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 K.-W. 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, V. Voitishchuk, Stephanus J. L. van Eijndhoven:
  IIR-based pure linear prediction. 68-75
  
• Joseph Tabrikian, Shlomo Dubnov, Yulya Dickalov:
  Maximum a-posteriori probability pitch tracking in noisy environments using harmonic model. 76-87
  

Volume 12, Number 2, March 2004

• Paavo Alku, Tomas Bäckström:
  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, James Droppo, Alex Acero:
  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, Wenyin Liu, Hong-Jiang Zhang:
  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, Hervé Bourlard:
  Speech recognition with auxiliary information. 189-203
  
• Assaf Ben-Yishai, David Burshtein:
  A discriminative training algorithm for hidden Markov models. 204-217
  
• Li Deng, James Droppo, Alex Acero:
  Estimating cepstrum of speech under the presence of noise using a joint prior of static and dynamic features. 218-233
  
• Vaibhava Goel, Shankar Kumar, William Byrne:
  Segmental minimum Bayes-risk decoding for automatic speech recognition. 234-249
  
• Vincent Vanhoucke, Ananth Sankar:
  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, Satoshi Nakamura, S. Narayanan:
  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, Yasuhiro Minami, Atsushi Nakamura, Naonori Ueda:
  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, Masahiro Hasegawa, Kazuya Shitaoka, Tasuku Kitade, Hiroaki Nanjo:
  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, Douglas W. Oard, Michael Picheny, Josef Psutka, Bhuvana Ramabhadran, Dagobert Soergel, Todd Ward, Wei-Jing Zhu:
  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. Sondhi, D. DeVries:
  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, Pak-Chung Ching, Ba-Ngu Vo:
  Crosstalk resilient interference cancellation in microphone arrays using Capon beamforming. 468-477
  
• Yahong Rosa Zheng, Rafik A. Goubran, Mohamed El-Tanany:
  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, Jingdong Chen, Yiteng Huang:
  Time-delay estimation via linear interpolation and cross correlation. 509-519
  
• Ilyas Potamitis, Huimin Chen, George Tremoulis:
  Tracking of multiple moving speakers with multiple microphone arrays. 520-529
  
• Hiroshi Sawada, Ryo Mukai, Shoko Araki, Shoji Makino:
  A robust and precise method for solving the permutation problem of frequency-domain blind source separation. 530-538
  
• Siow Yong Low, Sven Nordholm, Roberto Togneri:
  Convolutive blind signal separation with post-processing. 539-548
  

Volume 12, Number 6, November 2004

• Isabel Trancoso:
  From the Editor-in-Chief. 553
  
• Tomas Bäckström, Paavo Alku, Tuomas Paatero, W. Bastiaan Kleijn:
  A time-domain interpretation for the LSP decomposition. 554-560
  
• Sharon Gannot, Israel Cohen:
  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, Pierre Ouellet, Pierre Dumouchel:
  Speaker adaptation using an eigenphone basis. 579-589