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ORCIDIEEE Transactions on Speech and Audio Processing, Volume 11
Volume 11, Number 1, January 2003
- Filiz Basbug, Kumar Swaminathan, Srinivas Nandkumar:
Noise reduction and echo cancellation front-end for speech codecs. 1-13 - Jan Mark de Haan, Nedelko Grbic, Ingvar Claesson, Sven E. Nordholm
:
Filter bank design for subband adaptive microphone arrays. 14-23 - Hong-Kwang Jeff Kuo, Chin-Hui Lee:
Discriminative training of natural language call routers. 24-35 - Axel Nackaerts, Bart De Moor, Rudy Lauwereins:
A formant filtered physical model for wind instruments. 36-44 - Ming Zhang, Hui Lan, Wee Ser:
A robust online secondary path modeling method with auxiliary noise power scheduling strategy and norm constraint manipulation. 45-53 - Martin Bouchard:
Multichannel affine and fast affine projection algorithms for active noise control and acoustic equalization systems. 54-60 - Upendra V. Chaudhari, Jirí Navrátil, Stéphane H. Maes:
Multigrained modeling with pattern specific maximum likelihood transformations for text-independent speaker recognition. 61-69 - Jen-Tzung Chien:
Linear regression based Bayesian predictive classification for speech recognition. 70-79 - Chulhee Lee, Donghoon Hyun, Euisun Choi, Jinwook Go, Chungyong Lee:
Optimizing feature extraction for speech recognition. 80-87 - Jonas Lindblom, Jonas Samuelsson:
Bounded support Gaussian mixture modeling of speech spectra. 88-99 - Raymond N. J. Veldhuis, Esther Klabbers:
On the computation of the Kullback-Leibler measure for spectral distances. 100-103
Volume 11, Number 2, March 2003
- Shoko Araki, Ryo Mukai, Shoji Makino, Tsuyoki Nishikawa, Hiroshi Saruwatari:
The fundamental limitation of frequency domain blind source separation for convolutive mixtures of speech. 109-116 - Sean A. Ramprashad:
The multimode transform predictive coding paradigm. 117-129 - Anand D. Subramaniam, Bhaskar D. Rao:
PDF optimized parametric vector quantization of speech line spectral frequencies. 130-142 - Koen Eneman, Marc Moonen:
Iterated partitioned block frequency-domain adaptive filtering for acoustic echo cancellation. 143-158
Volume 11, Number 3, May 2003
- Christopher J. C. Burges, John C. Platt, Soumya Jana:
Distortion discriminant analysis for audio fingerprinting. 165-174 - Aggelos Pikrakis, Sergios Theodoridis, Dimitris Kamarotos:
Recognition of isolated musical patterns using context dependent dynamic time warping. 175-183 - Jürgen Tchorz, Birger Kollmeier:
SNR estimation based on amplitude modulation analysis with applications to noise suppression. 184-192 - Sven E. Nordholm, Ingvar Claesson, Nedelko Grbic:
Performance limits in subband beamforming. 193-203 - Futoshi Asano, Shiro Ikeda, Michiaki Ogawa, Hideki Asoh, Nobuhiko Kitawaki:
Combined approach of array processing and independent component analysis for blind separation of acoustic signals. 204-215 - Ran Yaniv, David Burshtein:
An enhanced dynamic time warping model for improved estimation of DTW parameters. 216-228 - Mingyang Wu, DeLiang Wang, Guy J. Brown:
A multipitch tracking algorithm for noisy speech. 229-241 - Davide Rocchesso, Julius O. Smith III:
Generalized digital waveguide networks. 242-254 - Stefan Bilbao, Julius O. Smith III:
Finite difference schemes and digital waveguide networks for the wave equation: stability, passivity, and numerical dispersion. 255-266 - Jerome R. Bellegarda, Kim E. A. Silverman:
Natural language spoken interface control using data-driven semantic inference. 267-277 - Lester S. H. Ngia:
Recursive identification of acoustic echo systems using orthonormal basis functions. 278-293
Volume 11, Number 4, July 2003
- Isabel Trancoso:
From the editor-in-chief. 297 - Xiaodong He, Yunxin Zhao:
Fast model selection based speaker adaptation for nonnative speech. 298-307 - Sin-Horng Chen, Wen-Hsing Lai, Yih-Ru Wang:
A new duration modeling approach for Mandarin speech. 308-320 - Diego H. Milone, Antonio J. Rubio:
Prosodic and accentual information for automatic speech recognition. 321-333 - Yi Hu, Philipos C. Loizou:
A generalized subspace approach for enhancing speech corrupted by colored noise. 334-341 - Peter Eneroth:
Joint filterbanks for echo cancellation and audio coding. 342-354 - Doh-Suk Kim:
Perceptual phase quantization of speech. 355-364 - Dai Yang, Hongmei Ai, Chris Kyriakakis, C.-C. Jay Kuo:
High-fidelity multichannel audio coding with Karhunen-Loeve transform. 365-380 - In-Kwon Yeo, Hyoung Joong Kim:
Modified patchwork algorithm: a novel audio watermarking scheme. 381-386
Volume 11, Number 5, September 2003
- Zoran Cvetkovic, James D. Johnston:
Nonuniform oversampled filter banks for audio signal processing. 393-399 - Farshad Lahouti, Amir K. Khandani:
Quantization of LSF parameters using a trellis modeling. 400-412 - Çagri Özgenc Etemoglu, Vladimir Cuperman:
Matching pursuits sinusoidal speech coding. 413-424 - Hui Jiang, Chin-Hui Lee:
A new approach to utterance verification based on neighborhood information in model space. 425-434 - Hong Kook Kim, Richard C. Rose:
Cepstrum-domain acoustic feature compensation based on decomposition of speech and noise for ASR in noisy environments. 435-446 - Bing Xiang, Toby Berger:
Efficient text-independent speaker verification with structural Gaussian mixture models and neural network. 447-456 - Yi Hu, Philipos C. Loizou:
A perceptually motivated approach for speech enhancement. 457-465 - Israel Cohen:
Noise spectrum estimation in adverse environments: improved minima controlled recursive averaging. 466-475 - James R. Hopgood, Peter J. W. Rayner:
Blind single channel deconvolution using nonstationary signal processing. 476-488 - Nikolaos Mitianoudis, Michael E. Davies:
Audio source separation of convolutive mixtures. 489-497 - Saeed Gazor, Wei Zhang:
A soft voice activity detector based on a Laplacian-Gaussian model. 498-505
Volume 11, Number 6, November 2003
- Frank Baumgarte, Christof Faller:
Binaural cue coding-Part I: psychoacoustic fundamentals and design principles. 509-519 - Christof Faller, Frank Baumgarte:
Binaural cue coding-Part II: Schemes and applications. 520-531 - Rongshan Yu, Chi Chung Ko:
Lossless compression of digital audio using cascaded RLS-LMS prediction. 532-537 - Biqing Wu, Marc Bodson:
Direct adaptive cancellation of periodic disturbances for multivariable plants. 538-548 - Jingdong Chen, Jacob Benesty, Yiteng Huang:
Robust time delay estimation exploiting redundancy among multiple microphones. 549-557 - Jen-Tzung Chien, Chih-Hsien Huang:
Bayesian learning of speech duration models. 558-567 - Li Deng, Jasha Droppo, Alex Acero:
Recursive estimation of nonstationary noise using iterative stochastic approximation for robust speech recognition. 568-580 - Chao-Shih Huang, Hsiao-Chuan Wang, Chin-Hui Lee:
A study on model-based error rate estimation for automatic speech recognition. 581-589 - Jeff Z. Ma, Li Deng:
Efficient decoding strategies for conversational speech recognition using a constrained nonlinear state-space model. 590-602 - Alexandros Potamianos, Shrikanth S. Narayanan:
Robust recognition of children's speech. 603-616 - Doroteo T. Toledano, Luis A. Hernández Gómez, Luis Villarrubia Grande:
Automatic phonetic segmentation. 617-625 - Wai C. Chu:
Window optimization in linear prediction analysis. 626-635 - Cheng-Chieh Lee, Yair Shoham:
Trellis code excited linear prediction (TCELP) speech coding. 636-647 - Ajit V. Rao, Sassan Ahmadi, Jan Linden, Allen Gersho, Vladimir Cuperman, Ryan Heidari:
Pitch adaptive windows for improved excitation coding in low-rate CELP coders. 648-659 - Mohamed Kamal Omar, Mark Hasegawa-Johnson:
Approximately independent factors of speech using nonlinear symplectic transformation. 660-671 - Alexandre Guérin, Gérard Faucon, Régine Le Bouquin-Jeannès:
Nonlinear acoustic echo cancellation based on Volterra filters. 672-683 - Israel Cohen:
Analysis of two-channel generalized sidelobe canceller (GSC) with post-filtering. 684-699 - Firas Jabloun, Benoît Champagne:
Incorporating the human hearing properties in the signal subspace approach for speech enhancement. 700-708 - Iain McCowan, Hervé Bourlard:
Microphone array post-filter based on noise field coherence. 709-716 - Ing Yann Soon, Soo Ngee Koh:
Speech enhancement using 2-D Fourier transform. 717-724 - Ka Fai Cedric Yiu, Xiaoqi Yang, Sven Nordholm, Kok Lay Teo:
Near-field broadband beamformer design via multidimensional semi-infinite-linear programming techniques. 725-732 - Xianxian Zhang, John H. L. Hansen:
CSA-BF: a constrained switched adaptive beamformer for speech enhancement and recognition in real car environments. 733-745 - Yannick Estève, Christian Raymond, Renato de Mori, David Janiszek:
On the use of linguistic consistency in systems for human-computer dialogues. 746-756 - Helen M. Meng, Carmen Wai, Roberto Pieraccini:
The use of belief networks for mixed-initiative dialog modeling. 757-773 - Federico Fontana:
Computation of linear filter networks containing delay-free loops, with an application to the waveguide mesh. 774-782 - Lauri Savioja, Vesa Välimäki:
Interpolated rectangular 3-D digital waveguide mesh algorithms with frequency warping. 783-790 - Tony Gustafsson, Bhaskar D. Rao, Mohan M. Trivedi:
Source localization in reverberant environments: modeling and statistical analysis. 791-803 - Anssi Klapuri:
Multiple fundamental frequency estimation based on harmonicity and spectral smoothness. 804-816 - Yoshikazu Seki, Kiyohide Ito:
Coloration perception depending on sound direction. 817-825 - Darren B. Ward, Eric A. Lehmann, Robert C. Williamson:
Particle filtering algorithms for tracking an acoustic source in a reverberant environment. 826-836
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