Abstract is missing.
- 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]
- On optimal and minimum-entropy decodingW. Bastiaan Kleijn. 1671-1674 [doi]
- A new sinusoidal phase modeling algorithmSassan Ahmadi, Andreas S. Spanias. 1675-1678 [doi]
- Recursive and adaptive predictive coding of speechKazuo Nakata, Kin-ichi Higure. 1679-1682 [doi]
- The multimodal multipulse excitation vocoderTakahiro Unno, Thomas P. Barnwell III, Mark A. Clements. 1683-1686 [doi]
- Minimum variance distortionless response (MVDR) modeling of voiced speechManohar N. Murthi, Bhaskar D. Rao. 1687-1690 [doi]
- Phase modelling of speech excitation for low bit-rate sinusoidal transform codingXiaoqin Sun, Fabrice Plante, Barry M. G. Cheetham, Kenneth W. T. Wong. 1691-1694 [doi]
- An adaptive-rate digital communication system for speechJohn E. Kleider, William M. Campbell. 1695-1698 [doi]
- Smoothing the evolution of the spectral parameters in linear prediction of speech using target matchingMohammad Zad-issa, Peter Kabal. 1699-1702 [doi]
- Comparative study of different parameters for temporal decomposition based speech codingShahrokh Ghaemmaghami, Mohamed Deriche, Boualem Boashash. 1703-1706 [doi]
- Efficient algorithm to compute LSP parameters from 10th-order LPC coefficientsSara Grassi, Alain Dufaux, Michael Ansorge, Fausto Pellandini. 1707-1710 [doi]
- Speech compression with preservation of speaker identityJohn Leis, Mark Phythian, Sridha Sridharan. 1711-1714 [doi]
- A method for measuring modulation transmission in speech transmitted via a nonlinear channelJuha Backman. 1715-1718 [doi]
- Perceptual entropy rate estimates for the phonemes of American EnglishVincent van de Laar, W. Bastiaan Kleijn, Ed F. Deprettere. 1719-1722 [doi]
- Rescoring under fuzzy measures with a multilayer neural network in a rule-based speech recognition systemOlivier Oppizzi, Régis Quélavoine. 1723-1726 [doi]
- Optimization of HMM by a genetic algorithmChak-Wai Chau, Sam Kwong, C. K. Diu, Wolfgang R. Fahrner. 1727-1730 [doi]
- Inference of variable-length acoustic units for continuous speech recognitionSabine Deligne, Frédéric Bimbot. 1731-1734 [doi]
- Comparative performance analysis of statistical trajectory models in cellular environmentBojan Petek, Ove Andersen, Paul Dalsgaard. 1735-1738 [doi]
- Inter-digit HMM: connected digit recognition using the Macrophone corpusYu-Hung Kao, Lorin Netsch. 1739-1742 [doi]
- Wide context acoustic modeling in read vs. spontaneous speechMichael Finke, Ivica Rogina. 1743-1746 [doi]
- Performance of hybrid MMI-connectionist/HMM systems on the WSJ speech databaseJörg Rottland, Christoph Neukirchen, Daniel Willett. 1747-1750 [doi]
- Statistical modeling of co-articulation in continuous speech based on data driven interpolationDon X. Sun. 1751-1754 [doi]
- Microsegment-based connected digit recognitionJohn J. Godfrey, Aravind Ganapathiraju, Coimbatore S. Ramalingam, Joseph Picone. 1755-1758 [doi]
- Context-dependent hybrid HME/HMM speech recognition using polyphone clustering decision treesJürgen Fritsch, Michael Finke, Alex Waibel. 1759-1762 [doi]
- Improved automatic recognition of Norwegian natural numbers by incorporating phonetic knowledgeKnut Kvale, Ingunn Amdal. 1763-1766 [doi]
- Hybrid HMM/ANN systems for training independent tasks: experiments on Phonebook and related improvementsStéphane Dupont, Hervé Bourlard, Olivier Deroo, Vincent Fontaine, Jean-Marc Boite. 1767-1770 [doi]
- European speech databases for telephone applicationsHarald Höge, Herbert S. Tropf, Richard Winski, Henk van den Heuvel, Reinhold Haeb-Umbach, Khalid Choukri. 1771-1774 [doi]
- Development of a large vocabulary speech database for CantonesePak-Chung Ching, Ka-Fai Chow, Tan Lee, Alfred Ying Pang Ng, Lai-Wan Chan. 1775-1778 [doi]
- An approach to continuous speech recognition based on layered self-adjusting decoding graphQiru Zhou, Wu Chou. 1779-1782 [doi]
- Look-ahead techniques for fast beam searchStefan Ortmanns, Andreas Eiden, Hermann Ney, Norbert Coenen. 1783-1786 [doi]
- An efficient search method for large-vocabulary continuous-speech recognitionKen Hanazawa, Yasuhiro Minami, Sadaoki Furui. 1787-1790 [doi]
- Extensions to the word graph method for large vocabulary continuous speech recognitionHermann Ney, Stefan Ortmanns, Ingo Lindam. 1791-1794 [doi]
- An O(N√E¯) Viterbi algorithmSarvar Patel. 1795-1798 [doi]
- CCLMDS'96: towards a speaker-independent large-vocabulary Mandarin dictation systemTung-Hui Chiang, Chung-Mou Pengwu, Shih-Chieh Chien, Chao-Huang Chang. 1799-1802 [doi]
- Japanese large-vocabulary continuous-speech recognition using a business-newspaper corpusTatsuo Matsuoka, Katsutoshi Ohtsuki, Takeshi Mori, Kotaro Yoshida, Sadaoki Furui, Katsuhiko Shirai. 1803-1806 [doi]
- A PC-based real-time large vocabulary continuous speech recognizer for GermanMeinrad Niemöller, Alfred Hauenstein, Erwin Marschall, Petra Witschel, Ulrike Harke. 1807-1810 [doi]
- Progress in recognizing conversational telephone speechBarbara Peskin, Larry Gillick, Natalie Liberman, Michael Newman, Paul van Mulbregt, Steven Wegmann. 1811-1814 [doi]
- Recognition of conversational telephone speech using the JANUS speech engineTorsten Zeppenfeld, Michael Finke, Klaus Ries, Martin Westphal, Alex Waibel. 1815-1818 [doi]
- Approaches to phoneme-based topic spotting: an experimental comparisonRoland Kuhn, Peter Nowell, Caroline Drouin. 1819-1822 [doi]
- A keyword selection strategy for dialogue move recognition and multi-class topic identificationPhilip N. Garner, Aidan Hemsworth. 1823-1826 [doi]
- Improved lexicon modeling for continuous speech recognitionSeong-Jin Yun, Yung-Hwan Oh, Gyung-Chul Shin. 1827-1830 [doi]
- Interpolation, spectrum analysis, error-control coding, and fault-tolerant computingJosé M. N. Vieira, Paulo Jorge S. G. Ferreira. 1831-1834 [doi]
- Analysis of the stability of time-domain source separation algorithms for convolutively mixed signalsYannick Deville, Nabil Charkani. 1835-1838 [doi]
- Time-varying reconstruction of stationary processes subjected to analogue periodic scramblingAlban Duverdier, Bernard Lacaze. 1839-1842 [doi]
- Signal recovery from grouped dataMiroslaw Pawlak, Ulrich Stadtmüller. 1843-1844 [doi]
- Two-dimensional pilot-symbol-aided channel estimation by Wiener filteringPeter A. Hoeher, Stefan Kaiser, Patrick Robertson. 1845-1848 [doi]
- Blind equalization of switching channels by ICA and learning of learning rateHoward Hua Yang, Shun-ichi Amari. 1849-1852 [doi]
- Adaptive soft-constraint satisfaction (SCS) algorithms for fractionally-spaced blind equalizersBuyurman Baykal, Oguz Tanrikulu, Jonathon A. Chambers. 1853-1856 [doi]
- Complete iterative reconstruction algorithms for irregularly sampled data in spline-like spacesAkram Aldroubi, Hans Georg Feichtinger. 1857-1860 [doi]
- Signal de-noising using the wavelet transform and regularizationA. Sony John, Uday B. Desai. 1861-1864 [doi]
- Exact multichannel deconvolution on radial domainsStephen D. Casey, Carlos Alberto Berenstein, David Francis Walnut. 1865-1868 [doi]
- Signal reconstruction from phase only information and application to blind system estimationHaralambos Pozidis, Athina P. Petropulu. 1869-1872 [doi]
- A fast Gauss-Newton parallel-cascade adaptive truncated Volterra filterThomas M. Panicker, V. John Mathews. 1873-1876 [doi]
- Sufficient stability bounds for slowly varying discrete-time recursive linear filtersAlberto Carini, V. John Mathews, Giovanni L. Sicuranza. 1877-1880 [doi]
- Spread spectrum interference suppression using adaptive time-frequency tilingsBrian S. Krongold, Michael Kramer, Kannan Ramchandran, Douglas L. Jones. 1881-1884 [doi]
- A DSP based long distance echo canceller using short length centered adaptive filtersPaulo A. C. Marques, Fernando Manuel Gomes de Sousa, José M. N. Leitão. 1885-1888 [doi]
- Optimal and robust shockwave detection and estimationBrian M. Sadler, Laurel C. Sadler, Tien Pham. 1889-1892 [doi]
- Automatic fault monitoring using acoustic emissionsGopal T. Venkatesan, Dennis West, Kevin M. Buckley, Ahmed H. Tewfik, Mostafa Kaveh. 1893-1896 [doi]
- A new algorithm for double talk detection and separation in the context of digital mobile radio telephoneHassan Ezzaidi, Ivan Bourmeyster, Jean Rouat. 1897-1900 [doi]
- Transmission of chosen transform coefficients of normalized cardiac beats for compressionSupratim Saha, Angarai Ganesan Ramakrishnan. 1901-1904 [doi]
- FIR filters in continuous-time envelope constrained filter designBa-Ngu Vo, Thi-Ngoc Ho, Antonio Cantoni, Victor Sreeram. 1905-1908 [doi]
- Modified cepstral analysis for accurate estimation of echo parameters in telecommunication networksMatteo Bertocco, Dennis Lorenzin, Pietro Paglierani. 1909-1912 [doi]
- Bispectral reconstruction using incomplete phase knowledge: a neuroelectric signal estimation applicationOlivier Meste. 1913-1916 [doi]
- Optimal phase-locked loop design with Kalman predictors for synchronous networksGustavo A. Hirchoren, Dalton S. Arantes. 1917-1920 [doi]
- Overparametrization in adaptive filtersAlbertus C. den Brinker. 1921-1924 [doi]
- Recursive estimation of linearly or harmonically modulated frequencies of multiple cisoids in noisePetr Tichavský, Peter Händel. 1925-1928 [doi]
- Selective coefficient update of gradient-based adaptive algorithmsTyseer Aboulnasr, Khaled A. Mayyas. 1929-1932 [doi]
- A pipelined architecture for LMS adaptive FIR filters without adaptation delayQuanhong Zhu, Scott C. Douglas, Kent F. Smith. 1933-1936 [doi]
- Deterministic stabilty analyses of unit-norm constrained algorithms for unbiased adaptive IIR filteringMarkus Rupp, Scott C. Douglas. 1937-1940 [doi]
- A modified normalized lattice adaptive filter for fast samplingParthapratim De, H. Howard Fan. 1941-1944 [doi]
- Symmetric alpha-stable filter theoryJohn S. Bodenschatz. 1945-1948 [doi]
- Adaptive channel equalization using context treesOwen E. Kelly, Don H. Johnson. 1949-1952 [doi]
- Subband adaptive filtering with time-varying nonuniform filter banksMichael L. McCloud, Delores M. Etter. 1953-1956 [doi]
- Best input for optimal tracking randomly time-varying systems: justification of adaptive predictive structureSofia Ben Jebara, Meriem Jaïdane-Saïdane. 1957-1960 [doi]
- Stability of variable and random stepsize LMSSaul B. Gelfand, Yongbin Wei, James V. Krogmeier. 1961-1963 [doi]
- Fast least-squares polynomial approximation in moving time windowsErich Fuchs, Klaus Donner. 1965-1968 [doi]
- An efficient Haar wavelet-based approach for the harmonic retrieval problemYi Chu, Wen-Hsien Fang, Shun-Hsyung Chang. 1969-1972 [doi]
- Wavelet transform based fast approximate Fourier transformHaitao Guo, C. Sidney Burrus. 1973-1976 [doi]
- On computing the 2-D extended lapped transformsDragutin Sevic, Miodrag Popovic. 1977-1980 [doi]
- Efficient computation of the discrete Wigner distribution function through a new iterative algorithmIsabel García, Consuelo Gonzalo, Margarita Pérez-Castellanos, José A. Moreno, José Sánchez-Dehesa. 1981-1984 [doi]
- Probabilistic complexity analysis of incremental DFT algorithmsJoseph M. Winograd, S. Hamid Nawab. 1985-1988 [doi]
- On the recursive total least-squaresCuong Pham, Tokunbo Ogunfunmi. 1989-1992 [doi]
- Parallel-recursive filter structures for the computation of discrete transformsRichard J. Kozick, Maurice F. Aburdene. 1993-1996 [doi]
- Basefield transforms derived from character tablesAndreas Klappenecker. 1997-2000 [doi]
- Block-recursive filters and filter-banksGábor Péceli, Annamária R. Várkonyi-Kóczy. 2001-2004 [doi]
- Fast approximate DCT: basic-idea, error analysis, applicationsAbdulnasir Hossen, Ulrich Heute. 2005-2008 [doi]
- Fast sliding transforms in transform-domain adaptive filteringAnnamária R. Várkonyi-Kóczy, Sergios Theodoridis. 2009-2012 [doi]
- Time dependent autoregressive spectrum estimation of heart wall vibrationsHiroshi Kanai, Michie Sato, Noriyoshi Chubachi. 2013-2016 [doi]
- Properties of the structured auto-regressive time-frequency distributionJakob Ängeby. 2017-2020 [doi]
- Zero-tracking time-frequency distributionsChenshu Wang, Moeness G. Amin. 2021-2024 [doi]
- Vector sampling expansion: deterministic and stochastic signalsDaniel Seidner, Meir Feder. 2025-2028 [doi]
- Optimal time segmentation for signal modeling and compressionPaolo Prandoni, Michael M. Goodwin, Martin Vetterli. 2029-2032 [doi]
- Pre-filtering for the initialization of multi-wavelet transformsMichael J. Vrhel, Akram Aldroubi. 2033-2036 [doi]
- Matching pursuit with damped sinusoidsMichael M. Goodwin. 2037-2040 [doi]
- Localized subclasses of quadratic time-frequency representationsAntonia Papandreou-Suppappola, Robin L. Murray, Gloria Faye Boudreaux-Bartels. 2041-2044 [doi]
- Class-dependent, discrete time-frequency distributions via operator theoryJack McLaughlin, James Droppo, Les E. Atlas. 2045-2048 [doi]
- Extending the characteristic function method for joint a-b and time-frequency analysisFranz Hlawatsch, Teresa Twaroch. 2049-2052 [doi]
- An architecture for realization of the cross-terms free polynomial Wigner-Ville distributionLJubisa Stankovic, Srdjan Stankovic, Igor Djurovic. 2053-2056 [doi]
- Understanding discrete rotationsMichael S. Richman, Thomas W. Parks. 2057-2060 [doi]
- Design of RNS frequency sampling filter banksUwe Meyer-Bäse, Jon Mellott, Fred J. Taylor. 2061-2064 [doi]
- Optimal design of multirate systems with constraintsWilliam M. Campbell, Thomas W. Parks. 2065-2068 [doi]
- Design of conjugate quadrature filters having specified zerosWayne Lawton, Charles A. Micchelli. 2069-2072 [doi]
- Design of paraunitary oversampled cosine-modulated filter banksJörg Kliewer, Alfred Mertins. 2073-2076 [doi]
- Time-domain design of linear-phase PR filter banksMasaaki Ikehara, Truong Q. Nguyen. 2077-2080 [doi]
- QMF filter bank design by a new global optimization methodBenjamin W. Wah, Yi Shang, Tao Wang, Ting Yu. 2081-2084 [doi]
- Unified approach to the design of quadrature-mirror filtersVijay K. Jain. 2085-2088 [doi]
- Inverse filter technique for high-precision ultrasonic pulsed wave range Doppler sensorsHeinrich Ruser, Martin Vossiek, Alexander von Jena, Valentin Mágori. 2089-2092 [doi]
- Classification of piano sounds using time-frequency signal analysisChristoph Delfs, Friedrich Jondral. 2093-2096 [doi]
- Transform/subband representations for signals with arbitrarily shaped regions of supportJohn G. Apostolopoulos, Jae S. Lim. 2097-2100 [doi]
- On optimum oversampling in the Gabor schemeMartin J. Bastiaans. 2101-2104 [doi]
- The discrete-time frequency warped wavelet transformsGianpaolo Evangelista, Sergio Cavaliere. 2105-2108 [doi]
- Aspects of spectrum and hybrid spectrum analysis for sensor SNR determinationArvid Breitenbach. 2109-2112 [doi]
- Generalized sampling without bandlimiting constraintsMichael Unser, Josiane Zerubia. 2113-2116 [doi]
- Wavelet packets and genetic algorithmsJaakko Astola, Karen Egiazarian, Heikki Huttunen. 2117-2120 [doi]
- An iterative algorithm for time-variant filtering in the discrete Gabor transform domainXiang-Gen Xia, Shie Qian. 2121-2124 [doi]
- Time-frequency analysis of acoustic transientsPatrick J. Loughlin, Dale Groutage, Robert Rohrbaugh. 2125-2128 [doi]
- Boundary-compensated wavelet basesMark Allan Coffey. 2129-2132 [doi]
- Eliminating interference terms in the Wigner distribution using extended libraries of basesIsrael Cohen, Shalom Raz, David Malah. 2133-2136 [doi]
- A flexible tiling of the time axis for adaptive wavelet packet decompositionsAntonio S. Pena, Nuria González Prelcic, Carlos A. Serantes. 2137-2140 [doi]
- Time delay calculation of stress waves using wavelet analysis application in canine edematous lungsMehran Jahed, Bizhan Najafi, Ali Khamene, Stephen J. Lai-Fook. 2141-2144 [doi]
- Affine order statistic filters: a data-adaptive filtering framework for nonstationary signalsAlexander Flaig, Gonzalo R. Arce, Kenneth E. Barner. 2145-2148 [doi]
- Limits of finite wordlength FIR digital filter designDusan M. Kodek. 2149-2152 [doi]
- Elimination of limit cycles due to two's complement quantization in normal form digital filtersGuo-Fang Xu, Tamal Bose, Jim Schroeder. 2153-2156 [doi]
- On the design of multidimensional FIR filters by transformationLina J. Karam. 2157-2160 [doi]
- Approximation of complex-valued 2-D frequency response specifications by separable-denominator digital filtersHartmut Brandenstein, Rolf Unbehauen. 2161-2164 [doi]
- Properties of approximate Parks-McClellan filtersLi Lee, Alan V. Oppenheim. 2165-2168 [doi]
- Design of nonlinear phase FIR digital filters using quadratic programmingMathias C. Lang. 2169-2172 [doi]
- Design of polar-separable FIR filters by radial slice approximationsRichard Rau, James H. McClellan. 2173-2176 [doi]
- Analysis of limit cycles in the direct form delta operator structure by computer-aided testJuha Kauraniemi. 2177-2180 [doi]
- The synthesis of sharp diamond-shaped filters using the frequency response masking approachYong Ching Lim, Seo-How Low. 2181-2184 [doi]
- Minimization of finite wordlength error in 2-D FIR digital filters in the frequency domainMitsuhiko Yagyu, Akinori Nishihara, Nobuo Fujii. 2185-2188 [doi]
- Tradeoff between roundoff and overflow errors in digital filter realizationsJosé L. Sanz-González. 2189-2192 [doi]
- Design of sharp FIR bandstop filters using quadrature masking filtersRui Yang, Yong Ching Lim, Maurice G. Bellanger. 2193-2196 [doi]
- Implementation options for block floating point digital filtersKamen R. Ralev, Peter H. Bauer. 2197-2200 [doi]
- Design of multiplierless elliptic IIR filtersLjiljana D. Milic, Miroslav D. Lutovac. 2201-2204 [doi]
- Realizable warped IIR filters and their propertiesMatti Karjalainen, Aki Härmä, Unto K. Laine. 2205-2208 [doi]
- New exchange rules for IIR filter designIvan W. Selesnick. 2209-2212 [doi]
- The design of polyphase-based IIR multiband filtersArtur Krukowski, Izzet Kale, Richard C. S. Morling. 2213-2216 [doi]
- A pipelined/interleaved IIR digital filter architectureZhongnong Jiang, Alan N. Willson Jr.. 2217-2220 [doi]
- A new approach to the phase error and THD improvement in linear phase IIR filtersBojan Djokic, Miroslav D. Lutovac, Miodrag Popovic. 2221-2224 [doi]
- Design of recursive digital filters with magnitude specificationsAshraf Alkhairy. 2225-2227 [doi]
- FIR compaction filters: new design methods and propertiesAhmet Kirac, Palghat P. Vaidyanathan. 2229-2232 [doi]
- Performance of fractional-delay filters using optimal offset windowsAnush Yardim, Gerald D. Cain, Arnaud Lavergne. 2233-2236 [doi]
- FIR filtering of nonuniformly sampled signalsAndrzej Tarczynski, Vesa Välimäki, Gerald D. Cain. 2237-2240 [doi]
- Fractional discrete-time linear systemsManuel Duarte Ortigueira. 2241-2244 [doi]
- An efficient fractional sample delayer for digital beam steeringN. Paul Murphy, Artur Krukowski, Andrzej Tarczynski. 2245-2248 [doi]
- Interactive DSP course development/teaching environmentChaouki T. Abdallah, Dalton S. Arantes, Gregory L. Heileman, Don R. Hush, Ramiro Jordan, Roberto de Alencar Lotufo, Neeraj Magotra, L. Howard Pollard, Edl Schamiloglu, Robert Whitman. 2249-2252 [doi]
- An interactive DSP tutorial on the WebMartti Rahkila, Matti Karjalainen. 2253-2256 [doi]
- Efficient realization of the block frequency domain adaptive filterDaniël W. E. Schobben, Gerard P. M. Egelmeers, Piet C. W. Sommen. 2257-2260 [doi]
- A complementary pair LMS algorithm for adaptive filteringWoo-Jin Song, Min-Soo Park. 2261-2264 [doi]
- Using a lattice algorithm to estimate the Kalman gain vector in fast Newton-type adaptive filteringMarc Moonen, Ian K. Proudler. 2265-2268 [doi]
- New Bussgang methods for blind equalizationJoão-Batista Destro-Filho, Gérard Favier, João M. T. Romano. 2269-2272 [doi]
- A refined class of cost functions in blind equalizationV. Shtrom, H. Howard Fan. 2273-2276 [doi]
- Novel blind variants of the OBE algorithmMajid Nayeri, T. M. Lin, John R. Deller Jr.. 2277-2280 [doi]
- Conjugate gradient method for adaptive direction-of-arrival estimation of coherent signalsPi Sheng Chang, Alan N. Willson Jr.. 2281-2284 [doi]
- A polyphase IIR adaptive filter: error surface analysis and applicationPhillip M. S. Burt, Max Gerken. 2285-2288 [doi]
- Adaptive periodic IIR filtersJ. William Whikehart, Soura Dasgupta. 2289-2292 [doi]
- Adaptive AR spectral estimation based on multi-band decomposition of the linear prediction error with variable forgetting factorsFernando Gil Vianna Resende Jr., Paulo S. R. Diniz, Mineo Kaneko, Akinori Nishihara. 2293-2296 [doi]
- Controlled convergence of QR least-squares adaptive algorithms-application to speech echo cancellationFrançois Capman, Jérôme Boudy, Philip Lockwood. 2297-2300 [doi]
- Iterative total least squares filter in robot navigationLaurence Tianruo Yang, Man Lin. 2301-2304 [doi]
- A new two-dimensional parallel block adaptive filter with reduced computational complexityShigenori Kinjo, Mirai Oshiro, Hiroshi Ochi. 2305-2308 [doi]
- An over-sampling subband adaptive filter with the optimal real filter bankYoshito Higa, Hiroshi Ochi, Shigenori Kinjo. 2309-2312 [doi]
- A hybrid LMS-LMF scheme for echo cancellationAzzedine Zerguine, Maamar Bettayeb, Colin F. N. Cowan. 2313-2316 [doi]
- A robust frequency-domain echo cancellerTomas Gänsler. 2317-2320 [doi]
- 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]
- Identification and compensation of the electrodynamic transducer nonlinearitiesHans Schurer, Alex G. J. Nijmeijer, Mark A. Boer, Cornelis H. Slump, Otto E. Herrmann. 2381-2384 [doi]
- 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]
- Using orthogonal least squares identification for adaptive nonlinear filtering of GSM signalsJean-Pierre Costa, Thierry Pitarque, Eric Thierry. 2397-2400 [doi]
- Numerical integration of nonlinear multidimensional systemsDaniel Homm, Rudolf Rabenstein. 2401-2404 [doi]
- Third order Volterra system identificationPanos Koukoulas, Nicholas Kalouptsidis. 2405-2408 [doi]
- Demodulation of discrete multicomponent AM-FM signals using periodic algebraic separation and energy demodulationBalasubramaniam Santhanam, Petros Maragos. 2409-2412 [doi]
- A new approach to optimal nonlinear filteringSubhash Challa, Farhan A. Faruqi. 2413-2416 [doi]
- Volterra series based modeling and compensation of nonlinearities in high power amplifiersMurali Tummla, Michael T. Donovan, Bruce E. Watkins, Richard North. 2417-2420 [doi]
- Lapped biorthogonal transforms for transform coding with reduced blocking and ringing artifactsHenrique S. Malvar. 2421-2424 [doi]
- Optimum low cost two channel IIR orthonormal filter bankJamal Tuqan, Palghat P. Vaidyanathan. 2425-2428 [doi]
- Direct design of nonuniform filter banksTakayuki Nagai, Takaaki Futie, Masaaki Ikehara. 2429-2432 [doi]
- Reconstruction for novel sampling structuresCormac Herley. 2433-2436 [doi]
- Comparison of two eigenstructure algorithms for lossless multirate filter optimizationDong-Yan Huang, Phillip A. Regalia, Maurice G. Bellanger. 2437-2440 [doi]
- Parametrization of discrete finite biorthogonal wavelets with linear phaseFrank Hartenstein. 2441-2444 [doi]
- Efficiently VLSI-realizable prototype filters for modulated filter banksTanja Karp, Alfred Mertins, Truong Q. Nguyen. 2445-2448 [doi]
- Theory of cyclic filter banksPalghat P. Vaidyanathan, Ahmet Kirac. 2449-2452 [doi]
- Oversampled filter banks: optimal noise shaping, design freedom, and noise analysisHelmut Bölcskei, Franz Hlawatsch. 2453-2456 [doi]
- A new approach to the compensation of aliasing in transform and subband codersFrank Heinle. 2457-2460 [doi]
- Parameterization of symmetric multiwaveletsPeter Rieder. 2461-2464 [doi]
- Downscaled inverses for M-channel lapped transformsRicardo L. de Queiroz, Reiner Eschbach. 2465-2468 [doi]
- Time-varying filter banks with variable system delayGerald Schuller. 2469-2472 [doi]
- Balanced multiwaveletsJérôme Lebrun, Martin Vetterli. 2473-2476 [doi]
- Performance costs for theoretical minimal-length equalizersMichael G. Larimore, Sally L. Wood, John R. Treichler. 2477-2480 [doi]
- Nonlinear channel equalizer using Gaussian sum approximationsPatrick Grohan, Sylvie Marcos. 2481-2484 [doi]
- 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]
- Adaptive DFE algorithms for IS-136 based TDMA cellular phonesAhmad Bahai, Markus Rupp. 2489-2492 [doi]
- Effective-fourth-order resonator based MASH bandpass sigma-delta modulatorsMohammed Al-Janabi, Izzet Kale, Richard C. S. Morling. 2493-2496 [doi]
- Equalisation of time variant multipath channels using amplitude banded techniquesTetsuya Shimamura, Colin F. N. Cowan. 2497-2500 [doi]
- 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.