Abstract is missing.
- Automatic indexing of lecture speech by extracting topic-independent discourse markersTatsuya Kawahara, Masahiro Hasegawa. 1-4 [doi]
- Audio Indexing of Arabic broadcast newsJay Billa, Mohammed Noamany, Amit Srivastava, D. Liu, R. Stone, J. Xu, John Makhoul, Francis Kubala. 5-8 [doi]
- Automatic speech summarization applied to English broadcast news speechChiori Hori, Sadaoki Furui, Robert Malkin, Hua Yu, Alex Waibel. 9-12 [doi]
- Transcribing audio-video archivesClaude Barras, Alexandre Allauzen, Lori Lamel, Jean-Luc Gauvain. 13-16 [doi]
- Learning new user formulations in automatic Directory AssistanceCosmin Popovici, M. Andorno, Pietro Laface, Luciano Fissore, Mario Nigra, Claudio Vair. 17-20 [doi]
- A scalable architecture for Directory Assistance automationPremkumar Natarajan, Rohit Prasad, Richard M. Schwartz, John Makhoul. 21-24 [doi]
- Combination of boosting and discriminative training for natural language call steering systemsImed Zitouni, Hong-Kwang Jeff Kuo, Chin-Hui Lee. 25-28 [doi]
- Combining prior knowledge and boosting for call classification in spoken language dialogueMarie Rochery, Robert E. Schapire, Mazin G. Rahim, Narendra K. Gupta, Giuseppe Riccardi, Srinivas Bangalore, Hiyan Alshawi, Shona Douglas. 29-32 [doi]
- Extracting descriptive noun phrases from conversational speechOwen Kimball, Rukmini Iyer, Herbert Gish, Scott Miller, Fred Richardson. 33-36 [doi]
- Adaptive language models for spoken dialogue systemsRoger Argiles Solsona, Eric Fosler-Lussier, Hong-Kwang Jeff Kuo, Alexandros Potamianos, Imed Zitouni. 37-40 [doi]
- Evaluation of spoken language grammar learning in the ATIS domainYe-Yi Wang, Alex Acero. 41-44 [doi]
- ASR system modeling for automatic evaluation and optimization of dialogue systemsOlivier Pietquin, Steve Renals. 45-48 [doi]
- Digit recognition in noisy environments via a sequential GMM/SVM systemShai Fine, George Saon, Ramesh A. Gopinath. 49-52 [doi]
- Robust speech recognition in Noisy Environments: The 2001 IBM spine evaluation systemBrian Kingsbury, George Saon, Lidia Mangu, Mukund Padmanabhan, Ruhi Sarikaya. 53-56 [doi]
- Uncertainty decoding with SPLICE for noise robust speech recognitionJasha Droppo, Alex Acero, Li Deng. 57-60 [doi]
- Accounting for uncertainity in observations: A new paradigm for Robust Automatic Speech RecognitionTrausti T. Kristjansson, Brendan J. Frey. 61-64 [doi]
- Missing data speech recognition in reverberant conditionsKalle J. Palomäki, Guy J. Brown, Jon P. Barker. 65-68 [doi]
- Combining the union model and missing feature method to improve noise robustness in ASRPeter Jancovic, Ji Ming. 69-72 [doi]
- Asymmetrical Support Vector Machines and applications in speech processingPeng Ding, Zhenbiao Chen, Yang Liu, Bo Xu. 73-76 [doi]
- Using SVMS and discriminative models for speech recognitionN. D. Smith, Mark J. F. Gales. 77-80 [doi]
- Maximum mutual information based acoustic-features representation of phonological features for speech recognitionMohamed Kamal Omar, Mark Hasegawa-Johnson. 81-84 [doi]
- Improved cross-task recognition using MMIE trainingR. Cordoba, Philip C. Woodland, Mark J. F. Gales. 85-88 [doi]
- Direct models for phoneme recognitionAnton Lilchododev, Yuqing Gao. 89-92 [doi]
- Structurally discriminative graphical models for automatic speech recognition - results from the 2001 Johns Hopkins Summer WorkshopGeoffrey Zweig, Jeff Bilmes, T. Richardson, Karim Filali, Karen Livescu, P. Xu, K. Jackson, Yigal Brandman, Eric D. Sandness, E. Holtz, J. Torres, B. Byrne. 93-96 [doi]
- A new algorithm for fast discriminative trainingQi Li, Biing-Hwang Juang. 97-100 [doi]
- An alternative approach of finding competing hypotheses for better minimum classification error trainingYik-Cheung Tam, Brian Mak. 101-104 [doi]
- Minimum Phone Error and I-smoothing for improved discriminative trainingDaniel Povey, Philip C. Woodland. 105-108 [doi]
- Utterance-level boosting of HMM speech recognizersCarsten Meyer. 109-112 [doi]
- A dynamic in-search discriminative training approach for large vocabulary speech recognitionHui Jiang 0001, Olivier Siohan, Frank K. Soong, Chin-Hui Lee. 113-116 [doi]
- Classifier design for verification of multi-class recognition decisionTomoko Matsui, Frank K. Soong, Biing-Hwang Juang. 117-120 [doi]
- Speaker indexing for retrieval of voicemail messagesDelphine Charlet. 121-124 [doi]
- Co-channel speaker segment separationBrett Y. Smolenski, Robert E. Yantorno, Daniel S. Benincasa, Stanley J. Wenndt. 125-128 [doi]
- Speaker detection and tracking for telephone transactionsJack McLaughlin, Douglas A. Reynolds. 129-132 [doi]
- Noise-robust open-set speaker recognition using noise-dependent Gaussian mixture classifierYifan Gong. 133-136 [doi]
- Automatic estimation of one's age with his/her speech based upon acoustic modeling techniques of speakersNobuaki Minematsu, Mariko Sekiguchi, Keikichi Hirose. 137-140 [doi]
- Using prosodic and lexical information for speaker identificationFrederick Weber, Linda Manganaro, Barbara Peskin, Elizabeth Shriberg. 141-144 [doi]
- Speaker identification using multilingual phone stringsQin Jin, Tanja Schultz, Alex Waibel. 145-148 [doi]
- Gender-dependent phonetic refraction for speaker recognitionWalter D. Andrews, Mary A. Kohler, Joseph P. Campbell, John J. Godfrey, Jaime Hernandez-Cordero. 149-152 [doi]
- Speaker identification using online, frame dependent, and diffusive variance adaptationScott Axelrod. 153-156 [doi]
- Speaker recognition using adaptively boosted decision tree classifierSay Wei Foo, Eng Guan Lim. 157-160 [doi]
- Generalized linear discriminant sequence kernels for speaker recognitionWilliam M. Campbell. 161-164 [doi]
- Mode adaptive Unequal Error Protection for transform predictive speech and audio codersHervé Taddei, Sean A. Ramprashad, Carl-Erik W. Sundberg, Hui-Ling Lou. 165-168 [doi]
- A packet loss recovery method using packet arrived behind the playout time for CELP decodingMasahiro Serizawa, Hironori Ito. 169-172 [doi]
- Packet loss concealment based on sinusoidal extrapolationJonas Lindblom, Per Hedelin. 173-176 [doi]
- Multiple description speech coding with diversitiesXin Zhong, Biing-Hwang Juang. 177-180 [doi]
- Pyramid CELP: Embedded speech coding for packet communicationsChristoph Erdmann, David Bauer, Peter Vary. 181-184 [doi]
- Context-based error recovery technique for GSM AMR speech codecM. K. Vinay, P. V. Suresh Babu. 185-188 [doi]
- Noise adaptive speech recognition in time-varying noise based on sequential kullback proximal algorithmKaisheng Yao, Kuldip K. Paliwal, Satoshi Nakamura. 189-192 [doi]
- Comparing Jacobian adaptation with cepstral mean normalization and parallel model combination for noise robust speech recognitionKimmo Parssinen, Petri Salmela, Mikko Harju, Imre Kiss. 193-196 [doi]
- Jacobian joint adaptation to noise, channel and vocal tract lengthHiroshi Shimodaira, Nobuyoshi Sakai, Mitsuru Nakai, Shigeki Sagayama. 197-200 [doi]
- Dynamic estimation of a noise over estimation factor for Jacobian-based adaptationChristophe Cerisara, Jean-Claude Junqua, Luca Rigazio. 201-204 [doi]
- Efficient second-order adaptation for large vocabulary distributed speech recognitionRobert W. Morris, Michael E. Deisher. 205-208 [doi]
- Cepstrum-domain model combination based on decomposition of speech and noise for noisy speech recognitionHong Kook Kim, Richard C. Rose. 209-212 [doi]
- Automatic confidence score mapping for adapted speech recognition systemsAnanth Sankar, Ashvin Kalman. 213-216 [doi]
- Confidence scoring for accurate HMM-based word recognition by using SM-based monophone score normalizationTakaharu Sato, Muhammad Ghulam, Takashi Fukuda, Tsuneo Nitta. 217-220 [doi]
- Confidence scoring based on backward language modelsJacques Duchateau, Kris Demuynck, Patrick Wambacq. 221-224 [doi]
- A concept graph based confidence measureKadri Hacioglu, Wayne Ward. 225-228 [doi]
- Probabilistic integration of multiple confidence measures and context information for concept verificationYi-Chung Lin, Huei-Ming Wang. 229-232 [doi]
- Estimating semantic confidence for spoken dialogue systemsSameer S. Pradhan, Wayne Ward. 233-236 [doi]
- An upper bound on the quality of artificial bandwidth extension of narrowband speech signalsPeter Jax, Peter Vary. 237-240 [doi]
- Enhancing quality of CELP coded speech via wideband extension by using voicing GMM interpolation and HNM re-synthesisDar Ghulam Raza, Cheung-fat Chan. 241-244 [doi]
- Speech signal band width extension and noise removal using subband HMNMitsuhiro Hosoki, Takayuki Nagai, Akira Kurematsu. 245-248 [doi]
- Gender-dependent and speaker-dependent speech enhancementIlyas Potamitis, Nikos Fakotakis, George Kokkinakis. 249-252 [doi]
- Speech enhancement using MMSE short time spectral estimation with gamma distributed speech priorsRainer Martin. 253-256 [doi]
- Speech enhancement based on auditory spectral changeThomas F. Quatieri, Robert B. Dunn. 257-260 [doi]
- Improved spectral tracking using interpolated linear prediction parametersWesley Pereira, Peter Kabal. 261-264 [doi]
- Improvement of the restricted temporal decomposition method for line spectral frequency parametersPhu Chien Nguyen, Masato Akagi. 265-268 [doi]
- Circular LPC modeling and constant pitch transform for accurate speech analysis and high quality speech synthesisS. Shukla, Ali Erdem Ertan, Thomas P. Barnwell III. 269-272 [doi]
- The relation between speech segment selectivity and source localization accuracyParham Aarabi, Albarz Mahdavi. 273-276 [doi]
- Joint optimization of model and excitation in parametric speech codersKhosrow Lashkari, Toshio Miki. 277-280 [doi]
- Robust formant tracking in noiseIan C. Bruce, Neel V. Karkhanis, Eric D. Young, Murray B. Sachs. 281-284 [doi]
- Noise-robust cellular phone speech recognition using codec-adapted speech and noise modelsTsuneo Kato, Masaki Naito, Tohru Shimizu. 285-288 [doi]
- On the joint use of noise reduction and MLLR adaptation for in-car hands-free speech recognitionMarco Matassoni, Maurizio Omologo, Alfiero Santarelli, Piergiorgio Svaizer. 289-292 [doi]
- Unified speech recognition for the landline and wireless environmentsRafid A. Sukkar, Rathi Chengalvarayan, John Jacob. 293-296 [doi]
- Robust HMM-based speech/music segmentationJitendra Ajmera, Iain McCowan, Hervé Bourlard. 297-300 [doi]
- Porting an audio partitioner across domainsMauro Cettolo. 301-304 [doi]
- Continuous audio-visual digit recognition using decision fusionGeorg Meyer, Jeff Mulligan. 305-308 [doi]
- Robust bi-modal speech recognition based on state synchronous modeling and stream weight optimizationSatoshi Nakamura, Ken'ichi Kumatani, Satoshi Tamura. 309-312 [doi]
- Enhanced posteriors bias prediction for robust multi-stream ASR combining voicing and estimate reliabilitiesHervé Glotin. 313-316 [doi]
- Robust phoneme discrimination using acoustic waveformsZoran Cvetkovic, Baltasar Beferull-Lozano, Andreas Buja. 317-320 [doi]
- Phone based voice activity detection using online Bayesian adaptation with conjugate normal distributionsJianping Zhang, Wayne Ward, Bryan L. Pellom. 321-324 [doi]
- Discriminative training of language models for speech recognitionHong-Kwang Jeff Kuo, Eric Fosler-Lussier, Hui Jiang 0001, Chin-Hui Lee. 325-328 [doi]
- Fundamental frequency estimation based on instantaneous frequency amplitude spectrumTomohiro Tanaka, Takao Kobayashi, Dhany Arifianto, Takashi Masuko. 329-332 [doi]
- Pitch determination and voice quality analysis using Subharmonic-to-Harmonic RatioXuejing Sun. 333-336 [doi]
- Spline-based continuous-time pitch estimationAndrei Jefremov, W. Bastiaan Kleijn. 337-340 [doi]
- Pitch estimation based on Circular AMDFWenyao Zhang, Gang Xu, Yuguo Wang. 341-344 [doi]
- Robust pitch determination using nonlinear state-space embeddingDmitry E. Terez. 345-348 [doi]
- The DYPSA algorithm for estimation of glottal closure instants in voiced speechAnastasis Kounoudes, Patrick A. Naylor, Mike Brookes. 349-352 [doi]
- Robust pitch tracking in the car environmentHolger Quast, Olaf Schreiner, Manfred R. Schroeder. 353-356 [doi]
- A statistical pitch detection algorithmYih-Ru Wang, I.-Je Wong, Teng-Chun Tsao. 357-360 [doi]
- Yet Another Algorithm for Pitch TrackingKavita Kasi, Stephen A. Zahorian. 361-364 [doi]
- Recovery of the harmonic fundamental from the mixed partial derivatives of the STFT phaseDouglas J. Nelson. 365-368 [doi]
- A multi-pitch tracking algorithm for noisy speechMingyang Wu, DeLiang Wang, Guy J. Brown. 369-372 [doi]
- Data-driven temporal filters for robust features in speech recognition obtained via Minimum Classification Error (MCE)Jeih-Weih Hung, Lin-Shan Lee. 373-376 [doi]
- Modulation features for speech recognitionDimitrios Dimitriadis, Petros Maragos, Alexandros Potamianos. 377-380 [doi]
- Discriminative auditory features for robust speech recognitionBrian Mak, Yik-Cheung Tam, Peter Qi Li. 381-384 [doi]
- Perceptual speech modeling for noisy speech recognitionChung-Hsien Wu, Yu-Hsien Chiu, Huigan Lim. 385-388 [doi]
- Acoustic analysis and recognition of whispered speechTaisuke Ito, Kazuya Takeda, Fumitada Itakura. 389-392 [doi]
- Blind channel estimation based on speech correlation structureYounes Souilmi, Luca Rigazio, Patrick Nguyen, David Kryze, Jean-Claude Junqua. 393-396 [doi]
- Noise-robust speech recognition using a new spectral estimation method "PHASOR"Kiyoaki Aikawa, Kentaro Ishizuka. 397-400 [doi]
- Non-linear transformations of the feature space for robust Speech RecognitionÁngel de la Torre, José C. Segura, M. Carmen Benítez, Antonio M. Peinado, Antonio J. Rubio. 401-404 [doi]
- Noise robust speech recognition using Gaussian basis functions for non-linear likelihood function approximationChris Pal, Brendan J. Frey, Trausti T. Kristjansson. 405-408 [doi]
- VTS residual noise compensationJosé C. Segura, M. Carmen Benítez, A. de la Torre, Stéphane Dupont, Antonio J. Rubio. 409-412 [doi]
- A comparative analysis of UK and US English accents in recognition and synthesisQin Yan, Saeed Vaseghi. 413-416 [doi]
- Improved duration control for speech synthesis using a multigram language modelMatthias Eichner, Matthias Wolff, Rüdiger Hoffmann. 417-420 [doi]
- Towards a multilingual prosody model for text-to-speechOliver Jokisch, Hongwei Ding, Hans Kruschke. 421-424 [doi]
- Pitch and speech-rate conversion using envelope modulation modelingKazuaki Yoshida, Michiko Kazama, Mikio Tohyama. 425-428 [doi]
- A phase generation method for speech reconstruction from spectral envelope and pitch intervalsHong-Goo Kang, Hong Kook Kim. 429-432 [doi]
- Real-time speech synthesis on an ultra low-resource, programmable DSP systemHamid Sheikhzadeh, Etienne Cornu, Robert L. Brennan, Todd Schneider. 433-436 [doi]
- Determination of articulatory movements from speech acoustics using an HMM-based speech production modelSadao Hiroya, Masaaki Honda. 437-440 [doi]
- Similarity structure in perceptual and physical measures for visual Consonants across talkersJintao Jiang, Abeer Alwan, Lyme E. Bernstein, Edward T. Auer, Patricia A. Keating. 441-444 [doi]
- Articulatory speech synthesis based upon fluid dynamic principlesJun Huang, Stephen E. Levinson, Donald Davis, Scott Slimon. 445-448 [doi]
- Analysis by synthesis of FM modulation and aspiration noise components in pathological voicesBrian Gabelman, Abeer Alwan. 449-452 [doi]
- Domain adaptation for TTS systemsMin Chu, Chun Li, Hu Peng, Eric Chang. 453-456 [doi]
- The synthesis unit generation algorithm for Mandarin TTSShaw-Hwa Hwang, Cheng-Yu Yei. 457-460 [doi]
- Robust splicing costs and efficient search with BMM Models for concatenative speech synthesisIvan Bulyko, Mari Ostendorf, Jeff Bilmes. 461-464 [doi]
- Unit selection algorithm for Japanese speech synthesis based on both phoneme unit and diphone unitTomoki Toda, Hisashi Kawai, Minoru Tsuzaki, Kiyohiro Shikano. 465-468 [doi]
- Synthesis of unseen context and spectral and pitch contour smoothing in concatenated text to speech synthesisPhuay Hui Low, Saeed Vaseghi. 469-472 [doi]
- Speech segment selection for concatenative synthesis based on prosody-aligned distance measureChih-Chung Kuo, Chi-Shiang Kuo. 473-476 [doi]
- A data-driven method for input feature selection within neural prosody generationÇaglayan Erdem, Hans-Georg Zimmermann. 477-480 [doi]
- Corpus based extraction of quantitative prosodic parameters of stress groups in SpanishDavid Escudero Mancebo, Valentín Cardeñoso-Payo, Antonio Bonafonte. 481-484 [doi]
- Assigning phrase accent to Chinese Text-to-Speech systemYao Qian, Fang Chen. 485-488 [doi]
- Generation of Chinese prosodic phrasing rules by an extension matrix algorithmWeijun Chen, Fuzong Lin, Jianmin Li, Bo Zhang. 489-492 [doi]
- Autoregressive acoustical modelling of free field cough soundAnnemie Van Hirtum, Daniel Berckmans, Kris Demuynck, Dirk Van Compernolle. 493-496 [doi]
- Analysis of syllable duration models for Mandarin speechWen-Hsing Lai, Sin-Homg Chen. 497-500 [doi]
- Study on prosodic boundary location in Chinaese mandarinWeixiang Hu, Taiyi Huang, Bo Xu. 501-504 [doi]
- Analysis of pitch contour of Thai tone using Fujisaki's modelPusadee Seresangtakul, Tomio Takara. 505-508 [doi]
- A method for automatic extraction of model parameters from fundamental frequency contours of speechShuichi Narusawa, Nobuaki Minematsu, Keikichi Hirose, Hiroya Fujisaki. 509-512 [doi]
- Robust parameters for automatic segmentation of speechA. K. V. Sai Jayram, V. Ramasubramanian, T. V. Sreenivas. 513-514 [doi]
- A simple approach to non-uniform vowel normalizationSrinivasan Umesh, S. V. Bharath Kumar, M. K. Vinay, Rajesh Sharma, Rohit Sinha. 517-520 [doi]
- DCT based pseudo complex cepstrumR. Muralishankar, A. G. Ramakrishnan. 521-524 [doi]
- Gaussian mixture model based mutual information estimation between frequency bands in speechMattias Nilsson, Harald Gustaftson, Soren Vang Andersen, W. Bastiaan Kleijn. 525-528 [doi]
- On the approximate W-disjoint orthogonality of speechScott Rickard, Özgür Yilmaz. 529-532 [doi]
- Speech analysis and feature extraction using chaotic modelsVassilis Pitsikalis, Petros Maragos. 533-536 [doi]
- Signal subspace speech enhancement with perceptual post-filteringMark Klein, Peter Kabal. 537-540 [doi]
- Speech enhancement using excitation source informationB. Yegnanarayana, S. R. Mahadeva Prasanna, K. Sreenivasa Rao. 541-544 [doi]
- Noise suppression with synthesis windowing and pseudo noise injectionAkihiko Sugiyama, Thanh Phong Hua, Masanori Kato, Masahiro Serizawa. 545-548 [doi]
- Speech enhancement by harmonic modeling via map pitch trackingJoseph Tabrikian, Shlomo Dubnov, Yulya Dickalov. 549-552 [doi]
- Monaural speech segregation based on pitch tracking and amplitude modulationGuoning Hu, DeLiang L. Wang. 553-556 [doi]
- Acoustic diversity for improved speech recognition in reverberant environmentsBradford W. Gillespie, Les E. Atlas. 557-560 [doi]
- Speech enhancement for non-stationary noise environment by adaptive wavelet packetSungwook Chang, Y. Kwon, Sung-il Yang, I.-jae Kim. 561-564 [doi]
- Noise suppression based on approximate KLT with wavelet packet expansionChung-Hsien Yang, Jhing-Fa Wang. 565-568 [doi]
- A perceptual signal subspace approach for speech enhancement in colored noiseFiras Jabloun, Benoît Champagne. 569-572 [doi]
- A subspace approach for enhancing speech corrupted by colored noiseYi Hu, Philipos C. Loizou. 573-576 [doi]
- Fast model adaptation and complexity selection for nonnative English speakersXiaodong He, Yunxin Zhao. 577-580 [doi]
- Rapid adaptation with linear combinations of rank-one matricesVaibhava Goel, Karthik Visweswariah, Ramesh Gopinath. 581-584 [doi]
- Structuring linear transforms for adaptation using training time informationK. Visweswariah, V. Goel, R. Gopinath. 585-588 [doi]
- Non-uniform scaling based speaker normalizationRohit Sinha, Srinivasan Umesh. 589-592 [doi]
- Acoustic-phonetic speech parameters for speaker-independent speech recognitionOm Deshmukh, Carol Y. Espy-Wilson, Amit Juneja. 593-596 [doi]
- Piecewise linear constraints for model space adaptationPatrick Nguyen, Luca Rigazio, Jean-Claude Junqua, Christian Wellekens. 597-600 [doi]
- On maximum mutual information speaker-adapted trainingJohn W. McDonough, Thomas Schaaf, Alex Waibel. 601-604 [doi]
- Supervised adaptation of MCE-trained CDHMMS using minimum classification error linear regressionJian Wu, Qiang Huo. 605-608 [doi]
- Speaker selection training for large vocabulary continuous speech recognitionChao Huang, Tao Chen, Eric Chang. 609-612 [doi]
- Efficient adaptation text design based on the Kullback-Leibler measureXiaodong Cui, Abeer Alwan. 613-616 [doi]
- Unsupervised incremental online adaptation to unknown environment and speakerDongsuk Yook. 617-620 [doi]
- Perception-based selective encryption of G.729 speechAntonio Servetti, Juan Carlos De Martin. 621-624 [doi]
- A methodology for speech codec FEC design and optimisation based upon the ITU-T Rec. P.861 objective speech quality measureJ. A. Gibbs, H. Fikretler. 625-628 [doi]
- A 4 kb/s hybrid MELP/CELP speech coding candidate for ITU standardizationAlan McCree, Jacek Stachurski, Takahiro Unno, Ali Erdem Ertan, Erdal Paksoy, Vishu Viswanathan, Ari Heikkinen, Anssi Rämö, Sakari Himanen, Peter Blocher, Oliver Dressler. 629-632 [doi]
- Generalized analysis-by-synthesis based on system identificationNam Soo Kim, Joon-Hyuk Chang. 633-636 [doi]
- Low complexity recursive coding of spectrum parametersAnand D. Subramaniam, William R. Gardner, Bhaskar D. Rao. 637-640 [doi]
- Spectral quantization by compandingTuraj Zakizadeh Shabestary, Per Hedelin. 641-644 [doi]
- KLT-based classified VQ for the speech signalMoo Young Kim 0001, W. Bastiaan Kleijn. 645-648 [doi]
- A novel quantization scheme for the noise-like component in waveform interpolation speech codingJani Nurminen, Ari Heikkinen, Jukka Saarinen. 649-652 [doi]
- Scalable innovation codingFredrik Norden, Per Hedelin. 653-656 [doi]
- A new bandwidth scalable wideband speech/audio coderKyung Tae Kim, Sung-Kyo Jung, Young-Cheol Park, Dae Hee Youn. 657-660 [doi]
- A time domain reformulation of linear prediction equivalent to the LSP decompositionTomas Bäckström, Paavo Alku, W. Bastiaan Kleijn. 661-664 [doi]
- All-pole modeling technique based on the Weighted Sum of the LSP polynomialsPaavo Alku, Tomas Bäckström. 665-668 [doi]
- Evaluation of kernel methods for speaker verification and identificationVincent Wan, Steve Renals. 669-672 [doi]
- Using nonstandard SVM for combination of Speaker Verification and Verbal Information Verification in speaker authentication systemYang Liu, Peng Ding, Bo Xu. 673-676 [doi]
- Speaker verification using text-constrained Gaussian Mixture ModelsDouglas E. Sturim, Douglas A. Reynolds, Robert B. Dunn, Thomas F. Quatieri. 677-680 [doi]
- Short-time Gaussianization for robust speaker verificationBing Xiang, Upendra V. Chaudhari, Jiri Navratil, Ganesh N. Ramaswamy, Ramesh A. Gopinath. 681-684 [doi]
- Automatic speechreading with application to speaker verificationCharles C. Broun, X. Zhang, Russell M. Mersereau, Mark A. Clements. 685-688 [doi]
- A Monte-Carlo method for score normalization in Automatic Speaker Verification using Kullback-Leibler distancesMathieu Ben, Raphaël Blouet, Frédéric Bimbot. 689-692 [doi]
- Improving speaker verification with figure of merit trainingXiaohan Li, Eric Chang, Bei-qian Dai. 693-696 [doi]
- The sphericity measure for cellular speaker verificationRan D. Zilca, Upendra V. Chaudhari, Ganesh N. Ramaswamy. 697-700 [doi]
- Combining stochastic feature transformation and handset identification for telephone-based speaker verificationMan-Wai Mak, Sun-Yuan Kung. 701-704 [doi]
- Fast LM look-ahead for large vocabulary continuous speech recognition using perfect hashingAntonio Cardenal López, Javier Dieguez-Tirado, Carmen García-Mateo. 705-708 [doi]
- Efficient language model lookahead through polymorphic linguistic context assignmentHagen Soltau, Florian Metze, Christian Fügen, Alex Waibel. 709-712 [doi]
- Recent advances in efficient decoding combining on-line transducer composition and smoothed language model incorporationDaniel Willett, Shigeru Katagiri. 713-716 [doi]
- Doing away with the Viterbi approximationKris Demuynck, Dirk Van Compernolle, Patrick Wambacq. 717-720 [doi]
- The 2001 BYBLOS English large vocabulary conversational speech recognition systemSpyros Matsoukas, Thomas Colthurst, Owen Kimball, Alex Solomonoff, Fred Richardson, Carl Quillen, Herbert Gish, Pierre L. Dognin. 721-724 [doi]
- Speaking-rate dependent decoding and adaptation for spontaneous lecture speech recognitionHiroaki Nanjo, Tatsuya Kawahara. 725-728 [doi]
- Analysis on individual differences in automatic transcription of spontaneous presentationsTakahiro Shinozaki, Sadaoki Furui. 729-732 [doi]
- Towards automatic corpus preparation for a German broadcast news transcription systemWolfgang Macherey, Hermann Ney. 733-736 [doi]
- Speech labeling and the most frequent phrase extraction using same section in a presentation speechYoshiaki Itoh, Kazuyo Tanaka. 737-740 [doi]
- Recognition of Chinese names in continuous speech for directory assistance applicationsYuan-Fu Liao, Georg Rose. 741-744 [doi]
- Dynamic generation of proper name pronunciations for directory assistanceFrédéric Béchet, Renato de Mori, Gérard Subsol. 745-748 [doi]
- Flemish accent identification based on formant and duration featuresPieter-Jan Ghesquiere, Dirk Van Compernolle. 749-752 [doi]
- Merging segmental and rhythmic features for Automatic Language IdentificationJérôme Farinas, François Pellegrino, Jean-Luc Rouas, Régine André-Obrecht. 753-756 [doi]
- Language identification using Gaussian mixture model tokenizationPedro A. Torres-Carrasquillo, Douglas A. Reynolds, J. R. Deller. 757-760 [doi]
- Mixed-memory Markov models for Automatic Language IdentificationKatrin Kirchhoff, Sonia Parandekar, Jeff Bilmes. 761-764 [doi]
- Connectionist language modeling for large vocabulary continuous speech recognitionHolger Schwenk, Jean-Luc Gauvain. 765-768 [doi]
- Fast update of latent semantic spaces using a linear transform frameworkJerome R. Bellegarda. 769-772 [doi]
- Language model adaptation through topic decomposition and MDI estimationMarcello Federico. 773-776 [doi]
- Building a topic-dependent maximum entropy model for very large corporaJun Wu, Sanjeev Khudanpur. 777-780 [doi]
- Turn-Based Language Modeling for spoken dialog systemsRuhi Sarikaya, Yuqing Gao, Hakan Erdogan, Michael Picheny. 781-784 [doi]
- Rescoring effectiveness of language models using different levels of knowledge and their integrationWen Wang, Yang Liu, Mary P. Harper. 785-788 [doi]
- Text normalization with varied data sources for conversational speech language modelingSarah E. Schwarm, Mari Ostendorf. 789-792 [doi]
- A low-power, fixed-point, front-end feature extraction for a distributed speech recognition systemBrian Delaney, Nikil Jayant, Mat Hans, Tajana Simunic, Andrea Acquaviva. 793-796 [doi]
- A comparison of front-end configurations for robust speech recognitionBen Milner. 797-800 [doi]
- Increased mfcc filter bandwidth for noise-robust phoneme recognitionMark D. Skowronski, John G. Harris. 801-804 [doi]
- Including detailed information feature in MFCC for large vocabulary contious speech recornitionLei Jia, Bo Xu. 805-808 [doi]
- Hierarchical tandem feature extractionSunil Sivadas, Hynek Hermansky. 809-812 [doi]
- Optimizing feature extraction for English word recognitionEuisun Choi, Donghoon Hyun, Chulhee Lee. 813-816 [doi]
- Towards knowledge-based features for HMM based large vocabulary automatic speech recognitionBenoit Launay, Olivier Siohan, Arun C. Surendran, Chin-Hui Lee. 817-820 [doi]
- On feature extraction by mutual information maximizationKari Torkkola. 821-824 [doi]
- Application of variational Bayesian PCA for speech feature extractionOh-Wook Kwon, Te-Won Lee, Kwokleung Chan. 825-828 [doi]
- A Bayesian approach to speech feature enhancement using the dynamic cepstral priorLi Deng, Jasha Droppo, Alex Acero. 829-832 [doi]
- Mandarin verbal information verificationXiaolong Li, Ke Chen. 833-836 [doi]
- Auditory-based acoustic distinctive features and spectral cues for automatic speech recognition using a multi-stream paradigmHesham Tolba, Sid-Ahmed Selouani, Douglas D. O'Shaughnessy. 837-840 [doi]
- Multi-dialectal Spanish speech recognitionAlbino Nogueiras, Mónica Caballero, Asunción Moreno. 841-844 [doi]
- Context-dependent acoustic modeling using graphemes for large vocabulary speech recognitionStephan Kanthak, Hermann Ney. 845-848 [doi]
- Training of across-word phoneme models for large vocabulary continuous speech recognitionAchim Sixtus, Hermann Ney. 849-852 [doi]
- Maximum entropy and MCE based HMM stream weight estimation for audio-visual ASRGuillaume Gravier, Scott Axelrod, Gerasimos Potamianos, Chalapathy Neti. 853-856 [doi]
- Implementation of automatic capitalisation generation systems for speech inputJi-Hwan Kim, Philip C. Woodland. 857-860 [doi]
- Transcription of out-of-vocabulary words in large vocabulary speech recognition based on phoneme-to-grapheme conversionBart Decadt, Jacques Duchateau, Walter Daelemans, Patrick Wambacq. 861-864 [doi]
- Automatic new word extraction methodQin Shi, Li Qin Shen, Hai Xin Chai. 865-868 [doi]
- Efficient reduction of Gaussian components using MDL criterion for HMM-based speech recognitionKoichi Shinoda, Ken-ichi Iso. 869-872 [doi]
- Compact decision trees with cluster validity for speech recognitionJen-Tzung Chien, Chih-Hsien Huang, Shun-Ju Chen. 873-876 [doi]
- Unsupervised acoustic model trainingLori Lamel, Jean-Luc Gauvain, Gilles Adda. 877-880 [doi]
- A beamforming approach to permutation alignment for multichannel frequency-domain blind speech separationMuhammad Z. Ikram, Dennis R. Morgan. 881-884 [doi]
- Soft constrained subband beamforming for hands-free speech enhancementNedelko Grbic, Sven Nordholm. 885-888 [doi]
- Hands-free continuous speech recognition in noise using a speaker beam-former based on spectrum-entropyGeorge Nokas, Evangelos Dermatas. 889-892 [doi]
- Talker localization in a real acoustic environment based on DOA estimation and statistical sound source identificationTakanobu Nishiura, Satoshi Nakamura, Kiyohiro Shikano. 893-896 [doi]
- Speech recognizer-based microphone array processing for robust hands-free speech recognitionMichael L. Seltzer, Bhiksha Raj, Richard M. Stern. 897-900 [doi]
- Microphone array post-filtering for non-stationary noise suppressionIsrael Cohen, Baruch Berdugo. 901-904 [doi]
- Microphone array post-filter for diffuse noise fieldIain McCowan, Hervé Bourlard. 905-908 [doi]
- Noise reduction using a small-scale microphone array in multi noise source environmentMasato Akagi, Takashi Kago. 909-912 [doi]
- Blind separation of non-linear convolved speech mixturesAthanasios Koutras. 913-916 [doi]
- Bund source separation based on Multi-Stage ICA combining frequency-domain ICA and time-domain ICATsuyoki Nishikawa, Hiroshi Saruwatari, Kiyohiro Shikano. 917-920 [doi]
- Blind source separation based on fast-convergence algorithm using ICA and beamforming for real convolutive mixtureHiroshi Saruwatari, Toshiya Kawamura, Katsuyuki Sawai, Atsunobu Kaminuma, Masao Sakata. 921-924 [doi]
- Adaptation experiments on the SPINE database with the Extended Maximum Likelihood Linear Transformation (EMLLT) modelRamesh Gopinath, Vaibhava Goel, Karthik Visweswariah, Peder A. Olsen. 925-928 [doi]
- Increasing speech recognition robustness with HMM2Katrin Weber, Samy Bengio, Hervé Bourlard. 929-932 [doi]
- Two dissimilarity measures for HMMS and their application in phoneme model clusteringMatti Vihola, Mikko Harju, Petri Salmela, Janne Suontausta, Janne Savela. 933-936 [doi]
- The HMM error modelMark J. F. Gales. 937-940 [doi]
- On application of the Faragó-Lugosi algorithm in speech recognitionWilliam J. J. Roberts, Yariv Ephraim. 941-944 [doi]
- Modeling inverse covariance matrices by basis expansionPeder A. Olsen, Ramesh A. Gopinath. 945-948 [doi]
- Factor analysed hidden Markov modelsAntti-Veikko I. Rosti, Mark J. F. Gales. 949-952 [doi]
- On the use of high order derivatives for high performance alphabet recognitionJoseph di Martino. 953-956 [doi]
- A recognition method with parametric trajectory synthesized using direct relations between static and dynamic feature vector time seriesYasuhiro Minami, Erik McDermott, Atsushi Nakamura, Shigeru Katagiri. 957-960 [doi]
- A mixture linear model with target-directed dynamics for spontaneous speech recognitionJeff Z. Ma, Li Deng. 961-964 [doi]
- A new combined model of statics-dynamics of speechZhijian Ou, Zuoying Wang. 965-968 [doi]
- A sequence-based generalization of mean-field annealing using the Forward/Backward algorithm: Application to image segmentationDavid J. Miller 0001, Piya Bunyaratavej, Qi Zhao. 969-972 [doi]
- Universal approximation of fully complex feed-forward neural networksTaehwan Kim, Tülay Adali. 973-976 [doi]
- A normalised complex backpropagation algorithmAndrew I. Hanna, Danilo P. Mandic. 977-980 [doi]
- Robust on-line Principal Component Analysis based on a fixed-point approachYadunandana N. Rao, Jose C. Principe. 981-984 [doi]
- Time series prediction based on the Relevance Vector Machine with adaptive kernelsJoaquin Quiñonero Candela, Lars Kai Hansen. 985-988 [doi]
- A Bayesian model selection criterion for HMM topology optimizationAlain Biem, Jin-Young Ha, Jayashree Subrahmonia. 989-992 [doi]
- Blind source separation of time-varying instantaneous mixtures using an on-line algorithmKenneth E. Hild II, Deniz Erdogmus, Jose C. Principe. 993-996 [doi]
- Robust blind source separation and dispersing algorithmsPando G. Georgiev, Andrzej Cichocki. 997-1000 [doi]
- Polar coordinate based nonlinear function for frequency-domain blind source separationHiroshi Sawada, Ryo Mukai, Shoko Araki, Shoji Makino. 1001-1004 [doi]
- Is gradient descent appropriate for entropy-based source separation?S. C. Douglas, Geoffrey C. Orsak. 1005-1008 [doi]
- On complex infomax applied to functional MRI dataVince D. Calhoun, Tülay Adali, Godfrey D. Pearlson, J. Pekar. 1009-1012 [doi]
- Location-based sound segregationNicoleta Roman, DeLiang Wang, Guy J. Brown. 1013-1016 [doi]
- On the training of DS-CDMA neural-network receiversJohn D. Matyjas, George N. Karystinos, Stella N. Batalama. 1017-1020 [doi]
- Extracting noise-robust features from audio dataChristopher J. C. Burges, John C. Platt, Soumya Jana. 1021-1024 [doi]
- A new subband non linear prediction coding algorithm for narrowband speech signal: The nADPCMB⊥MLT coding schemeGuido D'Alessandro, Marcos Faúndez-Zanuy, Francesco Piazza. 1025-1028 [doi]
- An artificial neural network-based scheme for robust watermarking of audio signalsHuijuan Yang, Jagdish Chandra Patra, Choong W. Chan. 1029-1032 [doi]
- An RNN-based channel classification for Mandarin speech recognition over GSM/PSTN transmission environmentsWei-Tyng Hong. 1033-1036 [doi]
- A new nonlinear prediction model based on the Recurrent Neural Predictive Hidden Markov Model for speech enhancementJoohim Lee, Changwoo Seo, Ki Yong Lee. 1037-1040 [doi]
- Hierarchical cluster model for perceptual image processingJonathan Randall, Ling Guan, Xing Zhang, Wanqing Li. 1041-1044 [doi]
- A human-perception-like image recognition system based on PAP vector representation with multi resolution conceptMasakazu Yagi, Tadashi Shibata. 1045-1048 [doi]
- Outlier estimation and detection application to skin lesion classificationSigurdur Sigurdsson, Jan Larsen, Lars Kai Hansen, Peter Alshede Philipsen, Hans-Christian Wulf. 1049-1052 [doi]
- Offline recognition of handwritten Chinese characters using Gabor features, CDHMM modeling and MCE trainingYong Ge, Qiang Huo, Zhi-Dan Feng. 1053-1056 [doi]
- Natural gradient learning neural networks for modeling and identification of nonlinear systems with memoryMohamed Ibnkahla, Benoit Pochon. 1057-1060 [doi]
- On sensitivity of neural adaptive filters with respect to the slope parameter of a neuron activation functionWarren Sherliker, Igor R. Krcmar, Milorad M. Bozic, Danilo P. Mandic. 1061-1064 [doi]
- A discriminative training criterion and an associated EM learning algorithmMohamed Afify, Olivier Siohan. 1065-1068 [doi]
- Simultaneous extraction of Principal Components using givens rotations and output variancesDeniz Erdogmus, Yadunandana N. Rao, Jose C. Principe, Jing Zhao, Kenneth E. Hild II. 1069-1072 [doi]
- Fast algorithm for adaptive estimation of principal and minor componentsShan Ouyang, P. C. Ching. 1073-1076 [doi]
- Lip segmentation by fuzzy clustering incorporating with shape functionS. L. Wang, S. H. Leung, W. H. Lau. 1077-1080 [doi]
- A recurrent RBF network for non-linear channel with time-varying characteristicMasayoshi Mimura, Nozomu Hamada, Toshihiro Furukawa. 1081-1084 [doi]
- FFT coefficient memory reduction technique for OFDM applicationsM. Hasan, T. Arslan. 1085-1088 [doi]
- Global stability of adaptive IIR filters for nonstationary signalsMiloje S. Radenkovic, Tamal Bose. 1089-1092 [doi]
- Stochastic analysis for the leaky Delayed LMS algorithm: A new model without invoking the independence assumptionOrlando José Tobias, Rui Seara. 1093-1096 [doi]
- Adaptation with constant gains: Analysis for fast variationsLars Lindbom, Mikael Sternad, Anders Ahlén. 1097-1100 [doi]
- An analysis of the fast subspace tracking algorithm NOjaJonathan H. Manton, Iven M. Y. Mareels, Samir Attallah. 1101-1104 [doi]
- Tracking analysis results for NLMS and APASundar G. Sankaran, A. A. (Louis) Beex. 1105-1108 [doi]
- A new efficient quadratic filter based on the Chen's LMS linear algorithm and its performance analysisMounir Sayadi, Farhat Fnaiech, Samir Sakrani, Mohamed Najim. 1109-1112 [doi]
- Channel equalization for multiuser chaotic communications systemsMahmut Ciftci, Douglas B. Williams. 1113-1116 [doi]
- Nonlinear line echo cancellation using a simplified second order Volterra filterF. Kuch, Walter Kellermann. 1117-1120 [doi]
- Recursive estimation of three-dimensional aircraft position using terrain-aided positioningPer-Johan Nordlund, Fredrik Gustafsson. 1121-1124 [doi]
- Fuzzy weighted median filtersYao Nie, Kenneth E. Barner. 1125-1128 [doi]
- Estimation of a certain class of chaotic signals: An em-based approachCarlos Pantaleón, Luis Vielva, David Luengo, Ignacio Santamaría. 1129-1132 [doi]
- Post-beamforming second-order Volterra filter for nonlinear pulse-echo imagingHui Yao, Pornchai Phukpattaranont, Emad S. Ebbini. 1133-1136 [doi]
- On lower bounds for deterministic parameter estimationPhilippe Forster, Pascal Larzabal. 1137-1140 [doi]
- Intrinsic distance lower bound for unbiased estimators on Riemannian manifoldsJoão Xavier, Victor A. N. Barroso. 1141-1144 [doi]
- A lower bound for the maximum likelihood ratio: Sparse antenna array applicationsYuri I. Abramovich, Nicholas K. Spencer, Alexei Y. Gorokhov. 1145-1148 [doi]
- Robust estimation with parametric score function estimationRamon F. Brcich, Abdelhak M. Zoubir. 1149-1152 [doi]
- Canonical coordinates for reduced-rank estimation of improper complex random vectorsPeter J. Schreier, Louis L. Scharf. 1153-1156 [doi]
- Joint mean and COvariance Matching Estimation Techniques: MCOMETH. Kassem, Philippe Forster, Pascal Larzabal. 1157-1160 [doi]
- Theory and lattice factorization of oversampled linear-phase perfect reconstruction filter banksLu Gan, Kai-Kuang Ma. 1161-1164 [doi]
- An octave-band family of non-redundant directional filter banksPaul S. Hong, Mark J. T. Smith. 1165-1168 [doi]
- DCT-based general structure for linear phase paraunitary filter banksJie Liang, Trac D. Tran. 1169-1172 [doi]
- Design and evaluation of nonuniform DFT filter banks in subband microphone arraysJ. M. de Haan, Nedelko Grbic, Ingvar Claesson, Sven Nordholm. 1173-1176 [doi]
- A fast design method for orthogonal modulated filter banksDidier Pinchon, Pierre Siohan, Cyrille Siclet. 1177-1180 [doi]
- Design of allpass-based non-uniform oversampled DFT filter banksEnisa Galijasevic, Jörg Kliewer. 1181-1184 [doi]
- The generalized discrete fractional fourier transformsSoontorn Oraintara. 1185-1188 [doi]
- Influence of a random integration width on periodic samplingBernard Lacaze, Corinne Mailhes. 1189-1192 [doi]
- An ellipsoidal approximation to the Hadamard product of ellipsoidsR. Lorenz, S. Boyd. 1193-1196 [doi]
- A sampling theorem for the Radon transform of finite complexity objectsIrena Maravic, Martin Vetterli. 1197-1200 [doi]
- Optimum compaction filters for cyclostationary signalsAshish Pandharipande, Soura Dasgupta. 1201-1204 [doi]
- Warped discrete Fourier transform: A new concept in digital signal processingStefan Franz, Sanjit K. Mitra, J. C. Schmidt, Gerhard Doblinger. 1205-1208 [doi]
- Spectral characterization of feedback linear periodically time-varying systemsCarlos Mosquera, Sandro Scalise, Giorgio Taricco. 1209-1212 [doi]
- Time-frequency projection filters: Online implementation, subspace tracking, and application to interference excisionGerald Matz, Franz Hlawatsch. 1213-1216 [doi]
- Effects of dicrete LPTV on stationary signalsDaniel Roviras, Bernard Lacaze, N. Thomas. 1217-1220 [doi]
- 2(ℝ) processesFrancisco M. Garcia, Isabel M. G. Lourtie, Jorge Buescu. 1221-1224 [doi]
- A time-frequency approach to blind deconvolution in multipath underwater channelsN. Martins, S. Jesus, Cedric Gervaise, André Quinquis. 1225-1228 [doi]
- A new approach to time-frequency localized signal designAhmet Kemal Özdemir, Zafer Aydin, Orhan Arikan. 1229-1232 [doi]
- Smooth orthogonal signal extensions for paraunitary tree-structured filter banksMaría Elena Domínguez Jiménez, Nuria González Prelcic. 1233-1236 [doi]
- On two-channel orthogonal and symmetric complex-valued FIR filter banks and their corresponding waveletsXiqi Gao, Truong Q. Nguyen, Gilbert Strang. 1237-1240 [doi]
- Frame bounds for biorthogonal cosine-modulated filter banksAlfred Mertins. 1241-1244 [doi]
- A unified approach for constructing multi-waveletsHüseyin Özkaramanli, Asim Bhatti. 1245-1248 [doi]
- Wavelet optimization for classificationMarie-Françoise Lucas, Christian Doncarli, Eric Hitti, Nicolas Dechamps. 1249-1252 [doi]
- Shiftable, projection-based complex wavelet transformsFelix C. A. Fernandes, Rutger L. van Spaendonck, C. Sidney Burrus. 1253-1256 [doi]
- Deconvolution with wavelet footprints for ill-posed inverse problemsPier Luigi Dragotti, Martin Vetterli. 1257-1260 [doi]
- A bivariate shrinkage function for wavelet-based denoisingLevent Sendur, Ivan W. Selesnick. 1261-1264 [doi]
- Recovering filamentary objects in severely degraded binary images using beamlet-driven partitioningXiaoming Huo, David L. Donoho. 1265-1268 [doi]
- Simultaneous seismic compression and denoising using a lapped transform coderLaurent C. Duval. 1269-1272 [doi]
- The generalized lapped pseudo-biorthogonal transformToshihisa Tanaka, Yukihiko Yamashita. 1273-1276 [doi]
- Theory of fractionally spaced cyclic-prefix equalizersP. P. Vaidyanathan, Bojan Vrcelj. 1277-1280 [doi]
- Amplitude and gain error influence on time error estimation algorithm for time interleaved A/D converter systemJonas Elbornsson, Fredrik Gustafsson, Jan-Erik Eklund. 1281-1284 [doi]
- Bayesian change detection for multi-temporal SAR imagesMartial Coulon, Jean-Yves Tourneret. 1285-1288 [doi]
- Bayesian Optimum Radar Detector in non-Gaussian noiseEmmanuelle Jay, Jean Philippe Ovarlez, David Declercq, Patrick Duvaut. 1289-1292 [doi]
- Embedding and detecting spread-spectrum watermarks under estimation attacksDarko Kirovski, Henrique S. Malvar. 1293-1296 [doi]
- Dual-band land mine detection using a Bayesian approachLennart Svensson, Magnus Lundberg. 1297-1300 [doi]
- Improved multiedge detection and reflectivity estimation for SAR imagesMarie Chabert, Franz Hlawatsch, Jean-Yves Tourneret. 1301-1304 [doi]
- Unicast-based inference of network link delay distributions using mixed finite mixture modelsMeng-Fu Shih, Alfred O. Hero III. 1305-1308 [doi]
- Bayesian deconvolution in nuclear spectroscopy using RJMCMCSirajudeen Gulam Razul, William J. Fitzgerald, Christophe Andrieu. 1309-1312 [doi]
- Detection of abrupt spectral changes using support vector machines an application to audio signal segmentationManuel Davy, Simon J. Godsill. 1313-1316 [doi]
- Mean frequency estimation of narrowband signals and its application to Doppler ultrasound blood velocity waveform estimationKumari L. Fernando, V. John Mathews, Edward B. Clark. 1317-1320 [doi]
- Road vehicle speed estimation from a two-microphone arrayFernando Pérez-González, Roberto López-Valcarce, Carlos Mosquera. 1321-1324 [doi]
- Acoustic velocity measurements in the air by means of Laser Doppler Velocimetry: Cramér-Rao Bounds and Maximum Likelihood EstimationAlain le Duff, Guy Plantier, Jean-Christophe Valière, Richard Perdriau. 1325-1328 [doi]
- Further results in multistage adaptive filteringSuleyman Serdar Kozat, Andrew C. Singer. 1329-1332 [doi]
- A new parameterization for designing stable adaptive IIR filtersGang Li, Lunji Qiu. 1333-1336 [doi]
- The eigen-delay all-pass correction algorithmsJavier J. Tapia, Joseph L. LoCicero, Guillermo E. Atkin. 1337-1340 [doi]
- New approach to multichannel linear prediction problemsJunyoung Nam. 1341-1344 [doi]
- A new open loop delayless subband adaptive filter structureMariane R. Petraglia, Rogerio Guedes Alves, M. N. S. Swamy. 1345-1348 [doi]
- A stable adaptive Hammerstein filter employing partial orthogonalization of the input signalsJanez Jeraj, V. John Mathews, Joel Dubow. 1349-1352 [doi]
- A general adaptive normalised nonlinear-gradient descent algorithm for nonlinear adaptive filtersDanilo P. Mandic, Andrew I. Hanna, Dai I. Kim. 1353-1356 [doi]
- Spectrum estimation of real vector wide sense stationary processes by the Hybrid Steepest Descent MethodKonstantinos Slavakis, Isao Yamada, Kohichi Sakaniwa. 1357-1360 [doi]
- Block time delay estimation by Newton's method with allpass constraintsX. Sun, S. C. Douglas. 1361-1364 [doi]
- New fast preconditioners for Toeplitz-like linear systemsAndrew E. Yagle. 1365-1368 [doi]
- Fast inversion of L-block banded matrices and their inversesAmir Asif, José M. F. Moura. 1369-1372 [doi]
- Optimum error nonlinearities for long adaptive filtersTareq Y. Al-Naffouri, Ali H. Sayed. 1373-1376 [doi]
- Generalized subband decomposition LMS algorithm employing selective partial updatesKutluyil Dogancay, Oguz Tanrikulu. 1377-1380 [doi]
- Gradient based variable forgetting factor nonlinear RLS algorithm using correlation function with nonzero lagsShu Hung Leung, C. F. So. 1381-1384 [doi]
- An improved model for the Normalized LMS algorithm with Gaussian inputs and large number of coefficientsMárcio Holsbach Costa, José Carlos M. Bermudez. 1385-1388 [doi]
- Robust ℋ℞ filtering for linear discrete-time state-space models with uncertain time-varying parametersKarina A. Barbosa, Alexandre Trofino, Carlos E. de Souza. 1389-1392 [doi]
- Optimum leakage factor for the MOV-LMS algorithm in nonlinear modeling and control systemsJosé Carlos M. Bermudez, Márcio Holsbach Costa. 1393-1396 [doi]
- On order statistic least mean square algorithmsRadu Ciprian Bilcu, Pauli Kuosmanen, Karen Egiazarian. 1397-1400 [doi]
- A variable weight mixed-norm adaptive algorithmAzzedine Zerguine, Tyseer Aboulnasr. 1401-1404 [doi]
- Normalized natural gradient adaptive filtering for sparse and non-sparse systemsSteven L. Gay, Scott C. Douglas. 1405-1408 [doi]
- Delayed Error and update scheduling of LMS algorithmsMile Milisavljevic. 1409-1412 [doi]
- Partial-update NLMS algorithms with data-selective updatingStefan Werner, Marcello L. R. de Campos, Paulo S. R. Diniz. 1413-1416 [doi]
- Adaptive update lifting with gradient criteria modeling high-order differencesBéatrice Pesquet-Popescu, Gemma Piella, Henk J. A. M. Heijmans. 1417-1420 [doi]
- Mixed H2/ H∞ algorithm for exponentially windowed adaptive filteringSandip D. Kothari, Ravi N. Banavar, Subhasis Chaudhuri. 1421-1424 [doi]
- A blind components separation procedure for FM signal analysisBraham Barkat, Karim Abed-Meraim. 1425-1428 [doi]
- EM algorithm applied to non-stationary noisy phase signalsAnisse Taleb. 1429-1432 [doi]
- Wigner-Ville and polynomial Wigner-Ville Transforms in the estimation of nonlinear FM signal parametersChristophe De Luigi, Eric Moreau. 1433-1436 [doi]
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- Automatic classification of auto-and cross-terms of time-frequency distributions in antenna arraysLuke A. Cirillo, Abdelhak M. Zoubir, Ning Ma, Moeness G. Amin. 1445-1448 [doi]
- On the signal interference structure generated by modified wigner-ville distributionCornel Ioana, André Quinquis. 1449-1452 [doi]
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- Theory of higher-order Rihaczek spectraAlfred Hanssen, Louis L. Scharf. 1457-1460 [doi]
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- Generalized time-bandwidth product optimal short-time fourier transformationLutfiye Durak, Orhan Arikan. 1465-1468 [doi]
- A fast sinusoidal signal analysis technique for the determination of complex frequenciesMerit Y. Hong. 1469-1472 [doi]
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- Adaptive time delay estimation method with signal selectivityYan Zhang, Chun-Mei Wang, Leslie M. Collins. 1477-1480 [doi]
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- Extension of the matrix bartlett's formula to the third order and to noisy modelsJean Pierre Delmas, Mhammed Loufti. 1485-1488 [doi]
- The multivariate normal inverse Gaussian heavy-tailed distribution; simulation and estimationTor Arne Øigård, Alfred Hanssen. 1489-1492 [doi]
- Multiresolution nonparametric intensity and density estimationRebecca M. Willett, Robert D. Nowak. 1493-1496 [doi]
- On the expected complexity of integer least-squares problemsBabak Hassibi, Haris Vikalo. 1497-1500 [doi]
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- Asymptotic rates of the information transfer ratioSinan Sinanovic, Don H. Johnson. 1505-1508 [doi]
- Optimal training for block transmissions over doubly-selective fading channelsXiaoli Ma, Georgios B. Giannakis, Shuichi Ohno. 1509-1512 [doi]
- Optimal transmit signal design for active sonar/radarSteven M. Kay, John H. Thanos. 1513-1516 [doi]
- On the structure, performance, and applications of recursive all-pass filters with adjustable and linear group delayChris Dick, Fred Harris. 1517-1520 [doi]
- New windows for tunable length fir filter designJean-Jacques Fuchs. 1521-1524 [doi]
- Mixed-norm fir filter optimization using second-order cone programmingDan P. Scholnik. 1525-1528 [doi]
- A nonlinear lowpass filter that eliminates peak attenuationJames McNames, Brahm Goldstein. 1529-1532 [doi]
- Low delay lowpass FIR filters with maximally flat characteristics in the passband and the transmission zeros in the stopbandNaoyui Alkawa, Masamitsu Sato. 1533-1536 [doi]
- Efficient sparse FIR filter designDavide Mattera, Francesco Palmieri, Simon Haykin. 1537-1540 [doi]
- Optimal nonlinear filter to remove random impulses from Gaussian noiseScott V. Notley, J. M. Harte, Stephen J. Elliott. 1541-1544 [doi]
- Digital interpolation in the passband domainRoland Zukunft, Sven Haar, Thomas Magesacher. 1545-1548 [doi]
- Boundary filters design for multiwaveletsTai-Chiu Hsung, Daniel Pak-Kong Lun. 1549-1552 [doi]
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- Closed-form design of allpass fractional delay filtersSoo-Chang Pei, Peng-Hua Wang. 1561-1564 [doi]
- Hierarchical Bayesian classification of chirp signalsChristian Doncarli, Manuel Davy, Jean-Yves Tourneret. 1565-1568 [doi]
- Changepoint detection using reversible jump MCMC methodsS. Suparman, Michel Doisy, Jean-Yves Tourneret. 1569-1572 [doi]
- Changepoint detection in multivariate Poisson distributionsJean-Yves Tourneret, Andre Ferrari, Gérard Letac. 1573-1576 [doi]
- Fast and accurate variance-segmentation of white Gaussian dataZhen Wang, Peter Willett. 1577-1580 [doi]
- Maximum likelihood noise estimation for spectrogram segmentation controlCyril Hory, Nadine Martin. 1581-1584 [doi]
- An active contour model for image segmentation: A variational perspectiveByeong Rae Lee, A. Ben Hamza, Hamid Krim. 1585-1588 [doi]
- Likelihood ratio detection of signals on reverberation noiseA. F. Gualtierotti, Adriana Climescu-Haulica, M. D. Pal. 1589-1592 [doi]
- Bayesian target detection and localization using a dual-mode sensorDarryl Morrell, Ya Xue. 1593-1596 [doi]
- Molecular signal processing and detection in T-cellsJohn F. Keane, Les E. Atlas. 1597-1600 [doi]
- Detection over Multiresolution analysis subspacesNurgun Erdol, Spyros Kyperountas. 1601-1604 [doi]
- Sequential Monte Carlo simulation of dynamical models with slowly varying parameters: Application to audioWilliam Fong, Simon J. Godsill. 1605-1608 [doi]
- Space-Time Block Codes: Trained, blind and semi-blind detectionPetre Stoica, Girish Ganesan. 1609-1612 [doi]
- On the tracking performance of certain algorithms based on a time-varying nonlinear Hammerstein modelAnders E. Nordsjö. 1613-1616 [doi]
- A new importance function for particle filtering and its application to blind detection in flat fading channelsYufei Huang, Petar M. Djuric. 1617-1620 [doi]
- Sequential particle filtering in the presence of additive Gaussian noise with unknown parametersPetar M. Djuric, Joaquín Míguez. 1621-1624 [doi]
- Efficient particle filtering for Jump Markov SystemsChristophe Andrieu, Manuel Davy, Arnaud Doucet. 1625-1628 [doi]
- Stable accelerated adaptive DOA tracking algorithmAkira Sano, Koich Hidaka, Hiromitsu Ohmori. 1629-1632 [doi]
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- Adaptive blind channel identification: Multi-channel least mean square and Newton algorithmsYiteng Huang, Jacob Benesty. 1637-1640 [doi]
- Blind source separation of convolutive nonlinear mixtures by flexible spline nonlinear functionsFabio Milani, Mirko Solazzi, Aurelio Uncini. 1641-1644 [doi]
- Optimum adaptive blind source separation algorithmsGeorge V. Moustakides. 1645-1648 [doi]
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- Blind ML detection of CPM signals via the EMV algorithmHoang Nguyen, Bernard C. Levy. 2457-2460 [doi]
- Utilizing channel coding information in CIVA-based blind sequence detectorsXiaohua Li. 2461-2464 [doi]
- Sequential detector for nonlinear channels with applications to satellite communicationsJayesh H. Kotecha, Petar M. Djuric. 2465-2468 [doi]
- SNR-independent velocity estimation for mobile cellular communications systemsWei Sheng, Steven D. Blostein. 2469-2472 [doi]
- Cramér-Rao Lower bound for geolocation in non-line-of-sight environmentYihong Qi, Hisashi Kobayashi. 2473-2476 [doi]
- Optimal prediction of the mobile-radio fading envelopeRaphael J. Lyman. 2477-2480 [doi]
- Prediction based direct blind desion feedback equalizationHwasun Yoo, Sunghwan Ong, Changeon Kang, Daesik Hong. 2481-2484 [doi]
- Maximum ratio Combining performance with imperfect channel estimatesY. Akyildiz, Bhaskar D. Rao. 2485-2488 [doi]
- A requantization-based method for recompressing JPEG imagesHeinz H. Bauschke, Chris H. Hamilton, Mason S. Macklem, Justin S. McMichael, Nicholas R. Swart. 2489-2492 [doi]
- Progressive source-channel coding of video for unknown noisy channelsZongping Zhang, Guizhong Liu, Yiwen Yang. 2493-2496 [doi]
- Optimal frequency-differential encoding of sinusoidal model parametersJesper Jensen, Richard Heusdens. 2497-2500 [doi]
- Convolutional decoding for channels with false alarmsMohamed F. Mansour, Ahmed H. Tewfik. 2501-2504 [doi]
- On joint source-channel decoding for correlated sourcesMarc Adrat, R. Haenel, Peter Vary. 2505-2508 [doi]
- Symmetry-preserving reversible integer-to-integer wavelet transformsMichael D. Adams, Rabab Kreidieh Ward. 2509-2512 [doi]
- Application layer error correction coding for rate-distortion optimized streaming to wireless clientsJacob Chakareski, Philip A. Chou. 2513-2516 [doi]
- Source-adaptive shifting of modulation signal points for improved transmission of waveform signals over noisy channelsNorbert Görtz. 2517-2520 [doi]
- Results on minimal tail-biting trellis representation of double-circulant wavelet codesHossein Pishro-Nik, Faramarz Fekri. 2521-2524 [doi]
- Optimal maximal and maximal prefix codes equivalent to Huffman codesDongyang Long, Weijia Jia. 2525-2528 [doi]
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- Comparison between unitary and causal approaches to backward adaptive transform coding of vectorial signalsDavid L. Mary, Dirk T. M. Slock. 2533-2536 [doi]
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- Wavelet packet waveforms for multicarrier CDMA communicationsHongbing Zhang, H. Howard Fan, Alan R. Lindsey. 2557-2560 [doi]
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- Correlation matching in channel estimation for multirate DS/CDMAPing Liu, Zhengyuan Xu. 2581-2584 [doi]
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- Reduced interference MC-CDMA system using Discrete Prolate codesIvana Raos, Santiago Zazo, José Manuel Páez-Borrallo. 2597-2600 [doi]
- Optimum and suboptimum CDMA detectors: Near-far resistance under multipathXiaodong Yue, H. Howard Fan. 2601-2604 [doi]
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- Novel blind and group-blind constant modulus multiuser detectors for CDMA systemsSeungjae Bahng, Anders Høst-Madsen. 2613-2616 [doi]
- Diversely polarised arrays in DS-CDMA: A space-time channel estimation approachJason W. P. Ng, A. Manikas. 2617-2620 [doi]
- Hybrid linear-iterative detection algorithms for MIMO CDMA systems in multipath channelsTarik Muharemovic, Eko N. Onggosanusi, Anand G. Dabak, Behnaam Aazhang. 2621-2624 [doi]
- "Fractional self-decorrelation" pre-processing to enhance the output SINR of single-user-type DS-CDMA detectors in blind space-time RAKE receivers for fading channels with delay spreads approaching a symbol's durationKainam Thomas Wong, Shun Keung Cheung. 2625-2628 [doi]
- Stochastic least-symbol-error-rate adaptive equalization for pulse-amplitude modulationSheng Chen 0001, Bernard Mulgrew, Lajos Hanzo. 2629-2632 [doi]
- Schroeder sequences for time dispersive frequency selective channel estimation using DFT and Least Sum of Squared Errors methodsMessaoud Ahmed-Ouameur, Daniel Massicotte. 2633-2636 [doi]
- A hybrid adaptive technique for dynamic channel equalizationsMoeness G. Amin, Lin He. 2637-2640 [doi]
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- Fast algorithm for adaptive blind equalization using order-α Renyi's entropyIgnacio Santamaría, Carlos Pantaleón, Luis Vielva, Jose C. Principe. 2657-2660 [doi]
- Strategy to transition from the constant modulus algorithm to a decision feedback blind equalizerNigel McGinty. 2661-2664 [doi]
- Analysis of the decision delay effect on the convergence of gradient recursive decision feedback equalizersFelip Riera-Palou, James M. Noras, David G. M. Cruickshank. 2665-2668 [doi]
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- Compressing multi-component digital maps using JBIG2Pasi Fränti, Eugene I. Ageenko, Pavel Kopylov, Sami Gröhn. 2677-2680 [doi]
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- Efficient implementation of digital compensation in IQ modulatorsG. C. Lee, John Tuthill, Antonio Cantoni. 2697-2700 [doi]
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- Internet voice playout scheduling for integrated delay and error concealment with time-scale modificationFang Liu, JongWon Kim 0001, C. C. Jay Kuo. 2709-2712 [doi]
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- Performance analysis of blind channel estimation for precoded multirate multiuser systemsTing Wang, Zhengyuan Xu. 2741-2744 [doi]
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- A Synergistic Text Compression Method-STCMJulio Blandon, Malek Adjouadi, Shahriar Emami. 2773-2776 [doi]
- A model-driven joint source and channel coder for MPEG-2 video transmissionChung-Lin Huang, Sling Liang. 2777-2780 [doi]
- Integrated iterative equalization for ARQ systemsHarvind Samra, Zhi Ding. 2781-2784 [doi]
- Accelerated predictive-transformErlan H. Feria, Sos S. Agaian. 2785-2788 [doi]
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