Volume 11, Number 1, January 2003

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  Filiz Basbug, Kumar Swaminathan, Srinivas Nandkumar: Noise reduction and echo cancellation front-end for speech codecs. 1-13
• 
  J. M. 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, Jiri Navratil, 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

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  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, S. Ikeda, M. 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
• 
  Cagri 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 Torre 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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