


default search action
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

, 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

, Tom 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, Jasha 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, Jasha 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, Shrikanth 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. 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

, 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 - Tom 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

manage site settings
To protect your privacy, all features that rely on external API calls from your browser are turned off by default. You need to opt-in for them to become active. All settings here will be stored as cookies with your web browser. For more information see our F.A.Q.


Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCID














