokuda, Takao Kobayashi, Satoshi Imai:
Adaptive cepstral analysis of speech. 481-489 - Michael Paraskevas, John Mourjopoulos:
A differential perceptual audio coding method with reduced bitrate requirements. 490-503 - Elias Bjarnason:
Analysis of the filtered-X LMS algorithm. 504-514 - Thomas F. Quatieri, Robert B. Dunn, Thomas E. Hanna:
A subband approach to time-scale expansion of complex acoustic signals. 515-519
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