Table of Contents

• Space-time processing for wireless communicationsArogyaswami Paulraj. 1-4 [doi]
• Variability of performance in video codingDon Pearson. 5-8 [doi]
• Expert SummariesRama Chellapa, Renato de Mori, Georgios B. Giannakis, Hans Georg Musmann, Hermann Ney, Mark J. T. Smith, John R. Treichler, Michael D. Zoltowski. 9-10 [doi]
• Expanding team experiences in DSP educationDelores M. Etter, Geoffrey C. Orsak. 11-14 [doi]
• Interactive classroom for DSP/communication coursesHüseyin Abut, Yusuf Öztürk. 15-18 [doi]
• Experiences in teaching DSP first in the ECE curriculumJames H. McClellan, Ronald W. Schafer, Mark A. Yoder. 19-22 [doi]
• Analog signal processing: a replacement for the sophomore-level circuit analysis courseDavid C. Munson Jr.. 23-26 [doi]
• Re-engineering the electrical engineering curriculumSanjit K. Mitra. 27-30 [doi]
• Structural subband decomposition: a new concept in digital signal processingSanjit K. Mitra. 31-34 [doi]
• A new algorithm for the generalized eigenvalue problemKnut Hüper, Uwe Helmke. 35-38 [doi]
• A lattice structure for perfect reconstruction linear time varying filter banks with all pass analysis banksSoura Dasgupta, Chris W. Schwarz, Minyue Fu. 39-42 [doi]
• Algorithm design for structured systems: application to pole placementSteffen Paul, Josef A. Nossek. 43-46 [doi]
• Actions of noncompact groups and algorithm design: a case studyKlaus Diepold, Rainer Pauli. 47-50 [doi]
• Discretization issues for the design of optimal blind algorithmsRodney A. Kennedy, Deva K. Borah, Zhi Ding. 51-54 [doi]
• ∞ optimization methodsZhuquan Zang, Antonio Cantoni, Kok Lay Teo. 55-58 [doi]
• Local adaptive algorithms for information maximization in neural networks, and application to source separationJeroen Dehaene, Nanayaa Twum-Danso. 59-62 [doi]
• Quick aggregation of Markov chain functionals via stochastic complementationKutluyil Dogançay, Vikram Krishnamurthy. 63-66 [doi]
• A rank preserving flow algorithm for quadratic optimization problems subject to quadratic equality constraintsJohn B. Moore, Danchi Jiang. 67-70 [doi]
• Verbmobil: the combination of deep and shallow processing for spontaneous speech translationThomas Bub, Wolfgang Wahlster, Alex Waibel. 71-74 [doi]
• Prosodic processing and its use in VERBMOBILHeinrich Niemann, Elmar Nöth, Andreas Kießling, Ralf Kompe, Anton Batliner. 75-78 [doi]
• The language components in VerbmobilHans Ulrich Block. 79-82 [doi]
• The Karlsruhe-Verbmobil speech recognition engineMichael Finke, Petra Geutner, Hermann Hild, Thomas Kemp, Klaus Ries, Martin Westphal. 83-86 [doi]
• An experiment on Korean-to-English and Korean-to-Japanese spoken language translationJae-Woo Yang, Jun Park. 87-90 [doi]
• Multilingual person to person communication at IRSTBianca Angelini, Mauro Cettolo, Anna Corazza, Daniele Falavigna, Gianni Lazzari. 91-94 [doi]
• Fast word-graph generation for spontaneous conversational speech translationTohru Shimizu, Harald Singer, Yoshinori Sagisaka. 95-98 [doi]
• Janus-III: speech-to-speech translation in multiple languagesAlon Lavie, Alex Waibel, Lori S. Levin, Michael Finke, Donna Gates, Marsal Gavaldà, Torsten Zeppenfeld, Puming Zhan. 99-102 [doi]
• State-transition cost functions and an application to language translationHiyan Alshawi, Adam L. Buchsbaum. 103-106 [doi]
• Hybrid language processing in the Spoken Language TranslatorManny Rayner, David M. Carter. 107-110 [doi]
• Finite-state speech-to-speech translationEnrique Vidal. 111-114 [doi]
• An experimental bidirectional Japanese/English interpreting video phone system using InternetShoji Hiraoka, Masakatsu Hoshimi, Kenji Matsui, Jean-Claude Junqua. 115-118 [doi]
• From neural networks to neural strategiesChristian Goerick, Bernhard Sendhoff, Werner von Seelen. 119-122 [doi]
• Neural and traditional techniques in diagnostic ECG classificationRosaria Silipo, Giovanni Bortolan. 123-126 [doi]
• Unsupervised learning for blind source separation: an information-theoretic approachDragan Obradovic, Gustavo Deco. 127-130 [doi]
• Applications of neural blind separation to signal and image processingJuha Karhunen, Aapo Hyvärinen, Ricardo Vigário, Jarmo Hurri, Erkki Oja. 131-134 [doi]
• Communications and neural networks: theory and practiceMark D. Plumbley. 135-138 [doi]
• Robust vector quantization by competitive learningJoachim M. Buhmann, Thomas Hofmann. 139-142 [doi]
• Recognizing faces from a new viewpointThomas Vetter. 143-146 [doi]
• Hybrid optimization of feedforward neural networks for handwritten character recognitionWolfgang Utschick, Josef A. Nossek. 147-150 [doi]
• Reading checks with multilayer graph transformer networksYann LeCun, Léon Bottou, Yoshua Bengio. 151-154 [doi]
• Neural networks for process control in steel manufacturingMartin Schlag, Einar Bröse, Björn Feldkeller, Otto Granckow, Michael Jansen, Thomas Poppe, Clemens Schäffner, Günter Sörgel. 155-158 [doi]
• A neuro-dynamic programming approach to admission control in ATM networks: the single link casePeter Marbach, John N. Tsitsiklis. 159-162 [doi]
• Issues in measuring the benefits of multimodal interfacesJames L. Flanagan, Ivan Marsic. 163-166 [doi]
• Multimodal interfaces for multimedia information agentsAlex Waibel, Bernhard Suhm, Minh Tue Vo, Jie Yang 0001. 167-170 [doi]
• Smart rooms, desks and clothesAlex Pentland. 171-174 [doi]
• Human machine interaction by voice and gestureNikil Jayant. 175-177 [doi]
• Audio-visual interaction in multimedia communicationTsuhan Chen, Ram Rao. 179-182 [doi]
• Lip motion modeling and speech driven estimationFabio Lavagetto, Skjalg Lepsøy, Carlo Braccini, Sergio Curinga. 183-186 [doi]
• Voice source localization for automatic camera pointing system in videoconferencingHong Wang, Peter Chu. 187-190 [doi]
• Video interface for spatiotemporal interactions based on multi-dimensional video computingAkihito Akutsu, Yoshinobu Tonomura, Hiroshi Hamada. 191-194 [doi]
• Indexing and search of multimodal informationAlexander G. Hauptmann, Howard D. Wactlar. 195-198 [doi]
• Acoustic indexing for multimedia retrieval and browsingSteve J. Young, Martin G. Brown, Jonathan Foote, Gareth J. F. Jones, Karen Spärck Jones. 199-202 [doi]
• Broadcast news transcriptionFrancis Kubala, Hubert Jin, Spyros Matsoukas, Long Nguyen, Richard M. Schwartz. 203-206 [doi]
• Image/speech processing that adopts an artistic approach-toward integration of art and technologyRyohei Nakatsu. 207-210 [doi]
• Noise cancelling for microphone arraysJens Meyer, Carsten Sydow. 211-213 [doi]
• A microphone array system for speech recognitionKenji Kiyohara, Yutaka Kaneda, Satoshi Takahashi, Hiroaki Nomura, Junji Kojima. 215-218 [doi]
• Strategies for combining acoustic echo cancellation and adaptive beamforming microphone arraysWalter Kellermann. 219-222 [doi]
• A steerable and variable first-order differential microphone arrayGary W. Elko, Anh-Tho Nguyen Pong. 223-226 [doi]
• Microphone array based speech recognition with different talker-array positionsMaurizio Omologo, Marco Matassoni, Piergiorgio Svaizer, Diego Giuliani. 227-230 [doi]
• Acoustic source location in a three-dimensional space using crosspower spectrum phasePiergiorgio Svaizer, Marco Matassoni, Maurizio Omologo. 231-234 [doi]
• Superdirective microphone array for a set-top videoconferencing systemPeter L. Chu. 235-238 [doi]
• Simultaneous echo cancellation and car noise suppression employing a microphone arrayMattias Dahl, Ingvar Claesson, Sven Nordebo. 239-242 [doi]
• Analytical evaluation of a self-calibrating microphone arraySven Nordholm, Ingvar Claesson. 243-246 [doi]
• Microphone array response to speaker movementsYves Grenier, Sofiène Affes. 247-250 [doi]
• A digital processing system for source location and sound capture by large microphone arraysHarvey F. Silverman, William R. Patterson III, James L. Flanagan, Daniel Rabinkin. 251-254 [doi]
• 3-D unitary ESPRIT for joint 2-D angle and carrier estimationMartin Haardt, Josef A. Nossek. 255-258 [doi]
• Quality enhancement of coded and corrupted speeches in GSM mobile systems using residual redundancyThomas Hindelang, Wen Xu, Christian Erben. 259-262 [doi]
• Pilot assisted coherent DS-CDMA reverse-link communications with optimal robust channel estimationFuyun Ling. 263-266 [doi]
• A new frequency estimator applied to burst transmissionChristian Bergogne, Philippe Sehier, Michel Bousquet. 267-270 [doi]
• Unified specification of control and data flowThorsten Grötker, Rainer Schoenen, Heinrich Meyr. 271-274 [doi]
• Reconfigurable processing: the solution to low-power programmable DSPJan M. Rabaey. 275-278 [doi]
• DSP cores for mobile communications: where are we going?Gerhard P. Fettweis. 279-282 [doi]
• DSPs in mobile communication in the United StatesSanjay Kasturia, Raziel Haimi-Cohen, Colin A. Warwick. 283-286 [doi]
• FRIDGE: an interactive code generation environment for HW/SW codesignMarkus Willems, Volker Bürsgens, Thorsten Grötker, Heinrich Meyr. 287-290 [doi]
• Staying ahead of the game in silicon for digital mobile communicationsRavi Subramanian, Marc Barberis, Herbert Dawid, Klaus-Jürgen Koch. 291-294 [doi]
• Approximation of optimal step size control for acoustic echo cancellationChristiane Antweiler, Jörn Grunwald, Holger Quack. 295-298 [doi]
• Subband stereo echo canceller using the projection algorithm with fast convergence to the true echo pathShoji Makino, Klaus Strauss, Suehiro Shimauchi, Yoichi Haneda, Akira Nakagawa. 299-302 [doi]
• A better understanding and an improved solution to the problems of stereophonic acoustic echo cancellationJacob Benesty, Dennis R. Morgan, M. Mohan Sondhi. 303-306 [doi]
• Comparison of three post-filtering algorithms for residual acoustic echo reductionValérie Turbin, André Gilloire, Pascal Scalart. 307-310 [doi]
• Audio coding using sinusoidal excitation representationWen-Whei Chang, De-Yu Wang, Li-Wei Wang. 311-314 [doi]
• Optimum bit allocation and decomposition for high quality audio codingXiang Wei, Martyn J. Shaw, Martin R. Varley. 315-318 [doi]
• 5 lattice quantization for 64 kbit/s low-delay subband audio coder with a 15 kHz bandwidthKarine Hay, Laurent Mainard, Samir Saoudi. 319-322 [doi]
• An experimental audio codec based on warped linear prediction of complex valued signalsAki Härmä, Unto K. Laine, Matti Karjalainen. 323-326 [doi]
• High quality low complexity scalable wavelet audio codingWilliam Kurt Dobson, Jienkan Jack Yang, Kevin J. Smart, Feng Kathy Guo. 327-330 [doi]
• An efficient tonal component coding algorithm for MPEG-2 Audio NBCYuichiro Takamizawa, Masahiro Iwadare, Akihiko Sugiyama. 331-334 [doi]
• Spectral amplitude warping (SAW) for noise spectrum shaping in audio codingRoch Lefebvre, Claude Laflamme. 335-338 [doi]
• A fast noise-scaling algorithm for uniform quantization in audio coding schemesCarlos A. Serantes, Antonio S. Pena, Nuria González Prelcic. 339-342 [doi]
• Pyramid vector coding for high quality audio compressionDaniele Cadel, Giorgio Parladori. 343-346 [doi]
• Subband audio coding with synthesis filters minimizing a perceptual distortionKarine Gosse, François Moreau de Saint-Martin, Xavier Durot, Pierre Duhamel, Jean-Bernard Rault. 347-350 [doi]
• New results in low bitrate audio coding using a combined harmonic-wavelet representationSimon Boland, Mohamed Deriche. 351-354 [doi]
• Adaptive inverse control of weakly nonlinear systemsWolfgang J. Klippel. 355-358 [doi]
• Broadband beamforming with adaptive postfiltering for speech acquisition in noisy environmentsSven Fischer, Karl-Dirk Kammeyer. 359-362 [doi]
• Near-field beamforming for microphone arraysJames G. Ryan, Rafik A. Goubran. 363-366 [doi]
• A robust adaptive microphone array with improved spatial selectivity and its evaluation in a real environmentOsamu Hoshuyama, Akihiko Sugiyama, Akihiro Hirano. 367-370 [doi]
• Tracking multiple talkers using microphone-array measurementsDouglas E. Sturim, Michael S. Brandstein, Harvey F. Silverman. 371-374 [doi]
• A robust method for speech signal time-delay estimation in reverberant roomsMichael S. Brandstein, Harvey F. Silverman. 375-378 [doi]
• A model-based approach to active noise cancellation using loudspeaker arrayJie Gu, Sze-Fong Yau. 379-382 [doi]
• Reverberant sound field analysis using a microphone arrayWolfgang Täger, Yannick Mahieux. 383-386 [doi]
• Minimisation of the maximum error signal in active controlAlberto González, Antonio Albiol, Steve J. Elliott. 387-390 [doi]
• Subband active noise control algorithm based on a delayless subband adaptive filter architectureJeong Hyeon Yun, Young-Cheol Park, Dae Hee Youn. 391-394 [doi]
• Nonlinear active noise control in a linear ductPaul Strauch, Bernard Mulgrew. 395-398 [doi]
• Fast exact filtered-X LMS and LMS algorithms for multichannel active noise controlScott C. Douglas. 399-402 [doi]
• A novel frequency domain filtered-X LMS algorithm for active noise reductionToshifumi Kosakat, Stephen J. Elliott, Christopher C. Boucher. 403-406 [doi]
• Practical supergrain head sized arraysDorra Masmoudi, Dominique Dallet, Jean Paul Dom. 407-410 [doi]
• A multichannel compression strategy for a digital hearing aidTodd Schneider, Robert L. Brennan. 411-414 [doi]
• Multi-microphone sub-band adaptive signal processing for improvement of hearing aid performance: primarily results using normal hearing volunteersPaul W. Shields, Douglas R. Campbell. 415-418 [doi]
• Environmental noise reduction based on speech/non-speech identification for hearing aidsKenzo Itoh, Masahide Mizushima. 419-422 [doi]
• Blind separation of multiple speakers in a multipath environmentRussell H. Lambert, Anthony J. Bell. 423-426 [doi]
• A single-chip 1, 200 sinusoid real-time generator for additive synthesis of musical signalsFernando De Bernardinis, Roberto Roncella, Roberto Saletti, Pierangelo Terreni, Graziano Bertini. 427-430 [doi]
• A generalized musical-tone generator with application to sound compression and synthesisCarlo Drioli, Davide Rocchesso. 431-434 [doi]
• A singing voice synthesis system based on sinusoidal modelingMichael W. Macon, Leslie Jensen-Link, James Oliverio, Mark A. Clements, E. Bryan George. 435-438 [doi]
• Time-scale modification of audio signals with combined harmonic and wavelet representationsKhaled N. Hamdy, Ahmed H. Tewfik, Ting Chen, Satoshi Takagi. 439-442 [doi]
• A waveguide model for slapbass synthesisErhard Rank, Gernot Kubin. 443-446 [doi]
• Minimum perceptual spectral distance FIR filter designShao-Po Wu, William Putnam. 447-450 [doi]
• A phase interpolation algorithm for sinusoidal model based music synthesisXiaoshu Qian, Yinong Ding. 451-454 [doi]
• Analytical approximations of fractional delays: Lagrange interpolators and allpass filtersStephan Tassart, Philippe Depalle. 455-458 [doi]
• Improved discrete-time modeling of multi-dimensional wave propagation using the interpolated digital waveguide meshLauri Savioja, Vesa Välimäki. 459-462 [doi]
• Generalized likelihood ratio test for selecting a geo-acoustic environmental modelChristoph F. Mecklenbräuker, Peter Gerstoft, Pei Jung Chung, Johann F. Böhme. 463-466 [doi]
• Tuning genetic algorithms for underwater acoustics using a priori statistical informationMaria-João Rendas, Georges Bienvenu. 467-470 [doi]
• Robust beamformer weight design for broadband matched-field processsingKerem Harmanci, Jeffrey L. Krolik. 471-474 [doi]
• Fastmap: a fast, approximate maximum a posteriori probability parameter estimator with application to robust matched-field processingBrian F. Harrison, Richard J. Vaccaro, Donald W. Tufts. 475-478 [doi]
• Electromagnetic matched field processing for source localizationDonald F. Gingras, Peter Gerstoft, Neil L. Gerr, Christoph F. Mecklenbräuker. 479-482 [doi]
• Power-law processors for detecting unknown signals in colored noiseIvars P. Kirsteins, Sanjay K. Mehta, John W. Fay. 483-486 [doi]
• Multitarget detection/tracking of echoes with known waveform: algorithm and applicationsVittorio Rampa, Umberto Spagnolini. 487-490 [doi]
• Detection of Gaussian bandpass transients under impulsive noise: a wavelet transform approachFrancisco M. Garcia, Isabel M. G. Lourtie. 491-494 [doi]
• Maximum likelihood estimator for magneto-acoustic localisationGilles Dassot, Roland Blanpain, Claude Jauffret. 495-498 [doi]
• Barankin bound for source localization in shallow waterJoseph Tabrikian, Jeffrey L. Krolik. 499-502 [doi]
• Underwater transient signal processing: marine mammal identification, localization, and source signal deconvolutionZoi-Heleni Michalopoulou. 503-506 [doi]
• Numerical optimization of non-adaptive microphone arraysAlexander Goldin. 507-510 [doi]
• Joint direction-of-arrival and array shape tracking for multiple moving targetsJason Goldberg, Ana I. Pérez-Neira, Miguel Angel Lagunas. 511-514 [doi]
• Comparison of probabilistic least squares and probabilistic multi-hypothesis tracking algorithms for multi-sensor trackingMark L. Krieg, Douglas A. Gray. 515-518 [doi]
• Direction finding with imperfect wavefront coherence: a matrix fitting approach using genetic algorithmAlex B. Gershman, Christoph F. Mecklenbräuker, Johann F. Böhme. 519-522 [doi]
• Design of an optimum wideband active sonar array with robustnessSaman S. Abeysekera, Yee Hong Leung. 523-526 [doi]
• Multipath time-delay estimationJean-Jacques Fuchs. 527-530 [doi]
• Fast maximum likelihood estimation with multiple signal initializationRobert B. MacLeod, Richard J. Vaccaro. 531-534 [doi]
• An algorithm for detecting closely spaced delay/Doppler componentsAmir W. Habboosh, Richard J. Vaccaro, Steven M. Kay. 535-538 [doi]
• Improvement of TDOA measurement using wavelet denoising with a novel thresholding techniqueShi-Quan Wu, Hing-Cheung So, Pak-Chung Ching. 539-542 [doi]
• A short-time Wiener filter for noise removal in underwater acoustic dataCharles W. Therrien, K. L. Frack Jr., Natanael Ruiz Fontes. 543-546 [doi]
• Fast approximate subspace tracking (FAST)Donald W. Tufts, Edward C. Real, James W. Cooley. 547-550 [doi]
• A subspace framework for fast parameter estimation with known waveformsBrian E. Freburger, Donald W. Tufts, Tom A. Palka. 551-554 [doi]
• Terrain classification in polarimetric SAR using wavelet packetsNirmal Keshava, José M. F. Moura. 555-558 [doi]
• Electromagnetic matched-field processing for target height finding with over-the-horizon radarMichael P. Papazoglou, Jeffrey L. Krolik. 559-562 [doi]
• Time-frequency classification using a multiple hypotheses test: an application to the classification of humpback whale signalsGeoff Roberts, Abdelhak M. Zoubir, Boualem Boashash. 563-566 [doi]
• Source classification using pole method of AR modelJianguo Huang, Jianping Zhao, Yiqing Xie. 567-570 [doi]
• The LMMSE estimate-based multiuser detector: performance analyses and adaptive implementationHongya Ge. 571-574 [doi]
• Improved Doppler tracking and correction for underwater acoustic communicationsMark Johnson, Lee E. Freitag, Milica Stojanovic. 575-578 [doi]
• A blind multichannel combiner for long range underwater communicationsBayan S. Sharif, Jeffrey A. Neasham, David Thompson, Oliver R. Hinton, Alan E. Adams. 579-582 [doi]
• Very long instruction word architectures for digital signal processingJon Mellott, Fred J. Taylor. 583-586 [doi]
• A novel 32 bit RISC architecture unifying RISC and DSPChristoph Baumhof, Frank Müller, Otto Müller, Manfred Schlett. 587-590 [doi]
• A dual-issue RISC processor for multimedia signal processingHisakazu Sato, Edgar Holmann, Toyohiko Yoshida, Masahito Matsuo, Toru Kengaku. 591-594 [doi]
• A processor-coprocessor architecture for high end video applicationsElmar Maas, Dirk Herrmann, Rolf Ernst, Peter Rüffer, Sieghard Hasenzahl, Martin Seitz. 595-598 [doi]
• An MPEG-2 encoder architecture based on a single-chip dedicated LSI with a control MPUYasushi Ooi, Osamu Ohnishi, Yutaka Yokoyama, Yoichi Katayama, Masayuki Mizuno, Masakazu Yamashina, Hideo Takano, Naoya Hayashi, Ichiro Tamitani. 599-602 [doi]
• An efficient and reconfigurable VLSI architecture for different block matching motion estimation algorithmsXiao-Dong Zhang, Chi-Ying Tsui. 603-606 [doi]
• An operation-saving VLSI geometry engine coreKonstantina Karagianni, George Diamantakos, Vassilis Paliouras, Thanos Stouraitis. 607-610 [doi]
• The FFT butterfly operation in 4 processor cycles on a 24 bit fixed-point DSP with a pipelined multiplierMartin Grajcar, Bernhard Sick. 611-614 [doi]
• New unified VLSI architectures for computing DFT and other transformsShen-Fu Hsiao, Chung-Yi Yen. 615-618 [doi]
• Half-rate GSM vocoder implementation on a dual mac digital signal processorMohit K. Prasad, Paul D'Arcy, Arup Gupta, Marc S. Diamondstein, Hosahalli R. Srinivas. 619-622 [doi]
• VLSI implementation of an area-efficient architecture for the Viterbi algorithmCarlos Cabrera, Montserrat Bóo, Javier D. Bruguera. 623-626 [doi]
• M)Leilei Song, Keshab K. Parhi. 627-630 [doi]
• m) on digital signal processorsWolfram Drescher, Kay Bachmann, Gerhard P. Fettweis. 631-634 [doi]
• A fast direction sequence generation method for CORDIC processorsSeunghyeon Nahm, Wonyong Sung. 635-638 [doi]
• A radix-4 redundant CORDIC algorithm with fast on-line variable scale factor compensationChieh-Chih Li, Sau-Gee Chen. 639-642 [doi]
• Pipelining of cordic based IIR digital filtersJun Ma, Keshab K. Parhi, Ed F. Deprettere. 643-646 [doi]
• An asynchronous implementation of the maxlist algorithmChris J. Myers, Hao Zheng 0001. 647-650 [doi]
• A novel systematic mapping approach for highly efficient multiplexed FIR-filter architecturesWolfgang Wilhelm, Tobias G. Noll. 651-654 [doi]
• An upper bound of the throughput of multirate multiprocessor schedulesRainer Schoenen, Vojin Zivojnovic, Heinrich Meyr. 655-658 [doi]
• Minimizing the number of operations in DSP computationsInki Hong, Miodrag Potkonjak. 659-662 [doi]
• BEEHIVE: an adaptive, distributed, embedded signal processing environmentShahram Famorzadeh, Vijay K. Madisetti, Thomas Egolf, Tuongvu Nguyen. 663-666 [doi]
• On objective function selection in list scheduling algorithms for digital signal processing applicationsJan Jonsson, Jonas Vasell. 667-670 [doi]
• VLSI high level synthesis of fast exact least mean square algorithms based on fast FIR filtersJean-Philippe Diguet, Olivier Sentieys, Daniel Chillet, Jean Luc Philippe. 671-674 [doi]
• Hierarchical VHDL libraries for DSP ASIC designJohn V. McCanny, Douglas Ridge, Yi Hu, Jill Hunter. 675-678 [doi]
• DSP Quant: design, validation, and applications of DSP hard real-time benchmarkChunho Lee, Darko Kirovski, Inki Hong, Miodrag Potkonjak. 679-682 [doi]
• Constructing memory layouts for address generation units supporting offset 2 accessBernhard Wess, Martin Gotschlich. 683-686 [doi]
• Modulo-addressing utilization in automatic software synthesis for digital signal processorsMarkus Willems, Holger Keding, Vojin Zivojnovic, Heinrich Meyr. 687-690 [doi]
• Cooperative register assignment and code compaction for digital signal processors with irregular datapathsWerner Kreuzer, Bernhard Wess. 691-694 [doi]
• Optimization of embedded DSP programs using post-pass data-flow analysisAshok Sudarsanam, Sharad Malik, Steven W. K. Tjiang, Stan Y. Liao. 695-698 [doi]
• Code positioning to reduce instruction cache misses in signal processing applications on multimedia RISC processorsHans-Joachim Stolberg, Masao Ikekawa, Ichiro Kuroda. 699-702 [doi]
• Code generation by using integer-controlled dataflow graphTakashi Miyazaki, Edward A. Lee. 703-706 [doi]
• Fixed-point C compiler for TMS320C50 digital signal processorJiyang Kang, Wonyong Sung. 707-710 [doi]
• Transcription of broadcast news-system robustness issues and adaptation techniquesRaimo Bakis, Scott Saobing Chen, Ponani S. Gopalakrishnan, Ramesh Gopinath, Stéphane H. Maes, Lazaros Polymenakos. 711-714 [doi]
• Transcribing broadcast news showsJean-Luc Gauvain, Gilles Adda, Lori Lamel, Martine Adda-Decker. 715-718 [doi]
• Broadcast news transcription using HTKPhilip C. Woodland, Mark J. F. Gales, David Pye, Steve J. Young. 719-722 [doi]
• Transcription of broadcast television and radio news: the 1996 ABBOT systemGary D. Cook, Dan J. Kershaw, James Christie, Carl W. Seymour, Steve R. Waterhouse. 723-726 [doi]
• Improved topic discrimination of broadcast news using a model of multiple simultaneous topicsToru Imai, Richard M. Schwartz, Francis Kubala, Long Nguyen. 727-730 [doi]
• Enhanced full rate speech codec for IS-136 digital cellular systemTero Honkanen, Janne Vainio, Kari Järvinen, Petri Haavisto, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul. 731-734 [doi]
• A CELP variable rate speech codec with low average rateLei Zhang, Tian Wang, Vladimir Cuperman. 735-738 [doi]
• HCELP: low bit rate speech coder for voice storage applicationsMustapha Bouraoui, Francois Druilhe, Gang Feng. 739-742 [doi]
• Low-rate CELP speech coding using an improved weighting functionChul Hong Kwon, Chong Kwan Un. 743-746 [doi]
• Toll quality variable-rate speech codecPasi Ojala. 747-750 [doi]
• A variable rate multimodal speech coder with gain-matched analysis-by-synthesisErdal Paksoy, Alan McCree, Vishu Viswanathan. 751-754 [doi]
• Design of a toll-quality 4-kbit/s speech coder based on phase-adaptive PSI-CELPKazunori Mano. 755-758 [doi]
• A high quality BI-CELP speech coder at 8 kbit/s and belowSoon Y. Kwon, Hochong Park, Hyokang Chang. 759-762 [doi]
• Low complexity VQ for multi-tap pitch predictor codingJayesh Patel. 763-766 [doi]
• A 4 kbit/s renewal code excited linear prediction speech coderHong Kook Kim, Yong Duk Cho, Moo-young Kim, Sang Ryong Kim. 767-770 [doi]
• GSM enhanced full rate speech codecKari Järvinen, Janne Vainio, Pekka Kapanen, Tero Honkanen, Petri Haavisto, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul. 771-774 [doi]
• Description of ITU-T Recommendation G.729 Annex A: reduced complexity 8 kbit/s CS-ACELP codecRedwan Salami, Claude Laflamme, Bruno Bessette, Jean-Pierre Adoul. 775-778 [doi]
• Semantic clustering for adaptive language modelingReinhard Kneser, Jochen Peters. 779-782 [doi]
• Task adaptation using MAP estimation in N-gram language modelingHirokazu Masataki, Yoshinori Sagisaka, Kazuya Hisaki, Tatsuya Kawahara. 783-786 [doi]
• Distant bigram language modelling using maximum entropyMichael Simons, Hermann Ney, Sven C. Martin. 787-790 [doi]
• Nonuniform Markov modelsEric Sven Ristad, Robert G. Thomas. 791-794 [doi]
• Modelling word-pair relations in a category-based language modelThomas Niesler, Philip C. Woodland. 795-798 [doi]
• Language model adaptation using mixtures and an exponentially decaying cachePhilip Clarkson, Anthony J. Robinson. 799-802 [doi]
• Confidence-driven estimator perturbation: BMPC [Best Model Perturbation within Confidence]Stefan Besling, Hans-Günter Meier. 803-806 [doi]
• Domain adaptation with clustered language modelsJoerg P. Ueberla. 807-810 [doi]
• Improving parsing of spontaneous speech with the help of prosodic boundariesRalf Kompe, Andreas Kießling, Heinrich Niemann, Elmar Nöth, Anton Batliner, Stefanie Schachtl, Tobias Ruland, Hans Ulrich Block. 811-814 [doi]
• Specialized language models using dialogue predictionsCosmin Popovici, Paolo Baggia. 815-818 [doi]
• K-TLSS(S) language models for speech recognitionGermán Bordel, Amparo Varona, M. Inés Torres. 819-822 [doi]
• Language model adaptation for conversational speech recognition using automatically tagged pseudo-morphological classesCarlos Crespo, Daniel Tapias, Gregorio Escalada, Jorge Alvarez. 823-826 [doi]
• Model adaptation based on HMM decomposition for reverberant speech recognitionTetsuya Takiguchi, Satoshi Nakamura, Qiang Hou, Kiyohiro Shikano. 827-830 [doi]
• Model compensation for noises in training and test dataDriss Matrouf, Jean-Luc Gauvain. 831-834 [doi]
• Jacobian approach to fast acoustic model adaptationShigeki Sagayama, Yoshikazu Yamaguchi, Satoshi Takahashi, Jun-ichi Takahashi. 835-838 [doi]
• A unified maximum likelihood approach to acoustic mismatch compensation: application to noisy Lombard speech recognitionMohamed Afify, Yifan Gong, Jean-Paul Haton. 839-842 [doi]
• Enhancement and recognition of noisy speech within an autoregressive hidden Markov model framework using noise estimates from the noisy signalBeth T. Logan, Anthony J. Robinson. 843-846 [doi]
• Fast speech recognition algorithm under noisy environment using modified CMS-PMC and improved IDMM+SQHiroki Yamamoto, Tetsuo Kosaka, Masayuki Yamada, Yasuhiro Komori, Minoru Fujita. 847-850 [doi]
• The effects of background music on speech recognition accuracyBhiksha Raj, Vipul N. Parikh, Richard M. Stern. 851-854 [doi]
• Joint model and feature space optimization for robust speech recognitionJenq-Neng Hwang, Chien-Jen Wang. 855-858 [doi]
• Co-channel speech separation for robust automatic speech recognition: stability and efficiencyKuan-Chieh Yen, Yunxin Zhao. 859-862 [doi]
• Missing data techniques for robust speech recognitionMartin P. Cooke, Andrew C. Morris, Phil D. Green. 863-866 [doi]
• Spectral subtraction and RASTA-filtering in text-dependent HMM-based speaker verificationDetlef Hardt, Klaus Fellbaum. 867-870 [doi]
• Noise robust speech recognition with state duration constraintsKari Laurila. 871-874 [doi]
• Confidence measures for spontaneous speech recognitionThomas Schaaf, Thomas Kemp. 875-878 [doi]
• A probabilistic approach to confidence estimation and evaluationLarry Gillick, Yoshiko Ito, Jonathan Young. 879-882 [doi]
• Word-based confidence measures as a guide for stack search in speech recognitionChalapathy Neti, Salim Roukos, Ellen Eide. 883-886 [doi]
• Neural-network based measures of confidence for word recognitionMitch Weintraub, Françoise Beaufays, Zeév Rivlin, Yochai Konig, Andreas Stolcke. 887-890 [doi]
• Improving utterance verification using hierarchical confidence measures in continuous natural numbers recognitionJavier Caminero, Luis A. Hernández Gómez, Celinda de la Torre, Cesar Martín del Alamo. 891-894 [doi]
• On the influence of frame-asynchronous grammar scoring in a CSR systemAntonio J. Rubio, Jesús E. Díaz-Verdejo, Pedro García Teodoro, José C. Segura. 895-898 [doi]
• A segment-based wordspotter using phonetic filler modelsAlexandros S. Manos, Victor W. Zue. 899-902 [doi]
• A multi-phase approach for fast spotting of large vocabulary Chinese keywords from Mandarin speech using prosodic informationBo-Ren Bai, Chiu-yu Tseng, Lin-Shan Lee. 903-906 [doi]
• Accurate keyword spotting using strictly lexical fillersRachida El Méliani, Douglas D. O'Shaughnessy. 907-910 [doi]
• Failure simulation for a phoneme HMM based keyword spotterMartin Holzapfel, Günther Ruske, Harald Höge. 911-914 [doi]
• Wordspotting using a predictive neural model for the telephone speech corpusSuhardi, Klaus Fellbaum. 915-918 [doi]
• Shape-invariant pitch and time-scale modification of speech by variable order phase interpolationMat P. Pollard, Barry M. G. Cheetham, Colin C. Goodyear, Mike D. Edgington. 919-922 [doi]
• A Chinese text-to-speech system based on part-of-speech analysis, prosodic modeling and non-uniform unitsFu-Chiang Chou, Chiu-yu Tseng, Keh-Jiann Chen, Lin-Shan Lee. 923-926 [doi]
• Automatic prosodic modeling for speaker and task adaptation in text-to-speechEduardo López Gonzalo, Jose M. Rodriguez-Garcia, Luis A. Hernández Gómez, Juan Manuel Villar-Navarro. 927-930 [doi]
• Prosody generation with a neural network: weighing the importance of input parametersGerit P. Sonntag, Thomas Portele, Barbara Heuft. 931-934 [doi]
• Evaluation of a speech synthesis method for nonlinear modeling of vocal folds vibration effectHiroshi Ohmura, Kazuyo Tanaka. 935-938 [doi]
• Generation of F0 contour using stochastic mapping and vector quantization control parametersHeo-Jin Byeon, Yeon-Jun Kim, Yung-Hwan Oh. 939-942 [doi]
• Spectral normalization employing hidden Markov modeling of line spectrum pair frequenciesBryan L. Pellom, John H. L. Hansen. 943-946 [doi]
• Time domain technique for pitch modification and robust voice transformationRivarol Vergin, Douglas D. O'Shaughnessy, Azarshid Farhat. 947-950 [doi]
• 0Kimihito Tanaka, Masanobu Abe. 951-954 [doi]
• Reliability assessment and evaluation of objectively measured descriptors for perceptual speaker characterizationBurhan Necioglu, Mark A. Clements, Thomas P. Barnwell III. 955-958 [doi]
• Recent improvements on Microsoft's trainable text-to-speech system-WhistlerXuedong Huang, Alex Acero, Hsiao-Wuen Hon, Yun-Cheng Ju, Jingsong Liu, Scott Meredith, Mike Plumpe. 959-962 [doi]
• Automatic generation of speech synthesis units based on closed loop trainingTakehiko Kagoshima, Masami Akamine. 963-966 [doi]
• Isolated word recognition using the HMM structure selected by the genetic algorithmTomio Takara, Kazuya Higa, Itaru Nagayama. 967-970 [doi]
• Discrete mixture HMMSatoshi Takahashi, Kiyoaki Aikawa, Shigeki Sagayama. 971-974 [doi]
• Using word temporal structure in HMM speech recognitionLuciano Fissore, Pietro Laface, Franco Ravera. 975-978 [doi]
• Smoothness analysis for trajectory featuresZhihong Hu, Etienne Barnard. 979-982 [doi]
• Frequency-warping and speaker-normalizationSrinivasan Umesh, Leon Cohen, Douglas J. Nelson. 983-986 [doi]
• Integrating syllable boundary information into speech recognitionSu-Lin Wu, Michael L. Shire, Steven Greenberg, Nelson Morgan. 987-990 [doi]
• Explicit, N-best formant features for vowel classificationPhilipp Schmid, Etienne Barnard. 991-994 [doi]
• Dual-channel auditory spectrum modelingJayadev Billa. 995-998 [doi]
• Direct identification vs. correlated models to process acoustic and articulatory informations in automatic speech recognitionRégine André-Obrecht, Bruno Jacob. 999-1002 [doi]
• Adapting PSN recognition models to the GSM environment by using spectral transformationThierry Soulas, Chafic Mokbel, Denis Jouvet, Jean Monné. 1003-1006 [doi]
• Integrated-multilingual speech recognition using universal phonological features in a functional speech production modelLi Deng. 1007-1010 [doi]
• Phone classification with segmental features and a binary-pair partitioned neural network classifierStephen A. Zahorian, Peter L. Silsbee, Xihong Wang. 1011-1014 [doi]
• Smoothed N-best-based speaker adaptation for speech recognitionTomoko Matsui, Tatsuo Matsuoka, Sadaoki Furui. 1015-1018 [doi]
• A fast algorithm for unsupervised incremental speaker adaptationMichael Schüßler, Florian Gallwitz, Stefan Harbeck. 1019-1022 [doi]
• Improved estimation of supervision in unsupervised speaker adaptationShigeru Homma, Kiyoaki Aikawa, Shigeki Sagayama. 1023-1026 [doi]
• Improved Bayesian learning of hidden Markov models for speaker adaptationJen-Tzung Chien, Chin-Hui Lee, Hsiao-Chuan Wang. 1027-1030 [doi]
• Studies in transformation-based adaptationVenkatesh Nagesha, Larry Gillick. 1031-1034 [doi]
• Speaker adaptation in the Philips system for large vocabulary continuous speech recognitionEric Thelen, Xavier L. Aubert, Peter Beyerlein. 1035-1038 [doi]
• Speaker normalization based on frequency warpingPuming Zhan, Martin Westphal. 1039-1042 [doi]
• Speaker adaptive training: a maximum likelihood approach to speaker normalizationTasos Anastasakos, John W. McDonough, John Makhoul. 1043-1046 [doi]
• Experiments in speaker normalisation and adaptation for large vocabulary speech recognitionDavid Pye, Philip C. Woodland. 1047-1050 [doi]
• Effectiveness of speaker normalized HMM by projection to speaker subspaceYasuo Ariki. 1051-1054 [doi]
• Speaker normalization and adaptation based on linear transformationJun Ishii, Masahiro Tonomura. 1055-1058 [doi]
• Speaker-adapted training on the Switchboard CorpusJohn W. McDonough, Tasos Anastasakos, George Zavaliagkos, Herbert Gish. 1059-1062 [doi]
• Model transformation for robust speaker recognition from telephone dataFrançoise Beaufays, Mitch Weintraub. 1063-1066 [doi]
• Speaker recognition with the Switchboard corpusLori Lamel, Jean-Luc Gauvain. 1067-1070 [doi]
• Handset-dependent background models for robust text-independent speaker recognitionLarry P. Heck, Mitch Weintraub. 1071-1074 [doi]
• Telephone based speaker recognition using multiple binary classifier and Gaussian mixture modelsPierre Castellano, Stefan Slomka, Sridha Sridharan. 1075-1078 [doi]
• Comparison of whole word and subword modeling techniques for speaker verification with limited training dataStephan Euler, Rainer Langlitz, Joachim Zinke. 1079-1082 [doi]
• A comparison of model estimation techniques for speaker verificationMichael J. Carey 0002, Eluned S. Parris, Stephen J. Bennett, Harvey Lloyd-Thomas. 1083-1086 [doi]
• Speaker verification using frame and utterance level likelihood normalizationSeiichi Nakagawa, Konstantin P. Markov. 1087-1090 [doi]
• A new codebook training algorithm for VQ-based speaker recognitionJialong He, Li Liu, Günther Palm. 1091-1094 [doi]
• Bispectrum features for robust speaker identificationStanley J. Wenndt, Sanyogita Shamsunder. 1095-1098 [doi]
• Speaker identification based text to audio alignment for an audio retrieval systemDeb K. Roy, Carl Malamud. 1099-1102 [doi]
• Robust speaker recognition through acoustic array processing and spectral normalizationJoaquín González Rodríguez, Javier Ortega-Garcia. 1103-1106 [doi]
• Providing single and multi-channel acoustical robustness to speaker identification systemsJavier Ortega-Garcia, Joaquín González Rodríguez. 1107-1110 [doi]
• Robust spoken language identification using large vocabulary speech recognitionJames Hieronymus, Shubha Kadambe. 1111-1114 [doi]
• Double bigram-decoding in phonotactic language identificationJirí Navrátil, Werner Zühlke. 1115-1118 [doi]
• Random walk theory applied to language identificationEtienne Marcheret, Michael I. Savic. 1119-1122 [doi]
• Frequency characteristics of foreign accented speechLevent M. Arslan, John H. L. Hansen. 1123-1126 [doi]
• A study on improving decisions in closed set speaker identificationMübeccel Demirekler, Afsar Saranli. 1127-1130 [doi]
• The use of harmonic features in speaker recognitionBojan Imperl, Zdravko Kacic, Bogomir Horvat. 1131-1134 [doi]
• An approach to speaker identification using multiple classifiersVlasta Radová, Josef Psutka. 1135-1138 [doi]
• Development and evaluation of the ATOS spontaneous speech conversational systemJorge Alvarez, Daniel Tapias, Carlos Crespo, Ismael Cortázar, Fernando Martinez. 1139-1142 [doi]
• A spoken language system for automated call routingGiuseppe Riccardi, Allen L. Gorin, Andrej Ljolje, Michael Riley. 1143-1146 [doi]
• Dialogos: a robust system for human-machine spoken dialogue on the telephoneDario Albesano, Paolo Baggia, Morena Danieli, Roberto Gemello, Elisabetta Gerbino, Claudio Rullent. 1147-1150 [doi]
• Surfin' the World Wide Web with JapaneseKazuhiro Kondo, Charles T. Hemphill. 1151-1154 [doi]
• Internet Chinese information retrieval using unconstrained Mandarin speech queries based on a client-server architecture and a PAT-tree-based language modelLee-Feng Chien, Sung-Chien Lin, Jenn-Chau Hong, Ming-Chiuan Chen, Hsin-Min Wang, Jia-lin Shen, Keh-Jiann Chen, Lin-Shan Lee. 1155-1158 [doi]
• Combining key-phrase detection and subword-based verification for flexible speech understandingTatsuya Kawahara, Chin-Hui Lee, Biing-Hwang Juang. 1159-1162 [doi]
• Controlling limited-domain applications by probabilistic semantic decoding of natural speechHolger Stahl, Johannes Müller, Manfred K. Lang. 1163-1166 [doi]
• Multi-channel speech enhancement in a car environment using Wiener filtering and spectral subtractionJörg Meyer, Klaus Uwe Simmer. 1167-1170 [doi]
• Weighted matching algorithms and reliability in noise cancelling by spectral subtractionNéstor Becerra Yoma, Fergus R. McInnes, Mervyn A. Jack. 1171-1174 [doi]
• HMM-based speech enhancement using harmonic modelingMichael E. Deisher, Andreas S. Spanias. 1175-1178 [doi]
• Model based speech pause detectionBruce L. McKinley, Gary. H. Whipple. 1179-1182 [doi]
• Integrated speech enhancement and coding in the time-frequency domainAndrzej Drygajlo, Benito Carnero. 1183-1186 [doi]
• Quality enhancement of narrowband CELP-coded speech via wideband harmonic re-synthesisCheung-fat Chan, Wai-Kwong Hui. 1187-1190 [doi]
• Speech enhancement using CSS-based array processingFutoshi Asano, Satoru Hayamizu. 1191-1194 [doi]
• Co-channel speaker separation using constrained nonlinear optimizationDaniel S. Benincasa, Michael I. Savic. 1195-1198 [doi]
• A contextual blind separation of delayed and convolved sourcesTe-Won Lee, Reinhold Orglmeister. 1199-1202 [doi]
• Segregation of concurrent speech with the reassigned spectrumGeorg F. Meyer, Fabrice Plante, Frédéric Berthommier. 1203-1206 [doi]
• Enhancement of esophageal speech by injection noise rejectionHector R. Javkin, Michael Galler, Nancy Niedzielski. 1207-1210 [doi]
• Real-time digital speech processing strategies for the hearing impairedNeeraj Magotra, Sudheer Sirivara. 1211-1214 [doi]
• Iterative-batch and sequential algorithms for single microphone speech enhancementSharon Gannot, David Burshtein, Ehud Weinstein. 1215-1218 [doi]
• Kalman filtering for low distortion speech enhancement in mobile communicationPatrick Sörqvist, Peter Händel, Björn E. Ottersten. 1219-1222 [doi]
• Exploiting the potential of auditory preprocessing for robust speech recognition by locally recurrent neural networksKlaus Kasper, Herbert Reininger, Dietrich Wolf. 1223-1226 [doi]
• Feature adaptation using deviation vector for robust speech recognition in noisy environmentTai-Hwei Hwang, Lee-Min Lee, Hsiao-Chuan Wang. 1227-1230 [doi]
• Binaural phoneme recognition using the auditory image model and cross-correlationKeith I. Francis, Timothy R. Anderson. 1231-1234 [doi]
• Utterance dependent parametric warping for a talker-independent HMM-based recognizerDaniel J. Mashao, John E. Adcock. 1235-1238 [doi]
• Phase-corrected RASTA for automatic speech recognition over the phoneJohan de Veth, Louis Boves. 1239-1242 [doi]
• A binaural speech processing method using subband-cross correlation analysis for noise robust recognitionShoji Kajita, Kazuya Takeda, Fumitada Itakura. 1243-1246 [doi]
• Modelling asynchrony in speech using elementary single-signal decompositionMichael J. Tomlinson, Martin J. Russell, Roger K. Moore, Andrew P. Buckland, Martin A. Fawley. 1247-1250 [doi]
• Subband-based speech recognitionHervé Bourlard, Stéphane Dupont. 1251-1254 [doi]
• Sub-band based recognition of noisy speechSangita Tibrewala, Hynek Hermansky. 1255-1258 [doi]
• Recognizing reverberant speech with RASTA-PLPBrian Kingsbury, Nelson Morgan. 1259-1262 [doi]
• Multi-resolution phonetic/segmental features and models for HMM-based speech recognitionSaeed Vaseghi, Naomi Harte, Ben Milner. 1263-1266 [doi]
• Maximum likelihood weighting of dynamic speech features for CDHMM speech recognitionJavier Hernando. 1267-1270 [doi]
• Speech recognition using automatically derived acoustic baseformsRichard C. Rose, Eduardo Lleida. 1271-1274 [doi]
• On combining frequency warping and spectral shaping in HMM based speech recognitionAlexandros Potamianos, Richard C. Rose. 1275-1278 [doi]
• Recursive linear prediction using OBE identification with automatic bound estimationJohn R. Deller Jr., Tsung-Ming Lin, Majid Nayeri. 1279-1282 [doi]
• Nonlinear long-term prediction of speech signalsMartin Birgmeier, Hans-Peter Bernhard, Gernot Kubin. 1283-1286 [doi]
• Vocal tract shape trajectory estimation using MLP analysis-by-synthesisHywel B. Richards, John S. Bridle, Melvyn J. Hunt, John S. Mason. 1287-1290 [doi]
• Fast and robust joint estimation of vocal tract and voice source parametersWen Ding, Nick Campbell, Norio Higuchi, Hideki Kasuya. 1291-1294 [doi]
• Spectral correlates of glottal waveform models: an analytic studyBoris Doval, Christophe d'Alessandro. 1295-1298 [doi]
• A time varying ARMAX speech modeling with phase compensation using glottal source modelKeiichi Funaki, Yoshikazu Miyanaga, Koji Tochinai. 1299-1302 [doi]
• Speech representation and transformation using adaptive interpolation of weighted spectrum: vocoder revisitedHideki Kawahara. 1303-1306 [doi]
• The weft: a representation for periodic soundsDan Ellis. 1307-1310 [doi]
• A computationally efficient algorithm for calculating loudness patterns of narrowband speechMarkus Hauenstein. 1311-1314 [doi]
• Two-channel blind deconvolution for non-minimum phase impulse responsesKen'ichi Furuya, Yutaka Kaneda. 1315-1318 [doi]
• Variable time-scale modification of speech using transient informationSung Joo Lee, Hee-Dong Kim, Hyung Soon Kim. 1319-1322 [doi]
• Speech enhancement with reduction of noise components in the wavelet domainJong Won Seok, Keun-Sung Bae. 1323-1326 [doi]
• Blind separation and restoration of signals mixed in convolutive environmentJiangtao Xi, James P. Reilly. 1327-1330 [doi]
• Construction and evaluation of a robust multifeature speech/music discriminatorEric D. Scheirer, Malcolm Slaney. 1331-1334 [doi]
• Encoding of speech spectral parameters using adaptive quantization methodsInsung Lee, Hang Chae Woo. 1335-1338 [doi]
• Optimal transformation of LSP parameters using neural networkHai Le Vu, László Lois. 1339-1342 [doi]
• Speech spectrum representation and coding using multigrams with distanceJan Cernocký, Geneviève Baudoin, Gérard Chollet. 1343-1346 [doi]
• Incorporating perception into LSF quantization some experimentsRonald P. Cohn, John S. Collura. 1347-1350 [doi]
• Predictive VQ for noisy channel spectrum coding: AR or MA?Jan Skoglund, Jan Linden. 1351-1354 [doi]
• Efficient encoding of mel-generalized cepstrum for CELP codersKazuhito Koishida, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai. 1355-1358 [doi]
• A candidate coder for the ITU-T's new wideband speech coding standardJuin-Hwey Chen. 1359-1362 [doi]
• Perceptual speech coding using time and frequency masking constraintsBenito Carnero, Andrzej Drygajlo. 1363-1366 [doi]
• A multi-band CELP wideband speech coderAnil Ubale, Allen Gersho. 1367-1370 [doi]
• A design of transform coder for both speech and audio signals at 1 bit/sampleTakehiro Moriya, Naoki Iwakami, Akio Jin, Kazunaga Ikeda, Satoshi Miki. 1371-1374 [doi]
• Speech quality assessment of compounded digital telecommunication systems; perceptual dimensionsKim T. Petersen, Steffen Duus Hansen, John Aasted Sørensen. 1375-1378 [doi]
• Performance assessment of tandem connection of cellular and satellite-mobile codersSimão F. Campos Neto, Franklin L. Corcoran, Ara Karahisar. 1379-1382 [doi]
• The consequences of linguistic perception on low-rate speech codingJohn J. Parry, Ian S. Burnett, Joe F. Chicharo. 1383-1386 [doi]
• Using a quantitative psychoacoustical signal representation for objective speech quality measurementMartin Hansen, Birger Kollmeier. 1387-1390 [doi]
• A method of extracting time-varying acoustic features effective for speech recognitionKazuyo Tanaka, Hiroaki Kojima. 1391-1394 [doi]
• Elimination of trajectory folding phenomenon: HMM, trajectory mixture HMM and mixture stochastic trajectory modelIrina Illina, Yifan Gong. 1395-1398 [doi]
• Linear dynamic segmental HMMs: variability representation and training procedureWendy J. Holmes, Martin J. Russell. 1399-1402 [doi]
• Model parameter estimation for mixture density polynomial segment modelsToshiaki Fukada, Yoshinori Sagisaka, Kuldip K. Paliwal. 1403-1406 [doi]
• The importance of segmentation probability in segment based speech recognizersJan P. Verhasselt, Irina Illina, Jean-Pierre Martens, Yifan Gong, Jean-Paul Haton. 1407-1410 [doi]
• Adaptation of polynomial trajectory segment models for large vocabulary speech recognitionAshvin Kannan, Mari Ostendorf. 1411-1414 [doi]
• Speaker adaptation experiments using nonstationary-state hidden Markov models: a MAP approachChengalv Rathinavelu, Li Deng. 1415-1418 [doi]
• Vocabulary optimization based on perplexityKyuwoong Hwang. 1419-1422 [doi]
• REMAP for video soundtrack indexingPhilippe Gelin, Christian Wellekens. 1423-1426 [doi]
• Robust pitch detection of speech signals using steerable filtersJinhai Cai, Zhi-Qiang Liu. 1427-1430 [doi]
• Evaluation of the relationship between emotional concepts and emotional parameters on speechTsuyoshi Moriyama, Hideo Saito, Shinji Ozawa. 1431-1434 [doi]
• Time-frequency analysis of the glottal openingWolfgang Wokurek. 1435-1438 [doi]
• Time-frequency structured decorrelation of speech signals via nonseparable Gabor framesWerner Kozek, Hans Georg Feichtinger. 1439-1442 [doi]
• Generalized mixture of HMMs for continuous speech recognitionFilipp Korkmazskiy, Biing-Hwang Juang, Frank K. Soong. 1443-1446 [doi]
• Writer adaptation of a HMM handwriting recognition systemAndrew W. Senior, Krishna S. Nathan. 1447-1450 [doi]
• In-service adaptation of multilingual hidden-Markov-modelsUdo Bub, Joachim Köhler, Bojan Imperl. 1451-1454 [doi]
• Development of dialect-specific speech recognizers using adaptation methodsVassilios Diakoloukas, Vassilios Digalakis, Leonardo Neumeyer, Jaan Kaja. 1455-1458 [doi]
• Syllable-based relevance feedback techniques for Mandarin voice record retrieval using speech queriesLin-Shan Lee, Bo-Ren Bai, Lee-Feng Chien. 1459-1462 [doi]
• Automatic alternative transcription generation and vocabulary selection for flexible word recognizersDoroteo Torre Toledano, Luis Villarrubia, Luis A. Hernández Gómez, Jose Maria Elvira. 1463-1466 [doi]
• An advanced system to generate pronunciations of proper nounsNeeraj Deshmukh, Julie Ngan, Jonathan Hamaker, Joseph Picone. 1467-1470 [doi]
• Automatic pronunciation scoring for language instructionHoracio Franco, Leonardo Neumeyer, Yoon Kim, Orith Ronen. 1471-1474 [doi]
• Speaker-independent name dialing with out-of-vocabulary rejectionCoimbatore S. Ramalingam, Lorin Netsch, Yu-Hung Kao. 1475-1478 [doi]
• Hidden understanding models for statistical sentence understandingRichard M. Schwartz, Scott Miller, David Stallard, John Makhoul. 1479-1482 [doi]
• An alternative scheme for perplexity estimationFrédéric Bimbot, Marc El-Bèze, Michèle Jardino. 1483-1486 [doi]
• Extensions to phone-state decision-tree clustering: single tree and tagged clusteringDouglas B. Paul. 1487-1490 [doi]
• Evaluation of fast algorithms for finding the nearest neighborStéphane Lubiarz, Philip Lockwood. 1491-1494 [doi]
• Fusion of visual and acoustic signals for command-word recognitionRudolf Kober, Ulrich Harz, Jutta Schiffers. 1495-1497 [doi]
• Difference in visual information between face to face and telephone dialoguesYuri Iwano, Yosuke Sugita, Yusuke Kasahara, Shu Nakazato, Katsuhiko Shirai. 1499-1502 [doi]
• Cepstrum-based filter-bank design using discriminative feature extraction training at various levelsAlain Biem, Shigeru Katagiri. 1503-1506 [doi]
• Minimum error rate training for designing tree-structured probability density functionWu Chou. 1507-1510 [doi]
• A frequency-weighted HMM based on minimum error classification for noisy speech recognitionHiroshi Matsumoto, Masanori Ono. 1511-1514 [doi]
• Dictionary-based discriminative HMM parameter estimation for continuous speech recognition systemsDaniel Willett, Christoph Neukirchen, Jörg Rottland. 1515-1518 [doi]
• A DFE-based algorithm for feature selection in speech recognitionÁngel de la Torre, Antonio M. Peinado, Antonio J. Rubio, Victoria E. Sánchez. 1519-1522 [doi]
• Robustness issues and solutions in speech recognition based telephony servicesVijay Raman, Vidhya Ramanujam. 1523-1526 [doi]
• Speaker-dependent speech recognition based on phone-like units models-application to voice diallingVincent Fontaine, Hervé Bourlard. 1527-1530 [doi]
• Enhanced control and estimation of parameters for a telephone based isolated digit recognizerJosef G. Bauer. 1531-1534 [doi]
• HTIMIT and LLHDB: speech corpora for the study of handset transducer effectsDouglas A. Reynolds. 1535-1538 [doi]
• Robustness improvements in continuously spelled names over the telephoneMichael Galler, Jean-Claude Junqua. 1539-1542 [doi]
• A fast algorithm for stochastic matching with application to robust speaker verificationQi Li, Sarangarajan Parthasarathy, Aaron E. Rosenberg. 1543-1546 [doi]
• A Bayesian predictive classification approach to robust speech recognitionQiang Huo, Hui Jiang 0001, Chin-Hui Lee. 1547-1550 [doi]
• Robust speech recognition based on Viterbi Bayesian predictive classificationHui Jiang 0001, Keikichi Hirose, Qiang Huo. 1551-1554 [doi]
• Efficient mixed excitation models in LPC based prototype interpolation speech codersCharalampos Papanastasiou, Costas S. Xydeas. 1555-1558 [doi]
• High quality split band LPC vocoder operating at low bit ratesIan A. Atkinson, Suat Yeldener, Ahmet M. Kondoz. 1559-1562 [doi]
• Non-linear techniques for pitch and waveform enhancement in PWI codersHui Li, Gordon Lockhart. 1563-1566 [doi]
• Multi-prototype waveform coding using frame-by-frame analysis-by-synthesisIan S. Burnett, Duong H. Pham. 1567-1570 [doi]
• Multiband prototype waveform analysis synthesis for very low bit rate speech codingKhashayar Yaghmaie, Ahmet M. Kondoz. 1571-1574 [doi]
• A formant vocoder based on mixtures of GaussiansParham Zolfaghari, Tony Robinson. 1575-1578 [doi]
• Natural quality variable-rate spectral speech coding below 3.0 kbpsEngin Erzin, Arun Kumar, Allen Gersho. 1579-1582 [doi]
• A new 2-kbit/s speech coder based on normalized pitch waveformYusuke Hiwasaki, Kazunori Mano. 1583-1586 [doi]
• A comparison of the new 2400 bps MELP Federal Standard with other standard codersMary A. Kohler. 1587-1590 [doi]
• MELP: the new Federal Standard at 2400 bpsLynn M. Supplee, Ronald P. Cohn, John S. Collura, Alan McCree. 1591-1594 [doi]
• Using a perception-based frequency scale in waveform interpolationJes Thyssen, W. Bastiaan Kleijn, Roar Hagen. 1595-1598 [doi]
• Very low complexity interpolative speech coding at 1.2 to 2.4 kbpsYair Shoham. 1599-1602 [doi]
• Modified multiband excitation model at 2400 bpsMichele L. Jamrozik, John N. Gowdy. 1603-1606 [doi]
• Variable bit rate MBELP speech coding via V/UV distribution dependent spectral quantizationEric W. M. Yu, Cheung-fat Chan. 1607-1610 [doi]
• Voice characteristics conversion for HMM-based speech synthesis systemTakashi Masuko, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai. 1611-1614 [doi]
• The perceptual importance of selected voice quality parametersGudrun Klasmeyer. 1615-1618 [doi]
• A parametric three-dimensional model of the vocal-tract based on MRI dataHani Yehia, Mark Tiede. 1619-1622 [doi]
• Inverse filter approach to pitch modification: application to concatenative synthesis of female speechRashid Ansari. 1623-1626 [doi]
• Vowel amplitude variation during sentence productionHelen M. Hanson. 1627-1630 [doi]
• Experiments in female voice speech synthesis using a parametric articulatory modelDongbing Wei, Colin C. Goodyear. 1631-1634 [doi]
• Articulatory speech synthesis using diphone unitsAndrew Richard Greenwood. 1635-1638 [doi]
• An auditory-based measure for improved phone segment concatenationDavid T. Chappell, John H. L. Hansen. 1639-1642 [doi]
• Correlation based speech formant recoveryDouglas J. Nelson. 1643-1646 [doi]
• The modulation spectrogram: in pursuit of an invariant representation of speechSteven Greenberg, Brian Kingsbury. 1647-1650 [doi]
• From vocalic detection to automatic emergence of vowel systemsFrançois Pellegrino, Régine André-Obrecht. 1651-1654 [doi]
• Acoustic characteristics of lexical stress in continuous speechDavid van Kuijk, Louis Boves. 1655-1658 [doi]
• Pole-zero modeling of vocal tract for fricative soundsMinsheng Liu, Arild Lacroix. 1659-1662 [doi]
• Quantitative characterization of functional voice disorders using motion analysis of high-speed video and modelingThomas Wittenberg, Patrick Mergell, Monika Tigges, Ulrich Eysholdt. 1663-1666 [doi]
• Robust speech decoding: a universal approach to bit error concealmentTim Fingscheidt, Peter Vary. 1667-1670 [doi]
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• A hybrid LMS-LMF scheme for echo cancellationAzzedine Zerguine, Maamar Bettayeb, Colin F. N. Cowan. 2313-2316 [doi]
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• Adaptive sub-channel equalization in multicarrier transmissionLing Qin, Maurice G. Bellanger. 2321-2324 [doi]
• Sub-RLS algorithm with an extremely simple update equationKensaku Fujii, Juro Ohga. 2325-2328 [doi]
• Analysis of a delayless subband adaptive filterNoriyuki Hirayama, Hideaki Sakai. 2329-2332 [doi]
• A new efficient method of convergence calculation for adaptive filters using the sign algorithm with digital data inputsShin'ichi Koike. 2333-2336 [doi]
• Performance of the a priori and a posteriori QR-LSL algorithms in a limited precision environmentMaria D. Miranda, Leonardo Aguayo, Max Gerken. 2337-2340 [doi]
• Robust stability of time-variant difference equations with restricted parameter perturbations: regions in coefficient-spaceKamal Premaratne, Mohamed Mansour. 2341-2344 [doi]
• Spherical subspace and eigen based affine projection algorithmsRonald D. Degroat, Dinko Begusic, Eric M. Dowling, Darel A. Linebarger. 2345-2348 [doi]
• On the convergence and MSE of Chen's LMS adaptive algorithmSau-Gee Chen, Yung-An Kao, Ching-Yeu Chen. 2349-2352 [doi]
• Noise constrained LMS algorithmYongbin Wei, Saul B. Gelfand, James V. Krogmeier. 2353-2356 [doi]
• The log-log LMS algorithmShivaling S. Mahant-Shetti, Srinath Hosur, Alan Gatherer. 2357-2360 [doi]
• Improved fault coverage for adaptive fault tolerant filtersJ. Jiang, C. D. Schmitz, Bernard A. Schnaufer, W. Kenneth Jenkins. 2361-2364 [doi]
• Optimal noise levels for stochastic resonanceAlfredo Restrepo, Luis F. Zuluaga, Luis E. Pino. 2365-2368 [doi]
• p norm design of weighted order statistic filtersC. Emanuel Savin, M. Omair Ahmad, M. N. Shanmukha Swamy. 2369-2372 [doi]
• A constrained optimisation approach to the blind estimation of Volterra kernelsTania Stathaki, Anne Scohyers. 2373-2376 [doi]
• Improved accuracy in the singularity spectrum of multifractal chaotic time seriesOlufemi Adeyemi, Gloria Faye Boudreaux-Bartels. 2377-2380 [doi]
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• Wavelet-based transformations for nonlinear signal processingRobert D. Nowak, Richard G. Baraniuk. 2385-2388 [doi]
• Signal restoration by statistical soft morphologyElena Stringa, Carlo S. Regazzoni. 2389-2392 [doi]
• Identification and quantification of nonstationary chaotic behaviorTed Frison, Henry D. I. Abarbanel. 2393-2396 [doi]
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• Third order Volterra system identificationPanos Koukoulas, Nicholas Kalouptsidis. 2405-2408 [doi]
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• Link adaptation to channel interference using multi-rate source and channel coding for CDMA mobile communicationsS. Tateesh, S. A. Atungsiri, Ahmet M. Kondoz. 2485-2488 [doi]
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• Efficient equalization of nonlinear communication channelsWalter Frank, Ulrich Appel. 2501-2504 [doi]
• A direct equalization methodWalter Y. Chen. 2505-2508 [doi]
• Non-Wiener effects in LMS-implemented adaptive equalizersMichael Reuter, James R. Zeidler. 2509-2512 [doi]
• Joint blind equalization with a shell partition-based CMA for QAM signal constellationsGi Hun Lee, Rae-Hong Park. 2513-2516 [doi]
• Constant modulus blind equalisation algorithms under soft constraint satisfactionOguz Tanrikulu, Buyurman Baykal, Anthony G. Constantinides, Jonathon A. Chambers. 2517-2520 [doi]
• CMA beamforming for multipath correlated sourcesTuan Nguyen, Zhi Ding. 2521-2524 [doi]
• CMA convergence for constant envelope, non-zero bandwidth signalsSally L. Wood, Michael G. Larimore, John R. Treichler. 2525-2528 [doi]
• An equalization technique for OFDM and MC-CDMA in a time-varying multipath fading channelsWon-Gi Jeon, Kyung Hi Chang, Yong Soo Cho. 2529-2532 [doi]
• Detecting human face from monocular image sequences by genetic algorithmsHertog Nugroho, Shigeyoshi Takahashi, Yoshiteru Ooi, Shinji Ozawa. 2533-2536 [doi]
• Rule-based face detection in frontal viewsConstantine Kotropoulos, Ioannis Pitas. 2537-2540 [doi]
• Hierarchical filtering scheme for the detection of facial keypointsMarkus Michaelis, Rainer Herpers, Lars Witta, Gerald Sommer. 2541-2544 [doi]
• Fast face recognition method using a multistage hierarchical networkMaxim A. Grudin, David M. Harvey, Leonid I. Timchenko, Paulo J. G. Lisboa. 2545-2548 [doi]
• Restoration and resolution enhancement of video sequencesVadim Avrin, Its'hak Dinstein. 2549-2552 [doi]
• Line registration of jittered videoAnil C. Kokaram, Peter J. W. Rayner, Peter M. B. van Roosmalen, Jan Biemond. 2553-2556 [doi]
• A new auto-regressive (AR) model-based algorithm for motion picture restorationShowbhik Kalra, Man-Nang Chong, Dilip Krishnan. 2557-2560 [doi]
• Image processing techniques for blind TV ghost cancellationZhichun Lei, Peter Appelhans, Hartmut Schröder. 2561-2564 [doi]
• Removal of blocking and ringing artifacts in transform coded imagesJianping Hu, Nadir Sinaceur, Fu Li, Kwok-Wai Tam, Zhigang Fan. 2565-2568 [doi]
• Hard-constrained signal feasibility problemsPatrick L. Combettes, Pascal Bondon. 2569-2572 [doi]
• Blind restoration of blurred and noisy imagesNader Moayeri, Konstantinos Konstantinides. 2573-2576 [doi]
• Maximum likelihood estimation of blur from multiple observationsAmbasamudram N. Rajagopalan, Subhasis Chaudhuri. 2577-2580 [doi]
• Image enhancement using color and spatial informationSpiros Fotopoulos, Dimitris Sindoukas, Nikolaos A. Laskaris, George Economou. 2581-2584 [doi]
• Image enhancement by morphological pyramid decomposition and modified reconstructionLars Floreby, Farook Sattar, Göran Salomonsson. 2585-2588 [doi]
• The learning type of mean and median hybrid filtersMitsuhiko Meguro, Akira Taguchi. 2589-2592 [doi]
• The closest-to-mean filter: an edge preserving smoother for Gaussian environmentsDaniel Leo Lau, Juan Guillermo González. 2593-2596 [doi]
• Audio as a support to scene change detection and characterization of video sequencesCaterina Saraceno, Riccardo Leonardi. 2597-2600 [doi]
• Motion and shape signatures for object-based indexing of MPEG-4 compressed videoAhmet Müfit Ferman, Bilge Günsel, A. Murat Tekalp. 2601-2604 [doi]
• Using feature selection to aid an iconic search through an image databaseKieron Messer, Josef Kittler. 2605-2608 [doi]
• Hidden Markov model parsing of video programsWayne Wolf. 2609-2611 [doi]
• A human-machine interface for medical image analysis and visualization in virtual environmentsChristian Krapichler, Michael Haubner, Andreas Lösch, Karl-Hans Englmeier. 2613-2616 [doi]
• Gaze tracking for multimodal human-computer interactionRainer Stiefelhagen, Jie Yang 0001. 2617-2620 [doi]
• Digital watermarking of MPEG-2 coded video in the bitstream domainFrank Hartung, Bernd Girod. 2621-2624 [doi]
• Error concealment improvements for MPEG-2 using enhanced error detection and early re-synchronizationSusanna Aign. 2625-2628 [doi]
• MPEG-2 nonlinear temporally scalable coding and hybrid quantizationSadik Bayrakeri, Russell M. Mersereau. 2629-2632 [doi]
• Transcoding of MPEG-2 video in the frequency domainPedro A. Amado Assunção, Mohammed Ghanbari. 2633-2636 [doi]
• Optimisation of two-layer SNR scalability for MPEG-2 videoDavid Wilson, Mohammed Ghanbari. 2637-2640 [doi]
• Nonlinear predictive rate control for constant bit rate MPEG video codersYoo-Sok Saw, Peter M. Grant, John M. Hannah, Bernard Mulgrew. 2641-2644 [doi]
• Combined affine and translational motion compensation scheme using triangular tessellationsDavid Benedict Bradshaw, Nick G. Kingsbury. 2645-2648 [doi]
• 2-D mesh-based synthetic transfiguration of an object with occlusionCandemir Toklu, A. Tanju Erdem, A. Murat Tekalp. 2649-2652 [doi]
• Robust object tracking based on spatial characterization of objects by additive invariantsHolger Eggers, Fabrice Moscheni, Roberto Castagno. 2653-2656 [doi]
• A noise robust method for segmentation of moving objects in video sequencesRoland Mech, Michael Wollborn. 2657-2660 [doi]
• Video segmentation based on spatial and temporal informationJae-Gark Choi, Si-Woong Lee, Seong Dae Kim. 2661-2664 [doi]
• Combined audio and visual streams analysis for video sequence segmentationJeho Nam, Ahmed H. Tewfik. 2665-2668 [doi]
• Facial features motion analysis for wire-frame tracking in model-based moving image codingPaul M. Antoszczyszyn, John M. Hannah, Peter M. Grant. 2669-2672 [doi]
• Feedback loop for coder control in a block-based hybrid coder with mesh-based motion compensationJörn Ostermann. 2673-2676 [doi]
• Efficient motion estimation for block based video compressionKrit Panusopone, K. R. Rao. 2677-2680 [doi]
• A flexible hardware-oriented fast algorithm for motion estimationFengqi Yu, Alan N. Willson Jr.. 2681-2683 [doi]
• Robust block-matching motion-estimation technique for noisy sourcesRobert M. Armitano, Ronald W. Schafer, Frederick L. Kitson, Vasudev Bhaskaran. 2685-2688 [doi]
• Robust estimation of multi-component motion in image sequences using the epipolar constraintEckehard G. Steinbach, Subhasis Chaudhuri, Bernd Girod. 2689-2692 [doi]
• Image compression using variable blocksize vector quantization based on rate-distortion decompositionChing Yang Wang, Chyuan-Huei Thomas Yang, Long-Wen Chang. 2693-2696 [doi]
• Design of successive approximation lattice vector quantizersStephan F. Simon, Lambert Bosse. 2697-2700 [doi]
• VQ-encoding of luminance parameters in fractal coding schemesHannes Hartenstein, Dietmar Saupe, Kai-Uwe Barthel. 2701-2704 [doi]
• A system/graph theoretical analysis of attractor codersMohammad Gharavi-Alkhansari, Thomas S. Huang. 2705-2708 [doi]
• Wavelet coding of image using quadtree representation and block entropy codingWenxing Zheng, Hua Nan. 2709-2712 [doi]
• Hybrid KLT-SVD image compressionPatrick Waldemar, Tor A. Ramstad. 2713-2716 [doi]
• Embedded DCT coding with significance maskingNeil K. Laurance, Donald M. Monro. 2717-2720 [doi]
• Adaptive block-size transform coding for image compressionJavier Bracamonte, Michael Ansorge, Fausto Pellandini. 2721-2724 [doi]
• Distortion/decoding time tradeoffs in software DCT-based image codingKrisda Lengwehasatit, Antonio Ortega. 2725-2728 [doi]
• Computation-distortion characteristics of block transform codingVivek K. Goyal, Martin Vetterli. 2729-2732 [doi]
• Conditional weighted universal source codes: second order statistics in universal codingMichelle Effros. 2733-2736 [doi]
• A low-complexity error-resilient H.263 coderMasoud R. K. Khansari, Vasudev Bhaskaran. 2737-2740 [doi]
• Target detection from coregistered visual-thermal-range imagesJorge E. Pérez-Jácome, Vijay K. Madisetti. 2741-2744 [doi]
• Integration of monocular cues to create depth effectCassandra T. Swain. 2745-2748 [doi]
• Synthesis of multi viewpoint images at non-intermediate positionsAndré Redert, Emile A. Hendriks, Jan Biemond. 2749-2752 [doi]
• Performance analysis and learning approaches for vehicle detection and counting in aerial imagesVasu Parameswaran, Philippe Burlina, Rama Chellappa. 2753-2756 [doi]
• Probabilistic shape models: the role of the partition functionArnaldo J. Abrantes, Jorge S. Marques. 2757-2760 [doi]
• Multiedge detection in SAR imagesRoger Fjørtoft, Philippe Marthon, Armand Lopes, Eliane Cubero-Castan. 2761-2764 [doi]
• Entropy and multiscale analysis: a new feature extraction algorithm for aerial imagesAlexandre Winter, Henri Maître, Nicole Cambou, Eric Legrand. 2765-2768 [doi]
• Detection of linear features using a localized Radon transform with a wavelet filterAbbie L. Warrick, Pamela A. Delaney. 2769-2772 [doi]
• Oriented texture classification based on self-organizing neural network and Hough transformAparecido Nilceu Marana, Luciano da Fontoura Costa, Sergio A. Velastin, Roberto de Alencar Lotufo. 2773-2775 [doi]
• Unsupervised image segmentation using a telegraph parameterization of Pickard random fieldsYves Goussard, Jérôme Idier, Alain De Cesare. 2777-2780 [doi]
• Unsupervised Markovian segmentation of sonar imagesMax Mignotte, Christophe Collet, Patrick Pérez, Patrick Bouthemy.