Journal: IEEE Transactions on Audio, Speech & Language Processing

Volume 11, Issue 6

509 -- 519Frank Baumgarte, Christof Faller. Binaural cue coding-Part I: psychoacoustic fundamentals and design principles
520 -- 531Christof Faller, Frank Baumgarte. Binaural cue coding-Part II: Schemes and applications
532 -- 537Rongshan Yu, Chi Chung Ko. Lossless compression of digital audio using cascaded RLS-LMS prediction
538 -- 548Biqing Wu, Marc Bodson. Direct adaptive cancellation of periodic disturbances for multivariable plants
549 -- 557Jingdong Chen, Jacob Benesty, Yiteng Huang. Robust time delay estimation exploiting redundancy among multiple microphones
558 -- 567Jen-Tzung Chien, Chih-Hsien Huang. Bayesian learning of speech duration models
568 -- 580Li Deng, James Droppo, Alex Acero. Recursive estimation of nonstationary noise using iterative stochastic approximation for robust speech recognition
581 -- 589Chao-Shih Huang, Hsiao-Chuan Wang, Chin-Hui Lee. A study on model-based error rate estimation for automatic speech recognition
590 -- 602Jeff Z. Ma, Li Deng. Efficient decoding strategies for conversational speech recognition using a constrained nonlinear state-space model
603 -- 616Alexandros Potamianos, Shrikanth S. Narayanan. Robust recognition of children s speech
617 -- 625Doroteo Torre Toledano, Luis A. Hernández Gómez, Luis Villarrubia Grande. Automatic phonetic segmentation
626 -- 635Wai C. Chu. Window optimization in linear prediction analysis
636 -- 647Cheng-Chieh Lee, Yair Shoham. Trellis code excited linear prediction (TCELP) speech coding
648 -- 659A. V. Rao, Sassan Ahmadi, Jan Linden, Allen Gersho, Vladimir Cuperman, Ryan Heidari. Pitch adaptive windows for improved excitation coding in low-rate CELP coders
660 -- 671Mohamed Kamal Omar, Mark Hasegawa-Johnson. Approximately independent factors of speech using nonlinear symplectic transformation
672 -- 683Alexandre Guerin, Gérard Faucon, Régine Le Bouquin-Jeannès. Nonlinear acoustic echo cancellation based on Volterra filters
684 -- 699Israel Cohen. Analysis of two-channel generalized sidelobe canceller (GSC) with post-filtering
700 -- 708Firas Jabloun, Benoît Champagne. Incorporating the human hearing properties in the signal subspace approach for speech enhancement
709 -- 716Iain McCowan, Hervé Bourlard. Microphone array post-filter based on noise field coherence
717 -- 724Ing Yann Soon, Soo Ngee Koh. Speech enhancement using 2-D Fourier transform
725 -- 732Ka Fai Cedric Yiu, Xiaoqi Yang, Sven Nordholm, Kok Lay Teo. Near-field broadband beamformer design via multidimensional semi-infinite-linear programming techniques
733 -- 745Xianxian Zhang, John H. L. Hansen. CSA-BF: a constrained switched adaptive beamformer for speech enhancement and recognition in real car environments
746 -- 756Yannick Estève, Christian Raymond, Renato de Mori, D. Janiszek. On the use of linguistic consistency in systems for human-computer dialogues
757 -- 773Helen M. Meng, Carmen Wai, Roberto Pieraccini. The use of belief networks for mixed-initiative dialog modeling
774 -- 782Federico Fontana. Computation of linear filter networks containing delay-free loops, with an application to the waveguide mesh
783 -- 790Lauri Savioja, Vesa Välimäki. Interpolated rectangular 3-D digital waveguide mesh algorithms with frequency warping
791 -- 803Tony Gustafsson, Bhaskar D. Rao, Mohan M. Trivedi. Source localization in reverberant environments: modeling and statistical analysis
804 -- 816Anssi Klapuri. Multiple fundamental frequency estimation based on harmonicity and spectral smoothness
817 -- 825Yoshikazu Seki, Kiyohide Ito. Coloration perception depending on sound direction
826 -- 836Darren B. Ward, Eric A. Lehmann, Robert C. Williamson. Particle filtering algorithms for tracking an acoustic source in a reverberant environment

Volume 11, Issue 5

393 -- 399Zoran Cvetkovic, James D. Johnston. Nonuniform oversampled filter banks for audio signal processing
400 -- 412Farshad Lahouti, Amir K. Khandani. Quantization of LSF parameters using a trellis modeling
413 -- 424Cagri Ö. Etemoglu, Vladimir Cuperman. Matching pursuits sinusoidal speech coding
425 -- 434Hui Jiang, Chin-Hui Lee. A new approach to utterance verification based on neighborhood information in model space
435 -- 446Hong Kook Kim, R. C. Rose. Cepstrum-domain acoustic feature compensation based on decomposition of speech and noise for ASR in noisy environments
447 -- 456Bing Xiang, Toby Berger. Efficient text-independent speaker verification with structural Gaussian mixture models and neural network
457 -- 465Yi Hu, Philipos C. Loizou. A perceptually motivated approach for speech enhancement
466 -- 475Israel Cohen. Noise spectrum estimation in adverse environments: improved minima controlled recursive averaging
476 -- 488James R. Hopgood, Peter J. W. Rayner. Blind single channel deconvolution using nonstationary signal processing
489 -- 497Nikolaos Mitianoudis, Michael E. Davies. Audio source separation of convolutive mixtures
498 -- 505Saeed Gazor, Wei Zhang. A soft voice activity detector based on a Laplacian-Gaussian model

Volume 11, Issue 4

297 -- 0Isabel Trancoso. From the editor-in-chief
298 -- 307Xiaodong He, Yunxin Zhao. Fast model selection based speaker adaptation for nonnative speech
308 -- 320Sin-Horng Chen, Wen-Hsing Lai, Yih-Ru Wang. A new duration modeling approach for Mandarin speech
321 -- 333Diego H. Milone, Antonio J. Rubio. Prosodic and accentual information for automatic speech recognition
334 -- 341Yi Hu, Philipos C. Loizou. A generalized subspace approach for enhancing speech corrupted by colored noise
342 -- 354Peter Eneroth. Joint filterbanks for echo cancellation and audio coding
355 -- 364Doh-Suk Kim. Perceptual phase quantization of speech
365 -- 380Dai Yang, Hongmei Ai, C. Kyriakakis, C. C. Jay Kuo. High-fidelity multichannel audio coding with Karhunen-Loeve transform
381 -- 386In-Kwon Yeo, Hyoung Joong Kim. Modified patchwork algorithm: a novel audio watermarking scheme

Volume 11, Issue 3

165 -- 174Christopher J. C. Burges, John C. Platt, Soumya Jana. Distortion discriminant analysis for audio fingerprinting
175 -- 183Aggelos Pikrakis, Sergios Theodoridis, Dimitris Kamarotos. Recognition of isolated musical patterns using context dependent dynamic time warping
184 -- 192Jürgen Tchorz, Birger Kollmeier. SNR estimation based on amplitude modulation analysis with applications to noise suppression
193 -- 203Sven E. Nordholm, Ingvar Claesson, Nedelko Grbic. Performance limits in subband beamforming
204 -- 215Futoshi Asano, S. Ikeda, M. Ogawa, Hideki Asoh, Nobuhiko Kitawaki. Combined approach of array processing and independent component analysis for blind separation of acoustic signals
216 -- 228Ran Yaniv, David Burshtein. An enhanced dynamic time warping model for improved estimation of DTW parameters
229 -- 241Mingyang Wu, DeLiang Wang, G. J. Brown. A multipitch tracking algorithm for noisy speech
242 -- 254Davide Rocchesso, Julius O. Smith III. Generalized digital waveguide networks
255 -- 266Stefan Bilbao, Julius O. Smith III. Finite difference schemes and digital waveguide networks for the wave equation: stability, passivity, and numerical dispersion
267 -- 277Jerome R. Bellegarda, Kim E. A. Silverman. Natural language spoken interface control using data-driven semantic inference
278 -- 293Lester S. H. Ngia. Recursive identification of acoustic echo systems using orthonormal basis functions

Volume 11, Issue 2

109 -- 116Shoko Araki, Ryo Mukai, Shoji Makino, Tsuyoki Nishikawa, Hiroshi Saruwatari. The fundamental limitation of frequency domain blind source separation for convolutive mixtures of speech
117 -- 129Sean A. Ramprashad. The multimode transform predictive coding paradigm
130 -- 142Anand D. Subramaniam, Bhaskar D. Rao. PDF optimized parametric vector quantization of speech line spectral frequencies
143 -- 158Koen Eneman, Marc Moonen. Iterated partitioned block frequency-domain adaptive filtering for acoustic echo cancellation

Volume 11, Issue 1

1 -- 13Filiz Basbug, Kumar Swaminathan, Srinivas Nandkumar. Noise reduction and echo cancellation front-end for speech codecs
14 -- 23J. M. de Haan, Nedelko Grbic, Ingvar Claesson, Sven E. Nordholm. Filter bank design for subband adaptive microphone arrays
24 -- 35Hong-Kwang Jeff Kuo, Chin-Hui Lee. Discriminative training of natural language call routers
36 -- 44Axel Nackaerts, Bart De Moor, Rudy Lauwereins. A formant filtered physical model for wind instruments
45 -- 53Ming Zhang, Hui Lan, Wee Ser. A robust online secondary path modeling method with auxiliary noise power scheduling strategy and norm constraint manipulation
54 -- 60Martin Bouchard. Multichannel affine and fast affine projection algorithms for active noise control and acoustic equalization systems
61 -- 69Upendra V. Chaudhari, Jiri Navratil, Stéphane H. Maes. Multigrained modeling with pattern specific maximum likelihood transformations for text-independent speaker recognition
70 -- 79Jen-Tzung Chien. Linear regression based Bayesian predictive classification for speech recognition
80 -- 87Chulhee Lee, Donghoon Hyun, Euisun Choi, Jinwook Go, Chungyong Lee. Optimizing feature extraction for speech recognition
88 -- 99Jonas Lindblom, Jonas Samuelsson. Bounded support Gaussian mixture modeling of speech spectra
100 -- 103Raymond N. J. Veldhuis, Esther Klabbers. On the computation of the Kullback-Leibler measure for spectral distances