Journal: IEEE Transactions on Audio, Speech & Language Processing

Volume 4, Issue 6

401 -- 411Akitoshi Kataoka, Takehiro Moriya, Shinji Hayashi. An 8-kb/s conjugate structure CELP (CS-CELP) speech coder
412 -- 419Mei-Yuh Hwang, Xuedong Huang, Fileno A. Alleva. Predicting unseen triphones with senones
420 -- 429Rafid A. Sukkar, Chin-Hui Lee. Vocabulary independent discriminative utterance verification for nonkeyword rejection in subword based speech recognition
430 -- 445Philipos C. Loizou, Andreas S. Spanias. High-performance alphabet recognition
446 -- 455Zhiqian Wang, Jezekiel Ben-Arie. Conveying visual information with spatial auditory patterns
456 -- 460Milan Z. Markovic, Branko D. Kovacevic, Milan M. Milosavljevic. A statistical pattern recognition approach to robust recursive identification of nonstationary AR model of speech production system

Volume 4, Issue 5

325 -- 336Fady Alajaji, Nam C. Phamdo, Thomas E. Fuja. Channel codes that exploit the residual redundancy in CELP-encoded speech
337 -- 351Peter L. Silsbee, Alan C. Bovik. Computer lipreading for improved accuracy in automatic speech recognition
352 -- 359Mark J. F. Gales, Stephen J. Young. Robust continuous speech recognition using parallel model combination
360 -- 378Mari Ostendorf, Vassilios Digalakis, Owen Kimball. From HMM's to segment models: a unified view of stochastic modeling for speech recognition
379 -- 382Hiroto Saito, Isao Umoto, Akira Sasou, Shogo Nakamura, Yoshihiko Horio, Tahiro Kubota. Subadaptive piecewise linear quantization for speech signal (64 kbit/s) compression
383 -- 389T. V. Sreenivas, Pradeep Kirnapure. Codebook constrained Wiener filtering for speech enhancement
389 -- 392Y.-K. Park, C. K. Un, O. W. Kwon. Modeling acoustic transitions in speech by modified hidden Markov models with state duration and state duration-dependent observation probabilities
392 -- 396Suresh Subramaniam, Athina P. Petropulu, Christopher Wendt. Cepstrum-based deconvolution for speech dereverberation

Volume 4, Issue 4

253 -- 0Qiaobing Xie, Rabab K. Ward, Charles A. Laszlo. Automatic Assessment of Infants' Levels-of-Distress from the Cry Signals
266 -- 280Roar Hagen. Robust LPC spectrum quantization-vector quantization by a linear mapping of a block code
281 -- 289Vassilios Digalakis, Peter Monaco, Hy Murveit. Genones: generalized mixture tying in continuous hidden Markov model-based speech recognizers
290 -- 293Y.-K. Park, C. K. Un. On the generation and use of a parallel-branch subunit model in continuous HMM
294 -- 300Vassilios Digalakis, Leonardo Neumeyer. Speaker adaptation using combined transformation and Bayesian methods
301 -- 306Li Deng, Hossein Sameti. Transitional speech units and their representation by regressive Markov states: applications to speech recognition
307 -- 313John H. L. Hansen, Brian D. Womack. Feature analysis and neural network-based classification of speech under stress
313 -- 316Joerg P. Ueberla. An extended clustering algorithm for statistical language models

Volume 4, Issue 3

157 -- 166Martin P. DeSimio, Timothy R. Anderson, John J. Westerkamp. Phoneme recognition with a model of binaural hearing
167 -- 189Tung-Hui Chiang, Yi-Chung Lin, Keh-Yih Su. On jointly learning the parameters in a character-synchronous integrated speech and language model
190 -- 202Ananth Sankar, Chin-Hui Lee. A maximum-likelihood approach to stochastic matching for robust speech recognition
203 -- 213Ta-Hsin Li, Jerry D. Gibson. Speech analysis and segmentation by parametric filtering
214 -- 223S. J. Elliott, T. J. Sutton. Performance of feedforward and feedback systems for active control
224 -- 230Boaz Rafaely, Miriam Furst. Audiometric ear canal probe with active ambient noise control
231 -- 234Evangelos Dermatas, George Kokkinakis. Algorithm for clustering continuous density HMM by recognition error
234 -- 239Minjie Xie, Jean-Pierre Adoul. Algebraic vector quantization of LSF parameters with low storage and computational complexity
240 -- 242David Burshtein. Robust parametric modeling of durations in hidden Markov models
243 -- 247Silvio Cucchi, Milan Fratti, M. Ronchi. On improving performance of analysis by synthesis speech coders

Volume 4, Issue 2

81 -- 88Il-Taek Lim, Byeong Gi Lee. Lossy pole-zero modeling for speech signals
89 -- 95Antonio M. Peinado, José C. Segura, Antonio J. Rubio, Pedro García Teodoro, José L. Pérez-Córdoba. Discriminative codebook design using multiple vector quantization in HMM-based speech recognizers
96 -- 103Sen M. Kuo, Minjiang Ji. Passband disturbance reduction in periodic active noise control systems
104 -- 114Aníbal J. S. Ferreira. Convolutional effects in transform coding with TDAC: an optimal window
115 -- 123Charles D. Creusere, Sanjit K. Mitra. Efficient audio coding using perfect reconstruction noncausal IIR filter banks
124 -- 132Wen-Whei Chang, Chin-Tun Wang. A masking-threshold-adapted weighting filter for excitation search
133 -- 137B. Yegnanarayana, P. Satyanarayana Murthy. Source-system windowing for speech analysis and synthesis
138 -- 140Shin-Lun Tung, I.-Shine Lei, Yau-Tarng Juang. Projection-based group delay scheme for speech recognition
141 -- 144Qiang Huo, Chorkin Chan, Chin-Hui Lee. On-line adaptation of the SCHMM parameters based on the segmental quasi-Bayes learning for speech recognition
144 -- 148Din-Yuen Chan, Jar-Ferr Yang, Chun-Chin Fang. Fast implementation of MPEG audio coder using recursive formula with fast discrete cosine transforms
148 -- 152Benoît Champagne, Stéphane Bédard, Alex Stephenne. Performance of time-delay estimation in the presence of room reverberation

Volume 4, Issue 1

1 -- 0Joseph P. Campbell. In Memory of Thomas E. Tremain 1934-1995
2 -- 0Tzyy-Ping Jung, Ashok K. Krishnamurthy, Stanley C. Ahalt, Mary E. Beckman, Sook-Hyang Lee. Deriving gestural score from articulator-movement records using weighted temporal decomposition
19 -- 0Biing-Hwang Juang, Mazin G. Rahim. Signal bias removal by maximum likelihood estimation for robust telephone speech recognition
31 -- 0Marc A. Zissman. Comparison of four approaches to automatic language identification of telephone speech
45 -- 0Dibyendu Nandy, Jezekiel Ben-Arie. Estimating the azimuth of a sound source from the binaural spectral amplitude
56 -- 0Chi-shi Liu, Hsiao-Chuan Wang, Chin-Hui Lee. Speaker verification using normalized log-likelihood score
60 -- 0Fu-Rong Jean, Hsiao-Chuan Wang. Transparent quantization of speech LSP parameters based on KLT and 2-D-prediction
66 -- 0Richard P. Lippmann. Accurate consonant perception without mid-frequency speech energy