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    Journal: IEEE Transactions on Audio, Speech & Language Processing

    Volume 12, Issue 6

    553 -- 0Isabel Trancoso. From the Editor-in-Chief
    554 -- 560Tomas Bäckström, Paavo Alku, Tuomas Paatero, W. Bastiaan Kleijn. A time-domain interpretation for the LSP decomposition
    561 -- 571Sharon Gannot, Israel Cohen. Speech enhancement based on the general transfer function GSC and postfiltering
    572 -- 578Mukund Padmanabhan, Satya Dharanipragada. Maximum-likelihood nonlinear transformation for acoustic adaptation
    579 -- 589Patrick Kenny, Gilles Boulianne, Pierre Ouellet, Pierre Dumouchel. Speaker adaptation using an eigenphone basis

    Volume 12, Issue 5

    449 -- 450Walter Kellermann, M. Sondhi, D. DeVries. Introduction to the Special Issue on Multichannel Signal Processing for Audio and Acoustics Applications
    451 -- 459Israel Cohen. Relative transfer function identification using speech signals
    460 -- 467Ingo Schwetz, Gerhard Gruhler, Klaus Obermayer. Correlation and stationarity of speech radiation: consequences for linear multichannel filtering
    468 -- 477Wing-Kin Ma, Pak-Chung Ching, Ba-Ngu Vo. Crosstalk resilient interference cancellation in microphone arrays using Capon beamforming
    478 -- 488Yahong Rosa Zheng, Rafik A. Goubran, Mohamed El-Tanany. Robust near-field adaptive beamforming with distance discrimination
    489 -- 498Michael L. Seltzer, Bhiksha Raj, Richard M. Stern. Likelihood-maximizing beamforming for robust hands-free speech recognition
    499 -- 508Dmitry N. Zotkin, Ramani Duraiswami. Accelerated speech source localization via a hierarchical search of steered response power
    509 -- 519Jacob Benesty, Jingdong Chen, Yiteng Huang. Time-delay estimation via linear interpolation and cross correlation
    520 -- 529Ilyas Potamitis, Huimin Chen, George Tremoulis. Tracking of multiple moving speakers with multiple microphone arrays
    530 -- 538Hiroshi Sawada, Ryo Mukai, Shoko Araki, Shoji Makino. A robust and precise method for solving the permutation problem of frequency-domain blind source separation
    539 -- 548Siow Yong Low, Sven Nordholm, Roberto Togneri. Convolutive blind signal separation with post-processing

    Volume 12, Issue 4

    349 -- 350Sadaoki Furui, Mary E. Beckman, Julia Hirschberg, SShuichi Itahashi, Tatsuya Kawahara, Satoshi Nakamura, S. Narayanan. Introduction to the Special Issue on Spontaneous Speech Processing
    351 -- 364Yi Liu, Pascale Fung. State-dependent phonetic tied mixtures with pronunciation modeling for spontaneous speech recognition
    365 -- 381Shinji Watanabe, Yasuhiro Minami, Atsushi Nakamura, Naonori Ueda. Variational bayesian estimation and clustering for speech recognition
    382 -- 390Kiyotaka Uchimoto, Kazuma Takaoka, Chikashi Nobata, Atsushi Yamada, Satoshi Sekine, Hitoshi Isahara. Morphological analysis of the corpus of spontaneous Japanese
    391 -- 400Hiroaki Nanjo, Tatsuya Kawahara. Language model and speaking rate adaptation for spontaneous presentation speech recognition
    401 -- 408Sadaoki Furui, Tomonori Kikuchi, Yosuke Shinnaka, Chiori Hori. Speech-to-text and speech-to-speech summarization of spontaneous speech
    409 -- 419Tatsuya Kawahara, Masahiro Hasegawa, Kazuya Shitaoka, Tasuku Kitade, Hiroaki Nanjo. Automatic indexing of lecture presentations using unsupervised learning of presumed discourse markers
    420 -- 435William 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
    436 -- 445Steffen Werner, Matthias Eichner, Matthias Wolff, Rüdiger Hoffmann. Toward spontaneous speech Synthesis-utilizing language model information in TTS

    Volume 12, Issue 3

    189 -- 203Todd A. Stephenson, Mathew Magimai-Doss, Hervé Bourlard. Speech recognition with auxiliary information
    204 -- 217Assaf Ben-Yishai, David Burshtein. A discriminative training algorithm for hidden Markov models
    218 -- 233Li Deng, James Droppo, Alex Acero. Estimating cepstrum of speech under the presence of noise using a joint prior of static and dynamic features
    234 -- 249Vaibhava Goel, Shankar Kumar, William Byrne. Segmental minimum Bayes-risk decoding for automatic speech recognition
    250 -- 264Vincent Vanhoucke, Ananth Sankar. Mixtures of inverse covariances
    265 -- 276Laurent Girin. Joint matrix quantization of face parameters and LPC coefficients for low bit rate audiovisual speech coding
    277 -- 289Moo-young Kim, W. Bastiaan Kleijn. KLT-based adaptive classified VQ of the speech signal
    290 -- 301Frank Norden, Thomas Eriksson. Time evolution in LPC spectrum coding
    302 -- 312Laurent Daudet, Mark B. Sandler. MDCT analysis of sinusoids: exact results and applications to coding artifacts reduction
    313 -- 322Debi Prasad Das, Ganapati Panda. Active mitigation of nonlinear noise Processes using a novel filtered-s LMS algorithm
    323 -- 333Lisa G. Huettel, Leslie M. Collins. A theoretical analysis of normal- and impaired-hearing intensity discrimination
    334 -- 342Sarah E. Schwarm, Ivan Bulyko, Mari Ostendorf. Adaptive language modeling with varied sources to cover new vocabulary items

    Volume 12, Issue 2

    93 -- 99Paavo Alku, Tomas Bäckström. Linear predictive method for improved spectral modeling of lower frequencies of speech with small prediction orders
    100 -- 109Mark A. Bartsch, Gregory H. Wakefield. Singing voice identification using spectral envelope estimation
    110 -- 120Rémy Boyer, Karim Abed-Meraim. Audio modeling based on delayed sinusoids
    121 -- 132Jesper Jensen, Richard Heusdens, Søren Holdt Jensen. A perceptual subspace approach for modeling of speech and audio signals with damped sinusoids
    133 -- 143Li 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
    144 -- 155Eric A. Durant, Gregory H. Wakefield, Dianne J. Van Tasell, Martin E. Rickert. Efficient perceptual tuning of hearing aids with genetic algorithms
    156 -- 167Lie Lu, Wenyin Liu, Hong-Jiang Zhang. Audio textures: theory and applications
    168 -- 174Xintian Wu, YongHong Yan. Speaker adaptation using constrained transformation
    175 -- 185Sadao Hiroya, Masaaki Honda. Estimation of articulatory movements from speech acoustics using an HMM-based speech production model

    Volume 12, Issue 1

    1 -- 8Venkatesh Krishnan, David V. Anderson, Kwan K. Truong. Optimal multistage vector quantization of LPC parameters over noisy channels
    9 -- 18Nam Soo Kim, Joon-Hyuk Chang. Signal modification for robust speech coding
    19 -- 26Mohamed Afify, Olivier Siohan. Sequential estimation with optimal forgetting for robust speech recognition
    27 -- 36Brian K.-W. Mak, Yik-Cheung Tam, Peter Qi Li. Discriminative auditory-based features for robust speech recognition
    37 -- 46Peder A. Olsen, Ramesh A. Gopinath. Modeling inverse covariance matrices by basis expansion
    47 -- 58Jeff Z. Ma, Li Deng. Target-directed mixture dynamic models for spontaneous speech recognition
    59 -- 67Yi Hu, Philipos C. Loizou. Speech enhancement based on wavelet thresholding the multitaper spectrum
    68 -- 75Albertus C. den Brinker, V. Voitishchuk, Stephanus J. L. van Eijndhoven. IIR-based pure linear prediction
    76 -- 87Joseph Tabrikian, Shlomo Dubnov, Yulya Dickalov. Maximum a-posteriori probability pitch tracking in noisy environments using harmonic model
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