Table of Contents

• On separating amplitude from frequency modulations using energy operatorsPetros Maragos, James F. Kaiser, Thomas F. Quatieri. 1-4 [doi]
• New uses for the N-Best sentence hypotheses within the BYBLOS speech recognition systemRichard M. Schwartz, Steve Austin, Francis Kubala, John Makhoul, Long Nguyen, Paul Placeway, George Zavaliagkos. 1-4 [doi]
• Selective nesting of circular convolution algorithmsRyszard Stasinski. 1-4 [doi]
• Wavenumber-domain SAR focusing from a nonuniform synthetic apertureThomas J. Flynn. 1-4 [doi]
• Coupled adaptive prediction and system identification: a statistical model and transient analysisMamadou Mboup, Madeleine Bonnet, Neil J. Bershad. 1-4 [doi]
• A gradient-based adaptive algorithm with reduced complexity, fast convergence and good tracking characteristicsJacob Benesty, Li Sheng Wen, Pierre Duhamel. 5-8 [doi]
• Standard and target driven AR-vector models for speech analysis and speaker recognitionFrédéric Bimbot, Luc Mathan, A. De Lima, Gérard Chollet. 5-8 [doi]
• A fast autofocus algorithm for synthetic aperture radar processingJørgen Dall. 5-8 [doi]
• Discrete representation of continuous signals: the Huggins transformDan Schonfeld. 5-8 [doi]
• Exploiting correlations among competing models with application to large vocabulary speech recognitionRonald Rosenfeld, Xuedong Huang 0001, Merrick L. Furst. 5-8 [doi]
• Low-delay frequency domain LMS algorithmOmar Ait Amrane, Eric Moulines, Maurice Charbit, Yves Grenier. 9-12 [doi]
• Synthetic-aperture optical array imaging system: selection of array data for single-point focusingOsamu Ikeda. 9-12 [doi]
• On inherent in-place and in-order features of the prime factor algorithmDaniel Pak-Kong Lun, Wan-Chi Siu. 9-12 [doi]
• Improvements in beam search for 10000-word continuous speech recognitionH. Ney, Reinhold Häb-Umbach, Bach-Hiep Tran, Martin Oerder. 9-12 [doi]
• Quadratic detectors for general nonlinear analysis of speechLes Atlas, Jing Fang. 9-12 [doi]
• New algorithms for the FFT computation of symmetric and translational complex conjugate sequencesChao Lu, Richard Tolimieri. 13-16 [doi]
• Conjugate gradient algorithm for adaptive echo cancellationG. K. Boray, M. D. Srinath. 13-16 [doi]
• Evaluation of a glottal ARMA model of speech productionA. P. Lobo, William A. Ainsworth. 13-16 [doi]
• Linear discriminant analysis for improved large vocabulary continuous speech recognitionReinhold Häb-Umbach, Hermann Ney. 13-16 [doi]
• A Fourier model of ISAR imaging of approaching targetsAndrew W. Krone, David C. Munson Jr.. 13-16 [doi]
• An efficient FFT algorithm for real-symmetric dataJianping Chen, Henrik Sorensen. 17-20 [doi]
• DP-based determination of F/sub 0/ contours from speech signalsAndreas Kießling 0001, Ralf Kompe, Heinrich Niemann, Elmar Nöth, Anton Batliner. 17-20 [doi]
• A fast match for continuous speech recognition using allophonic modelsLalit R. Bahl, Peter V. de Souza, P. S. Gopalakrishnan, David Nahamoo, Michael A. Picheny. 17-20 [doi]
• Motion estimation and compensation in SAR/ISAR imagingH. Yang, Mehrdad Soumekh. 17-20 [doi]
• A general form for recursive adaptive algorithms leading to an exact recursive CMAKatia Hilal, Pierre Duhamel. 17-20 [doi]
• An extended displacement operator for weakly structured covariance matricesFrançois Desbouvries, Claude Guéguen. 21-24 [doi]
• HMM-based noisy-speech pitch contour estimationYu-Hua Gu. 21-24 [doi]
• A fast least squares algorithm for constrained adaptive filteringLeonardo Silva Resende, João Marcos Travassos Romano, Maurice G. Bellanger. 21-24 [doi]
• Estimating rotation speed from projections in SARWeitong Chuang, Thomas S. Huang. 21-24 [doi]
• Continuous speech recognition by context-dependent phonetic HMM and an efficient algorithm for finding N-Best sentence hypothesesKatunobu Itou, Satoru Hayamizu, Hozumi Tanaka. 21-24 [doi]
• Application of the Dirichlet transform in analysis of nonuniformly sampled signalsAndrzej Wojtkiewicz, Michal Tuszynski. 25-28 [doi]
• Improvements to and applications of analysis of stressed speech using glottal waveformsKathleen E. Cummings, Mark A. Clements. 25-28 [doi]
• An efficient A* stack decoder algorithm for continuous speech recognition with a stochastic language modelDouglas B. Paul. 25-28 [doi]
• Statistical averaging and PARTAN-some alternatives to LMS and RLSK. Mike Tao. 25-28 [doi]
• A Bayes classifier for change detection in synthetic aperture radar imageryEric J. M. Rignot, Rama Chellappa. 25-28 [doi]
• An automatic method to estimate the time-based parameters of the glottal pulseformPaavo Alku. 29-32 [doi]
• Parametrisation of sea state from SAR imagesYves Delignon, René Garello, Alain Hillion. 29-32 [doi]
• Dividing the distributions of HMM and linear interpolation in speech recognitionKiyoshi Asai, Satoru Hayamizu, Ken'ichi Handa. 29-32 [doi]
• Use of the Gabor representation for Wigner distribution crossterm suppressionRichard S. Orr, Joel M. Morris, Shie Qian. 29-31 [doi]
• Performance and implementation of the inverse QR adaptive filterAvinash L. Ghirnikar, S. T. Alexander. 29-32 [doi]
• Optimization of acoustic-to-articulatory mappingP. P. L. Prado, E. H. Shiva, D. G. Childers. 33-36 [doi]
• Subphonetic modeling with Markov states-SenoneMei-Yuh Hwang, Xuedong Huang 0001. 33-36 [doi]
• Comparison of adaptive lattice filters to LMS transversal filters for sinusoidal cancellationRichard C. North, James R. Zeidler, Terence R. Albert, Walter H. Ku. 33-36 [doi]
• Efficient mapping scheme for the prime factor discrete Hartley transformWan-Chi Siu, Daniel Pak-Kong Lun. 33-36 [doi]
• Near field synthetic aperture ultrasonic imaging: non-destructive testingSeth D. Silverstein, Lewis J. Thomas. 33-36 [doi]
• Complexity reduction in fast RLS transversal adaptive filters with application to acoustic echo cancellationThieny Petillon, André Gilloire, Sergios Theodoridis. 37-40 [doi]
• Synthetic aperture active sonar imagingBrett L. Douglas, Hua Lee. 37-40 [doi]
• Tailored orthonormal sets for expansion of specific classes of functionsLonnie C. Ludeman. 37-40 [doi]
• Japanese dictation system using character source modelingTomokazu Yamada, Shouichi Matsunaga, Kiyohiro Shikano. 37-40 [doi]
• A new method for determining the vocal tract transfer function and its excitation from voiced speechMark M. Thomson. 37-40 [doi]
• Techniques for improving the quality of LD-CELP coders at 8 kb/sR. Soheili, Ahmet M. Kondoz, Barry G. Evans. 41-44 [doi]
• Implementation of waveform files for synthesis-by-ruleTakashi Hayashi 0005, Ken'ya Murakami. 41-44 [doi]
• DFT, convolution and error correcting codesHomayoun Shahri. 41-44 [doi]
• On the parameter estimation of the harmonic, evanescent and purely indeterministic components of homogeneous random fieldsJoseph M. Francos, Diego P. de Garrido, John W. Woods. 41-44 [doi]
• Genetic and learning automata algorithms for adaptive digital filtersR. Nambiar, C. K. K. Tang, P. Mars. 41-44 [doi]
• A method to evaluate accuracy of FFT-based periodicity analysis for short length signal in low SNRHiroshi Kanai, Noriyoshi Chubachi, Hideo Suzuki. 45-48 [doi]
• Results on local stability of fixed step size recursive algorithmsJames A. Bucklew, Thomas G. Kurtz, William A. Sethares. 45-48 [doi]
• Low-delay VXC at 8 kbit/s with interframe codingJey-Hsin Yao, John J. Shynk, Allen Gersho. 45-48 [doi]
• A first study on neural net based generation of prosodic and spectral information for Mandarin text-to-speechSin-Horng Chen, Shaw-Hwa Hwang, Chun-Yu Tsai. 45-48 [doi]
• Gibbs random fields: temperature and parameter analysisRosalind W. Picard. 45-48 [doi]
• Displacement rank of pseudo-inversesPierre Comon. 49-52 [doi]
• Optimization of intonation control using statistical F/sub 0/ resetting characteristicsYoshinori Sagisaka, Nobuyoshi Kaiki. 49-52 [doi]
• On reducing the bit rate of a CELP-based speech coderY. J. Liu. 49-52 [doi]
• QRD-based sliding window adaptive LS lattice algorithmsKarl Zhao, Fuyun Ling, Hanoch Lev-Ari, John G. Proakis. 49-52 [doi]
• MAP region segmentation based on composite random field modelsAly A. Farag. 49-52 [doi]
• A fast input reordering algorithm for the discrete cosine transformAthanassios N. Skodras. 53-55 [doi]
• Hierarchical segmentation using compound Gauss-Markov random fieldsFerran Marqués, Jordi Cunillera, Antoni Gasull. 53-56 [doi]
• Two-stage F/sub 0/ control model using syllable based F/sub 0/ unitsMasanobu Abe, Hirokazu Sato. 53-56 [doi]
• Reduced complexity CELP coderMichel Mauc, Geneviève Baudoin. 53-56 [doi]
• Neural filters: a class of filters unifying FIR and median filtersLin Yin, Jaakko Astola, Yrjö Neuvo. 53-56 [doi]
• Use of the mean-field approximation in an EM-based approach to unsupervised stochastic model-based image segmentationDavid A. Langan, Karl J. Molnar, James W. Modestino, Jun Zhang 0006. 57-60 [doi]
• An articulatory speech synthesizer based on a frequency-domain simulation of the vocal tractQiguang Lin, Gunnar Fant. 57-60 [doi]
• Application of an optimally localized and fast wavelet transform in image compressionTouradj Ebrahimi, Murat Kunt. 57-60 [doi]
• Adaptive stack filtering by LMS and perceptron learningNirwan Ansari, Yuchou Huang, Jean-Hsang Lin. 57-60 [doi]
• A multi-stage perspective on CELP speech codingPer Hedelin. 57-60 [doi]
• Exact expectation analysis of the LMS adaptive filter without the independence assumptionScott C. Douglas, Teresa Huai-Ying Meng. 61-64 [doi]
• Evidential signal processing for low-level sensor fusionScott Shaw, Tom Garvey. 61-64 [doi]
• Inference of letter-phoneme correspondences by delimiting and dynamic time warping techniquesRobert W. P. Luk, Robert I. Damper. 61-64 [doi]
• Successive orthogonalizations in the multistage CELP coderNicolas Moreau, Przemyslaw Dymarski. 61-64 [doi]
• Multiscale Markov random fields and constrained relaxation in low level image analysisPatrick Pérez, Fabrice Heitz. 61-64 [doi]
• A new excitation model for LPC vocoder at 2.4 kb/sXiongwei Zhang, Chen Xianzhi. 65-68 [doi]
• Concatenative speech synthesis by minimum distortion criteriaNaoto Iwahashi, Nobuyoshi Kaiki, Yoshinori Sagisaka. 65-68 [doi]
• Monitoring rotating machine signalsW. Armin Kittel, Monson H. Hayes. 65-68 [doi]
• Stability analysis of the noncanonical LMS (NCLMS) algorithmWoon-Seng Gan, John J. Soraghan, Robert W. Stewart, Tariq S. Durrani. 65-68 [doi]
• 2D analysis of Gabor-filter output signatures for texture segmentationDennis F. Dunn, William E. Higgins, Joseph Wakeley. 65-68 [doi]
• Rotation and gray-scale transform invariant texture recognition using hidden Markov modelJia-Lin Chen, Amlan Kundu. 69-72 [doi]
• SOCS: A speech output system from concept representationYoichi Yamashita, Riichiro Mizoguchi. 69-72 [doi]
• Improving the performance of the 16 kb/s LD-CELP speech coderJuin-Hwey Chen, Nikil Jayant, Richard V. Cox. 69-72 [doi]
• Equal convergence conditions for normal- and partitioned-frequency domain adaptive filtersPiet C. W. Sommen, E. de Wilde. 69-72 [doi]
• DSP subsystem for knowledge based health monitoring of gas turbine enginesM. N. Brown, Robert W. Stewart, Tariq S. Durrani, T. W. Buggy. 69-72 [doi]
• Extended model variety analysis for integrated processing and understanding of signalsErkan Dorken, S. Hamid Nawab, Victor R. Lesser. 73-76 [doi]
• Multimedia self-study courses in DSP and speech processingAndrew Sekey. 73-76 [doi]
• On the effectiveness of parameter reoptimization in multipulse based codersMarco Fratti, Gian Antonio Mian, Giuseppe Riccardi. 73-76 [doi]
• A rule-based text-to-speech system for PortugueseLuís C. Oliveira, Maria do Céu Guerreiro Viana Ribeiro, Isabel Trancoso. 73-76 [doi]
• Unsupervised textured image segmentationGeorge K. Gregoriou, Oleh J. Tretiak. 73-76 [doi]
• A hybrid neural network, dynamic programming word spotterTorsten Zeppenfeld, Alexander H. Waibel. 77-80 [doi]
• Tutorial visualization software for concept reinforcement in digital signal processing educationSally L. Wood. 77-80 [doi]
• Kalman filtering techniques in speech codingSam Crisafulli, James D. Mills, Robert R. Bitmead. 77-80 [doi]
• Neural network recognition of textured images using third order cumulants as functional linksF. A. DeCosta, M. F. Chouikha. 77-80 [doi]
• Optimization of IIR filter coefficients from FIR filter taps by mean field annealingRamin A. Nobakht. 77-80 [doi]
• A design lab for statistical signal processingEdward A. Lee. 81-84 [doi]
• A whole word recurrent neural network for keyword spottingK. P. Li, J. A. Naylor, M. L. Rossen. 81-84 [doi]
• Correcting slice selection axis motion artifacts in MR imagingMark Hedley, Hong Yan 0001. 81-84 [doi]
• Butterfly orthogonal structure for fast transforms, filter banks and waveletsAndrzej Drygajlo. 81-84 [doi]
• Time-scale modification of speech using an incremental time-frequency approach with waveform structure compensationBenoit Sylvestre, Peter Kabal. 81-84 [doi]
• Signal reconstruction from modified wavelet transform-An application to auditory signal processingToshio Irino, Hideki Kawahara. 85-88 [doi]
• Modeling and correction of artifacts due to z-motion in 2-D MRIJonathan K. Riek, A. Murat Tekalp, Warren E. Smith. 85-88 [doi]
• Keyword-spotting in noisy continuous speech using word pattern vector subabstraction and noise immunity learningYoichi Takebayashi, Hiroyuki Tsuboi, Hiroshi Kanazawa. 85-88 [doi]
• Support constraints for M-band alias-free filter banksStephen A. Martucci, Russell M. Mersereau. 85-88 [doi]
• Stochastic signalsManfred Herbert, Hans Wilhelm Schüßler. 85-88 [doi]
• A novel algorithm for HMM word spotting performance evaluation and error analysisJeffrey N. Marcus. 89-92 [doi]
• Digital implementation of a laser scanning confocal microscope using image enhancement algorithmsJohn Kesterson, David Koenig. 89-92 [doi]
• An efficient approximation-elimination algorithm for fast-nearest-neighbour search (speech coding)V. Ramasubramanian 0001, Kuldip K. Paliwal. 89-92 [doi]
• Computer algebra and fast algorithmsGerald E. Sobelman. 89-92 [doi]
• Sigma-delta encoding for efficient equalization of prefilter based FIR filtersScott R. Powell, Paul M. Chau. 89-92 [doi]
• Design of stable two-dimensional IIR digital filters with arbitrary magnitude functionTakao Kobayashi, Kazuyoshi Fukushi, Keiichi Tokuda, Satoshi Imai. 93-96 [doi]
• Rejection and keyword spotting algorithms for a directory assistance city name recognition applicationBenjamin Chigier. 93-96 [doi]
• An autofocus technique for imaging microscopyE. Hughlett, P. Kaiser. 93-96 [doi]
• Teaching the FFT using MatlabC. Sidney Burrus. 93-96 [doi]
• Speech coding by the efficient transformation of the spectral envelope of subwordsV. Ralph Algazi, David H. Irvine, C. Caldwell, Michael J. Ready, Kathy L. Brown, Sang Chung. 93-96 [doi]
• Training and search algorithms for an interactive wordspotting systemLynn D. Wilcox, Marcia A. Bush. 97-100 [doi]
• Technical training issues and solutions: perspectives of a DSP chip manufacturerMerle B. 'Tim' Grady. 97-100 [doi]
• Low bit-rate quantization of LSP parameters using two-dimensional differential codingChih-Chung Kuo, Fu-Rong Jean, Hsiao-Chuan Wang. 97-100 [doi]
• Design of digital filters, their implementation on a DSP-chip, measuring their performanceKarl Schwarz, Richard Reng, Hans Wilhelm Schüßler. 97-100 [doi]
• Analysis of fundamental resolution limit of ionospheric tomographyHelen Na, Hua Lee. 97-100 [doi]
• Adaptive vector quantization for waveform codingTzung Kwang Wang, John Foster, Shahryar Ardalan. 101-104 [doi]
• Complex approximation with additional constraintsMatthias Schulist. 101-104 [doi]
• A software environment for teaching image processingG. F. McLean, E. D. Graham, M. E. Jernigan. 101-104 [doi]
• Techniques for task independent word spotting in continuous speech messagesEdward M. Hofstetter, Richard C. Rose. 101-104 [doi]
• Reduction of dispersive ground-roll using time delay spectrometryPeter K. Møller, Henrik C. Larsen. 101-104 [doi]
• A digital signal processing course based on lecture/laboratory integrationVinay K. Ingle, John G. Proakis. 105-108 [doi]
• Discriminant wordspotting techniques for rejecting non-vocabulary utterances in unconstrained speechRichard C. Rose. 105-108 [doi]
• Seismic horizon picking using an artificial neural networkE. Harrigan, J. R. Kroh, William A. Sandham, Tariq S. Durrani. 105-108 [doi]
• Novel use of symmetries in linear phase digital filtersZhi-Jian (Alex) Mou. 105-108 [doi]
• Tree searched multi-stage vector quantization of LPC parameters for 4 kb/s speech codingBhaskar Bhattacharya, Wilf P. LeBlanc, Samy A. Mahmoud, Vladimir Cuperman. 105-108 [doi]
• Preconditioned iterative methods for solving Toeplitz-plus-Hankel systemsTa-Kang Ku, C. C. Jay Kuo. 109-112 [doi]
• A fast VQ codebook design algorithm for a large number of dataMitsuru Nakai, Hiroshi Shimodaira, Masayuki Kimura. 109-112 [doi]
• Improvement in finite wordlength FIR digital filter design by cascadingChristopher Young, Douglas L. Jones. 109-112 [doi]
• ADAPTIVE: a CAE tool for initial, continuous, and self-education in adaptive signal processingGlenn S. Zelniker, Maurice G. Bellanger, Hassan Mimoun, Henry A. Gancedo, Monica Murphy. 109-112 [doi]
• Robust mapping of noisy speech parameters for HMM word spottingKenney Ng, Herbert Gish, Jan Robin Rohlicek. 109-112 [doi]
• Chaotic signals and physical systemsHenry D. I. Abarbanel. 113-116 [doi]
• Synthesis/coding of audio signals using optimized waveletsDeepen Sinha, Ahmed H. Tewfik. 113-116 [doi]
• What is a multicomponent signal?Leon Cohen. 113-116 [doi]
• Gisting conversational speechJan Robin Rohlicek, Damaris M. Ayuso, Madeleine Bates, Robert J. Bobrow, A. Boulanger, Herbert Gish, Philippe Jeanrenaud, Marie Meteer, Man-Hung Siu. 113-116 [doi]
• Near-field acoustical imaging with boundary arraysSaleem A. Kassam, Richard J. Kozick. 113-116 [doi]
• Performance evaluation of analysis-by-synthesis homomorphic vocodersJae H. Chung, Ronald W. Schafer. 117-120 [doi]
• Computation of affine time-frequency distributions using the fast Mellin transformJean Philippe Ovarlez, Jacqueline Bertrand, Pierre Bertrand. 117-120 [doi]
• A novel approach to 3-dimensional holographic television display: principles and simulationsGozde Bozdagi, Levent Onural, Abdullah Atalar. 117-120 [doi]
• Signal processing in the context of chaotic signalsAlan V. Oppenheim, Gregory W. Wornell, Steven H. Isabelle, Kevin M. Cuomo. 117-120 [doi]
• Thinned lattice filter for LPC analysisCheung-fat Chan, Kwok-Wah Law. 117-120 [doi]
• A subspace fitting approach to super resolution multi-line fitting and straight edge detectionHamid K. Aghajan, Thomas Kailath. 121-124 [doi]
• RASTA-PLP speech analysis techniqueHynek Hermansky, Nelson Morgan, Aruna Bayya, Phil Kohn. 121-124 [doi]
• Improving the speech quality of cellular mobile systems under heavy fadingHuan-yu Su, Paul Mermelstein. 121-124 [doi]
• Modeling of chaotic time series for prediction, interpolation, and smoothingJohn J. SIDorowich. 121-124 [doi]
• Cycle frequency estimation of time/frequency diversity transmissions in a dense noise environmentJoEllen Wilbur. 121-124 [doi]
• Model reconstruction of chaotic dynamics: first preliminary radar resultsSimon Haykin, Henry Leung. 125-128 [doi]
• Evolution equations for continuous-scale morphologyRoger W. Brockett, Petros Maragos. 125-128 [doi]
• Exploiting recursive parameter trajectories in speech analysisNancy Hubing, Kyung Y. Yoo. 125-128 [doi]
• Study of voice packet reconstruction methods applied to CELP speech codingMei Yong. 125-128 [doi]
• An information-theoretic approach to positive time-frequency distributionsPatrick J. Loughlin, James W. Pitton, Les E. Atlas. 125-128 [doi]
• Real-time robust pitch detectorMithat C. Dogan, Jerry M. Mendel. 129-132 [doi]
• Signal separation for nonlinear dynamical systemsCory S. Myers, Steven Kay, Michael Richard. 129-132 [doi]
• Comparative study of error correction coding schemes for the GSM half-rate channelHenrik Nielsen, Keld B. Mikkelsen, Henrik B. Hansen, Yuhang Wu, Knud J. Larsen, John Aasted Sørensen. 129-132 [doi]
• Morphology on detection of calcifications in mammogramsDongming Zhao 0001, M. Shridhar, David G. Daut. 129-132 [doi]
• Hit arrays: a tool for signal design for Wigner distribution functions for frequency hop codesEdward L. Titlebaum, Sanjay K. Mehta. 129-132 [doi]
• Efficient tracking of time-varying signal subspacesCarlos E. Davila, M. S. Mobin. 133-136 [doi]
• Pitch determination of noisy speech using higher order statisticsAsunción Moreno, José A. R. Fonollosa. 133-136 [doi]
• Analysis of bone X-rays using morphological fractalsJagath Samarabandu, Raj Acharya, Ernest Hausmann, Kristin M. Allen. 133-136 [doi]
• Effects of convolution on chaotic signalsSteven H. Isabelle, Alan V. Oppenheim, Gregory W. Wornell. 133-136 [doi]
• Finite-state vector quantization over noisy channels and its application to LSP parametersYunus Hussain, Nariman Farvardin. 133-136 [doi]
• An efficient method for the detection of multiple concentric circlesXing Cao, Farzin Deravi. 137-140 [doi]
• An adaptive algorithm for mel-cepstral analysis of speechToshiaki Fukada, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai. 137-140 [doi]
• Synchronized chaotic signals and systemsLouis M. Pecora, Thomas L. Carroll. 137-140 [doi]
• Improving the performance of a mixed excitation LPC vocoder in acoustic noiseAlan McCree, Thomas P. Barnwell III. 137-140 [doi]
• New dimensions in wavelet analysisRichard G. Baraniuk, Douglas L. Jones. 137-140 [doi]
• Signal approximation via data-adaptive normalized Gaussian functions and its applications for speech processingShie Qian, Dapang Chen, Kebo Chen. 141-144 [doi]
• Signal reconstruction from Fourier transform magnitude using Markov random fields in X-ray crystallographyPeter C. Doerschuk. 141-144 [doi]
• A robust adaptive multi-spectral object detection by using wavelet transformXiaoli Yu, Irving S. Reed, W. Kraske, Alan D. Stocker. 141-144 [doi]
• A robust 2400 bit/s MBE-LPC speech coder incorporating joint source and channel codingD. Rowe, P. Secker. 141-144 [doi]
• Functional templates and their application to 3-D object recognitionRichard L. Delanoy, Jacques G. Verly, Dan E. Dudgeon. 141-144 [doi]
• Robust digital communication in time-varying noiseNickos S. Kontoyannis, Jerry R. Mitchell, A. A. (Louis) Beex. 145-148 [doi]
• Voice transformation using PSOLA techniqueHélène Valbret, Éric Moulines, Jean-Pierre Tubach. 145-148 [doi]
• A fuzzy approach to hand-written rotation-invariant character recognitionLi-Xin Wang, Jerry M. Mendel. 145-148 [doi]
• Signal reconstruction from windowed Fourier phaseJuyang Weng. 145-148 [doi]
• Improvements in 2.4 kbps high-quality speech codingJesper Haagen, Henrik Nielsen, Steffen Duus Hansen. 145-148 [doi]
• Parametric analysis methods of time-variant waveform and its patternYumi Takizawa, Atsushi Fukasawa. 149-152 [doi]
• A scheme for pitch extraction of speech using autocorrelation function with frame length proportional to the time lagKeikichi Hirose, Hiroya Fujisaki, Shigenobu Seto. 149-152 [doi]
• Phase retrieval using a window functionWooshik Kim, Monson H. Hayes. 149-152 [doi]
• Voice transmission at a very low bit rate on a noisy channel: 800 bps vocoder with error protection to 1200 bpsBenoit M. Mouy, Pierre E. de la Noue. 149-152 [doi]
• Dynamic planar warping for optical character recognitionEsther Levin, Roberto Pieraccini. 149-152 [doi]
• Multitone tracking with coupled EKFs and high order learningMiguel Angel Lagunas, Alba Pagès-Zamora. 153-156 [doi]
• Text-independent speaker recognition using neural networksHiroaki Hattori. 153-156 [doi]
• Handwritten word recognition using HMM with adaptive length Viterbi algorithmYang He 0001, Mou-Yen Chen, Amlan Kundu 0001. 153-156 [doi]
• Cinematic techniques for speech processing: temporal decomposition and multivariate linear predictionClaude Montacié, Paul Deléglise, Frédéric Bimbot, Marie-José Caraty. 153-156 [doi]
• Deconvolution/identification techniques for nonnegative signalsDennis M. Goodman, David R. Yu. 153-156 [doi]
• Comparison of text-independent speaker recognition methods using VQ-distortion and discrete/continuous HMMsTomoko Matsui, Sadaoki Furui. 157-160 [doi]
• A knowledge model based on-line recognition systemChang-Keng Lin, Kuo-Sen Chou, Bor-Shenn Jeng, Chun-Hsi Shih, Tzu-Kai Su, Tzu-I Fan. 157-160 [doi]
• On the application of Cadzow's extrapolation method of BL signalsIlan Sadka, Hanoch Ur. 157-160 [doi]
• Scaling exponents estimation from time-scale energy distributionsPaulo Gonçalves, Patrick Flandrin. 157-160 [doi]
• An automatic technique to include grammatical and morphological information in a trigram-based statistical language modelGiulio Maltese, Federico Mancini 0002. 157-160 [doi]
• Segmentation of nonstationary signalsPetar M. Djuric, Steven M. Kay, Gloria Faye Boudreaux-Bartels. 161-164 [doi]
• Continuous probabilistic acoustic map for speaker recognitionBelle L. Tseng, Frank K. Soong, Aaron E. Rosenberg. 161-164 [doi]
• On the accuracy of scanning colour imagesH. Joel Trussell, Poorvi L. Vora. 161-164 [doi]
• Cooccurrence smoothing for stochastic language modelingUte Essen, Volker Steinbiss. 161-164 [doi]
• Reconstruction of oversampled band-limited signals from Sigma Delta encoded binary sequencesSøren Hein, Avideh Zakhor. 161-164 [doi]
• Space scale analysis for image sampling and interpolationGary E. Ford, R. R. Estes, Hong Chen. 165-168 [doi]
• Least squares channel estimation for a channel with fast time variationsNing Zhou, Nils Holte. 165-168 [doi]
• Phoneme based speaker verificationMichael I. Savic, Jeffrey Sorensen. 165-168 [doi]
• Optimal MSE signal reconstruction in oversampled A/D conversion using convexityNguyen T. Thao, Martin Vetterli. 165-168 [doi]
• Task adaptation in stochastic language models for continuous speech recognitionShoichi Matsunaga, Tomokazu Yamada, Kiyohiro Shikano. 165-168 [doi]
• Adaptive IIR filtering and system identification via rational subspace methodsPhillip A. Regalia. 169-172 [doi]
• High-resolution image reconstruction from lower-resolution image sequences and space-varying image restorationA. Murat Tekalp, Mehmet K. Özkan, M. Ibrahim Sezan. 169-172 [doi]
• An adaptive spline method for signal restorationPierre Moulin. 169-172 [doi]
• Augmented phonetic map for voice verificationHarry M. Chang. 169-172 [doi]
• Hybrid grammar-bigram speech recognition system with first-order dependence modelJeremy H. Wright, Gareth J. F. Jones, E. N. Wrigley. 169-172 [doi]
• Nonlinear vectorial interpolation for speaker recognitionYifan Gong, Jean-Paul Haton. 173-176 [doi]
• Improved definition image expansionRichard R. Schultz, Robert L. Stevenson. 173-176 [doi]
• Automatic training of stochastic finite-state language models for speech understandingEgidio P. Giachin. 173-176 [doi]
• A combined Wigner-Ville and Hough transform for cross-terms suppression and optimal detection and parameter estimationSergio Barbarossa, A. Zanalda. 173-176 [doi]
• Distance matrices and modified cyclic projections for molecular conformationLee C. Potter, Da-Ming Chiang. 173-176 [doi]
• Fast computation of the Wigner distribution for finite-length signalsGregory S. Cunningham, William J. Williams. 177-180 [doi]
• Free-text speaker identification over long distance telephone channel using hypothesized phonetic segmentationYu-Hung Kao, Periagaram K. Rajasekaran, John S. Baras. 177-180 [doi]
• Polynomial spline signal processing algorithmsMichael Unser, Akram Aldroubi. 177-180 [doi]
• Systolic array implementations for Chebyshev nonuniform samplingY. S. Zhu, S. W. Leung. 177-180 [doi]
• Hidden Markov estimation for unrestricted stochastic context-free grammarsJulian Kupiec. 177-180 [doi]
• PEL-adaptive model-based interpolation of spatially subsampled imagesBabak Ayazifar, Jae S. Lim. 181-184 [doi]
• Speaker verification using temporal decorrelation post-processingLorin Netsch, George R. Doddington. 181-184 [doi]
• Distributions for time-frequency analysis: a generalization of Choi-Williams and the Butterworth distributionAntonia Papandreou, Gloria Faye Boudreaux-Bartels. 181-184 [doi]
• Integrating probabilistic LR parsing into speech understanding systemsDavid Goddeau, Victor Zue. 181-184 [doi]
• Time-varying frequency tracker with adaptive pole contractionMarina V. Dragosevic, Srdjan S. Stankovic. 181-184 [doi]
• An integrated speech-background model for robust speaker identificationDouglas A. Reynolds, Richard C. Rose. 185-188 [doi]
• A unified framework to incorporate speech and language information in spoken language processingKeh-Yih Su, Tung-Hui Chiang, Yi-Chung Lin 0001. 185-188 [doi]
• Adaptive bilinear inverse filteringHassan M. Ahmed, Fawad Rauf. 185-188 [doi]
• Affine Wigner distributionsRamachandra G. Shenoy, Thomas W. Parks. 185-188 [doi]
• Robust and physically-constrained interpolation of fluid flow fieldsJialin Zhong, Juyang Weng, Thomas S. Huang. 185-188 [doi]
• An unsupervised, sequential learning algorithm for the segmentation of speech waveforms with multiple speakersMan-Hung Siu, George Yu, Herbert Gish. 189-192 [doi]
• One-dimensional least-squares model-based halftoningDavid L. Neuhoff, Thrasyvoulos N. Pappas, Nambi Seshadri. 189-192 [doi]
• Analysis of gradient-based adaptation algorithms for linear and nonlinear recursive filtersChristophe Vignat, Christine Uhl, Sylvie Marcos. 189-192 [doi]
• An efficient algorithm for slowly-varying frequency estimationJ. I. Portillo García, José R. Casar Corredera, Gonzalo de Miguel. 189-192 [doi]
• Robust parsing for spoken language systemsStephanie Seneff. 189-192 [doi]
• Power-spectrum shaping of halftone patterns and its effect on visual appearanceTheophano Mitsa, Kevin J. Parker. 193-196 [doi]
• A speech understanding system based on statistical representation of semanticsRoberto Pieraccini, Evelyne Tzoukermann, Zakhar Gorelov, Jean-Luc Gauvain, Esther Levin, Chin-Hui Lee, Jay G. Wilpon. 193-196 [doi]
• A low cost adaptive transform decoder implementation for high-quality audioGrant Davidson, Wallace W. Anderson, Al Lovrich. 193-196 [doi]
• Time-varying higher-order spectra, generalised Wigner-Ville distribution and the analysis of underwater acoustic dataBoualem Boashash, Gordon Frazer. 193-196 [doi]
• CALF: a CORDIC adaptive lattice filterY. H. Hu, H. E. Liao. 193-196 [doi]
• Turning blue sound into blue noise (image halftoning)Thierry M. Bernard. 197-200 [doi]
• Subband coding of high-fidelity quality audio signals at 128 kbpsDo-Hui Teh, Ah-Peng Tan, Soo Ngee Koh. 197-200 [doi]
• Normalized lattice pole-zero adaptive filtersKai X. Miao, Hong Fan. 197-200 [doi]
• Analysis of transient signals using higher-order time-frequency distributionsJavier Rodríguez Fonollosa, Chrysostomos L. Nikias. 197-200 [doi]
• A real-time task-oriented speech understanding system using keyword-spottingHiroyuki Tsuboi, Yoichi Takebayashi. 197-200 [doi]
• Adaptive DCT image coding based on a three-component image modelXiaonong Ran, Nariman Farvardin. 201-204 [doi]
• High-fidelity audio compression: fractional-band waveletsChris Heegard, Talal Shamoon. 201-203 [doi]
• Coherent harmonic detection using non-stationary higher order spectraGary R. Wilson, Keith R. Hardwicke, Robert T. Trochta. 201-204 [doi]
• A structured network architecture for adaptive language acquisitionLaura G. Miller, Allen L. Gorin. 201-204 [doi]
• Performance analysis of a single-layer perceptron for a nonseparable data modelNeil J. Bershad, John J. Shynk. 201-204 [doi]
• Gauss-Newton based adaptive subspace estimationGeorge Mathew, Vellenki U. Reddy, Soura Dasgupta. 205-208 [doi]
• Assessment of cumulant-based approaches to harmonic retrievalDai C. Shin, Jerry M. Mendel. 205-208 [doi]
• Matrixing of bit rate reduced audio signalsW. R. Th. ten Kate, P. M. Boers, A. Mäkivirta, J. Kuusama, K. E. Christensen, E. Sørensen. 205-208 [doi]
• PARSEC: a structured connectionist parsing system for spoken languageAjay N. Jain, Alex Waibel, David S. Touretzky. 205-208 [doi]
• Real-time recursive two-dimensional DCT for HDTV systemsC. T. Chiu, K. J. Ray Liu. 205-208 [doi]
• Two-tone vs. random process inputs for nonlinear distortion estimationYong Soo Cho, Edward J. Powers. 209-212 [doi]
• DOA estimation via higher-order cumulants: a generalized approachZhenghao Shi, Frederick W. Fairman. 209-212 [doi]
• Selective decompression on a hierarchically coded imageH. Torbey, D. Freidlander, J. Barda. 209-212 [doi]
• Testing generality in JANUS: a multi-lingual speech translation systemLouise Osterholtz, Charles Augustine, Arthur E. McNair, Ivica Rogina, Hiroaki Saito, Tilo Sloboda, Joe Tebelskis, Alex Waibel. 209-212 [doi]
• New adaptive IIR lattice filters: input injected latticeX. Q. Liu, H. Fan, K. X. Miao. 209-212 [doi]
• Efficient grammar processing for a spoken language translation systemDavid B. Roe, Fernando C. N. Pereira, Richard Sproat, Michael D. Riley, Pedro J. Moreno, Alejandro Macarrón. 213-216 [doi]
• Adaptive state space filtering with FIR convergence behaviourWilliam W. Edmonson, Winser E. Alexander, Mohamed F. Chouikha. 213-216 [doi]
• Position, rotation, and scale invariant recognition of images using higher-order spectraVinod Chandran, Stephen L. Elgar. 213-216 [doi]
• A multi-start algorithm for signal adaptive subband systems (image coding)David Taubman, Avideh Zakhor. 213-216 [doi]
• Modeling of a room transfer function using common acoustical polesYoichi Haneda, Shoji Makino, Yutaka Kaneda. 213-216 [doi]
• Equalizability of room acousticsAnees S. Munshi. 217-220 [doi]
• The gradient adaptive split lattice algorithmChi-hsin Wu, Andrew E. Yagle. 217-219 [doi]
• Biperiodogram frequency estimates: asymptotic and finite sample sizeErdogan Dilaveroglu, Mark A. Wickert. 217-220 [doi]
• New subband decompositions and coders for image and video compressionRoberto H. Bamberger. 217-220 [doi]
• Accent phrase segmentation using pitch pattern clusteringHiroshi Shimodaira, Masayuki Kimura. 217-220 [doi]
• Signal separation in a symmetric adaptive noise canceler by output decorrelationDirk Van Compernolle, Stefaan Van Gerven. 221-224 [doi]
• An analysis/synthesis approach to real-time artificial reverberationJean-Marc Jot. 221-224 [doi]
• Automatic classification of communication signals using higher order statisticsJürgen Reichert. 221-224 [doi]
• Hierarchical video coding using a spatio-temporal subband decompositionFrank Bosveld, Reginald L. Lagendijk, Jan Biemond. 221-224 [doi]
• Automatic recognition of intonational featuresColin W. Wightman, Mari Ostendorf. 221-224 [doi]
• Ill-conditioned inverse problems: a solution using the complex cepstrum of higher order spectraDana H. Brooks. 225-228 [doi]
• Architecture for the real-time implementation of three-dimensional subband video codingJohn Hartung. 225-228 [doi]
• Analysis of the performance of adaptive IIR filters with fixed polesJ. M. Skutnik, G. Coutu, Gerard M. Exley. 225-228 [doi]
• Use of acoustic sentence level and lexical stress in HSMM speech recognitionJim L. Hieronymus, David McKelvie, Fergus R. McInnes. 225-227 [doi]
• Programmable audiogram matching using a frequency sampling filter implemented on the Texas TMS 320C30Norman D. Black, M. Lydon, N. Waterman, M. Powderly. 225-228 [doi]
• Optimal variable step LMS look-up-table plus transversal filter nonlinear echo cancellersLuis Weruaga-Prieto, Jesús Cid-Sueiro, Aníbal R. Figueiras-Vidal. 229-232 [doi]
• Temporal domain sub-band video coding with motion compensationJens-Rainer Ohm. 229-232 [doi]
• The use of emphasis to automatically summarize a spoken discourseFrancine R. Chen, Margaret Withgott. 229-232 [doi]
• Adaptive feedforward multiple-input, multiple-output active noise controlDouglas E. Melton, Richard A. Greiner. 229-232 [doi]
• Cyclic-cumulant based identification of almost periodically time-varying systems: parametric methodsAmod V. Dandawate, Georgios B. Giannakis. 229-232 [doi]
• On unique localization of constrained-signals sourcesMati Wax. 233-236 [doi]
• Subband coding of video using an edge-based vector quantization technique for compression of the upper bandsNader Mohsenian, Nasser M. Nasrabadi. 233-236 [doi]
• A vectorized systolic array for block RLS using inverse factorizationsHideaki Sakai. 233-236 [doi]
• An improved approach to the hidden Markov model decomposition of speech and noiseM. J. F. Gales, Steve J. Young. 233-236 [doi]
• A broadband pseudo-cascade active control systemJames C. Thi, Dennis R. Morgan. 233-236 [doi]
• Adaptive algorithms for underdetermined active control problemsS. J. Elliott, J. Rex. 237-240 [doi]
• Further results on smoothed reduced rank subspacesHamid Krim, John G. Proakis. 237-240 [doi]
• Speech recognition in noise using a projection-based likelihood measure for mixture density HMM'sBeth A. Carlson, Mark A. Clements. 237-240 [doi]
• Subband/VQ coding in perceptually uniform color spacesRobert E. Van Dyck, Sarah A. Rajala. 237-240 [doi]
• Interference cancellation system based on the LMS algorithmPatrícia Fernandes Campos. 237-240 [doi]
• Maximum likelihood estimation for array processing in unknown noise environmentsQiang Wu 0006, Kon Max Wong, James P. Reilly. 241-244 [doi]
• Transformations of speech spectra to a two-dimensional continuous-valued phonetic feature space for vowel trainingAndrew E. Beck, Stephen A. Zahorian. 241-244 [doi]
• A multi-frame pel-recursive algorithm for varying frame-to-frame displacement estimationJ. Huang, S. Liu, Monson H. Hayes III, Russell M. Mersereau. 241-244 [doi]
• A novel orthogonal set adaptive line enhancer tuned with fourth-order cumulantsA. G. Constantinides, Kate M. Knill, Jonathon A. Chambers. 241-244 [doi]
• Spectral contrast normalization and other techniques for speech recognition in noiseD. C. Bateman, David Bye, M. J. Hunt. 241-244 [doi]
• Adaptive multiple-input constrained pel-recursive displacement estimationSerafim N. Efstratiadis, Aggelos K. Katsaggelos. 245-248 [doi]
• A robust connected-words recognizerStefan Dobler, Peter Meyer, Hans-Wilhelm Rühl. 245-248 [doi]
• Generalized feedforward structures: a new class of adaptive filtersJosé C. Príncipe, Bert de Vries, Pedro Guedes de Oliveira. 245-248 [doi]
• Realization of cochlear filters by VLT switched capacitor biquadsJyhfong Lin, Wing-Hung Ki, K. Thompson, Shihab Shamma. 245-248 [doi]
• Direction finding with uniform circular arrays via phase mode excitation and beamspace root-MUSICMichael D. Zoltowski, Cherian P. Mathews. 245-248 [doi]
• A new eigenstructure-based parameter estimation of multichannel moving average processesLang Tong, Ruey-Wen Liu. 249-252 [doi]
• HMM modeling for speaker independent voice dialing in car environmentLorenzo Fissore, Pietro Laface, P. Ruscitti. 249-252 [doi]
• A real-time DSP implementation of a flute modelVesa Välimäki, Matti Karjalainen, Zoltán Jánosy, Unto K. Laine. 249-252 [doi]
• Implementation of the stabilized fast transversal filter algorithm on fixed point DSPRachid Atay, Pierre Baylou, Mohamed Najim. 249-252 [doi]
• The 'orthogonal algorithm' for optical flow detection using dynamic programmingGeorges Quénot. 249-252 [doi]
• ARMA filter design for music analysis/synthesisVirginia L. Stonick, Dana Massie. 253-256 [doi]
• Motion estimation optimizationSarah A. Rajala, Ikhlas M. Abdelqader, Griff L. Bilbro, Wesley E. Snyder. 253-256 [doi]
• Minimum variance signal estimation with adaptive order statistic filtersPeter M. Clarkson, Geoffrey A. Williamson. 253-256 [doi]
• Speech recognition using hidden Markov model decomposition and a general background speech modelM. Q. Wang, S. J. Young. 253-256 [doi]
• Detection and estimation of an unknown narrow-band signal in severely nonstationary noiseJ. Oli Jonsson, Allan O. Steinhardt. 253-256 [doi]
• Radon transformation of time-frequency distributions for analysis of multicomponent signalsJohn C. Wood, Daniel T. Barry. 257-260 [doi]
• Efficient joint compensation of speech for the effects of additive noise and linear filteringFu-Hua Liu, Alejandro Acero, Richard M. Stern. 257-260 [doi]
• Sound restoration in damaged acoustical recordingK. S. Jog, Ajay Ingle. 257-260 [doi]
• Effects of mutual coupling on super-resolution DF in linear arraysC. Roller, Wasyl Wasylkiwskyj. 257-260 [doi]
• On optimal brightness functions for optical flowThomas S. Denney Jr., Jerry L. Prince. 257-260 [doi]
• Telephone channel normalization for automatic speech recognitionSolomon Lerner, Baruch Mazor. 261-264 [doi]
• FSF (frequency sampling filter) bank for adaptive system identificationHitoshi Kiya, Satoshi Yamaguchi. 261-264 [doi]
• Motion estimation using lapped transformsRobert W. Young, Nick G. Kingsbury. 261-264 [doi]
• A beamformer based upon the random coefficient modelBruce Jost, Douglas B. Williams. 261-264 [doi]
• A Bayesian approach to the detection and correction of error bursts in audio signalsSimon J. Godsill, Peter J. W. Rayner. 261-264 [doi]
• Resolution analysis and enhancement of the eigenspace based rooting method of source localizationJoseph A. Marks, Kevin M. Buckley. 265-268 [doi]
• Region-based motion estimation using deterministic relaxation schemes for image sequence codingHenri Nicolas, Claude Labit. 265-268 [doi]
• Some further results on modulated/extended lapped transformsMukund Padmanabhan, Ken Martin. 265-268 [doi]
• Non-linear spectral subtraction (NSS) and hidden Markov models for robust speech recognition in car noise environmentsPhilip Lockwood, Jérôme Boudy, Marc Blanchet. 265-268 [doi]
• Analyses of the genetic algorithms in the continuous spaceXiaofeng Qi, Francesco Palmieri. 265-268 [doi]
• Models of dynamic complexity for time-series prediction (neural networks)Visakan Kadirkamanathan, Mahesan Niranjan, Frank Fallside. 269-272 [doi]
• Motion estimation with detection of occlusion areasR. Depommier, E. Dubois. 269-272 [doi]
• A robust speech/non-speech detection algorithm using time and frequency-based featuresBrian Mak, Jean-Claude Junqua, Ben Reaves. 269-272 [doi]
• Constrained signal reconstruction from wavelet transform coefficientsChristopher M. Brislawn. 269-272 [doi]
• Schur parametrization of symmetric matrices with any rank profileKlaus Diepold, Rainer Pauli. 269-272 [doi]
• Perfect reconstruction modulated filter banksJoël Mau. 273-276 [doi]
• Pitch dependent phone modelling for HMM based speech recognitionHarald Singer, Shigeki Sagayama. 273-276 [doi]
• Left ventricle motion analysis by hierarchical decompositionChang Wen Chen, Thomas S. Huang. 273-276 [doi]
• Parameter estimation for superimposed chirp signalsR. M. Liang, K. S. Arun. 273-276 [doi]
• Generalization and learning in Volterra and radial basis function networks: a theoretical analysisSean B. Holden, Peter J. W. Rayner. 273-276 [doi]
• Single sensor active noise cancellation based on the EM algorithmAlan V. Oppenheim, Ehud Weinstein, Kambiz C. Zangi, Meir Feder, D. Gauger. 277-280 [doi]
• Constrained total least squares: signal and system modelingJames A. Cadzow, D. Mitchell Wilkes. 277-280 [doi]
• Characterization of Weyl-Heisenberg frames via Poisson summation relationshipsRichard Tolimieri, Richard S. Orr. 277-280 [doi]
• Extending information sent to the input of a neural networkYvon Autret, Basel Solaiman. 277-280 [doi]
• A new approach to the time-sequential sampling problemN. Parker Willis, Yoram Bresler. 277-280 [doi]
• Oversampling in the Gabor schemeMeir Tzur, Yehoshua Y. Zeevi. 281-284 [doi]
• Hands-free voice communication in an automobile with a microphone arrayStephen Oh, Vishu Viswanathan, Panos Papamichalis. 281-284 [doi]
• Relations between fault tolerance and internal representations for multi-layer perceptronsMartin D. Emmerson, Robert I. Damper. 281-284 [doi]
• Extraction of periodic signals in colored noiseSteven Kay, Venkatesh Nagesha. 281-284 [doi]
• Property of circular convolution for subband image codingKiyoshi Nishikawa, Hitoshi Kiya, Masahiko Sagawa. 281-284 [doi]
• A family of wavelets which are dilatable by simple IIR filters (EEG analysis)Hubert Edward Hanrahan. 285-288 [doi]
• A new approach to identify the nature of the noise affecting an imageK. Chehdi, M. Sabri. 285-288 [doi]
• Beamforming microphone arrays for speech enhancementKevin R. Farrell, Richard J. Mammone, James L. Flanagan. 285-288 [doi]
• Optimal minmax estimation and the development of minmax estimation error boundsJames C. Preisig. 285-288 [doi]
• Learning the time-delay characteristics in a neural networkP. W. Rander, K. P. Unnikrishnan. 285-288 [doi]
• Time delay estimation for filtered Poisson processes using an EM-type algorithmNikolaos Antoniadis, Alfred O. Hero III. 289-292 [doi]
• A scheme for acquiring very high resolution images using multiple camerasKiyoharu Aizawa, Takashi Komatsu, Takahiro Saito. 289-292 [doi]
• A minimum classification error, maximum likelihood, neural networkHerbert Gish. 289-292 [doi]
• Speech enhancement using state dependent dynamical system modelYariv Ephraim. 289-292 [doi]
• Target description using wavelet transformIsmail Jouny. 289-292 [doi]
• Analysis and interpretation of a constrained notch filterBengt Carlsson, Peter Händel. 293-296 [doi]
• Adaptive distributed orthogonalization processing for principal components analysisHong Chen, Ruey-Wen Liu. 293-296 [doi]
• A wideband blind identification approach to speech acquisition using a microphone arrayV. C. Soon, L. Tong, Y. F. Huang, R. Liu. 293-296 [doi]
• Motion-compensated multiframe Wiener restoration of blurred and noisy image sequencesA. Tanju Erdem, M. Ibrahim Sezan, Mehmet K. Özkan. 293-296 [doi]
• Minimum-norm updating with the rank-revealing URV decompositionM. F. Griffin, E. C. Boman, G. W. Stewart. 293-296 [doi]
• An adaptive approach for optimal data reduction using recursive least squares learning methodSami Bannour, Mahmood R. Azimi-Sadjadi. 297-300 [doi]
• Realization of very high speed real-time FIR digital filtersKaushal K. Dhar. 297-300 [doi]
• The performance of Bayesian estimators in the superresolution of signal parametersAnthony Quinn. 297-300 [doi]
• Image identification and restoration in the subband domainJohn W. Woods, Jaemin Kim. 297-300 [doi]
• Dual-channel speech enhancement with auditory spectrum based constraintsSrinivas Nandkumar, John H. L. Hansen. 297-300 [doi]
• A new designing method of filter banks using all-pass polyphase filtersWenbo Wang, Dejung Wang. 301-304 [doi]
• Adaptive equalization with neural networks: new multi-layer perceptron structures and their evaluationMarcia Peng, Chrysostomos L. Nikias, John G. Proakis. 301-304 [doi]
• Vector equalization in hidden Markov models for noisy speech recognitionBiing-Hwang Juang, Kuldip K. Paliwal. 301-304 [doi]
• A nonlinear approach to estimate the amplitude of a signalPascal Bondon, Messaoud Benidir, Bernard C. Picinbono. 301-304 [doi]
• Optimal regularized image restoration with constraintsStanley J. Reeves. 301-304 [doi]
• The equivalence of TLS and correspondence analysisKung Yao, Flavio Lorenzelli, J. Kong. 305-308 [doi]
• Efficient synthesis and high-speed implementation of look-ahead recursive filtersChien-Piao Lan, Shih-Chieh Wen, Chein-Wei Jen. 305-308 [doi]
• Image restoration by complexity regularization via dynamic programmingSze-Fong Yau, Yoram Bresler. 305-308 [doi]
• A microphone array for car environmentsYves Grenier. 305-308 [doi]
• Complex neuron model with its applications to M-QAM data communications in the presence of co-channel interferencesZengjun Xiang, Guangguo Bi. 305-308 [doi]
• An efficient algorithm for joint estimation of differential time delays and frequency offsetsAhmad Reza Naghsh-Nilchi, V. John Mathews. 309-312 [doi]
• A nonlinear digital filter using fuzzy clusteringKaoru Arakawa, Yasuhiko Arakawa. 309-312 [doi]
• Robust estimation of AR parameters and its application for speech enhancementKi Yong Lee, Byung-Gook Lee, Iickho Song, SouGuil Ann. 309-312 [doi]
• ROC performance evaluation of multilayer perceptrons in the detection of one of M orthogonal signalsZoi-Heleni Michalopoulou, Loren W. Nolte, Dimitri Alexandrou. 309-312 [doi]
• Removal of 'staircase' effects in coarsely quantized video sequencesVictor Ramamoorthy. 309-312 [doi]
• A neural network for direction finding in array processingJiankan Yang, Qiang Wu 0006, James P. Reilly. 313-316 [doi]
• New results on reconstruction of continuous-tone from halftoneMostafa Analoui, Jan P. Allebach. 313-316 [doi]
• Modeling nonlinear time seriesAndrew M. Fraser. 313-316 [doi]
• CELP coding at 4.0 kb/sec and below: improvements to FS-1016Richard L. Zinser, Steven R. Koch. 313-316 [doi]
• A ring-structured adaptive notch filterWei-Ji Shyu, Jenho Tsao. 313-316 [doi]
• An investigation of chaos-oriented dimensionality algorithms applied to AR(1) processesOlivier J. J. Michel, Patrick Flandrin. 317-320 [doi]
• The design of low sensitivity digital filters using multi-criterion optimization strategiesVictor E. DeBrunner. 317-320 [doi]
• High resolution adaptive bearing estimation using a complex-weighted neural networkYupeng Chen, Chaohuan Hou. 317-320 [doi]
• Estimation of a random variable based on multidimensional dataGary L. Wise, Eric B. Hall. 317-320 [doi]
• Ultra-fast CELP coding using deterministic multi-codebook innovationsDaniel Lin. 317-320 [doi]
• IIR filters with reduced multipliers using cyclotomic polynomial numeratorsRichard J. Hartnett, Leland B. Jackson, Gloria Faye Boudreaux-Bartels. 321-324 [doi]
• Parameter estimation for linear multidimensional non-Gaussian signalsJitendra K. Tugnait. 321-324 [doi]
• Nonlinear signal processing using empirical global dynamical equationsJeffrey S. Brush, James B. Kadtke. 321-324 [doi]
• Super directive sensor array with neural network structureHidefumi Kobatake, Wataru Morita, Yoshiharu Yano. 321-324 [doi]
• Improved 4.8 kb/s CELP coding using two-stage vector quantization with multiple candidates (LCELP)Toshiki Miyano, Masahiro Serizawa, Junichi Takizawa, Shigeji Ikeda, Kazunori Ozawa. 321-324 [doi]
• Tree-structured delta codebook for an efficient implementation of CELPT. Taniguchi, Y. Tanaka, Yasuji Ohta. 325-328 [doi]
• Two-dimensional linear prediction and spectral estimation on a polar rasterWen-Hsien Fang. 325-328 [doi]
• Bootstrap: a fast blind adaptive signal separatorAbdulkadir Dinc, Yeheskel Bar-Ness. 325-328 [doi]
• Codebook prediction: a nonlinear signal modeling paradigmAndrew C. Singer, Gregory W. Wornell, Alan V. Oppenheim. 325-328 [doi]
• Analog to digital converter requirements and implementations for narrowband channelization applicationsDavid B. Chester, David H. Damerow, Clay Olmstead. 325-328 [doi]
• Resonator-in-a-loop filter banks-based on a Lerner grouping of outputsKen Martin, Mukund Padmanabhan. 329-332 [doi]
• Fractional excitation and other efficient transformed codebooks for CELP coding of speechM. Delprat, C. Gruet, F. Dervaux, C. Baroux. 329-332 [doi]
• A family of quantization based piecewise linear filter networksJohn Aasted Sørensen. 329-332 [doi]
• A new 2D fast lattice RLS algorithmX. Liu, Pierre Baylou, Mohamed Najim. 329-332 [doi]
• On error function selection for the analysis of nonlinear time seriesDaniel F. Drake, Douglas B. Williams. 329-332 [doi]
• Two-dimensional Prony modeling and parameter estimationJoseph J. Sacchini, William M. Steedly, Randolph L. Moses. 333-336 [doi]
• Excitation modeling based on speech residual informationPeter Lupini, Vladimir Cuperman. 333-336 [doi]
• Artificial neural network based stability testing of two-dimensional infinite impulse response digital filtersSeyed M. Aghili, Michael H. Thursby, D. Rao Marpaka. 333-336 [doi]
• Optimal design of 2-D IIR filters-strictly proper caseArnab K. Shaw, Pradeep Misra. 333-336 [doi]
• Optimal generalized regularization approach to parameter identification by using band-limited input signalAkira Sano, Hiroyuki Tsuji, Hiromitsu Ohmori. 333-336 [doi]
• Generalized analysis-by-synthesis coding and its application to pitch predictionW. Bastiaan Kleijn, Ravi P. Ramachandran, Peter Kroon. 337-340 [doi]
• Phoneme recognition using an auditory model and a recurrent self-organizing neural networkTimothy R. Anderson 0001. 337-340 [doi]
• Analog optical processor architectures for sinusoidal signalsJerome Tiemann, Joohwan Chun. 337-340 [doi]
• Convergence and colored noise issues in bounding ellipsoid identificationMajid Nayeri, John R. Deller Jr., M. M. Krunz. 337-340 [doi]
• 2-D system identification from bispectrum samplesA. David Salvia, Hector M. Valenzuela. 337-340 [doi]
• Using SOMs as feature extractors for speech recognitionJari Kangas, Kari Torkkola, Mikko Kokkonen. 341-344 [doi]
• Total least squares with linear constraintsEric M. Dowling, Ronald D. Degroat, Darel A. Linebarger. 341-344 [doi]
• On the design of finite wordlength IIR filters for video applicationsJanusz Konrad, Jan Radecki, Eric Dubois 0002. 341-344 [doi]
• A deconvolution-based efficient method for generating the excitation in linear predictive speech codingDomingo Docampo, Victoria Abreu-Sernández, Fernando Pérez-Cruz, Francisco González. 341-344 [doi]
• MAP estimation using a Viterbi approach with continuous-state representationCary R. Champlin, Darryl Morrell. 341-344 [doi]
• Parameter estimation in 2D fieldsNikhil Balram, José M. F. Moura. 345-348 [doi]
• Adaptive nonlinear image restoration filterRisto Suoranta, Kari-Pekka Estola. 345-348 [doi]
• Detection of the number of signals using predictive stochastic complexityShahrokh Valaee, Peter Kabal. 345-348 [doi]
• Mixture excitations and finite-state CELP speech codersAdil Benyassine, Hüseyin Abut. 345-348 [doi]
• A neural tree network for phoneme classification with experiments on the TIMIT databaseMazin G. Rahim. 345-348 [doi]
• Improved phonetically-segmented vector excitation coding at 3.4 kb/sShihua Wang, Allen Gersho. 349-352 [doi]
• Detection of number of sources via exploitation of centro-symmetry propertyGuanghan Xu, Richard H. Roy III, Thomas Kailath. 349-352 [doi]
• Multiplierless signal processors using table look-ups and residue arithmeticAlexander Skavantzos. 349-352 [doi]
• CDNN: a context dependent neural network for continuous speech recognitionHervé Bourlard, Nelson Morgan, Chuck Wooters, Steve Renals. 349-352 [doi]
• A general framework for the incorporation of uncertainty in set theoretic estimationPatrick L. Combettes, Messaoud Benidir, Bernard C. Picinbono. 349-352 [doi]
• Adaptive subspace nulling based on Karhunen-Loeve expansionWilson S. So, Allan O. Steinhardt. 353-356 [doi]
• Predictor codebooks for speaker-independent speech recognitionTakeshi Kawabata. 353-356 [doi]
• Improved adaptive detection performance via subspace processingKeith A. Burgess, Barry D. Van Veen. 353-356 [doi]
• Parallel adaptive decision feedback equalizersKalavai J. Raghunath, Keshab K. Parhi. 353-356 [doi]
• Self-structuring hidden control neural model for speech recognitionHelge B. D. Sørensen, Uwe Hartmann. 353-356 [doi]
• An improved VQ codebook design algorithm for HMMJun Mo Koo, Hwang Soo Lee, C. K. Un. 357-360 [doi]
• Image compression using block pattern-vector quantization with variable codevector dimensionsSherif A. Mohamed, Moustafa M. Fahmy. 357-360 [doi]
• A maximal invariant framework for adaptive detection with arraysSandip Bose, Allan O. Steinhardt. 357-360 [doi]
• A high-speed synthetic aperture radar processorTheodore J. Robnett, Bentsian Charny. 357-360 [doi]
• Back-propagation training of a neural network for word spottingThomas M. English, Lois C. Boggess. 357-360 [doi]
• On noisy pattern matching under geometrical constraintsSalvatore D. Morgera. 361-364 [doi]
• Mixture density estimators in Viterbi trainingChristian Wellekens. 361-364 [doi]
• Real-time detection of transient signals using spline-waveletsA. C. Cheung, Charles K. Chui, Andrew K. Chan. 361-364 [doi]
• Performance analysis for innovations-based detectionMichael J. Sousa. 361-364 [doi]
• A new connectionist architecture for word spottingMichael A. Franzini. 361-364 [doi]
• Comparison of Gaussian and neural network classifiers on vowel recognition using the discrete cosine transformDavid J. Burr. 365-368 [doi]
• HMM based on pair-wise Bayes classifiersTatsuya Kawahara, Shuji Doshita. 365-368 [doi]
• Transient signal detection in unknown colored noise fieldsVenkatesh Nagesha, Steven Kay, John Salisbury. 365-368 [doi]
• Real time implementation of wavelet transform in a dual DSP56001 systemY. Luo, D. N. Pinder, R. C. O'Driscoll. 365-368 [doi]
• Vector quantization of images using visual masking functionsRamin Baseri, V. John Mathews. 365-368 [doi]
• A hybrid neural system for driving a parallel synthesiser so as to synthesise high quality accented speechK. Mervyn Curtis, J. Burniston. 369-372 [doi]
• Application of multi-window detection to arraysRuth Onn, Allan O. Steinhardt. 369-372 [doi]
• Image coding with variable rate RVQFaouzi Kossentini, Mark J. T. Smith, Christopher F. Barnes. 369-372 [doi]
• Temporal decomposition for the initialization of a HMM isolated word-recognizerM. Taylor, Frédéric Bimbot. 369-372 [doi]
• TMS320C30 DSP based implementation of a half rate CELP coderYi-Sheng Wang, Brian M. McCarthy. 369-372 [doi]
• Robust CFAR detection in nonhomogeneous correlated interferenceHong Wang, Lujing Cai. 373-376 [doi]
• Fast multiscale statistical signal processing algorithmsAhmed H. Tewfik, M.-J. Kim. 373-376 [doi]
• Modeling improvement of the continuous hidden Markov model for speech recognitionZhi-ping Hu, Satoshi Imai. 373-376 [doi]
• Nonlinear neural network filters for image processingKamyar Rohani, Michael T. Manry. 373-376 [doi]
• Multirate codebook design with the entropy-constrained pairwise nearest neighbor (ECPNN) algorithmDiego P. de Garrido, William A. Pearlman. 373-376 [doi]
• A family of parallel hidden Markov modelsFabio Brugnara, Renato de Mori, Diego Giuliani, Maurizio Omologo. 377-380 [doi]
• Joint source-channel vector quantization using deterministic annealingDavid J. Miller 0001, Kenneth Rose. 377-380 [doi]
• Image compression using an outer product neural networkL. E. Russo, Edward C. Real. 377-380 [doi]
• Detection of the number of signals in noise with unknown, non-white covariance matricesW. G. Chen, James P. Reilly, Kon Max Wong. 377-380 [doi]
• Singularities and noise discrimination with waveletsWen-Liang Hwang, Stéphane Mallat. 377-380 [doi]
• Modeling state durations in hidden Markov models for automatic speech recognitionPadma Ramesh, Jay G. Wilpon. 381-384 [doi]
• Multiresolution representations using the auto-correlation functions of compactly supported waveletsNaoki Saito 0001, Gregory Beylkin. 381-384 [doi]
• Universal adaptive vector quantization using codebook quantization with application to image compressionKenneth Zeger, Anurag Bist. 381-384 [doi]
• A new information-theoretic criterion for detection of the number of signals in spatially correlated noise with unknown covariance matrixQ. T. Zhang, Kon Max Wong. 381-384 [doi]
• Adaptive image coding using multilayer neural networksFabio Arduini, Stefano Fioravanti, Daniele D. Giusto. 381-384 [doi]
• Stochastic vector quantization of imagesLuis Torres, E. Arias. 385-388 [doi]
• Wavelet-based lowpass/bandpass interpolationRamesh A. Gopinath, C. Sidney Burrus. 385-388 [doi]
• Adaptive vector quantizer for image compression using self-organization approachOscal T.-C. Chen, Bing J. Sheu, Wai-Chi Fang. 385-388 [doi]
• Representing dynamic features of phonetic segment in an orthogonalized codebook of HMM based speech recognition systemTsuneo Nitta, Jun'ichi Iwasaki, Yasuyuki Masai, Hiroshi Matsu'ura. 385-388 [doi]
• Bearing estimation in a colored noise background using the method of multiple windowsA. Drosopoulos, Simon Haykin. 385-388 [doi]
• Statistical evaluation of beam-space direction-of-arrival estimatorsHui Liu, Fu Li. 389-392 [doi]
• The design of generalized product-code vector quantizersWai-Yip Chan. 389-392 [doi]
• Design of multidimensional non-separable regular filter banks and waveletsJelena Kovacevic, Martin Vetterli. 389-392 [doi]
• Context modeling with the stochastic segment modelMari Ostendorf, Ibrahim Bechwati, Owen Kimball. 389-392 [doi]
• Neural network approach for adaptive vector quantization of imagesRosa Lancini, F. Perego, Stefano Tubaro. 389-392 [doi]
• Unsupervised information theory-based training algorithms for multilayer neural networksGerhard Rigoll. 393-396 [doi]
• Anisotropic edge detection using mean field annealingScott T. Acton, Alan C. Bovik. 393-396 [doi]
• Estimation of the signal component of a data vectorAbhijit A. Shah, Donald W. Tufts. 393-396 [doi]
• Fast computation of wavelet transforms with the extended lapped transformHenrique S. Malvar. 393-396 [doi]
• Rate-distortion optimization for tree-structured source coding with multi-way node decisionsGary J. Sullivan, Richard L. Baker. 393-396 [doi]
• Paraunitary filter banks and wavelet packetsAnand K. Soman, P. P. Vaidyanathan. 397-400 [doi]
• A translation/rotation/scaling/occlusion invariant neural network for 2D/3D object classificationJenq-Neng Hwang, Hang Li. 397-400 [doi]
• Systematic development of architectures for multidimensional DSP using the residue number systemDimitrios Soudris, Vassilis Paliouras, Thanos Stouraitis. 397-400 [doi]
• Context-dependent hidden control neutral network architecture for continuous speech recognitionBojan Petek, Joe Tebelskis. 397-400 [doi]
• Sequential optimization: robust subspace fitting for multiple source locationDavid J. Yemc. 397-400 [doi]
• A technique for defining the architecture and weights of a neural image classifierR. Re, Fabio Roli, Sebastiano B. Serpico, Gianni Vernazza. 401-404 [doi]
• A pyramidal scheme for lattice vector quantization of wavelet transform coefficients applied to image codingMichel Barlaud, Patrick Solé, Marc Antonini, Pierre Mathieu. 401-404 [doi]
• Minimization of frequency-weighting sensitivity in 2 D systems based on the Fornasini-Marchesini second modelTakao Hinamoto, Toshiaki Takao. 401-404 [doi]
• Maximum likelihood DOA estimation and detection without eigendecompositionA. Lee Swindlehurst. 401-404 [doi]
• Fast learning for multi-layer perceptrons using statistical techniquesEric R. Buhrke, Joseph L. LoCicero. 401-404 [doi]
• Design of signal-subspace cost functionals for parameter estimationW. Xu, Mostafa Kaveh. 405-408 [doi]
• A neural fuzzy training approach for continuous speech recognition improvementYasuhiro Komori. 405-408 [doi]
• A rank property of the generalized Hankel matrix for 2D sinusoidal sequencesJames V. Krogmeier. 405-408 [doi]
• Visual pattern recognition using morphological methodsIoannis N. M. Papadakis, James G. Reisman, Stelios C. A. Thomopoulos. 405-408 [doi]
• Wavelet transform image coding using trellis coded vector quantizationNader Moayeri, Ingrid Daubechies, Qing Song, Hong Shen Wang. 405-408 [doi]
• The fast discrete Radon transformBrian T. Kelley, Vijay K. Madisetti. 409-412 [doi]
• Image restoration using neural networksM. Teles de Figueiredo, José M. N. Leitão. 409-412 [doi]
• On the quantization efficiency of independent and uncorrelated random variablesChung J. Kuo, Yuh F. Hsu. 409-412 [doi]
• Speaker-independent phoneme recognition using large-scale neural networksSatoru Nakamura, Hidefumi Sawai, Masahide Sugiyama. 409-412 [doi]
• A new adaptive eigendecomposition algorithm based on a first-order perturbation criterionBenoît Champagne. 409-412 [doi]
• Eigen based methods to jointly estimate frequency and timing in PSK and MSK signalsMargarita Cabrera, Miguel Angel Lagunas. 413-416 [doi]
• Implantation of a high-resolution method in a sonar systemMichel Bouvet, Marc Di Martino. 413-416 [doi]
• A new algorithm to compute the discrete inverse Radon transformJoseph Segman. 413-416 [doi]
• The effect of various floating point formats on the absolute error bound in recursive filteringPeter H. Bauer, Jie Wang. 413-416 [doi]
• A multi-task neural network approach to speech recognitionE. L. Richards. 413-416 [doi]
• Adaptive beamforming with the multichannel least squares lattice using beam-space constraintsM. Wazenski, D. Alexandrou, B. Breed, D. DeFatta. 417-420 [doi]
• Time-frequency perspectives: the 'chirplet' transformSteve Mann, Simon Haykin. 417-420 [doi]
• Low roundoff noise augmented IIR filtersChimin Tsai, Adly T. Fam. 417-420 [doi]
• Prototype-based discriminative training for various speech unitsErik McDermott, Shigeru Katagiri. 417-420 [doi]
• Performance analysis for angle and polarization estimation using ESPRITJian Li 0001, R. T. Compton Jr.. 417-420 [doi]
• Incorporating acoustic-phonetic knowledge in hybrid TDNN/HMM frameworksChristian Dugast, Laurence Devillers. 421-424 [doi]
• A fast algorithm for the generation of orthogonal base functions on an arbitrarily shaped regionWilfried Philips. 421-424 [doi]
• The suboptimality of angle coding in the CORDIC algorithmDavid H. Kitabjian, Prabhakar R. Chitrapu. 421-424 [doi]
• An instrumental variable approach to array processing in spatially correlated noise fieldsPetre Stoica, Björn E. Ottersten, Mats Viberg. 421-424 [doi]
• Synthesis of spectral densities using finite automataCarlo M. Monti, Gianfranco L. Pierobon, Umberto Viaro. 421-424 [doi]
• Simultaneous estimation of time-of-arrival, pole, and amplitude parameters for transient exponential signalsJ. D. George, R. R. Muise, Jonathan S. Abel. 425-428 [doi]
• Error-correcting training for phoneme spottingLes T. Niles, Lynn D. Wilcox, Marcia A. Bush. 425-428 [doi]
• Quantization noise in sigma-delta modulators driven by deterministic inputsDavid F. Delchamps. 425-428 [doi]
• An ML algorithm for outliers detection and source localizationVictor A. N. Barroso, José M. F. Moura. 425-428 [doi]
• A new method for non-orthogonal decompositionNikolay Polyak, William A. Pearlman. 425-428 [doi]
• Parallel sequential running neural network and its application to automatic speech recognitionHuaiyu Zeng, Tiecheng Yu. 429-432 [doi]
• Alternate windows for multi-window spectral analysisThomas P. Bronez, Derrick S. Brown. 429-432 [doi]
• The influence of extended sources on the theoretical performance of the MUSIC and ESPRIT methods: narrow-band sourcesTimo-Pekka Jäntti. 429-432 [doi]
• Perfect arrays with small phase alphabetLeopold Bömer, Markus Antweiler, Hans D. Schotten. 429-432 [doi]
• Optimal synthetic FWI design of state-space digital filtersGang Li, Michel Gevers. 429-432 [doi]
• A segment-based speaker adaptation neural network applied to continuous speech recognitionKeiji Fukuzawa, Yasuhiro Komori, Hidefumi Sawai, Masahide Sugiyama. 433-436 [doi]
• Aperture extension for a towed array using an acoustic synthetic aperture or a linear prediction methodStergios Stergiopoulos, Nathan T. Allcott. 433-436 [doi]
• Roundoff and coefficient quantization noise of pipelined scattered look-ahead filters with decompositionKyungHi Chang, William G. Bliss. 433-436 [doi]
• The role of the velocity-slowness mapping in fan filtering of image sequencesThomas L. Marzetta. 433-435 [doi]
• Adaptive weighted norm linear prediction for sinusoids incorporating a priori frequency informationJan-Ti Yang, Sergio D. Cabrera. 433-436 [doi]
• Quantization of ordered dataPrashant P. Gandhi. 437-440 [doi]
• Analysis of I/O efficient order-statistic-based noise power estimatorsGeorge A. Zimmerman, Edward T. Olsen. 437-440 [doi]
• A combined analog-digital technique for normalizing video signals for the detection of moving objectsGregory W. Donohoe, Cheol-Ho Jeong. 437-440 [doi]
• Direction finding algorithms using fourth order statistics: asymptotic performance analysisEric Moulines, Jean-François Cardoso. 437-440 [doi]
• A Bayesian approach to speaker adaptation for the stochastic segment modelBurhan F. Necioglu, Mari Ostendorf, Jan Robin Rohlicek. 437-440 [doi]
• Coherent interference suppression via partially adaptive beamformingFeng Qian, Barry D. Van Veen. 441-444 [doi]
• General method for sinusoidal frequencies estimation using ARMA algorithms with nonlinear prediction error transformationAnatoliy Platonov, Zbigniew K. Gajo, Jerzy Szabatin. 441-444 [doi]
• Robust source localization based on local array response modelingBjörn E. Ottersten, Mats Viberg, Bo Wahlberg. 441-444 [doi]
• Fast facet edge detection in image sequences using vector quantizationMysore Y. Jaisimha, Eve A. Riskin, Robert M. Haralick. 441-444 [doi]
• An LVQ based reference model for speaker-adaptive speech recognitionOtto Schmidbauer, Joe Tebelskis. 441-444 [doi]
• Bidirectional motion estimation based on P frame motion vectors and area overlapWilliam E. Lynch. 445-448 [doi]
• On the optimization of spatio-temporal analysis for source motion estimationOlivier Zugmeyer, Jean-Pierre Le Cadre. 445-448 [doi]
• Robust speaker adaptation using a piecewise linear acoustic mappingJerome R. Bellegarda, Peter V. de Souza, Arthur Nádas, David Nahamoo, Michael A. Picheny, Lalit R. Bahl. 445-448 [doi]
• Approximate distribution of the parameter of a complex first-order autoregressive processMariano García Otero. 445-448 [doi]
• On the sensitivity of covariance based direction finding to gain and phase perturbationsBenjamin Friedlander. 445-448 [doi]
• Data association and tracking using hidden Markov models and dynamic programmingF. Martinerie, P. Forster. 449-452 [doi]
• Fast algorithms for block motion estimationAndré Zaccarin, Bede Liu. 449-452 [doi]
• Generalized autoregressive spectral estimationJenho Tsao, Wei-Ji Shyu. 449-452 [doi]
• A piecewise linear spectral mapping for supervised speaker adaptationHiroshi Matsukoto, Hirowo Inoue. 449-452 [doi]
• Fundamental limitations of diversely polarized antenna arraysAnthony J. Weiss, Benjamin Friedlander. 449-452 [doi]
• Rapid connectionist speaker adaptationMichael Witbrock, Patrick Haffner. 453-456 [doi]
• Further results on tradeoffs between detection and estimationBülent Baygün, Alfred O. Hero III. 453-456 [doi]
• Radar low-angle tracking with subarray level ML tracking algorithmsShi-Wei Gao, Zheng Bao. 453-456 [doi]
• Robust FSK sinusoidal frequency estimationJim Schroeder, Jim Lansford. 453-456 [doi]
• Motion compensated frame rate conversion of motion picturesReginald L. Lagendijk, M. Ibrahim Sezan. 453-46 [doi]
• Joint motion compensated prediction and interpolation of video sequencesSmita Gupta, Allen Gersho. 457-460 [doi]
• Performance analysis of wideband direction finding using interpolated arraysBenjamin Friedlander, Anthony J. Weiss. 457-460 [doi]
• Modeling exponential signals in a dispersive multipath environmentPhillip L. Ainsleigh, James D. George. 457-460 [doi]
• Speaker adaptive phoneme recognition based on feature mapping from spectral domain to probabilistic domainT. Kobayashi, Y. Uchiyama, J. Osada, K. Shirai. 457-460 [doi]
• Signal to noise ratio improvement by likelihood ratio techniques under model and environmental mismatchIsabel M. G. Lourtie, G. Clifford Carter, Sankar Basu. 457-460 [doi]
• A new criterion for adaptive beamsummingR. A. DeLap, Alfred O. Hero III. 461-464 [doi]
• Noise reduction of image sequences using adaptive motion compensated frame averagingJill M. Boyce. 461-464 [doi]
• Cross-bispectrum computation for multichannel quadratic phase coupling estimationRaghuveer M. Rao, Sohail A. Dianat. 461-464 [doi]
• Detection of weak, broadband signals under Doppler-scaled, multipath propagationHenry S. Chang. 461-464 [doi]
• Fast speaker adaptation combined with soft vector quantization in an HMM speech recognition systemFritz Class, Alfred Kaltenmeier, Peter Regel-Brietzmann, Karl Trottler. 461-464 [doi]
• A generalized normalization technique for signal detection in nonstationary correlated noiseQ. T. Zhang, H. S. Miao. 465-468 [doi]
• Comparison of motion compensation using different degrees of sub-pixel accuracy for interfield/interframe hybrid coding of HDTV image sequencesSiu-Leong Iu. 465-468 [doi]
• High-order polynomial root tracking algorithmDavid Starer, Arye Nehorai. 465-468 [doi]
• Approximating nonlinear systems by nonlinear ARMA and AR modelsRonald D. Degroat, Louis R. Hunt, Darel A. Linebarger. 465-468 [doi]
• Speaker normalization for speech recognitionXuedong Huang. 465-468 [doi]
• Pre-whitening for detection in correlated plus impulsive noiseAdam J. Efron, H. Jeen. 469-472 [doi]
• Distributed filtering without knowledge of noise distributionLang Hong. 469-472 [doi]
• Speaker independent speech recognition method using training speech from a small number of speakersMasakatsu Hoshimi, Maki Miyata, Shoji Hiraoka, Katsuyuki Niyada. 469-472 [doi]
• A framework for anisotropic adaptive filtering and analysis of image sequences and volumesHans Knutsson, Leif Haglund, Håkan Bårman, Gösta H. Granlund. 469-472 [doi]
• Detection of amplitude modulation using bispectraTaikang Ning, S. M. Gao. 469-472 [doi]
• Modified Kalman filtering with an optimal target functionLiang Li, Simon Haykin. 473-476 [doi]
• Segmental GPD training of HMM based speech recognizerWu Chou, Biing-Hwang Juang, Chin-Hui Lee. 473-476 [doi]
• Design of high-order conditional entropy coding for imagesShawmin Lei, Kou-Hu Tzou. 473-476 [doi]
• Performance analysis of a class of transient detection algorithmsBoaz Porat, Benjamin Friedlander. 473-476 [doi]
• FIR system identification using a linear combination of cumulantsJosé A. R. Fonollosa, Josep Vidal, Asunción Moreno. 473-476 [doi]
• A novel systolic array processor for MVDR beamformingC. F. T. Tang, K. J. Ray Liu. 477-480 [doi]
• Detection of transients using discrete wavelet transformAthina P. Petropulu. 477-480 [doi]
• Self-synchronizing variable-length codes for image transmissionWai-Man Lam, Amy R. Reibman. 477-480 [doi]
• Edgeworth series expansion of the conditional mean and the optimality of non-linear Volterra filtersPierre-Olivier Amblard, Daniel Baudois, Jean-Louis Lacoume. 477-480 [doi]
• Adaptation of large vocabulary recognition system parametersLalit R. Bahl, Peter V. de Souza, David Nahamoo, Michael A. Picheny, Salim Roukos. 477-480 [doi]
• Identification of two-dimensional systems using sum-of-cumulantsLuis F. Chaparro, L. Luo. 481-484 [doi]
• Classification of whale and ice sounds with a cochlear modelThomas W. Parks, Beth A. Weisburn. 481-484 [doi]
• Improved acoustic modeling with Bayesian learningJean-Luc Gauvain, Chin-Hui Lee. 481-484 [doi]
• Availability and approximation of signals in adaptive time delay estimationXuan Kong. 481-484 [doi]
• Morphological methods in image codingZiheng Zhou, Anastasios N. Venetsanopoulos. 481-484 [doi]
• Interpretation of underwater scene data acquired by a 3-D acoustic cameraFrancesco G. B. De Natale, Stefano Fioravanti, Daniele D. Giusto, Gianni Vernazza. 485-488 [doi]
• Vocabulary learning and environment normalization in vocabulary-independent speech recognitionHsiao-Wuen Hon, Kai-Fu Lee. 485-488 [doi]
• Use of higher order statistics to discriminate breaking wavesAsoke K. Nandi, C. A. Greated. 485-488 [doi]
• Asynchronous adaptive equalization in voiceband data modemsBiswa R. Ghosh, David G. Messerschmitt. 485-488 [doi]
• Fractal approximation of image blocksDonald M. Monro, Frank Dudbridge. 485-488 [doi]
• Adaptation of the HMM distributions: Application to a VQ codebook and to a noisy environmentEleftherios D. Frangoulis, Dimitrios A. Gaganelis. 489-492 [doi]
• Maximum likelihood localization of wideband sourcesMiriam A. Doron, Anthony J. Weiss. 489-492 [doi]
• Three-source boundary interpolating image coderS. Lucke, T. Rao, Russell M. Mersereau. 489-492 [doi]
• Adaptive deconvolution and identification of nonminimum phase FIR systems using Kalman filterBahram Shafai, Shaomin Mo. 489-492 [doi]
• Allocation of adaptivity of multistage digital filtersJohn R. Treichler, Michael G. Larimore, Sally L. Wood. 489-492 [doi]
• Interpolative BTC image codingBing Zeng, Yrjö Neuvo, Anastasios N. Venetsanopoulos. 493-496 [doi]
• Utilization of conditioned higher-order spectra for nonlinear system identificationSung Bae Kim, Edward J. Powers. 493-496 [doi]
• Focusing matrices for wideband array processing with no a priori angle estimatesWooyoung Hong, Ahmed H. Tewfik. 493-496 [doi]
• Discriminative template training for dynamic programming speech recognitionPao-Chung Chang, Biing-Hwang Juang. 493-496 [doi]
• A within-burst adaptive MLSE receiver for mobile TDMA cellular systemsEnrico Del Re, Guido Castellini, Laura Pierucci, Fabrizio Conti. 493-496 [doi]
• On the theory for autoregressive processesPiet M. T. Broersen, H. Einar Wensink. 497-500 [doi]
• More on the concentric ordered modulus algorithm for blind equalization of QAM and QPR modulations, with results for 64 QAM, 25 QPR and 49 QPRFernando López de Victoria. 497-500 [doi]
• An algorithm for joint vector quantizer and halftoner designRick A. Vander Kam, Philip A. Chou, Eve A. Riskin, Robert M. Gray. 497-500 [doi]
• Coherent wideband array processingEyal Doron, Miriam A. Doron. 497-500 [doi]
• Application of a generalized probabilistic descent method to dynamic time warping-based speech recognitionTakashi Komori, Shigeru Katagiri. 497-500 [doi]
• A decision directed adaptive correlator for a direct sequence spread spectrum receiverCharles N. Pateros, Gary J. Saulnier. 501-504 [doi]
• Discriminative analysis for feature reduction in automatic speech recognitionEnrico Bocchieri, Jay G. Wilpon. 501-504 [doi]
• The application of Laguerre partial series expansions for broadband array processingGeorge Henry Niezgoda, Jeffrey L. Krolik. 501-504 [doi]
• Optimal design of windows for spectral analysis of mono and bidimensional sampled signalsJoël Le Roux. 501-504 [doi]
• An interframe dynamic FSVQ codec for video sequence codingQiang Guo, Nasser M. Nasrabadi, Nader Mohsenian. 501-504 [doi]
• High performance connected digit recognition using codebook exponentsRégis Cardin, Yves Normandin, Renato de Mori. 505-508 [doi]
• Invariance of the generalized coherence estimate with respect to reference channel statisticsDana Sinno, Douglas Cochran. 505-508 [doi]
• Moving object detection and trajectory estimation in the transform/spatiotemporal mixed domainKnud Steven Knudsen, Leonard T. Bruton. 505-508 [doi]
• Computer simulation study of MTI using FFT and nonlinear processorsK. M. M. Prabhu, K. Bhoopathy Bagan. 505-508 [doi]
• Adaptive implementation of minimum-error-rate equalizers via backpropagation neural networksXiao-Hu Yu, Shi Xin Cheng. 505-508 [doi]
• Hidden Markov models using vector linear prediction and discriminative output distributionsPhilip C. Woodland. 509-512 [doi]
• Deterministic scanning and hybrid algorithms for fast decoding of IFS (iterated function system) encoded image setsHarvey A. Cohen. 509-512 [doi]
• Ambiguity structure of multipath channelsMaria-João D. Rendas, José M. F. Moura. 509-512 [doi]
• A simplified structure for echo cancellers with phase-roll compensationMahdi Y. Zaidan, Mohamed S. El-Hennawey. 509-512 [doi]
• Multiple frequencies and AR parameters estimation from one bit quantized signal via the EM algorithmIlan Ziskind, David Hertz. 509-512 [doi]
• Cost/benefit selection of spectral estimators for use with ultrasonic Doppler blood flow instrumentsM. Graça Ruano, Peter J. Fish. 513-516 [doi]
• Fast echo cancellation in a voice-processing systemVijay R. Raman, Mark R. Cromack. 513-516 [doi]
• On the performance of polynomial and HMM whole-word classifiers for digit recognition over telephoneHarald Katterfeldt, Peter Regel-Brietzmann, B. Vater. 513-516 [doi]
• Use of a pyramidal and polynomial transform for image sequence codingF. García-Ugalde, Claude Labit, V. Nzomigni. 513-516 [doi]
• Cramer Rao bounds on direction estimates for closely spaced emitters in multi-dimensional applicationsJack Jachner, Harry B. Lee. 513-516 [doi]
• Estimation of the fundamental frequency of a noisy sum of cisoids with harmonic related frequenciesAndré Ferrari, Gérard Alengrin, Céline Theys. 517-520 [doi]
• SWITCHBOARD: telephone speech corpus for research and developmentJohn J. Godfrey, Edward Holliman, Jane McDaniel. 517-520 [doi]
• Ultimate array accuracy and resolution in the presence of near-field emittersHamid R. Karimi, Athanassios Manikas. 517-520 [doi]
• Fuzzy subimage classification in image sequence codingSeong-Gon Kong, Bart Kosko. 517-520 [doi]
• Multichannel general order FTF algorithm and its application to the equalization of mobile communication channelsTülay Adali, Sasan H. Ardalan, Ali S. Sadri. 517-520 [doi]
• Bias analysis of source location estimates from eigenspace-based spatial-spectrum estimatorsXiao-Liang Xu, Kevin Buckley. 521-524 [doi]
• An efficient fractionally spaced equalizer with low computations for data transmissionS. H. Leung, Bao Ling Chan, S. M. Lau. 521-524 [doi]
• Recognition of hesitations in spontaneous speechDouglas D. O'Shaughnessy. 521-524 [doi]
• Spectral estimation based on AR-model excited by t-distribution processJunibakti Sanubari, Keiichi Tokuda, Mahoki Onoda. 521-524 [doi]
• Bit allocation and rate control based on human visual sensitivity for interframe codersHiroshi Watanabe, Sharad Singhal. 521-524 [doi]
• An improved speech detection algorithm for isolated Korean utterancesMinsoo Hahn, Chan Kyung Park. 525-528 [doi]
• Complex system identification methods for fast echo canceler initializationSyed Arif Ahmed, J. R. Cruz. 525-528 [doi]
• Localization and CRLB validity with Doppler and bearing measurementsJ. J. Towers, Yiu-Tong Chan. 525-528 [doi]
• Segmentation-based coding of motion difference and motion field images for low bit-rate video compressionSam Liu, Monson Hayes. 525-528 [doi]
• ARMA model order determination and MDL: a new perspectiveD. Mitchell Wilkes, Gang Liang, James A. Cadzow. 525-528 [doi]
• Super high definition image digitizing systemIsao Furukawa, Kazunobu Kashiwabuchi, Sadayasu Ono. 529-532 [doi]
• An alternative DFE adaptation methodW.-Y. Chen. 529-532 [doi]
• Static representation of speech dynamics for isolated word recognitionChorkin Chan, Jian-Xiong Wu. 529-532 [doi]
• Recursive 'ML' bearing estimation: initialization and sources number updatePascal Larzabal, Henri Clergeot. 529-532 [doi]
• Block time and frequency domain modified covariance algorithmsAndreas Spanias, Gim Lim, Philipos C. Loizou, Michael E. Deisher. 529-532 [doi]
• Robust automatic time alignment of orthographic transcriptions with unconstrained speechBarbara Wheatley, George R. Doddington, Charles T. Hemphill, John J. Godfrey, Edward Holliman, Jane McDaniel, Drew Fisher. 533-536 [doi]
• A low rank weighted matrix approximation method for robust estimation of sinusoid parametersGeoffrey S. Edelson, Ramdas Kumaresan, Donald W. Tufts. 533-536 [doi]
• Fast algorithms for the computation of the finite length decision feedback equalizerNaofal Al-Dhahir, John M. Cioffi. 533-536 [doi]
• Optimal bearing and bearing rate estimation using an ambiguity surfaceD. E. Ohlms, D. N. Nitka. 533-536 [doi]
• A 12.8 GFLOPS multi-DSP system for super high definition image processingTomoko Sawabe, Tetsurou Fujii, Hiroshi Nakada, Naohisa Ohta, Sadayasu Ono. 533-536 [doi]
• Decimation filter compiler for oversampling A/D applicationsSoei-Shin Hang, Rajeev Jain. 537-540 [doi]
• Gradient-based adaptive filters for non-Gaussian noise environmentsGeoffrey A. Williamson, Peter M. Clarkson. 537-540 [doi]
• The development of super high definition image storage systemTokumichi Murakami, Hideo Ohira, Okikazu Tanno, Ryuuta Suzuki, Minoru Wada, Taku Saito, Koji Ogura, Kohtaro Asai. 537-540 [doi]
• Performance analysis for DOA estimation algorithms using physical parametersFu Li, Hui Liu, Richard J. Vaccaro. 537-540 [doi]
• On increasing structural complexity of finite state speech modelsSaeed Vaseghi, P. N. Conner. 537-540 [doi]
• A new fine-frequency estimation algorithm based on parabolic regressionC. Mark McIntyre, David A. Dermott. 541-544 [doi]
• Application of the modulation model to speech recognitionAdam B. Fineberg, Richard J. Mammone, James L. Flanagan. 541-544 [doi]
• A VLSI chip set for ghost cancellation and waveform equalization of analog television signalsCole Erskine, Sergio Kusevitzky, Junichi Orihara, Hakuo Watanabe. 541-544 [doi]
• Automatic generation of architectural models for designing dedicated VLIW signal processorsG. Menez, Michel Auguin, Fernand Boéri, C. Carrière. 541-544 [doi]
• Conventional interference cancellation for minimum redundancy array structureErdal Panayirci, Yeheskel Bar-Ness, Wan-Ling Chen. 541-544 [doi]
• SDS: a framework for the design of DSP ASICsMagdy A. Bayoumi, N. A. Ramakrishna, V. Israni, R. K. Jayam. 545-548 [doi]
• Resynchronization of motion compensated video affected by ATM cell lossPaul Haskell, David G. Messerschmitt. 545-548 [doi]
• Comparison of adaptive blind equalizersYuang Lou. 545-548 [doi]
• HMM representation of quantized articulatory features for recognition of highly confusable wordsKevin Erler, Li Deng 0001. 545-548 [doi]
• New results of the twin processor method of measurement of the channel and/or scattering functionsSanjay K. Mehta, Edward L. Titlebaum. 545-548 [doi]
• Asynchronous multirate system design for programmable DSPsIchiro Kuroda, Takao Nishitani. 549-552 [doi]
• Automatic monitoring of the quality of cable television picturesQ. Zhang, R. Ward. 549-552 [doi]
• Lattice filter interpretations of the Chandrasekhar recursions for estimation and spectral factorizationAli H. Sayed, Hanoch Lev-Ari, Thomas Kailath. 549-552 [doi]
• On the use of acoustic-phonetic features in interactive labelling of multi-lingual speech corporaPaul Dalsgaard, Ove Andersen, William J. Barry, R. Jørgensen. 549-552 [doi]
• Time-delay estimation for deterministic transient signals in a multipath environmentRichard J. Vaccaro, Thulasinath G. Manickam. 549-552 [doi]
• Multidimensional nonlinear systems and structure theoremsIrwin W. Sandberg. 553-555 [doi]
• Phonemic HMM constrained by statistical VQ-code transitionSatoshi Takahashi, Tatsuo Matsuoka, Kiyohiro Shikano. 553-556 [doi]
• Globally convergent blind equalization algorithms for complex data systemsKen Yamazaki, Rodney A. Kennedy, Zhi Ding 0001. 553-556 [doi]
• Direct synthesis of optimized DSP assembly code from signal flow block diagramsDouglas B. Powell, Edward A. Lee, William C. Newman. 553-556 [doi]
• Resolving the components of transient signals by a multistage procedureSrinivasan Umesh, Donald W. Tufts. 553-556 [doi]
• Multiple parameter estimation of composite signals in uncharacterized impulsive noiseAlfred Mertins, Norbert J. Fliege. 557-560 [doi]
• Analog median filteringSteffen Paul, Knut Hüper, Josef A. Nossek. 557-560 [doi]
• A class of robust nonlinear filters for signal decomposition and filtering, utilizing the Haar basisHarold G. Longbotham. 557-560 [doi]
• A group-theoretic framework for fault-tolerant computationPaul E. Beckmann, Bruce R. Musicus. 557-560 [doi]
• Experiments on speaker-independent phone recognition using BREFLori Faith Lamel, Jean-Luc Gauvain. 557-560 [doi]
• Identification of Volterra systems with a polynomial neural networkRobert E. Parker Jr., Murali Tummala. 561-564 [doi]
• Image partition using an iterative multi-resolution smoothing algorithmJosé Crespo, Ronald W. Schafer. 561-564 [doi]
• Well-behaved dataflow programs for DSP computationGuang R. Gao, R. Govindarajan, Prakash Panangaden. 561-564 [doi]
• Experiments on stress-dependent phone modelling for continuous speech recognitionMartine Adda-Decker, Gilles Adda. 561-564 [doi]
• An image processing approach to frequency trackingJonathan S. Abel, Ho John Lee, Augustus P. Lowell. 561-564 [doi]
• Use of semi-Markov models for speaker-independent phoneme recognitionNimal Ratnayake, Michael I. Savic, Jeffrey Sorensen. 565-568 [doi]
• Relation of SNR thresholds for time delay estimation to available prior informationAriela Zeira, Peter M. Schultheiss. 565-568 [doi]
• Multispectral image segmentation using a multiscale modelCharles A. Bouman, Michael Shapiro. 565-568 [doi]
• Unfolding and retiming data-flow DSP programs for RISC multiprocessor schedulingLiang-Fang Chao, Edwin Hsing-Mean Sha. 565-568 [doi]
• Modeling chaotic systems with hidden Markov modelsCory S. Myers, Andrew Singer, Frances Bongjoo Shin, Eugene Church. 565-568 [doi]
• Distributed hidden Markov model training on loosely-coupled multiprocessor networksJ. T. Foote, Michael M. Hochberg, Peter M. Athanas, Aaron Smith, Michael E. Wazlowski, Harvey F. Silverman. 569-572 [doi]
• The general use of tying in phoneme-based HMM speech recognisersSteve J. Young. 569-572 [doi]
• Segmentation and mapping of highly convoluted contours with applications to medical imagesChristos Davatzikos, Jerry L. Prince. 569-572 [doi]
• Sum-difference stereo transform codingJames D. Johnston, A. J. Ferreira. 569-572 [doi]
• Fast implementation of recursive programs using transformationsMiodrag Potkonjak, Jan M. Rabaey. 569-572 [doi]
• A successive state splitting algorithm for efficient allophone modelingJun-ichi Takami, Shigeki Sagayama. 573-576 [doi]
• Inductive techniques for formal verification of systolic array designs in DSP applicationsNam Ling, Timothy K. Shih, Jonathan Huang. 573-576 [doi]
• Analysis of cardiac late potentials using the Wigner-Ville distributionD. J. Waldo, B. Ravi Sekar Reddy, A. J. Greenspon, G. A. Kidwell, Prabhakar R. Chitrapu. 573-576 [doi]
• Satellite image classification using a modified Metropolis dynamicsZoltan Kato, Josiane Zerubia, Marc Berthod. 573-576 [doi]
• Wake detection: a multichannel approach (SAR image)J. V. Candy. 577-580 [doi]
• Acoustic modelling of subword units in the Isadora speech recognizerErnst Günter Schukat-Talamazzini, Heinrich Niemann, Wieland Eckert, Thomas Kuhn 0002, Stefan Rieck. 577-580 [doi]
• The time-sequenced adaptive filter for linear prediction of the intraventricular electrogramCynthia J. Finelli, Gregory H. Wakefield, Janice M. Jenkins, Lorenzo A. DiCarlo. 577-580 [doi]
• Generation and implementation of DSP parallel programs from a signal processing design environmentGagan Mirchandani, Robert B. Pegram III, Jonathan Michel, Peter A. Twombly. 577-580 [doi]
• On compressing method of EEG data for their digital databaseYasuhiro Ohtaki, Kazuo Toraichi, Y. Ishiyama. 581-584 [doi]
• Recursive estimation techniques for detection of small objects in infrared image dataTarun Soni, James R. Zeidler, Walter H. Ku. 581-584 [doi]
• Recognition of demisyllable based units using semicontinuous hidden Markov modelsBernd Plannerer, Günther Ruske. 581-584 [doi]
• A compiler for multiprocessor DSP implementationPhu Hoang, Jan M. Rabaey. 581-584 [doi]
• A systematic partitioning approach for LS and SVD problems to fixed size arrays with constraintsFlavio Lorenzelli, Kung Yao. 585-588 [doi]
• Integration of STFT and Wigner analysis in a knowledge-based system for sound understandingNabil N. Bitar, S. Hamid Nawab, Erkan Dorken, D. E. Paneras. 585-588 [doi]
• A new algorithm for extracting fetal ECG signal using singular value decomposition methodShuqiu Li, Zi-Qiang Hou, Qi-Hu Li. 585-588 [doi]
• The automatic recognition of stop consonants using hidden Markov modelsT. Waardenburg, Johan A. du Preez, M. W. Coetzer. 585-588 [doi]
• Relationship among phoneme/word recognition rate, perplexity and sentence recognition and comparison of language modelsSeiichi Nakagawa, Isao Murase. 589-592 [doi]
• A tactile sensing algorithm based on elastic transfer function of surface deformationHiroyuki Shinoda, Shigeru Ando. 589-592 [doi]
• VLSI architectures for Dirichlet arithmeticGary A. Ray. 589-592 [doi]
• Task scheduling in the Georgia Tech digital signal multiprocessorBryce A. Curtis, Vijay K. Madisetti. 589-592 [doi]
• Hybrid segmental-LVQ/HMM for large vocabulary speech recognitionYan Ming Cheng, Douglas D. O'Shaughnessy, Vishwa Gupta, Patrick Kenny, Matthew Lennig, Paul Mermelstein, Sarangarajan Parthasarathy. 593-596 [doi]
• Recursive estimation of facial expression and movementHaibo Li 0001, Pertti Roivainen, Robert Forchheimer. 593-596 [doi]
• Blind deconvolution of sparse spike trains using stochastic optimizationYves Goussard. 593-596 [doi]
• Optimal automatic periodic multiprocessor compiler for multi-bus networksPedro R. Gelabert, Thomas P. Barnwell III. 593-596 [doi]
• Optimal wavelets for signal decomposition and the existence of scale-limited signalsJ. E. Odegard, Ramesh A. Gopinath, C. Sidney Burrus. 597-600 [doi]
• Continuous speech recognition with modified learning vector quantization algorithm and two-level DP-matchingShozo Makino, Mitsuru Endo, Toshio Sone, Ken'iti Kido. 597-600 [doi]
• Macropipelining based heterogeneous multiprocessor schedulingTakeo Hamada, Sati Banerjee, Paul M. Chau, Ronald D. Fellman. 597-600 [doi]
• Connectionist probability estimation in the DECIPHER speech recognition systemSteve Renals, Nelson Morgan, Michael Cohen, Horacio Franco. 601-604 [doi]