Table of Contents

• Deterministic characteristics of LPC distances: An inconsistency with perceptual evidenceHaiyan Yé, Denis Tuffelli. 1-4 [doi]
• Diagnostic information from subjective and objective intelligibility testsHerman J. M. Steeneken. 5-8 [doi]
• Vocal tract simulation: Implementation of continuous variations of the length in a Kelly-Lochbaum model, effects of area function spatial samplingH. Y. Wu, Pierre Badin, Y. M. Cheng, Bernard Guérin. 9-12 [doi]
• Modeling spectral speech transitions using temporal decomposition techniquesGunnar Ahlbom, Frédéric Bimbot, Gérard Chollet. 13-16 [doi]
• Information-theoretic compressibility of speech dataL. Thomas Ramsey, David Gribble. 17-20 [doi]
• Complexity reduced lattice filters for digital speech processingYuval Bistritz, Hanoch Lev-Ari, Thomas Kailath. 21-24 [doi]
• A minimum discrimination information approach for hidden Markov modelingYariv Ephraim, Amir Dembo, Lawrence R. Rabiner. 25-28 [doi]
• On the distributions of small-sample estimates of second-order AR process parcor coefficientsMariano García Otero, José R. Casar Corredera. 29-32 [doi]
• On the statistical sufficiency of the coherently averaged covariance matrix for the estimation of the parameters of wideband sourcesHsien-Sen Hung, Mostafa Kaveh. 33-36 [doi]
• Properties and generation of non-Gaussian time seriesDon H. Johnson, P. Srinivasa Rao. 37-40 [doi]
• Fast algorithms for QR and Cholesky factors of Toeplitz operatorsChristophe P. Rialan, Louis L. Scharf. 41-44 [doi]
• Statistically/Computationally efficient estimation of non-Gaussian autoregressive processesSteven Kay, Debasis Sengupta. 45-48 [doi]
• Maximum likelihood estimation of a class of non-Gaussian densities with application to deconvolutionTrung T. Pham, Rui J. P. deFigueiredo. 49-52 [doi]
• Incorporated robustness in narrow-band signal subspace spatial spectral estimatorsKevin M. Buckley. 53-56 [doi]
• On identification of non-Gaussian time seriesRonald R. Mohler, Z. Tang. 57-60 [doi]
• ARMA Modeling using cumulant and autocorrelation statisticsGeorgios B. Giannakis, Jerry M. Mendel, W. Wang. 61-64 [doi]
• The four tones recognition of continuous Chinese speechHengjie Ma. 65-68 [doi]
• Dynamic programming speech recognition using a context-free grammarHermann Ney. 69-72 [doi]
• A stochastic segment model for phoneme-based continuous speech recognitionSalim E. Roucos, Mari O. Dunham. 73-76 [doi]
• On the automatic segmentation of speech signalsTorbjørn Svendsen, Frank K. Soong. 77-80 [doi]
• Phoneme classification for real time speech recognition of ItalianPaolo D'Orta, Marco Ferretti, Stefano Scarci. 81-84 [doi]
• A connected speech recognition system based on spotting diphone-like segments-Preliminary resultsA. E. Rosenberg, A. M. Colla. 85-88 [doi]
• BYBLOS: The BBN continuous speech recognition systemYen-Lu Chow, Mari O. Dunham, Owen Kimball, Michael A. Krasner, Francis Kubala, John Makhoul, P. J. Price, Salim E. Roucos, Richard M. Schwartz. 89-92 [doi]
• Continuous speech recognition by means of acoustic/ Phonetic classification obtained from a hidden Markov modelStephen E. Levinson. 93-96 [doi]
• An early-decision, real-time, connected-speech recognizerHarvey F. Silverman, David P. Morgan, Susan H. Miller. 97-100 [doi]
• A performance evaluation of a connected digit recognizerLawrence R. Rabiner, Jay G. Wilpon, Biing-Hwang Juang. 101-104 [doi]
• Adaptive filtering without a desired signalLloyd J. Griffiths, Michael J. Rude. 105-108 [doi]
• "On the probability density function of the LMS adaptive filter weights"Neil J. Bershad, Lian Zuo Qu. 109-112 [doi]
• Non-linear adaptive signal processorMiguel Angel Lagunas, Francesc Vallverdú, M. E. Santamaria. 113-116 [doi]
• Suppression and detection of impulse type interference using adaptive median hybrid filtersAri Nieminen, Pekka Heinonen, Yrjö Neuvo. 117-120 [doi]
• A criticism of the parametric EEG spike detectorMichael P. Beddoes, L. Panych, Juan Qian, Juhn A. Wada. 121-124 [doi]
• Misadjustment expressions for infinite impulse response adaptive filtersGuy R. L. Sohie, Saud A. Alshibani. 125-128 [doi]
• The use of large adaptation gains to remove the SPR condition from recursive adaptive algorithmsKun Tang, Charles E. Rohrs. 129-132 [doi]
• Exciting conditions for quantized state adaptive algorithmsWilliam A. Sethares, C. Richard Johnson Jr.. 133-136 [doi]
• Assessment of finite precision limitations in LMS and BLMS adaptive algorithmsGanapati Panda, C. F. N. Cowan, Peter M. Grant. 137-140 [doi]
• OCF-A new coding algorithm for high quality sound signalsKarlheinz Brandenburg. 141-144 [doi]
• Causal constraints on the active control of soundPhilip A. Nelson, J. K. Hammond, Stephen J. Elliott. 145-148 [doi]
• Expansion of listening area with good localization in audio conferencingShigeaki Aoki, Nobuo Koizumi. 149-152 [doi]
• "A methodology for evaluating the performance of dynamic range control algorithms for speech enhancement"John T. Lynch. 153-156 [doi]
• Spectral distribution and damping factors measurements of musical strings using FFT techniquesAntoine J. Chaigne. 157-160 [doi]
• Neural network in the auditory system: Influence of the temporal context on the response represented by a random fieldThierry Hervé, Jean Marc Dolmazon, Jacques Demongeot. 161-164 [doi]
• Cancellation of acoustic noise in a pipe using digital adaptive filtersStuart J. Flockton. 165-168 [doi]
• Adaptive noise reduction in aircraft communication systemsJeffrey J. Rodriguez, Jae S. Lim, Elliot Singer. 169-172 [doi]
• Practical adaptive noise reduction in the aircraft cockpit environmentG. A. Powell, P. Darlington, P. D. Wheeler. 173-176 [doi]
• A speech enhancement method based on Kalman filteringKuldip K. Paliwal, Anjan Basu. 177-180 [doi]
• Co-Channel speech separationD. G. Childers, C. K. Lee. 181-184 [doi]
• Reducing cocktail party noise by adaptive array filteringH. Liang, N. Malik. 185-188 [doi]
• Iterative speech enhancement with spectral constraintsJohn H. L. Hansen, Mark A. Clements. 189-192 [doi]
• Enhancement of block-coded speechDale E. Veeneman, Baruch Mazor. 193-196 [doi]
• A multivariate voicing decision rule adapts to noise, distortion, and spectral shapingDavid L. Thomson. 197-200 [doi]
• Methods for noise cancellation based on the EM algorithmMeir Feder, Alan V. Oppenheim, Ehud Weinstein. 201-204 [doi]
• Techniques for suppression of an interfering talker in co-channel speechJ. A. Naylor, Steven F. Boll. 205-208 [doi]
• An algorithm of signal approximation by hybrid splineKazuo Toraichi, Iwao Sekita, Ryoichi Mori. 209-212 [doi]
• Feature extraction by spline function for relaxation matchingIwao Sekita, Kazuo Toraichi, Masaru Kamada, Ryoichi Mori, Kazuhiko Yamamoto, Hiromitsu Yamada. 213-216 [doi]
• Invariant planar shape recognition using dynamic alignmentLalit Gupta, Mandyam D. Srinath. 217-220 [doi]
• A correlation based technique for shift, scale, and rotation independent object identificationLicia Capodiferro, Roberto Cusani, Giovanni Jacovitti, M. Vascotto. 221-224 [doi]
• Object classification and registration by Radon transform based invariantsHans J. Dohse, Jorge L. C. Sanz, Anil K. Jain 0002. 225-228 [doi]
• Automatic recognition of isolated or occluded planar objects by a two steps processingJianming Song, Paul Gaillard. 229-232 [doi]
• Fast image segmentation for some machine vision applicationsEric B. Hinkle, Jorge L. C. Sanz. 233-236 [doi]
• Lemniscate transform: A new efficient technique for shape coding and representationAmlan Kundu 0001. 237-240 [doi]
• Pattern spectrum of images and morphological shape-size complexityPetros Maragos. 241-244 [doi]
• Some applications of mathematical morphology to range imageryT. R. Esselman, Jacques G. Verly. 245-248 [doi]
• Differential operator based edge and line detectionJ. E. Bevington, Russell M. Mersereau. 249-252 [doi]
• 2G filtered images by their zero crossingsJ. A. Reimer, P. D. Lawrence. 253-256 [doi]
• Aspects of directional filtering and applications in image processingA. Ikonomopoulos. 257-260 [doi]
• A method of acquiring 3-D data of an object from stereo imagesYu-ning Dong, Zhen-Ya He. 261-264 [doi]
• A concavity based algorithm for the recognition of partially occulded 3-dimensional objectsB. K. Miller, R. A. Jones. 265-268 [doi]
• Knowledge based object detection using SAR imagesDavid L. Wang. 269-272 [doi]
• A hierarchical representation of random waveforms by scale-space filteringMakoto Sato, Toshikazu Wada, Hiroshi Kawarada. 273-276 [doi]
• Structural processing of waveforms as treesScott W. Shaw, Rui J. P. deFigueiredo. 277-280 [doi]
• A new algorithm for estimating optic flow for low-level vision systemsN. A. Chalabi, T. S. Durrani. 281-284 [doi]
• A new heuristic edge extraction techniqueJoseph S. P. Shu. 285-288 [doi]
• Robust linear prediction for speech analysisChin-Hui Lee. 289-292 [doi]
• Factors in voice quality: Acoustic features related to genderD. G. Childers, Ke Wu, D. M. Hicks. 293-296 [doi]
• Estimation of noise variance from the noisy AR signal and its application in speech enhancementKuldip K. Paliwal. 297-300 [doi]
• Reconstruction from Fourier transform phase with applications to speech analysisB. Yegnanarayana, S. Tanveer Fathima, Hema A. Murthy. 301-304 [doi]
• A new approach to noise-robust LPCA. A. Wrench, C. F. N. Cowan. 305-307 [doi]
• Speech parameter estimation using a vocal tract/Cord modelJuergen Schroeter, Jerry N. Larar, M. Mohan Sondhi. 308-311 [doi]
• High resolution frequency analysis of voices-Feature extraction of nasal consonantsTetsuya Harada, Hiroshi Kawarada. 312-315 [doi]
• Speech spectral segmentation for spectral estimation and formant modellingHarprit S. Chhatwal, Anthony G. Constantinides. 316-319 [doi]
• Discrete all-pole modeling for voiced speechAmro El-Jaroudi, John Makhoul. 320-323 [doi]
• Fitting noncausal autoregressive signal plus noise models to noisy non-Gaussian linear processesJitendra K. Tugnait. 324-327 [doi]
• Jump detection and fast parameter tracking for piecewise AR processes using adaptive lattice filtersShiping Li, Bradley W. Dickinson. 328-331 [doi]
• Adaptive line enhancement using a random AR modelAhmed S. Abutaleb. 332-335 [doi]
• An analysis of finite precision effects for the autocorrelation method and Burg's method of linear predictionS. Thomas Alexander, Zong M. Rhee. 336-339 [doi]
• Extended matrix formulation for the Marple algorithmWilliam J. Vetter, Milton J. Porsani. 340-343 [doi]
• Objective methods for comparing autoregressive order-determining criteriaSverre Holm. 344-347 [doi]
• Some properties of prediction and interpolation errorsBernard Picinbono, Jean-Marc Kerilis. 348-351 [doi]
• On the extreme accuracy of maximum entropy spectrum estimation from an error-free autocorrelation functionPaul F. Fougere. 352-355 [doi]
• Autocorrelation distortion function for improved AR modelingV. K. Jain, B. L. Xu. 356-359 [doi]
• Context-dependent phonetic Markov models for large vocabulary speech recognitionAnne-Marie Derouault. 360-363 [doi]
• Speech recognition with very large size dictionaryBernard Mérialdo. 364-367 [doi]
• A continuous speech dialog system for the oral control of a sonar consolePierre Alinat, Evelyne Gallais, Jean-Paul Haton, Jean-Marie Pierrel, Pascal Richard. 368-371 [doi]
• Speaker-dependent speech recognition as the basis for a speech training aidDiane Kewley-Port, Charles S. Watson, Daniel Maki, Daniel Reed. 372-375 [doi]
• The lexical access component of the CMU continuous speech recognition systemAlexander I. Rudnicky, Lynn K. Baumeister, Kevin H. DeGraaf, Eric Lehmann. 376-379 [doi]
• Sentence parsing with weak grammatical constraintsRichard Stern, Wayne H. Ward, Alexander G. Hauptmann, Juan Leon. 380-383 [doi]
• Explicit time correlation in hidden Markov models for speech recognitionC. J. Wellekens. 384-386 [doi]
• REMORA A software architecture for the collaboration of different knowledge sources in phonetic decoding of continuous speechThérèse Martelli, Laurent Miclet, Jean-Pierre Tubach. 387-390 [doi]
• Syllable-based segment-hypotheses generation in fluently spoken speech using gross articulatory featuresOtto Schmidbauer. 391-394 [doi]
• On the stability of adaptive lattice filtersHanoch Lev-Ari, K.-F. Chiang, Thomas Kailath. 395-398 [doi]
• Stability of adaptively controlled systems-A graphical approachP. Darlington, S. J. Elliott. 399-402 [doi]
• Stabilization of fast recursive least-squares transversal filters for adaptive filteringJean-Luc botto. 403-406 [doi]
• A fast QR/Frequency-domain RLS adaptive filterJohn M. Cioffi. 407-410 [doi]
• On frequency-domain least squares adaptive algorithmsJae C. Lee, Sanjit K. Mitra. 411-414 [doi]
• Continuous-time least-squares fast transversal filtersHanoch Lev-Ari, John M. Cioffi, Thomas Kailath. 415-418 [doi]
• A fast transversal filter for adaptive line enhancementDirk T. M. Slock, John M. Cioffi, Thomas Kailath. 419-422 [doi]
• Numerical properties of a hyperbolic rotation method for windowed RLS filteringS. T. Alexander, C. T. Pan, Robert J. Plemmons. 423-426 [doi]
• K frequency-adaptive transversal filters in plant identificationJosé Manuel Páez-Borrallo, Aníbal R. Figueiras-Vidal, Luis Vergara-Dominguez. 427-430 [doi]
• Adaptive transversal filters with delayed coefficient adaptationGuozhu Long, Fuyun Ling, John G. Proakis. 431-434 [doi]
• Design of continuous gain adaptive αβ trackers for passive sonar applicationCarl J. Wenk, Harold F. Jarvis Jr.. 435-438 [doi]
• True angle estimation from a line array using time-delay estimates over a known rotationG. W. Johnson, E. J. Modugno. 439-442 [doi]
• Applications of error intensity measures to bearing estimationAlfred O. Hero III. 443-446 [doi]
• High resolution multipath time delay estimation for broadband random signalsJohn P. Ianniello. 447-450 [doi]
• High resolution time delay estimationIvars P. Kirsteins. 451-454 [doi]
• Time delay estimation by autoregressive modelizationM. A. Pallas, N. Martin, J. Martin. 455-458 [doi]
• Passive ranging in a multipath dominated environmentMiriam Hamilton, Peter M. Schultheiss. 459-462 [doi]
• Passive multipath target tracking in inhomogeneous acoustic mediumAmnon Shefi, Charles W. Therrien, Donald E. Kirk, Rigoberto Saez, Benjamin Friedlander. 463-466 [doi]
• A practical approach to the estimation of amplitude and time delay parameters of a composite signal in non-white Gaussian noiseRoger J. Tremblay, G. Clifford Carter, Dean W. Lytle. 467-470 [doi]
• The spherical interpolation method for closed-form passive source localization using range difference measurementsJonathan S. Abel, Julius O. Smith III. 471-474 [doi]
• A quadratic residue processor for complex DSP applicationsMagdy A. Bayoumi. 475-478 [doi]
• Custom VLSI design of a single chip multi-channel ADPCM processorJames D. Beatty, Richard D. Calder Jr., Perry Farazi, Daniel P. Kelly, James L. Melsa. 479-482 [doi]
• A bit-serial floating-point complex multiplier-accumulator for fault-tolerant digital signal processing arraysPaul M. Chau, Kay-Cheung Chew, Walter H. Ku. 483-486 [doi]
• A second generation Silicon compiler for bit-serial signal processing architectureY. S. Cheung, S. C. Leung. 487-490 [doi]
• A parameterized VLSI video-rate histogram processorBrian C. Richards, Alex Sherstinsky, Robert W. Brodersen. 491-494 [doi]
• Full-span structural compilation of DSP hardwareSteven G. Smith, Peter B. Denyer, David S. Renshaw, K. Asada, K. P. Coplan, M. Keightley, Javeed I. Mhar. 495-498 [doi]
• A flexible linear array oriented VLSI processor for continuous speech recognitionJun-ichi Takahashi, Takashi Kimura, Shigetatsu Hamaguchi, Naotaka Omiya. 499-502 [doi]
• A reconfigurable binary/RNS/LNS architecture for DSPFred J. Taylor. 503-506 [doi]
• A VLSI implementation of the partial rank algorithm for adaptive signal processingFathy F. Yassa, Steven G. Kratzer. 507-510 [doi]
• A split control store VLSI for 32 kbps ADPCM transcodingLuis Bonet, Tim A. Williams. 511-514 [doi]
• A cascadable adaptive FIR filter VLSI ICDavid E. Borth, Ira A. Gerson, John R. Haug. 515-518 [doi]
• The graph search machine (GSM): A programmable processor for connected word speech recognition and other applicationsStephen C. Glinski, T. Mariano Lalumia, Dan Cassiday, Taiho Koh, Christine Gerveshi, Gene A. Wilson, Jit Kumar. 519-522 [doi]
• The architecture and applications of the motorola DSP56000 digital signal processor familyKevin Kloker. 523-526 [doi]
• A 15 nanosecond complex multiplier-accumulator for FFTSRobert E. Owen. 527-530 [doi]
• A novel VLSI digital signal processor architecture for high-speed vector and transform operationsDavid M. Taylor, Rafi Retter. 531-534 [doi]
• A 40 MFLOPS digital signal processor: The first supercomputer on a chipRay Simar Jr., Tony Leigh, Peter Koeppen, Jerald Leach, Jim Potts, Denise Blalock. 535-538 [doi]
• Serial/Parallel architectures for area-efficient vector multiplicationSteven G. Smith, Peter B. Denyer. 539-542 [doi]
• Techniques to increase the computational throughput of bit-serial architecturesSteven G. Smith, M. S. McGregor, Peter B. Denyer. 543-546 [doi]
• An optimized VLSI architecture for a multiformat discrete cosine transformNicolas Demassieux, Gilles Concordel, Jean-Pierre Durandeau, Francis Jutand. 547-550 [doi]
• Segmentation of noisy images modelled by Markov random fields with Gibbs distributionM. El-Gabali, Malayappan Shridhar, Majid Ahmadi. 551-554 [doi]
• Linear feature extraction based on an AR model edge detectorYi-Tong Zhou, Rama Chellappa. 555-558 [doi]
• The modeling and segmentation of speckled imagesKeith D. Hartt, Patrick A. Kelly, Haluk Derin. 559-562 [doi]
• Segmentation of noisy textured images using simulated annealingChee Sun Won, Haluk Derin. 563-566 [doi]
• Semi-Markov random field models for image segmentationJohn Goutsias, Jerry M. Mendel. 567-570 [doi]
• Texture segmentation using a class of narrowband filtersMarianna Clark, Alan C. Bovik, Wilson S. Geisler. 571-574 [doi]
• Texture recognition using final-prediction-error criterionHidefumi Kobatake, Toshihiro Watanabe. 575-578 [doi]
• Unified approach for low level image analysisDong-Seok Jeong, Paul M. Lapsa. 579-582 [doi]
• CARMA Model method of two-dimensional shape classification: An eigensystem approach vs. the LP normMohammad V. Malakooti, Keith A. Teague. 583-586 [doi]
• Efficient CFAR detection of line segments in a 2-D imagePeter L. Chu. 587-590 [doi]
• A frequency domain approach to multiframe detection and estimation of dim targetsBoaz Porat, Benjamin Friedlander. 591-594 [doi]
• A new algorithm for maximum entropy image reconstructionGong Wei, Zhen-Ya He. 595-597 [doi]
• Thermal imaging techniques for the non destructive inspection of composite materials in real timeTariq S. Durrani, A. Rauf, K. Boyle, Franco Lotti, Stefano Baronti. 598-601 [doi]
• Mask extraction from optical images of VLSI circuitsHong Jeong, Bruce R. Musicus. 602-605 [doi]
• Spatial and temporal analysis of weather radar reflectivity imagesMark W. Merritt. 606-609 [doi]
• Automatic diagnosis of pneumoconiosis by texture analysis of chest X-ray imagesHidefumi Kobatake, Kunio Oh'ishi, Juichi Miyamichi. 610-613 [doi]
• Measurement of two-dimensional movement of traffic by image processingHidefumi Kobatake, Yoshiaki Inoue, Tatsuro Namai, Nobuhiro Hamba. 614-617 [doi]
• The automatic classification of the welding defectsChen Su-xian, Li Xiao-song. 618-621 [doi]
• Real-time analysis of fuel spray imagesAmira M. Badreldin. 622-624 [doi]
• A frequency-weighted Itakura spectral distortion measure and its application to speech recognition in noiseFrank K. Soong, M. Mohan Sondhi. 625-628 [doi]
• Errors in determining vocal tract shape from the acoustic signalRoman Kuc, Hee Han. 629-632 [doi]
• Rapid speaker adaptation using a probabilistic spectral mappingRichard M. Schwartz, Yen-Lu Chow, Francis Kubala. 633-636 [doi]
• Estimation of voice source and vocal tract parameters based on ARMA analysis and a model for the Glottal source waveformHiroya Fujisaki, Mats Ljungqvist. 637-640 [doi]
• Fuzzy vector quantazation applied to hidden Markov modelingHo-Ping Tseng, Michael J. Sabin, Edward A. Lee. 641-644 [doi]
• Signal modeling by exponential segments and application in voiced speech analysisSarangarajan Parthasarathy, Donald W. Tufts. 645-648 [doi]
• Mixed-phase deconvolution of speech based on a sine-wave modelThomas F. Quatieri, Robert J. McAulay. 649-652 [doi]
• Set-membership theory applied to linear prediction analysis of speechJohn R. Deller Jr., T. C. Luk. 653-656 [doi]
• Beyond quasi-stationarity: Designing time-frequency representations for speech signalsMichael D. Riley. 657-660 [doi]
• Power spectrum estimation with uncertainty in the sample location of correlation measurementsTaikang Ning, Chrysostomos L. Nikias. 661-664 [doi]
• An accuracy analysis of the Kumaresan-Tufts method for estimating complex damped exponentialsBenjamin Friedlander, Boaz Porat. 665-668 [doi]
• Accurate estimation of closely spaced, real, decaying exponentials in noiseGuy R. L. Sohie, Arun Mirchandani. 669-672 [doi]
• Signal enhancement using canonical projection operatorsJames A. Cadzow. 673-676 [doi]
• A SVD-based transient error method for analyzing noisy multicomponent exponential signalsM. J. E. Salami, S. T. Nichols, Michael R. Smith 0001. 677-680 [doi]
• A high resolution data-adaptive time-frequency representationDouglas L. Jones, Thomas W. Parks. 681-684 [doi]
• Time-frequency signal synthesis on signal subspacesFranz Hlawatsch, Werner Krattenthaler. 685-688 [doi]
• Digital spectra of non-uniformly sampled signals with applications to digitally synthesized sinusoidsY. C. Jenq. 689-692 [doi]
• A training procedure for a segment-based-network approach to isolated word recognitionFrank K. Soong. 693-696 [doi]
• Integration of acoustic information in a large vocabulary word recognizerVishwa Gupta, Matthew Lennig, Paul Mermelstein. 697-700 [doi]
• Experiments with the Tangora 20, 000 word speech recognizerA. Averbuch, Lalit R. Bahl, Raimo Bakis, P. Brown, G. Daggett, S. Das, Ken Davies, Steven V. De Gennaro, P. de Souza, E. Epstein, D. Fraleigh, Frederick Jelinek, B. Lewis, Robert L. Mercer, J. Moorhead, Arthur Nádas, David Nahamoo, M. Picheny, G. Shichman, P. Spinelli, D. Van Compernolle, H. Wilkens. 701-704 [doi]
• Multi-style training for robust isolated-word speech recognitionRichard P. Lippmann, Edward A. Martin, Douglas B. Paul. 705-708 [doi]
• Two-stage discriminant analysis for improved isolated-word recognitionEdward A. Martin, Richard P. Lippmann, Douglas B. Paul. 709-712 [doi]
• A speaker-stress resistant HMM isolated word recognizerDouglas B. Paul. 713-716 [doi]
• Cepstral domain stress compensation for robust speech recognitonYeunung Chen. 717-720 [doi]
• A new time-scale warping algorithm and associated modules for single dimensional and multidimensional speech parameter contoursA. Maheswaran, Robert E. Bogner. 721-724 [doi]
• Fast algorithms for vector quantization picture codingWilliam Equitz. 725-728 [doi]
• Optimum rate allocation in pyramid vector quantizer transform coding of imageryMary E. Blain, Thomas R. Fischer. 729-732 [doi]
• Image compression using vector quantization of linear (one- step) prediction errorsV. John Mathews, Randall W. Waite, Thao Duy Tran. 733-736 [doi]
• Laplacian pyramid image data compressionC.-H. Chen. 737-739 [doi]
• Efficient vector quantization for color image encodingJ. Barrilleaux, R. Hinkle, S. Wells. 740-743 [doi]
• An efficient pyramid image coding systemAnh Tran, Kwun-Min Liu, Kou-Hu Tzou, Eileen B. Vogel. 744-747 [doi]
• Low bit-rate image coding techniquesK. S. Thyagarajan, Mahesh Viswanathan. 748-751 [doi]
• Image coding based on segmentation using region growingK. S. Thyagarajan, Helge Bohlmann, Hüseyin Abut. 752-755 [doi]
• Vector quantizer architectures for speech and image codingHüseyin Abut, Bertram P. M. Tao, Jack L. Smith. 756-759 [doi]
• A finite-state vector quantizer for low-rate image sequence codingRichard L. Baker, Hsiao-hui Shen. 760-763 [doi]
• A design method of systolic arrays under the constraint of the number of the processorsSatoshi Horiike, Shogo Nishida, Toshiaki Sakaguchi. 764-767 [doi]
• A systolic algorithm for cyclic-by-rows SVDUwe Schwiegelshohn, Lothar Thiele. 768-770 [doi]
• Systolic ROM arrays for implementing RNS FIR filtersMajid Taheri, Graham A. Jullien, William C. Miller. 771-774 [doi]
• A speech feature extraction system with a linear processor arrayTeiji Emori, Masatoshi Tachibana. 775-778 [doi]
• Parallel VLSI computing array implementation for signal subspace updating algorithmAli H. Abdallah, Yu Hen Hu. 779-782 [doi]
• Implementation of Kalman filters using systolic arraysGeorge M. Papadourakis, Fred J. Taylor. 783-786 [doi]
• Bit serial systolic chip set for real-time image codingP. A. Ramamoorthy, Brahmaji Potu. 787-790 [doi]
• One- and two-dimensional systolic arrays for least-squares problemsUwe Schwiegelshohn, Lothar Thiele. 791-794 [doi]
• On a programmable signal processor for VLSINagaraja Srinivasa, Kasi Rajgopal, K. R. Ramakrishnan. 795-796 [doi]
• A hidden Markov model applied to Chinese four-tone recognitionXi-Xian Chen, Changnian Cai, Peng Guo, Ying Sun. 797-800 [doi]
• A real-time evaluation system for a real-time connected-speech recognizerSusan M. Miller, David P. Morgan, Harvey F. Silverman, Michael N. Karam, N. Rex Dixon. 801-804 [doi]
• Japanese linguistic processing for continuous speech recognitionToshiaki Tsuboi, Akihiro Tomihisa, Noboru Sugamura. 805-808 [doi]
• Experimental results on a large lexicon access taskPietro Laface, Giorgio Micca, Roberto Pieraccini. 809-812 [doi]
• Speech recognition using an auditory model with pitch-synchronous analysisMelvyn J. Hunt, Claude Lefebvre. 813-816 [doi]
• Continuous digit recognition using coarse phonetic segmentationDavid Lubensky. 817-820 [doi]
• An investigation on the use of acoustic sub-word units for automatic speech recognitionJay G. Wilpon, Biing-Hwang Juang, Lawrence R. Rabiner. 821-824 [doi]
• Extraction of phonemic variation rules in continuous speech spoken by multiple speakersShinta Kimura, Yasuhiro Nara. 825-828 [doi]
• Spoken sentence recognition by time-synchronous parsing algorithm of context-free grammarSei-Ichi Nakagawa. 829-832 [doi]
• A data-driven organization of the dynamic programming beam search for continuous speech recognitionHermann Ney, Dieter Mergel, Andreas Noll, Annedore Paeseler. 833-836 [doi]
• Lexical access with lattice inputHy Murveit, Mitchel Weintraub, Michael Cohen, Jared Bernstein, Alex Rudnicky. 837-840 [doi]
• A speech recognition system for the Italian languagePaolo D'Orta, Marco Ferretti, Alex Martelli, Sergio Melecrinis, Stefano Scarci, Giampiero Volpi. 841-843 [doi]
• Construction of language models for spoken database queriesDieter Mergel, Annedore Paeseler. 844-847 [doi]
• Large vocabulary word detection by searching in a tree-structural word dictionaryKaichiro Hatazaki, Takao Watanabe. 848-851 [doi]
• Efficient implementation of continuous speech recognition on a large scale parallel processorOwen Kimball, Lynn Cosell, Richard M. Schwartz, Michael A. Krasner. 852-855 [doi]
• Prosodic knowledge sources for word hypothesization in a continuous speech recognition systemAlex Waibel. 856-859 [doi]
• Acoustic-phonetic segment classification and scale-space filteringMeg Withgott, Steven C. Bagley, Richard F. Lyon, Marcia A. Bush. 860-863 [doi]
• Speaker-independent recognition of stop consonantsSarah K. Yoder, Leah H. Jamieson. 864-867 [doi]
• Interaction between stochastic modeling and knowledge-based techniques in acoustic-phonetic decoding of speechJean-Paul Haton, Noëlle Carbonell, Dominique Fohr, Jean-François Mari, Abdelaziz Kriouille. 868-871 [doi]
• A novel approach for complex Chebyshev-approximation with FIR filters using the Remez exchange algorithmKlaus P. Preuss. 872-875 [doi]
• On the design of recursive Hilbert-transformersHans Wilhelm Schüßler, J. Weith. 876-879 [doi]
• On the stability of linear shift variant digital filtersTamal Bose, David P. Brown. 880-883 [doi]
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• The design and application of optimal FIR fractional-slope phase filtersMark F. Pyfer, Rashid Ansari. 896-899 [doi]
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• Hankel approximation methods of IIR filter designK. S. Arun, M. Reuter. 920-923 [doi]
• Finite limiting effects for a band-limited Gaussian random process with applications to A/D conversionDennis R. Morgan. 924-927 [doi]
• Phase noise in quantized sine wavesFrederick A. Williams. 928-931 [doi]
• Direct form sensitivity reduction by order increaseA. A. (Louis) Beex, Victor E. DeBrunner. 932-935 [doi]
• A new scaling procedure for cascade digital filtersK. P. Prasad, P. Sathyanarayana. 936-939 [doi]
• Elimination of the pitch bias in the non-stationary characterization of speechSusanna Ragazzini, Lucio Prina Ricotti, Gianni Orlandi, Giuseppe Martinelli. 940-943 [doi]
• Spectral estimation of quasi-periodic dataShankar S. Narayan, John Parker Burg. 944-947 [doi]
• Speech analysis and reconstruction using short-time, elementary waveformsJean-Sylvain Liénard. 948-951 [doi]
• A lattice filter model with accurate lip impedance for dynamic articuratory movementNobuhiro Miki, Kunitoshi Motoki, Nobuo Nagai. 952-955 [doi]
• Median type filters with linear predictive substructuresPekka Heinonen, Yrjö Neuvo. 956-959 [doi]
• A single board multirate APC speech coding terminalK. Field, A. Derr, Lynn Cosell, C. Henry, Michael A. Krasner, J. Tiao. 960-963 [doi]
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• Implementation of a multi-pulse speech codec with pitch prediction on a single chip floating-point signal processorA. Fukui, K. Shibagaki. 968-971 [doi]
• Algorithm selection and software time/Space optimization for a DSP micro-based speech processor for a multi-electrode cochlear implantL. Robert Morris, Peter Barszczewski, Jonathan Bosloy. 972-975 [doi]
• The ASPEN parallel computer, speech recognition and parallel dynamic programmingAllen L. Gorin, Richard R. Shively. 976-979 [doi]
• Nonminimum phase system identification via cepstrum modeling of higher-order cumulantsChrysostomos L. Nikias, Renlong Pan. 980-983 [doi]
• Performance analysis of new least squares ARMA lattice modeling algorithmsErlendur Karlsson, Monson H. Hayes. 984-987 [doi]
• Estimating decay rates of single-frequency causal AR filters using a "Decimation" methodNeil J. Grossbard, John M. Retterer. 988-990 [doi]
• Exact recursive least squares algorithms for ARMA modelingSurendra Prasad, Shiv Dutt Joshi. 991-994 [doi]
• On matching correlation sequences by parametric spectral modelsJohn Makhoul, Allan O. Steinhardt. 995-998 [doi]
• A principal component approach for adaptive ARMA model identificationKambiz Heidarian, Yu Hen Hu. 999-1002 [doi]
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• Performance characteristics and speed-up rates of the NEC mPD7281 data flow processor in parallel processingGagan Mirchandani, Gerald A. McGuire. 1015-1018 [doi]
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• High-speed Wigner processing based on a single modulus quadratic residue numbering systemJoEllen Wilbur, Fred J. Taylor. 1031-1034 [doi]
• A signal processing cell architectureMohsin M. Jamali, M. M. Hussain, Graham A. Jullien. 1035-1038 [doi]
• Combined dynamic data analysis and process variable prediction approach for system fault detectionB. R. Upadhyaya, O. Glöckler, F. P. Wolvaardt. 1039-1042 [doi]
• The decomposition of long FFT's for high throughput implementationRivka Shenhav. 1043-1046 [doi]
• A multiple microprocessor system for general DSP operationGregory Y. Tang, Brian K. Lien. 1047-1050 [doi]
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• Color image-sequence compression using adaptive binary-tree vector quantization with codebook replenishmentChia-Lung Yeh. 1059-1062 [doi]
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• Experiments on video teleconferencing algorithms at 56 kilobits/secMichael Maragoudakis, Jerry D. Gibson. 1071-1074 [doi]
• Luminance adaptive chrominance codingBodo Braun. 1075-1078 [doi]
• 2- and 3-D Nonlinear predictorsGiovanni Ramponi, Giovanni L. Sicuranza, Silvio Cucchi. 1079-1082 [doi]
• Multi-pulse and regular-pulse LP coding of imagesCaspar Horne, Ed F. Deprettere. 1083-1086 [doi]
• Optimal detection in colored non-Gaussian noise with unknown parametersSteven Kay, Debasis Sengupta. 1087-1090 [doi]
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• Thresholds in combined detection and source motion estimationG. W. Johnson, W. A. Bradford. 1095-1098 [doi]
• Three-dimensional motion estimation by synthetic aperture underwater acoustic systemsHua Lee, Thomas S. Huang. 1099-1102 [doi]
• The effects of cross-correlated noise and multi-channel signal on ORing lossGregory E. Bottomley. 1103-1106 [doi]
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• 2-test in signal detectionQ. T. Zhang, P. Yip. 1111-1114 [doi]
• A fast prediction-error detector for estimating sparse-spike sequencesGeorgios B. Giannakis, Jerry M. Mendel, Xiaofeng Zhao. 1115-1118 [doi]
• A transform based covariance differencing approach to bearing estimationSurendra Prasad, Ronald T. Williams, A. K. Mahalanabis, Leon H. Sibul. 1119-1122 [doi]
• Statistical features versus word templates for speaker independent digit recognition over long distance telephone connectionsEnrico Bocchieri, George R. Doddington. 1123-1126 [doi]
• A VQ-based preprocessor using cepstral dynamic features for large vocabulary word recognitionSadaoki Furui. 1127-1130 [doi]
• Experiments in isolated digit recognition with a cochlear modelEric P. Loeb, Richard F. Lyon. 1131-1134 [doi]
• Speaker-independent isolated word recognition using word-based vector quantization and hidden Markov modelsY. S. Cheung, S. T. Leung. 1135-1138 [doi]
• Speech recognition with a noise-adapting codebookDavid B. Roe. 1139-1142 [doi]
• Increased noise immunity in large vocabulary speech recognition with the aid of spectral subtractionDirk Van Compernolle. 1143-1146 [doi]
• Hidden Markov model speech recognition based on Kalman filteringMark A. Clements, Sungjae Lim. 1147-1150 [doi]
• Connected word recognizer on a multiprocessor systemBasavaraj I. Pawate, M. L. McMahan, R. H. Wiggins, George R. Doddington, Periagaram K. Rajasekaran. 1151-1154 [doi]
• Weighted cepstral distance measures in vector quantization based speech recognizersTed H. Applebaum, Brian A. Hanson, Hisashi Wakita. 1155-1158 [doi]
• An efficient speaker-independent automatic speech recognition by simulation of some properties of human auditory perceptionHynek Hermansky. 1159-1162 [doi]
• HMM-Based speech recognition using multi-dimensional multi-labelingMasafumi Nishimura, Koichi Toshioka. 1163-1166 [doi]
• A large-vocabulary Chinese speech recognition systemXue-Dong Huang, Lian-Hong Cai, Ditang Fang, Bian-Jin Ci, Li Zhou, Li Jian. 1167-1170 [doi]
• Adaptive noise cancellation for speech with a TMS32020Hen-Geul Yeh. 1171-1174 [doi]
• A telephone speech recognition system using word spotting technique based on statistical measureTatsuya Kimura, Katsuyuki Niyada, Shoji Hiraoka, Shuji Morii, Taisuke Watanabe. 1175-1178 [doi]
• Multiple input adaptive iterative image restoration algorithmsAggelos K. Katsaggelos. 1179-1182 [doi]
• On increasing the convergence rate of regularized iterative image restoration algorithmsReginald L. Lagendijk, Russell M. Mersereau, Jan Biemond. 1183-1186 [doi]
• An image reconstruction from limited view angle projection dataTsuneo Saito, Hiroyuki Kudo. 1187-1190 [doi]
• A new iterative method with histogram equalization constraint for reconstructing image from phaseWang Yenping, Li Han, Zhu Wenchun. 1191-1194 [doi]
• Image reconstruction from one-bit Fourier phase: Theory, sampling, and coherent image modelThomas T. Huang, Jorge L. C. Sanz, W. E. Blanz. 1195-1198 [doi]
• Comparison of phase retrieval algorithmsThomas S. Huang, K. A. Rinaldi, Hua Lee. 1199-1200 [doi]
• Effects of constraints, initialization, and finite-word length in blind deblurring of images by convex projectionsCheng-Tie Chen, M. Ibrahim Sezan, A. Murat Tekalp. 1201-1204 [doi]
• Error bounds for iterative reprojection methods in computerized tomographyH. Joel Trussell, Hatice Örün Öztürk, M. Reha Civanlar. 1205-1208 [doi]
• Considerations for the restoration of stochastic degradationsH. Joel Trussell, Patrick L. Combettes. 1209-1212 [doi]
• Consistency of the minimum mean square error estimateH. Joel Trussell, M. Reha Civanlar. 1213-1216 [doi]
• ECT Image enhancementStephen A. Laico, Barry J. Sullivan. 1217-1220 [doi]
• Reconstruction of bandlimited signals from their unevenly-spaced sampled dataM. Soumekh. 1221-1224 [doi]
• A new approach for reconstruction of the left ventricle from biplane angiocardiogramsZhi-Dong Bai, Paruchuri R. Krishnaiah, C. R. Rao 0001, P. S. Reddy, Yung-Nien Sun, Lin-Cheng Zhao. 1225-1228 [doi]
• Effects of modeling domains on recursive color image restorationDenise L. Angwin, Howard Kaufman. 1229-1231 [doi]
• A new approach to 2-D Kalman filteringJiang Min, Chen Su Xiang. 1232-1235 [doi]
• Stochastic relaxation for MAP restoration of gray level images with multiplicative noiseHao Jinchi, Tal Simchony, Rama Chellappa. 1236-1239 [doi]
• Extrapolation of multi-dimensional bandlimited sequences using energy concentration informationLee C. Potter, K. S. Arun. 1240-1243 [doi]
• Digital restoration of multi-channel imagesNikolas P. Galatsanos, Roland T. Chin. 1244-1247 [doi]
• Order statistic last output reference filtersAdly T. Fam, Yong-Hoon Lee, Sung Jea Ko. 1248-1251 [doi]
• A minimum-risk quantizer for noisy sourcesMark K. Cook, Richard A. Jones. 1252-1255 [doi]
• Distance measure for speech recognition based on the smoothed group delay spectrumFumitada Itakura, Taizo Umezaki. 1257-1260 [doi]
• Improvement of word recognition results by trigram modelKiyohiro Shikano. 1261-1264 [doi]
• Speech recognition in scale spaceRichard F. Lyon. 1265-1268 [doi]
• Duration modelling in finite state automata for speech recognition and fast speaker adaptationM. Codogno, Luciano Fissore. 1269-1272 [doi]
• Some experiments on HMM speaker adaptationA. Jarre, R. Pieraccini. 1273-1276 [doi]
• Training of phoneme models in a sentence recognition systemAndreas Noll, Hermann Ney. 1277-1280 [doi]
• Unsupervised bootstrapping of diphone-like templates for connected speech recognitionAnna Maria Colla, Aaron E. Rosenberg. 1281-1284 [doi]
• Automatic speech recognition via pseudo-independent marginal mixturesArthur Nádas, David Nahamoo. 1285-1287 [doi]
• ADPCM Coding of speech with backward-adaptive algorithms for noise feedback and postfilteringN. S. Jayant. 1288-1291 [doi]
• Effect of signal bandwidth on the accuracy of adaptive interpolation of discrete-time signalsHenry M. Dante. 1292-1295 [doi]
• Estimating the number of sinusoids in additive white-noiseJean-Jacques Fuchs. 1296-1299 [doi]
• Selective deconvolution: A new approach to extrapolation and spectral analysis of discrete signalsUwe Franke. 1300-1303 [doi]
• A state-space approach to positive sequencesRichard J. Vaccaro, Fu Li. 1304-1307 [doi]
• On estimation of the number of signals and frequencies of multiple sinusoidsZhi-Dong Bai, Paruchuri R. Krishnaiah, Lin-Cheng Zhao. 1308-1311 [doi]
• On predictive least squares filteringTie-Jun Shan. 1312-1315 [doi]
• A Kalman filter algorithm for estimating sinusoids in colored noiseCarlos E. Davila, Ashley J. Welch, H. Grady Rylander III. 1316-1319 [doi]
• Convergence rate of adaptive line enhancer with tap failuresLarry Pearlstein, Bede Liu. 1320-1323 [doi]
• A linear predictive front-end processor for speech recognition in noisy environmentsYariv Ephraim, Jay G. Wilpon, Lawrence R. Rabiner. 1324-1327 [doi]
• Waveform coding of voiceband data signals at 16 kb/sD. O. Anderton, Craig K. Rushforth. 1328-1331 [doi]
• Modulo-PCM with multi-quantizerKiyoshi Mizui, Masafumi Hagiwara, Masao Nakagawa. 1332-1335 [doi]
• A bound on predictor misadjustment in ADPCMKhalid Sayood, David C. Farden. 1336-1339 [doi]
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• A fast algorithm for uniform vector quantizationT.-C. Chen. 1344-1347 [doi]
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• Speech coding using efficient pseudo-stochastic block codesDaniel Lin. 1354-1357 [doi]
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• Sub-band coding of images using predictive vector quantizationPeter H. Westerink, Jan Biemond, Dick E. Boekee. 1378-1381 [doi]
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• Transform coding of speech with weighted vector quantizationTakehiro Moriya, Masaaki Honda. 1629-1632 [doi]
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• Systolic array realization of digital filtersGeorge A. Lampropoulos. 1661-1664 [doi]
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• A fast algorithm for linear estimation of three-dimensional homogeneous anisotropic random fieldsAndrew E. Yagle. 1712-1715 [doi]
• Robust estimation of the acoustic attenuation parameterSteven W. Patton. 1716-1719 [doi]
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