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ORCIDIEEE Transactions on Speech and Audio Processing, Volume 6
Volume 6, Number 1, January 1998
- B. Yegnanarayana, Christophe d'Alessandro, Vassilios Darsinos:
An iterative algorithm for decomposition of speech signals into periodic and aperiodic components. 1-11 - Christophe d'Alessandro, Vassilios Darsinos, B. Yegnanarayana:
Effectiveness of a periodic and aperiodic decomposition method for analysis of voice sources. 12-23 - Masato 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. 24-35 - Lutz Welling, Hermann Ney:
Formant estimation for speech recognition. 36-48 - Li Lee, Richard C. Rose:
A frequency warping approach to speaker normalization. 49-60 - Olivier Cappé, Chafic Mokbel, Denis Jouvet, Eric Moulines:
An algorithm for maximum likelihood estimation of hidden Markov models with unknown state-tying. 61-70 - Mukund Padmanabhan, Lalit R. Bahl, David Nahamoo, Michael A. Picheny:
Speaker clustering and transformation for speaker adaptation in speech recognition systems. 71-77 - Markus Rupp, Ali H. Sayed:
Robust FxLMS algorithms with improved convergence performance. 78-85 - Sin-Horng Chen, Yuan-Fu Liao
, Song-Mao Chiang, Saga Chang:
An RNN-based preclassification method for fast continuous Mandarin speech recognition. 86-90 - Hamid Sheikhzadeh, Li Deng:
Speech analysis and recognition using interval statistics generated from a composite auditory model. 90-94
Volume 6, Number 2, March 1998
- Jianping Pan, Thomas R. Fischer:
Vector quantization of speech line spectrum pair parameters and reflection coefficients. 106-115 - Redwan 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. 116-130 - Yannis Stylianou, Olivier Cappé, Eric Moulines:
Continuous probabilistic transform for voice conversion. 131-142 - Patrick A. Naylor, Oguz Tanrikulu, Anthony G. Constantinides:
Subband adaptive filtering for acoustic echo control using allpass polyphase IIR filterbanks. 143-155 - Jacob Benesty, Dennis R. Morgan, Man Mohan Sondhi:
A better understanding and an improved solution to the specific problems of stereophonic acoustic echo cancellation. 156-165 - Miller S. Puckette, Judith C. Brown:
Accuracy of frequency estimates using the phase vocoder. 166-176 - Xiao Ming Gao, Seppo J. Ovaska, Mikko Lehtokangas, Jukka Saarinen:
Modeling of speech signals using an optimal neural network structure based on the PMDL principle. 177-180 - Kadri Hacioglu, Allam Hasib:
Pulse-by-pulse reoptimization of the synthesis filter in pulse-based coders. 180-185 - Faouzi Kossentini, Michael W. Macon, Mark J. T. Smith:
Audio coding using variable-depth multistage quantization. 186-189 - Ole Kirkeby, Philip Arthur Nelson, Hareo Hamada, Felipe Orduña-Bustamante:
Fast deconvolution of multichannel systems using regularization. 189-194
Volume 6, Number 3, May 1998
- Sahar E. Bou-Ghazale, John H. L. Hansen:
HMM-based stressed speech modeling with application to improved synthesis and recognition of isolated speech under stress. 201-216 - Robert W. P. Luk, Robert I. Damper:
Computational complexity of a fast Viterbi decoding algorithm for stochastic letter-phoneme transduction. 217-225 - Sin-Horng Chen, Shaw-Hwa Hwang, Yih-Ru Wang:
An RNN-based prosodic information synthesizer for Mandarin text-to-speech. 226-239 - Claude Marro, Yannick Mahieux, Klaus Uwe Simmer:
Analysis of noise reduction and dereverberation techniques based on microphone arrays with postfiltering. 240-259 - Mihailo S. Zilovic, Ravi Prakash Ramachandran, Richard J. Mammone:
Speaker identification based on the use of robust cepstral features obtained from pole-zero transfer functions. 260-267 - Alberto González, Antonio Albiol, Steve J. Elliott:
Minimization of the maximum error signal in active control. 268-281 - Fabrice Plante, Georg F. Meyer, William A. Ainsworth:
Improvement of speech spectrogram accuracy by the method of reassignment. 282-287 - Zenton Goh, Kah-Chye Tan, B. T. G. Tan:
Postprocessing method for suppressing musical noise generated by spectral subtraction. 287-292 - Ren-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. 293-299 - Nam Soo Kim, Chong Kwan Un:
Deleted strategy for MMI-based HMM training. 299-303 - Ashvin Kannan, Mari Ostendorf:
A comparison of constrained trajectory segment models for large vocabulary speech recognition. 303-306
Volume 6, Number 4, July 1998
- B. Yegnanarayana, Raymond N. J. Veldhuis:
Extraction of vocal-tract system characteristics from speech signals. 313-327 - Boh Lim Sim, Yit Chow Tong, Joseph Sylvester Chang, Chin-Tuan Tan:
A parametric formulation of the generalized spectral subtraction method. 328-337 - Julie E. Greenberg:
Modified LMS algorithms for speech processing with an adaptive noise canceller. 338-351 - Simon J. Godsill, Peter J. W. Rayner:
Statistical reconstruction and analysis of autoregressive signals in impulsive noise using the Gibbs sampler. 352-372 - Sharon Gannot, David Burshtein, Ehud Weinstein:
Iterative and sequential Kalman filter-based speech enhancement algorithms. 373-385 - Qiang Huo, Chin-Hui Lee:
On-line adaptive learning of the correlated continuous density hidden Markov models for speech recognition. 386-397 - Andrew Horner:
Nested modulator and feedback FM matching of instrument tones. 398-409 - Levent M. Arslan, John H. L. Hansen:
Likelihood decision boundary estimation between HMM pairs in speech recognition. 410-414 - Darren B. Ward:
Technique for broadband correlated interference rejection in microphone arrays. 414-417
Volume 6, Number 5, September 1998
- Fábio Violaro, Olivier Boëffard:
A hybrid model for text-to-speech synthesis. 426-434 - Ladan Baghai-Ravary, Steve W. Beet:
Multistep coding of speech parameters for compression. 435-444 - Hossein Sameti, Hamid Sheikhzadeh, Li Deng, Robert L. Brennan:
HMM-based strategies for enhancement of speech signals embedded in nonstationary noise. 445-455 - Jerome R. Bellegarda:
A multispan language modeling framework for large vocabulary speech recognition. 456-467 - Jacob Benesty, Dennis R. Morgan, Man Mohan Sondhi:
A hybrid mono/stereo acoustic echo canceler. 468-475 - C. Phillip Brown, Richard O. Duda:
A structural model for binaural sound synthesis. 476-488 - John H. L. Hansen, David T. Chappell:
An auditory-based distortion measure with application to concatenative speech synthesis. 489-495 - Sassan Ahmadi, Andreas S. Spanias:
A new phase model for sinusoidal transform coding of speech. 495-501
Volume 6, Number 6, November 1998
- Rathinavelu Chengalvarayan, Li Deng:
Speech trajectory discrimination using the minimum classification error learning. 505-515 - William Turin:
Unidirectional and parallel Baum-Welch algorithms. 516-523 - Mohamed Afify, Yifan Gong, Jean Paul Haton:
A general joint additive and convolutive bias compensation approach applied to noisy Lombard speech recognition. 524-538 - Yoshihiko Gotoh, Michael M. Hochberg, Harvey F. Silverman:
Efficient training algorithms for HMMs using incremental estimation. 539-548 - Tom Claes, Ioannis Dologlou, Louis ten Bosch, Dirk Van Compernolle:
A novel feature transformation for vocal tract length normalization in automatic speech recognition. 549-557 - Tatsuya Kawahara, Chin-Hui Lee, Biing-Hwang Juang:
Flexible speech understanding based on combined key-phrase detection and verification. 558-568 - Jan S. Erkelens, Piet M. T. Broersen:
LPC interpolation by approximation of the sample autocorrelation function. 569-573 - Bryan L. Pellom, John H. L. Hansen:
An improved (Auto: I, LSP: T) constrained iterative speech enhancement for colored noise environments. 573-579 - Néstor Becerra Yoma, Fergus R. McInnes, Mervyn A. Jack:
Improving performance of spectral subtraction in speech recognition using a model for additive noise. 579-582
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