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IEEE Transactions on Speech and Audio Processing, Volume 12
Volume 12, Number 1, January 2004
- Venkatesh Krishnan, David V. Anderson, Kwan K. Truong:
Optimal multistage vector quantization of LPC parameters over noisy channels. 1-8 - Nam Soo Kim, Joon-Hyuk Chang:
Signal modification for robust speech coding. 9-18 - Mohamed Afify, Olivier Siohan:
Sequential estimation with optimal forgetting for robust speech recognition. 19-26 - Brian Kan-Wing Mak
, Yik-Cheung Tam, Peter Qi Li:
Discriminative auditory-based features for robust speech recognition. 27-36 - Peder A. Olsen, Ramesh A. Gopinath:
Modeling inverse covariance matrices by basis expansion. 37-46 - Jeff Z. Ma, Li Deng:
Target-directed mixture dynamic models for spontaneous speech recognition. 47-58 - Yi Hu, Philipos C. Loizou:
Speech enhancement based on wavelet thresholding the multitaper spectrum. 59-67 - Albertus C. den Brinker
IIR-based pure linear prediction. 68-75 - Joseph Tabrikian
Maximum a-posteriori probability pitch tracking in noisy environments using harmonic model. 76-87
Volume 12, Number 2, March 2004
- Paavo Alku
Linear predictive method for improved spectral modeling of lower frequencies of speech with small prediction orders. 93-99 - Mark A. Bartsch, Gregory H. Wakefield:
Singing voice identification using spectral envelope estimation. 100-109 - Rémy Boyer, Karim Abed-Meraim:
Audio modeling based on delayed sinusoids. 110-120 - Jesper Jensen, Richard Heusdens, Søren Holdt Jensen:
A perceptual subspace approach for modeling of speech and audio signals with damped sinusoids. 121-132 - Li Deng, Jasha Droppo
Enhancement of log Mel power spectra of speech using a phase-sensitive model of the acoustic environment and sequential estimation of the corrupting noise. 133-143 - Eric A. Durant, Gregory H. Wakefield, Dianne J. Van Tasell, Martin E. Rickert:
Efficient perceptual tuning of hearing aids with genetic algorithms. 144-155 - Lie Lu
Audio textures: theory and applications. 156-167 - Xintian Wu, Yonghong Yan:
Speaker adaptation using constrained transformation. 168-174 - Sadao Hiroya, Masaaki Honda:
Estimation of articulatory movements from speech acoustics using an HMM-based speech production model. 175-185
Volume 12, Number 3, May 2004
- Todd A. Stephenson, Mathew Magimai-Doss
Speech recognition with auxiliary information. 189-203 - Assaf Ben-Yishai, David Burshtein:
A discriminative training algorithm for hidden Markov models. 204-217 - Li Deng, Jasha Droppo
Estimating cepstrum of speech under the presence of noise using a joint prior of static and dynamic features. 218-233 - Vaibhava Goel
Segmental minimum Bayes-risk decoding for automatic speech recognition. 234-249 - Vincent Vanhoucke
Mixtures of inverse covariances. 250-264 - Laurent Girin:
Joint matrix quantization of face parameters and LPC coefficients for low bit rate audiovisual speech coding. 265-276 - Moo Young Kim, W. Bastiaan Kleijn
KLT-based adaptive classified VQ of the speech signal. 277-289 - Frank Norden, Thomas Eriksson
Time evolution in LPC spectrum coding. 290-301 - Laurent Daudet, Mark B. Sandler:
MDCT analysis of sinusoids: exact results and applications to coding artifacts reduction. 302-312 - Debi Prasad Das, Ganapati Panda
Active mitigation of nonlinear noise Processes using a novel filtered-s LMS algorithm. 313-322 - Lisa G. Huettel, Leslie M. Collins:
A theoretical analysis of normal- and impaired-hearing intensity discrimination. 323-333 - Sarah E. Schwarm, Ivan Bulyko, Mari Ostendorf:
Adaptive language modeling with varied sources to cover new vocabulary items. 334-342
Volume 12, Number 4, July 2004
- Sadaoki Furui, Mary E. Beckman, Julia Hirschberg, Shuichi Itahashi, Tatsuya Kawahara
Introduction to the Special Issue on Spontaneous Speech Processing. 349-350 - Yi Liu, Pascale Fung:
State-dependent phonetic tied mixtures with pronunciation modeling for spontaneous speech recognition. 351-364 - Shinji Watanabe
Variational bayesian estimation and clustering for speech recognition. 365-381 - Kiyotaka Uchimoto, Kazuma Takaoka, Chikashi Nobata, Atsushi Yamada, Satoshi Sekine, Hitoshi Isahara:
Morphological analysis of the corpus of spontaneous Japanese. 382-390 - Hiroaki Nanjo, Tatsuya Kawahara
Language model and speaking rate adaptation for spontaneous presentation speech recognition. 391-400 - Sadaoki Furui, Tomonori Kikuchi, Yosuke Shinnaka, Chiori Hori:
Speech-to-text and speech-to-speech summarization of spontaneous speech. 401-408 - Tatsuya Kawahara
Automatic indexing of lecture presentations using unsupervised learning of presumed discourse markers. 409-419 - William Byrne, David S. Doermann, Martin Franz, Samuel Gustman, Jan Hajic
Automatic recognition of spontaneous speech for access to multilingual oral history archives. 420-435 - Steffen Werner, Matthias Eichner, Matthias Wolff, Rüdiger Hoffmann:
Toward spontaneous speech Synthesis-utilizing language model information in TTS. 436-445
Volume 12, Number 5, September 2004
- Walter Kellermann, M. Mohan Sondhi, Diemer de Vries:
Introduction to the Special Issue on Multichannel Signal Processing for Audio and Acoustics Applications. 449-450 - Israel Cohen:
Relative transfer function identification using speech signals. 451-459 - Ingo Schwetz, Gerhard Gruhler, Klaus Obermayer:
Correlation and stationarity of speech radiation: consequences for linear multichannel filtering. 460-467 - Wing-Kin Ma
Crosstalk resilient interference cancellation in microphone arrays using Capon beamforming. 468-477 - Yahong Rosa Zheng
Robust near-field adaptive beamforming with distance discrimination. 478-488 - Michael L. Seltzer, Bhiksha Raj, Richard M. Stern
Likelihood-maximizing beamforming for robust hands-free speech recognition. 489-498 - Dmitry N. Zotkin, Ramani Duraiswami
Accelerated speech source localization via a hierarchical search of steered response power. 499-508 - Jacob Benesty
Time-delay estimation via linear interpolation and cross correlation. 509-519 - Ilyas Potamitis
Tracking of multiple moving speakers with multiple microphone arrays. 520-529 - Hiroshi Sawada, Ryo Mukai, Shoko Araki
A robust and precise method for solving the permutation problem of frequency-domain blind source separation. 530-538 - Siow Yong Low, Sven Nordholm
Convolutive blind signal separation with post-processing. 539-548
Volume 12, Number 6, November 2004
- Isabel Trancoso
From the Editor-in-Chief. 553 - Tom Bäckström
A time-domain interpretation for the LSP decomposition. 554-560 - Sharon Gannot
Speech enhancement based on the general transfer function GSC and postfiltering. 561-571 - Mukund Padmanabhan, Satya Dharanipragada:
Maximum-likelihood nonlinear transformation for acoustic adaptation. 572-578 - Patrick Kenny, Gilles Boulianne
Speaker adaptation using an eigenphone basis. 579-589
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