IEEE 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
  
• 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

• 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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