Journal: IEEE Transactions on Audio, Speech & Language Processing

Volume 6, Issue 6

505 -- 515Rathinavelu Chengalvarayan, Li Deng. Speech trajectory discrimination using the minimum classification error learning
516 -- 523William Turin. Unidirectional and parallel Baum-Welch algorithms
524 -- 538Mohamed Afify, Yifan Gong, Jean-Paul Haton. A general joint additive and convolutive bias compensation approach applied to noisy Lombard speech recognition
539 -- 548Yoshihiko Gotoh, Michael M. Hochberg, Harvey F. Silverman. Efficient training algorithms for HMMs using incremental estimation
549 -- 557Tom Claes, Ioannis Dologlou, Louis ten Bosch, Dirk Van Compernolle. A novel feature transformation for vocal tract length normalization in automatic speech recognition
558 -- 568Tatsuya Kawahara, Chin-Hui Lee, Biing-Hwang Juang. Flexible speech understanding based on combined key-phrase detection and verification
569 -- 573Jan S. Erkelens, Piet M. T. Broersen. LPC interpolation by approximation of the sample autocorrelation function
573 -- 579Bryan L. Pellom, John H. L. Hansen. An improved (Auto: I, LSP: T) constrained iterative speech enhancement for colored noise environments
579 -- 582Néstor Becerra Yoma, Fergus R. McInnes, Mervyn A. Jack. Improving performance of spectral subtraction in speech recognition using a model for additive noise

Volume 6, Issue 5

426 -- 434Fábio Violaro, Olivier Boëffard. A hybrid model for text-to-speech synthesis
435 -- 444Ladan Baghai-Ravary, Steve W. Beet. Multistep coding of speech parameters for compression
445 -- 455Hossein Sameti, Hamid Sheikhzadeh, Li Deng, Robert L. Brennan. HMM-based strategies for enhancement of speech signals embedded in nonstationary noise
456 -- 467Jerome R. Bellegarda. A multispan language modeling framework for large vocabulary speech recognition
468 -- 475Jacob Benesty, Dennis R. Morgan, Man Mohan Sondhi. A hybrid mono/stereo acoustic echo canceler
476 -- 488C. Phillip Brown, Richard O. Duda. A structural model for binaural sound synthesis
489 -- 495John H. L. Hansen, David T. Chappell. An auditory-based distortion measure with application to concatenative speech synthesis
495 -- 501Sassan Ahmadi, Andreas S. Spanias. A new phase model for sinusoidal transform coding of speech

Volume 6, Issue 4

313 -- 327B. Yegnanarayana, Raymond N. J. Veldhuis. Extraction of vocal-tract system characteristics from speech signals
328 -- 337Boh Lim Sim, Yit-Chow Tong, Joseph Sylvester Chang, Chin Tuan Tan. A parametric formulation of the generalized spectral subtraction method
338 -- 351Julie E. Greenberg. Modified LMS algorithms for speech processing with an adaptive noise canceller
352 -- 372Simon J. Godsill, Peter J. W. Rayner. Statistical reconstruction and analysis of autoregressive signals in impulsive noise using the Gibbs sampler
373 -- 385Sharon Gannot, David Burshtein, Ehud Weinstein. Iterative and sequential Kalman filter-based speech enhancement algorithms
386 -- 397Qiang Huo, Chin-Hui Lee. On-line adaptive learning of the correlated continuous density hidden Markov models for speech recognition
398 -- 409Andrew Horner. Nested modulator and feedback FM matching of instrument tones
410 -- 414Levent M. Arslan, John H. L. Hansen. Likelihood decision boundary estimation between HMM pairs in speech recognition
414 -- 417Darren B. Ward. Technique for broadband correlated interference rejection in microphone arrays

Volume 6, Issue 3

201 -- 216Sahar E. Bou-Ghazale, John H. L. Hansen. HMM-based stressed speech modeling with application to improved synthesis and recognition of isolated speech under stress
217 -- 225Robert W. P. Luk, Robert I. Damper. Computational complexity of a fast Viterbi decoding algorithm for stochastic letter-phoneme transduction
226 -- 239Sin-Horng Chen, Shaw-Hwa Hwang, Yih-Ru Wang. An RNN-based prosodic information synthesizer for Mandarin text-to-speech
240 -- 259Claude Marro, Yannick Mahieux, Klaus Uwe Simmer. Analysis of noise reduction and dereverberation techniques based on microphone arrays with postfiltering
260 -- 267Mihailo S. Zilovic, Ravi P. Ramachandran, Richard J. Mammone. Speaker identification based on the use of robust cepstral features obtained from pole-zero transfer functions
268 -- 281Alberto Gonzalez, Antonio Albiol, Steve J. Elliott. Minimization of the maximum error signal in active control
282 -- 287F. Plante, Georg F. Meyer, William A. Ainsworth. Improvement of speech spectrogram accuracy by the method of reassignment
287 -- 292Zenton Goh, Kah-Chye Tan, B. T. G. Tan. Postprocessing method for suppressing musical noise generated by spectral subtraction
293 -- 299Ren-Yuan Lyu, I.-Chung Hong, Jia-lin Shen, Ming-Yu Lee, Lin-Shan Lee. Isolated Mandarin base-syllable recognition based upon the segmental probability model
299 -- 303Nam Soo Kim, Chong Kwan Un. Deleted strategy for MMI-based HMM training
303 -- 306Ashvin Kannan, Mari Ostendorf. A comparison of constrained trajectory segment models for large vocabulary speech recognition

Volume 6, Issue 2

106 -- 115Jianping Pan, Thomas R. Fischer. Vector quantization of speech line spectrum pair parameters and reflection coefficients
116 -- 130Redwan Salami, Claude Laflamme, Jean-Pierre Adoul, Akitoshi Kataoka, Shinji Hayashi, Takehiro Moriya, Claude Lamblin, Dominique Massaloux, Stéphane Proust, Peter Kroon, Yair Shoham. Design and description of CS-ACELP: a toll quality 8 kb/s speech coder
131 -- 142Yannis Stylianou, Olivier Cappé, Eric Moulines. Continuous probabilistic transform for voice conversion
143 -- 155Patrick A. Naylor, Oguz Tanrikulu, Anthony G. Constantinides. Subband adaptive filtering for acoustic echo control using allpass polyphase IIR filterbanks
156 -- 165Jacob Benesty, Dennis R. Morgan, Man Mohan Sondhi. A better understanding and an improved solution to the specific problems of stereophonic acoustic echo cancellation
166 -- 176Miller S. Puckette, Judith C. Brown. Accuracy of frequency estimates using the phase vocoder
177 -- 180Xiao Ming Gao, Seppo J. Ovaska, Mikko Lehtokangas, Jukka Saarinen. Modeling of speech signals using an optimal neural network structure based on the PMDL principle
180 -- 185Kadri Hacioglu, Allam Hasib. Pulse-by-pulse reoptimization of the synthesis filter in pulse-based coders
186 -- 189Faouzi Kossentini, Michael W. Macon, Mark J. T. Smith. Audio coding using variable-depth multistage quantization
189 -- 194Ole Kirkeby, Philip A. Nelson, Hareo Hamada, Felipe Orduna-Bustamante. Fast deconvolution of multichannel systems using regularization

Volume 6, Issue 1

1 -- 11B. Yegnanarayana, Christophe d'Alessandro, Vassilios Darsinos. An iterative algorithm for decomposition of speech signals into periodic and aperiodic components
12 -- 23Christophe d'Alessandro, Vassilios Darsinos, B. Yegnanarayana. Effectiveness of a periodic and aperiodic decomposition method for analysis of voice sources
24 -- 35Masato Abe, Kiyohito Fujii, Yoshifumi Nagata, Toshio Sone, Ken'iti Kido. Estimation of the waveform of a sound source by using an iterative technique with many sensors
36 -- 48Lutz Welling, Hermann Ney. Formant estimation for speech recognition
49 -- 60Li Lee, Richard C. Rose. A frequency warping approach to speaker normalization
61 -- 70Olivier Cappé, Chafic Mokbel, Denis Jouvet, Eric Moulines. An algorithm for maximum likelihood estimation of hidden Markov models with unknown state-tying
71 -- 77Mukund Padmanabhan, Lalit R. Bahl, David Nahamoo, Michael A. Picheny. Speaker clustering and transformation for speaker adaptation in speech recognition systems
78 -- 85Markus Rupp, Ali H. Sayed. Robust FxLMS algorithms with improved convergence performance
86 -- 90Sin-Horng Chen, Yuan-Fu Liao, Song-Mao Chiang, Saga Chang. An RNN-based preclassification method for fast continuous Mandarin speech recognition
90 -- 94Hamid Sheikhzadeh, Li Deng. Speech analysis and recognition using interval statistics generated from a composite auditory model