Table of Contents

• Isolated word recognition using hidden Markov modelsKazuhide Sugawara, Masafumi Nishimura, Koichi Toshioka, Masaaki Okochi, Toyohisa Kaneko. 1-4 [doi]
• Explicit modelling of state occupancy in hidden Markov models for automatic speech recognitionMartin J. Russell, Roger K. Moore. 5-8 [doi]
• Recent developments in the application of hidden Markov models to speaker-independent isolated word recognitionBiing-Hwang Juang, Lawrence R. Rabiner, Stephen E. Levinson, M. Mohan Sondhi. 9-12 [doi]
• Training of HMM recognizers by simulated annealingDouglas B. Paul. 13-16 [doi]
• Experiments on speaker-independent recognition of hand-segmented French vowelsMatthew Lennig. 17-20 [doi]
• Discrete utterance speech recognition without time alignmentKathy L. Brown, V. Ralph Algazi. 21-24 [doi]
• Comparative study of several distortion measures for speech recognitionN. Nocerino, Frank K. Soong, Lawrence R. Rabiner, Dennis H. Klatt. 25-28 [doi]
• Applying matrix quantization to isolated word recognitionDavid K. Burton. 29-32 [doi]
• Spoken word recognition based on top-down phoneme segmentationKiyoaki Aikawa, Masahide Sugiyama, Kiyohiro Shikano. 33-36 [doi]
• On associative recognition of isolated Chinese wordZhao Guo-tian. 37-40 [doi]
• New design methods for FIR filters with equiripple stopbands and prescribed degrees of passband flatnessP. P. Vaidyanathan. 41-44 [doi]
• Efficient FIR filter design using differential coding of filter coefficientsWonyong Sung, Sanjit K. Mitra. 45-48 [doi]
• Smoothed median filters with FIR substructuresPekka Heinonen, Yrjö Neuvo. 49-52 [doi]
• Residue feedback in ladder and lattice filter structuresDarrell Williamson, S. Sridharan. 53-56 [doi]
• A new S-Z replacement design technique of IIR digital filtersZ. S. Li, P. C. Hutchinson. 57-60 [doi]
• An approach for designing systems with prescribed behaviour at distinct frequencies regarding additional constraintsHans Wilhelm Schüßler, Peter Steffen. 61-64 [doi]
• Design of linear-phase partly digital anti-aliasing filtersTapio Saramäki, Kari-Pekka Estola. 65-68 [doi]
• Wave digital filters: Voltage, current, or power waves?Gernot Kubin. 69-72 [doi]
• A novel approach to the design of filters for filter banksGuenter Wackersreuther. 73-76 [doi]
• Cross spectrum ML estimateMiguel Angel Lagunas, M. E. Santamaria, Antoni Gasull, Asunción Moreno. 77-80 [doi]
• Adaptive spectral estimation by the conjugate gradient methodHuanqun Chen, Tapan K. Sarkar, Soheil A. Dianat, John D. Brule. 81-84 [doi]
• Window functions obtained from B-SFrancesco Palmieri, L. P. Bolgiano Jr.. 85-88 [doi]
• Spectrum estimation of time series with missing dataHenry M. Dante. 89-92 [doi]
• Spectral estimation statistics for noise corrupted autoregressive series-First-order caseDonald F. Gingras. 93-96 [doi]
• Spectral estimation via minimum energy correlation extensionAllan O. Steinhardt, Robert K. Goodrich, Richard A. Roberts. 97-100 [doi]
• Superresolution autoregressive spectral estimation technique using multiple step predictionShu Hung Leung, Wel-Yau Horng. 101-104 [doi]
• Adaptive methods for real time Pisarenko spectrum estimateYu Hen Hu. 105-108 [doi]
• Adaptive recursive scheme for spectral analysis of sinusoids in signals with unknown colored spectrumAkira Sano, Koh-ichi Hashimoto. 109-112 [doi]
• Hierarchical predictive approach to image codingAlberto Sanz, Carlos Muñoz, Narciso García. 113-116 [doi]
• Ordering predictive coding of digital imageShoji Mizuno, Kazumoto Iinuma. 117-120 [doi]
• A hybrid image coding technique using a noncausal stochastic modelYasuo Yoshida, Akira Nakamura, Hisanao Ogura. 121-124 [doi]
• Use of vector quantizers in image codingNasser M. Nasrabadi. 125-128 [doi]
• Compression of color digital images using vector quantization in product codesScott E. Budge, Richard L. Baker. 129-132 [doi]
• Adaptive vector quantization by progressive codevector replacementAllen Gersho, Mitsuharu Yano. 133-136 [doi]
• A fast matrix quantizer for image encodingH. Bheda, K. S. Thyagarajan, Hüseyin Abut. 137-140 [doi]
• A matrix quantizer incorporating the human visual modelK. S. Thyagarajan, S. Parthasarathy, Hüseyin Abut. 141-144 [doi]
• Adaptive transform tree coding of imagesWilliam A. Pearlman, May M. Leung, Priyadarshan Jakatdar. 145-148 [doi]
• A hybrid image compression techniqueCharles F. Hall. 149-152 [doi]
• Space-time processing of multisensor time delay data in bearing estimationJoseph C. Hassab. 153-156 [doi]
• Robust adaptive Kalman filtering for systems with unknown step inputs and non-Gaussian measurement errorsR. Lynn Kirlin, Alireza Moghaddamjoo. 157-160 [doi]
• Time delay estimation in presence of jitterMehdi Hatamian, David J. Anderson. 161-164 [doi]
• Application of robust filtering techniques to time delay estimation in the Arctic environmentThomas E. McCannon. 165-167 [doi]
• Time delay estimation for spread spectrum signals using generalized cross correlationWilliam E. Ryan, J. M. Schumpert. 168-171 [doi]
• Time-delay-of-arrival resolution of multipath communication signalsKeith A. Struckman. 172-175 [doi]
• 1-D and 2-D syntactic pattern recognition for the detection of bright spotsKou-Yuan Huang, King-sun Fu. 176-179 [doi]
• Predictive deconvolution of seismic array data for inversionAlastair D. McAulay. 180-183 [doi]
• Linear time-invariant space-variant filters and the WKB approximationLawrence J. Ziomek, Jan Vos. 184-187 [doi]
• Image enhancement using coherence processing with applications to seismogramsE. A. Quincy, D. J. Tomich. 188-191 [doi]
• A VLSI implementation of an FFT/NTT computational unitMagdy A. Bayoumi, Graham A. Jullien, William C. Miller. 192-195 [doi]
• A 32 bit 15M flop floating point programmable signal processor architecture for VLSI implementationJames H. Hesson. 196-199 [doi]
• Design and VLSI implementation of a concurrent solver for N coupled least-squares fitting problemsEd F. Deprettere, Kishan Jainandunsing. 200-203 [doi]
• An 18-bit floating-point signal processor VLSI with an on-chip 512W dual-port RAMHironori Yamauchi, Takao Kaneko, Tsutomu Kobayashi, Atsushi Iwata, Sadayasu Ono. 204-207 [doi]
• Very fast sequencing and data addressing made easy with new CMOS VLSI componentsPaul M. Toldalagi. 208-211 [doi]
• New CMOS chip family facilitates design of high speed DSP hardwareRobert Fine, Ted Dintersmith. 212-215 [doi]
• Application of an advanced signal processing engine to adaptive techniquesGeorge G. Ricker, Manuel F. Richey. 216-219 [doi]
• A custom-designed integrated circuit for the realization of residue number digital filtersW. K. Jenkins, E. S. Davidson, D. F. Paul. 220-223 [doi]
• High resolution digital filter chipAmnon Aliphas, William F. Ganong III, Peter Stonestrom, Dan Perkins. 224-227 [doi]
• Architecture and applications of a second-generation digital signal processorCole Erskine, Surendar Magar, Edward R. Caudel, Daniel Essig. 228-231 [doi]
• A systolic array computerEmmanuel A. Arnould, H. T. Kung 0001, Onat Menzilcioglu, Ken Sarocky. 232-235 [doi]
• The waveform segment vocoder: A new approach for very-low-rate speech codingSalim E. Roucos, Alexander MacLeod Wilgus. 236-239 [doi]
• A low bit rate segment vocoder based on line spectrum pairsJoel Crosmer, Thomas P. Barnwell III. 240-243 [doi]
• Application of line-spectrum pairs to low-bit-rate speech encodersGeorge S. Kang, Lawrence J. Fransen. 244-247 [doi]
• A multiple rate low rate voice codecJoseph Rothweiler, Jack Carmody. 248-251 [doi]
• Vector quantization and perceptual criteria for low-rate coding of speechMaurizio Copperi, Daniele Sereno. 252-255 [doi]
• Generalization of the multipulse coding for low bit rate coding purposes: The generalized decimationJean-Pierre Adoul, F. Didelot, Philippe Mabilleau, Sarto Morissette. 256-259 [doi]
• Pole-zero multipulse speech representation using harmonic modelling in the frequency domainIsabel Trancoso, Luís B. Almeida, José M. Tribolet. 260-263 [doi]
• Optimal initial conditions and pulse values for multipulse speech codingH. Joel Trussell, M. Reha Civanlar. 264-267 [doi]
• Bit rate reduction by Markov-Huffman coding of speech parametersPanos E. Papamichalis. 268-271 [doi]
• A multi-processor cellular automaton chipKenneth Steiglitz, Ronald R. Morita. 272-275 [doi]
• Towards an FIR filter tissuePeter R. Cappello. 276-279 [doi]
• A reconfigurable systolic primitive processor for signal processingThanos Stouraitis, S. Natarajan, Fred J. Taylor. 280-283 [doi]
• VLSI Architecture for solving covariance eigen systemYu Hen Hu. 284-287 [doi]
• Hierarchical flowgraph integration for VLSI array processorsSun-Yuan Kung, Jurgen Annevelink, Patrick M. Dewilde, S. C. Lo. 288-291 [doi]
• Design strategies for implementing systolic and wavefront arrays using OCCAMRoy Chapman, Tariq S. Durrani, T. Willey. 292-295 [doi]
• A practical comparison of the systolic and wavefront array processing architecturesDavid S. Broomhead, J. G. Harp, John G. McWhirter, K. J. Palmer, J. B. G. Roberts. 296-299 [doi]
• Systematic approaches to the design of algorithmically specified systolic arraysJosé A. B. Fortes, King-sun Fu, Benjamin W. Wah. 300-303 [doi]
• 2-Norm minimization in parametric spectral analysis: A general discussionAníbal R. Figueiras-Vidal, José R. Casar Corredera, Ramón García Gómez, José Manuel Páez-Borrallo. 304-307 [doi]
• An adaptive orthogonal maximum likelihood algorithm for parameter estimationXian-Ci Xiao. 308-311 [doi]
• Solving a class of nonlinear least squares problemsStephen W. Lang. 312-315 [doi]
• Non-recursive frequency estimation for closely spaced sinusoidsBoris Golubev, Steven R. Rogers. 316-319 [doi]
• Simple, effective computation of principal eigenvectors and their eigenvalues and application to high-resolution estimation of frequenciesDonald W. Tufts, Costas D. Melissinos. 320-323 [doi]
• Approximate factorization of unfactorable spectral modelsLeland B. Jackson. 324-326 [doi]
• Efficient generation of ARMA cross covariance sequencesA. A. (Louis) Beex. 327-330 [doi]
• Signal detection using autoregressive parametersJohn W. Ketchum, David Herrick. 331-334 [doi]
• Adaptive comb filtering for harmonic signal enhancementArye Nehorai, Boaz Porat. 335-338 [doi]
• Adaptive vector quantization for image sequence codingH. F. Sun, M. Goldberg. 339-342 [doi]
• An improved transform coder for image sequences using attributes of difference picturesM. Reha Civanlar, P. Santago. 343-346 [doi]
• Motion-compensated interframe codingRobert J. Moorhead II, Sarah A. Rajala. 347-350 [doi]
• Transform/Time domain coding using the method of projection onto convex setsA. Farid Faryar, Sarah A. Rajala. 351-354 [doi]
• A minimum risk quantizer for motion compensated image codingRichard A. Jones, Carl D. Bowling. 355-358 [doi]
• Block-recursive matching algorithm (BRMA) for displacement estimation of video imagesKou-Hu Tzou, To R. Hsing, Nancy A. Daly. 359-362 [doi]
• Motion-compensated adaptive intra-interframe predictive coding algorithmToshio Koga, A. Hirano, Yukihiko Iijima, Kazumoto Iinuma. 363-366 [doi]
• Motion-compensated hybrid coding at 50 kb/sStaffan Ericsson. 367-370 [doi]
• A new motion-compensated transform coding scheme R. H. J. M. Plompen, B. F. Schuurink, Jan Biemond. 371-374 [doi]
• Implicit motion compensated noise reduction of motion video scenesDennis Martinez, Jae S. Lim. 375-378 [doi]
• Investigation of text-independent speaker indentification over telephone channelsHerbert Gish, Kenneth F. Karnofsky, Michael A. Krasner, Salim E. Roucos, Richard M. Schwartz, Jared J. Wolf. 379-382 [doi]
• Automatic speaker recognition using vocoded speechStephanie S. Everett. 383-386 [doi]
• A vector quantization approach to speaker recognitionFrank K. Soong, Aaron E. Rosenberg, Lawrence R. Rabiner, Biing-Hwang Juang. 387-390 [doi]
• Text-dependent speaker recognition using vector quantizationJoseph T. Buck, David K. Burton, John E. Shore. 391-394 [doi]
• Periodicity estimation by hypothesis-directed searchJohn Amuedo. 395-398 [doi]
• A speaker verification system for access-controlW. Feix, M. DeGeorge. 399-402 [doi]
• A robust realtime pitch extraction from the ACF of LPC residual error signalsSoon-young Kwon, Aaron J. Goldberg, D. Ng, K. Ouellette. 403-406 [doi]
• Improved pitch detection algorithm for noisy speechSanguoon Chung, V. Ralph Algazi. 407-410 [doi]
• New objective measures for the evaluation of pitch extractorsVishu R. Viswanathan, William Russell. 411-414 [doi]
• Design and evaluation of double-transform pitch determination algorithms with nonlinear distortion in the frequency domain-preliminary resultsHelge Indefrey, Wolfgang Hess, Günter Seeser. 415-418 [doi]
• Objective estimation of perceptually specific subjective qualitiesSchuyler R. Quackenbush, Thomas P. Barnwell III. 419-422 [doi]
• Quantification of dysphony with allowance for inter-utterance variationS. Feijóo, C. Hernández. 423-426 [doi]
• A computational model of the cochlea used with cochlear prosthesis patientsDirk Van Compernolle. 427-429 [doi]
• The effects of controlled speech level input on the intelligibility testing of speech compression algorithmsJames T. Sims, John D. Tardelli. 430-433 [doi]
• Iterative deconvolution in noncoherent systemsMark A. Richards. 434-437 [doi]
• Deconvolution in the sequency domainB. W. Dahanayake, K. M. Wong. 438-441 [doi]
• Deconvolution without system model or a new blind deconvolutionG. Thomas, C. Dussaussois, F. Buret, Ph. Auriol. 442-444 [doi]
• Deconvolution by the conjugate gradient methodTapan K. Sarkar, Fung I. Tseng, Soheil A. Dianat, Bruce Z. Hollmann. 445-448 [doi]
• Time and frequency selective deconvolution using optimal controlPeter M. Clarkson, Joe K. Hammond. 449-452 [doi]
• 1 deconvolution and its application to seismic signal processingJ. Bee Bednar, Rao K. Yarlagadda, Terry L. Watt. 453-456 [doi]
• Signal deconvolution using fuzzy setsM. Reha Civanlar, H. Joel Trussell. 457-460 [doi]
• On absolute value minimization approaches to tauberian modellingJesús M. Alcázar-Fernández, José R. Casar Corredera, Aníbal R. Figueiras-Vidal. 461-464 [doi]
• On descrete band-limited signal extrapolationHua Lee, Thomas S. Huang. 465-468 [doi]
• A comparison of three methods for discrete-time signal extrapolationBarry J. Sullivan. 469-472 [doi]
• Robust LPC analysis of speech by extended correlation matchingV. K. Jain, Bishnu S. Atal. 473-476 [doi]
• Perceptual weightings and optimal pulse positioning in multipulse LPC speech codingH. A. Hawkins, D. Mitchell Wilkes, Mark A. Clements, Monson H. Hayes III. 477-480 [doi]
• All-pole speech modeling with a maximally pulse-like residualRichard C. Rose, Mark A. Clements. 481-484 [doi]
• 1 normRichard J. Mammone, Kent Wang, Steven Gay. 485-488 [doi]
• Speech transformations based on a sinusoidal representationThomas F. Quatieri, Robert J. McAulay. 489-492 [doi]
• High quality time-scale modification for speechSalim E. Roucos, Alexander MacLeod Wilgus. 493-496 [doi]
• Speech signal analysis and synthesis via Fourier-Bessel representationC. S. Chen, Kaliappan Gopalan, Pallabi Mitra. 497-500 [doi]
• Speech analysis and restitution using time-depedent autoregressive modelsMarie-Christine Omnes-Chevalier, C. Chollet, Yves Grenier. 501-504 [doi]
• A measure of in-synchrony regions in the auditory nerve firing patterns as a basis for speech vocodingOded Ghitza. 505-508 [doi]
• Perceptually based linear predictive analysis of speechHynek Hermansky, Brian A. Hanson, Hisashi Wakita. 509-512 [doi]
• A new model-based speech analysis/Synthesis systemDaniel W. Griffin, Jae S. Lim. 513-516 [doi]
• Statistical design of analysis/Synthesis systems with quantizationAmir Dembo, David Malah. 517-520 [doi]
• A unifying framework for analysis/Synthesis systems based on maximally decimated filter banksMark J. T. Smith, Thomas P. Barnwell III. 521-524 [doi]
• Quadrature mirror filters with perfect reconstruction and reduced computational complexityClaude R. Galand, Henri J. Nussbaumer. 525-528 [doi]
• Complementary IIR digital filter banksSanjit K. Mitra, Yrjö Neuvo, P. P. Vaidyanathan. 529-532 [doi]
• A filter/Detector interpretation of the short-time Fourier transform magnitudeJames C. Anderson. 533-536 [doi]
• Design of filters for discrete short-time Fourier transform synthesisZ. Shpiro, David Malah. 537-540 [doi]
• Flexible design of computationaly efficient nearly perfect QMF filter banksJ. Masson, Z. Picel. 541-544 [doi]
• Real-time telephone channel simulationKamal Jabbour, Jose Fernando Vega-Riveros. 545-547 [doi]
• Properties of the positive time-frequency distribution functionsLeon Cohen. 548-551 [doi]
• Properties of eigenanalysis methods for bearing estimation algorithmsDon H. Johnson. 552-555 [doi]
• Extending the threshold of the eigenstructure methodsMati Wax, Thomas Kailath. 556-559 [doi]
• High resolution spectral method for spatial discrimination of closely spaced correlated sourcesHenri Clergeot, Abdelaziz Ouamri, Sara Tressens. 560-563 [doi]
• On beamforming in presence of multipathArogyaswami Paulraj, Thomas Kailath. 564-567 [doi]
• Normal mode propagation and high resolution methodsEven Borten Lunde. 568-571 [doi]
• On the probability density of signal-to-noise ratio in an improved adaptive detectorIvars P. Kirsteins, Donald W. Tufts. 572-575 [doi]
• High resolution bearing estimation without eigen decompositionRamdas Kumaresan, Arnab K. Shaw. 576-579 [doi]
• Observation of impulse response by two relatively prime pseudorandom sequencesHikaru Date, Kimitoshi Fukudome, Masakazu Oda, Setsuya Tokuriki. 580-583 [doi]
• Catholic electroacoustical difficultiesJ. Robert Ashley, J. Matthew Ashley. 584-587 [doi]
• Omnidirectional coded planar loudspeaker arraysSaid E. El-Khamy, Onsy A. Abdel-Alim. 588-591 [doi]
• Design and optimization of a digital FM receiver using DPLL techniquesWerner Rosenkranz. 592-595 [doi]
• Implementation of a real-time, digital vocoder for tactile hearing prosthesesA. Maynard Engebretson, Michael P. O'Connell. 596-599 [doi]
• Signal processing architecture for loudspeaker array directivity controlDavid G. Meyer. 600-603 [doi]
• Reconstruction of tonal sequencesI. L. Veach, J. J. Narraway. 604-607 [doi]
• A new auditory model for the evaluation of sound quality of audio systemsMatti Karjalainen. 608-611 [doi]
• The maximum tolerable delay of speech and musicJohn Charles Cox. 612-615 [doi]
• Bounds for ARMA spectral analysis based on sample covariancesBenjamin Friedlander, Boaz Porat. 616-619 [doi]
• The role of spectral decomposition in the pattern recognition of narrowband signalsT. Hediger, A. Passamante. 620-623 [doi]
• Perturbation analysis of a SVD based method for the harmonic retrieval problemD. V. Bhaskar Rao. 624-627 [doi]
• Statistical properties of coherence and power contribution ratios via multivariate autoregressive modelingHideaki Sakai, Hidekatsu Tokumaru. 628-631 [doi]
• Event detection using recursively updated lattice filtersAnders Johansson, L. Gunnar Ahlbom, Lars-Henning Zetterberg. 632-635 [doi]
• Sensitivity and performance analysis of coherent signal-subspace processing for multiple wideband sourcesHong Wang, Mos Kaveh. 636-639 [doi]
• Direction of arrival estimation by eigenstructure methods with unknown sensor gain and phaseArogyaswami Paulraj, Thomas Kailath. 640-643 [doi]
• Detection algorithms based on prediction error - additive noise - data orthogonalityM. K. Ibrahim, Costas E. Goutis. 644-647 [doi]
• On nonlinear estimation in presence of non-Gaussian NoiseRavi Kumar. 648-651 [doi]
• A nonrecursive filter for edge preserving image restorationRama Chellappa, Hao Jinchi. 652-655 [doi]
• Identification of image and blur parameters for the restoration of noncausal blursA. Murat Tekalp, Howard Kaufman, John W. Woods. 656-659 [doi]
• Image restoration using a parallel indentification and filtering procedureJan Biemond, F. G. van der Putten. 660-663 [doi]
• Optimal techniques for constraint based signal restoration and image reconstructionRichard M. Leahy, Costas E. Goutis. 664-667 [doi]
• The Wilcoxon filter: A robust filtering schemeRegis J. Crinon. 668-671 [doi]
• Edge preserving signal enhancement using generalizations of order statistic filteringSteven R. Peterson, Saleem A. Kassam. 672-675 [doi]
• Constrained optimization for image restoration using nonlinear programmingChia-Lung Yeh, Roland T. Chin. 676-679 [doi]
• Post-reconstruction correction in quantitative single photon ECTYu-Shan Fong, A. Saha, Carlos A. Pomalaza-Raez. 680-683 [doi]
• Adaptive noise cancelling applied to image restorationK. Mike Tao, Fred M. Weinhaus. 684-687 [doi]
• Hierarchical approach to image estimationJohn W. Woods, Fure-Ching Jeng. 688-691 [doi]
• Boundary value problem in image restorationJohn W. Woods, Jan Biemond, A. Murat Tekalp. 692-695 [doi]
• Nonstationary iterative image restorationAggelos K. Katsaggelos, Jan Biemond, Russell M. Mersereau, Ronald W. Schafer. 696-699 [doi]
• A general formulation of constrained iterative restoration algorithmsAggelos K. Katsaggelos, Jan Biemond, Russell M. Mersereau, Ronald W. Schafer. 700-703 [doi]
• Adaptive least-squares technique for multi-band image enhancementVictor T. Tom, Mark J. Carlotto. 704-707 [doi]
• A digital filtering system for decoding in-line hologramsLevent Onural, Peter D. Scott. 708-711 [doi]
• Noise-immune speech transduction using multiple sensorsVishu R. Viswanathan, Claudia M. Henry, Alan G. Derr. 712-715 [doi]
• Adaptive noise reduction in aircraft communication systemsP. Darlington, P. D. Wheeler, G. A. Powell. 716-719 [doi]
• Processing of noisy speech using group delay functionsJoy A. Thomas, B. Yegnanarayana, Raghuram Karinthi, V. Venkateswar. 720-723 [doi]
• Cross-channel correlation for the enhancement of noisy speechLes E. Atlas, Leo M. Hengky. 724-727 [doi]
• Adaptive reduction of interfering speaker noise using the least mean squares algorithmS. T. Alexander. 728-731 [doi]
• Bandwidth design for speech-seeking microphone arraysJ. L. Flanagan. 732-735 [doi]
• Synthesis by rule: LPC diphones and calculation of formant trajectoriesXavier Rodet, Philippe Depalle. 736-739 [doi]
• Diphone synthesis using multipulse coding and a phase vocoderM. G. Stella, F. J. Charpentier. 740-743 [doi]
• GRAPHON-the Vienna speech systhesis system for arbitrary German textGeorg Dorffner, Markus Kommenda, Gernot Kubin. 744-747 [doi]
• Voice conversion: Factors responsible for qualityD. G. Childers, B. Yegnanarayana, Ke Wu. 748-751 [doi]
• Concatenation rules for demisyllable speech synthesisHelmut Dettweiler, Wolfgang Hess. 752-755 [doi]
• Optimum binary windows for discrete Fourier transformK. M. M. Prabhu. 756-759 [doi]
• Exact fast digital convolution by using P-adic numbers and polynomial transformationsSoo-Chang Pei, Ja-Ling Wu. 760-763 [doi]
• Complex digital signal processing using quadratic residue number systemsR. Krishnan, Graham A. Jullien, William C. Miller. 764-767 [doi]
• Efficient architectures for implementing the prime-factor Fourier transform modulesR. Kumaresan. 768-771 [doi]
• Prime factor decomposition of the discrete cosine transform and its hardware realizationPaul P. N. Yang, M. J. Narasimha. 772-775 [doi]
• DIF Algorithms for DCT and DSTPatrick C. Yip, Kamisetty Ramamohan Rao. 776-779 [doi]
• nFFT algorithmsMichael T. Heideman, C. Sidney Burrus. 780-783 [doi]
• Implementation of "Split-radix" FFT algorithms for complex, real, and real symmetric dataPierre Duhamel, Henk Hollmann. 784-787 [doi]
• The chord property speeds finite field FFTsG. Robert Redinbo, Kotesh K. Rao. 788-791 [doi]
• Sufficient condition for extendibility and two-dimensional power spectrum estimationChrysostomos L. Nikias, Anastasios N. Venetsanopoulos. 792-795 [doi]
• On 2-D prony methodsJosé R. Casar Corredera, Jesús M. Alcázar-Fernández, Luis Alfonso Hernández Gómez. 796-799 [doi]
• The optimum approach to multichannel AR spectrum estimationTaikang Ning, Chrysostomos L. Nikias. 800-803 [doi]
• Simultaneous confidence bands for a class of 2-D spectral estimatesGovind Sharma 0002, Rama Chellappa. 804-807 [doi]
• Frequency-wavenumber spectral estimation of non-planar random fieldsGregory H. Wakefield, M. Kaveh. 808-811 [doi]
• Phase errors in the cross spectrum estimate due to misalignmentSverre Holm. 812-815 [doi]
• Multidimensional maximum-entropy covariance extensionH. Lev-Ari. 816-819 [doi]
• Resolution property of the improved maximum likelihood methodFarid U. Dowla, Jae S. Lim. 820-822 [doi]
• Under-ice reverberation rejectionWilliam S. Hodgkiss, Dimitrios Alexandrou. 823-825 [doi]
• Application of a super resolution algorithm to acoustic tomography multipath dataJules S. Jaffe, Lisa Strong. 826-829 [doi]
• Bayesian estimation of medium properties in wavefield inversion techniquesIoannis Pitas, Anastasios N. Venetsanopoulos. 830-833 [doi]
• Inversion of modified beamformed array dataDavid J. Scheibner, Thomas W. Parks. 834-837 [doi]
• Identification of the modulation type of a signalY. T. Chan, L. G. Gadbois, P. Yansouni. 838-841 [doi]
• Speaker independent telephone speech recognitionHirosi Iizuka. 842-845 [doi]
• Evaluation of a network-based isolated digit recognizer using the TI multi-dialect databaseMarcia A. Bush, Gary E. Kopec. 846-849 [doi]
• Isolated word recognition using a segmental approachLouis C. Sauter. 850-853 [doi]
• Phonetically guided clustering for isolated word recognitionDieter Mergel, Hermann Ney. 854-857 [doi]
• A real-time, isolated-word, speech recognition system for dictation transcriptionF. Jelinek. 858-861 [doi]
• A coarse phonetic knowledge source for template independent large vocabulary word recognitionHelmut Lagger, Alex Waibel. 862-865 [doi]
• Large vocabulary speaker-independent Japanese speech recognition systemShuji Morii, Katsuyuki Niyada, Satoru Fujii, Masakatsu Hoshimi. 866-869 [doi]
• Recognition error measurements from parameterized distance distributionsA. E. Rosenberg. 870-873 [doi]
• An efficient vector-quantization preprocessor for speaker independent isolated word recognitionKuk-Chin Pan, Frank K. Soong, Lawrence R. Rabiner, A. F. Bergh. 874-877 [doi]
• A new algorithm for isolated word recognition with large within-class variabilityWissam W. Ahmed, George A. Bekey. 878-881 [doi]
• Speech recognition in the F-16 cockpit using principal spectral componentsPeriagaram K. Rajasekaran, George R. Doddington. 882-885 [doi]
• Hierarchical DP for word recognitionMartin S. Glassman. 886-889 [doi]
• Knowledge base enhancement of visual trackingHarley R. Myler, Wiley E. Thompson, Gerald M. Flachs. 890-892 [doi]
• Boundary detection in speckle imagesAlan C. Bovik, David C. Munson Jr.. 893-896 [doi]
• Adaptive hierarchical algorithm for accurate image segmentationFernand Cohen. 897-900 [doi]
• Segmentation of multilevel images using Gibbs distributionR. Cristi, M. Shridhar, M. V. Prasadarao. 901-904 [doi]
• Fast object segmentation in textured backgroudsJorge L. C. Sanz. 905-908 [doi]
• Knowledge based object detectionD. B. Sharman, Tariq S. Durrani. 909-912 [doi]
• A new approach to parameter estimation for Gibbs random fieldsHaluk Derin, Howard Elliott, J. Kuang. 913-916 [doi]
• 3-D Motion parameters from contours using a canonic differentialDavid Cyganski, John A. Orr. 917-920 [doi]
• Determination of affine transforms from object contours with no point correspondence informationJohn A. Orr, David Cyganski, Richard F. Vaz. 921-924 [doi]
• Segmentation of three dimensional range dataFernand S. Cohen, Raymond D. Rimey, Jean-François Cayula. 925-928 [doi]
• Two view motion analysis under a small perturbationXinhua Zhuang, Robert M. Haralick. 929-932 [doi]
• A silhouette-slice theorem for opaque 3-D objectsPatrick L. Van Hove, Jacques G. Verly. 933-936 [doi]
• Code-excited linear prediction(CELP): High-quality speech at very low bit ratesManfred R. Schroeder, Bishnu S. Atal. 937-940 [doi]
• A stochastic model excitation source for linear prediction speech analysis-synthesisGuen Oyama. 941-944 [doi]
• Mid-rate coding based on a sinusoidal representation of speechRobert J. McAulay, Thomas F. Quatieri. 945-948 [doi]
• A new approach to multipulse LPC coder designAmir Dembo, David Malah. 949-952 [doi]
• A linear programming approach to multipulse speech codingR. García Gómez, J. M. Alcázar-Fernández. 953-956 [doi]
• Efficient algorithms for obtaining multipulse excitation for LPC codersJean-Paul Lefèvre, Olivier Passien. 957-960 [doi]
• A speech coding method using thinned-out residualAkira Ichikawa, Shoichi Takeda, Yoshiaki Asakawa. 961-964 [doi]
• Regular excitation reduction for effective and efficient LP-coding of speechEd F. Deprettere, Peter Kroon. 965-968 [doi]
• An efficient, pitch-aligned high-frequency regeneration technique for RELP vocodersRichard L. Zinser. 969-972 [doi]
• The transmission of in-band signaling for medium band voice codec implementationsP. J. Wilson, J. M. Puetz, M. D. Dankberg. 973-976 [doi]
• A systolic array for linearly constrained least squares filteringSeth Z. Kalson, Kung Yao. 977-980 [doi]
• Processing speech with the multi-serial signal processorRichard F. Lyon, Niels Lauritzen. 981-984 [doi]
• A VLSI dynamic time warp processor for connected and isolated word speech recognitionRobert E. Owen. 985-988 [doi]
• A multi-pulse LPC synthesizer for telecommunications useZ. A. Putnins, G. A. Wilson, J. Kumar, R. D. Trupp. 989-992 [doi]
• Image processing address generator chipJohn A. Eldon, Zoltan Stroll, Earl E. Swartzlander Jr.. 993-996 [doi]
• VLSI Implementation of two-dimensional generalized mean filterAmlan Kundu, Gururaj Singh, Steven Butner. 997-1000 [doi]
• VLSI Architecture for a one chip video median filterNicolas Demassieux, Francis Jutand, Marc Saint-Paul, Michel Dana. 1001-1004 [doi]
• A two-dimensional digital filter chip set for modular two-dimensional filter implementationS. Lee, Anastasios N. Venetsanopoulos. 1005-1008 [doi]
• Systolic architectures based on barrel shifters for real-time signal and image processingP. A. Ramamoorthy, Tau Chen. 1009-1012 [doi]
• Adaptive restoration of unknown samples in certain time-discrete signalsRaymond N. J. Veldhuis, A. J. E. M. Janssen, Lodewijk B. Vries. 1013-1016 [doi]
• On the optimality of the Wigner distribution for detectionSteven Kay, Gloria Faye Boudreaux-Bartels. 1017-1020 [doi]
• An expansion of Wigner distribution and its applicationsN. M. Marinovic, G. Eichmann. 1021-1024 [doi]
• Wigner-Ville and evolutionary spectra for covariance equivalent nonstationary random processesJoe K. Hammond, R. F. Harrison. 1025-1028 [doi]
• Transformation, inversion and conversion of bilinear signal representationsFranz Hlawatsch. 1029-1032 [doi]
• Generalized ambiguity functionsLeon Cohen, Theodore E. Posch. 1033-1036 [doi]
• Signal synthesis from Wigner distributionKai-Bor Yu, Siuling Cheng. 1037-1040 [doi]
• A generalized delay operator for shift-variant systemsDean J. Schmidlin. 1041-1044 [doi]
• Time and spatial varying CAM and AI signal analysis using the Wigner distributionDavid Chester, JoEllen Wilbur. 1045-1048 [doi]
• Autoregressive models with time-dependent log area ratiosMarie-Christine Omnes-Chevalier, Yves Grenier. 1049-1052 [doi]
• A method of image reconstruction in fan beam tomographyMehrdad Soumekh. 1053-1056 [doi]
• Reconstruction of two-dimensional signals from threshold crossingsSusan R. Curtis, Alan V. Oppenheim, Jae S. Lim. 1057-1060 [doi]
• Incorporation of a priori moment constraints into signal recovery and synthesis problems via the method of convex projectionsM. Ibrahim Sezan, Henry Stark. 1061-1064 [doi]
• On the stability and sensitivity of multidimensional signal reconstruction from Fourier transform magnitudeJorge L. C. Sanz, Thomas S. Huang. 1065-1068 [doi]
• Nearest neighbor and generalized inverse distance interpolation for Fourier domain image reconstructionW. K. Jenkins, B. C. Mather, David C. Munson Jr.. 1069-1072 [doi]
• An inner product framework for image reconstructionBarry P. Medoff. 1073-1076 [doi]
• Image reconstruction of the heart using a priori information and spatiotemporal estimationR. S. Acharya, P. B. Heffernan, Richard A. Robb. 1077-1080 [doi]
• An image reconstruction from incomplete observation by constrained spectrum extrapolationTsuneo Saito, Takayoshi Chiba. 1081-1084 [doi]
• Aerodynamic aspects of voicing: Glottal pulse skewing revisitedBert Cranen, Louis Boves. 1085-1088 [doi]
• Variability in closed phase analysis of speechJ. N. Larar, Y. A. Alsaka, D. G. Childers. 1089-1092 [doi]
• Measuring source-tract interaction from speechA. S. Ananth, D. G. Childers, B. Yegnanarayana. 1093-1096 [doi]
• Analysis of vocal tract lip reflectance by linear predictionF. J. Charpentier. 1097-1100 [doi]
• Determining vocal tract shape by applying dynamic constraintsRoman Kuc, Franz Tuteur, J. Rimas Vaisnys. 1101-1104 [doi]
• A new articulatory model for speech productionTrevor Thomas, Frank Fallside. 1105-1108 [doi]
• Speech processing by a model of the auditory peripheryKaren L. Payton. 1109-1112 [doi]
• Formant tracking using hidden Markov modelsGary E. Kopec. 1113-1116 [doi]
• A robust formant-based speech spectrum comparison measureMelvyn J. Hunt. 1117-1120 [doi]
• Instantaneous-frequency distribution vs. time: An interpretation of the phase structure of speechDavid H. Friedman. 1121-1124 [doi]
• Cubic spline modeling of speech spectraC. David Covington. 1125-1128 [doi]
• A compensated-Kalman speech parameter estimatorG. A. Mack, Vijay K. Jain. 1129-1132 [doi]
• Spectrum and pitch estimation of speech using a time-varying ARMA estimation algorithmNobuhiro Miki, Yoshikazu Miyanaga, Sato Saga, Nobuo Nagai. 1133-1136 [doi]
• Bayesian deconvolution of speech containing pulsed excitationG. A. Mack, Vijay K. Jain. 1137-1140 [doi]
• Acoustic transmission-line analysis of formants in hyperbaric Helium speechPer Lunde. 1141-1144 [doi]
• Steady-state behavior of RLS adaptive algorithmsEvangelos Eleftheriou, David D. Falconer. 1145-1148 [doi]
• Fast recursive least-squares algorithms: Preventing divergencePhilippe Fabre, Claude Gueguen. 1149-1152 [doi]
• An efficient step size adaptation technique for LMS adaptive filtersMaurice G. Bellanger, Cumhur Cengiz Evci. 1153-1156 [doi]
• Convergence rates for the constant modulus algorithm with sinusoidal inputsJohn R. Treichler, Michael G. Larimore. 1157-1160 [doi]
• Global convergence of the constant modulus algorithmJulius O. Smith III, Benjamin Friedlander. 1161-1164 [doi]
• Noise capture properties of the constant modulus algorithmMichael G. Larimore, John R. Treichler. 1165-1168 [doi]
• Identification of sparse impulse response systems using an adaptive delay filterD. M. Etter. 1169-1172 [doi]
• "Noise cancellation studies using a least-squares lattice filter"T. Gardiner, John G. McWhirter, T. J. Shepherd. 1173-1176 [doi]
• A generalized filter structure for IIR adaptive filtersFathy F. Yassa. 1177-1180 [doi]
• An efficient algorithm for lattice filter/PredictorDae Hee Youn, V. John Mathews, Sung Ho Cho. 1181-1184 [doi]
• A minimal parameter adaptive notch filter with constrained poles and zerosArye Nehorai. 1185-1188 [doi]
• The unbiased least-squares latticeDavid C. Swanson, Frank W. Symons Jr.. 1189-1192 [doi]
• Detection and identification of sinusoids in broadband noise via a parallel recursive ALEDon R. Hush, Nasir Ahmed 0001. 1193-1196 [doi]
• Network-based connected digit recognition using vector quantizationMarcia A. Bush, Gary E. Kopec. 1197-1200 [doi]
• Application of a sequential pattern learning system to connected speech recognitionA. Richard Smith, J. N. Denenberg, T. B. Slack, C. C. Tan, Robert E. Wohlford. 1201-1204 [doi]
• Context-dependent modeling for acoustic-phonetic recognition of continuous speechRichard M. Schwartz, Yen-Lu Chow, Owen Kimball, Salim E. Roucos, Michael A. Krasner, John Makhoul. 1205-1208 [doi]
• A script-guided algorithm for the automatic segmentation of continuous speechHermann Ney. 1209-1212 [doi]
• Speaker dependent connected speech recognition via phonetic Markov modelsHervé Bourlard, Yves G. Kamp, Christian Wellekens. 1213-1216 [doi]
• Integration of segmenting and nonsegmenting approaches in continuous speech recognitionJean-Paul Brassard. 1217-1220 [doi]
• Continuous speech recognition on the bases of vector field model for segmentation and feature extraction, and continuous dynamic programming for pattern matchingRyu-ichi Oka. 1221-1224 [doi]
• Multi-speaker computer recognition of ten connectedly spoken lettersRenato de Mori, Giorgio Rossi, Jianli Sun. 1225-1228 [doi]
• A connected speech recognition system using a diphone-based language modelAnna Maria Colla, Carlo Scagliola, Donatella Sciarra. 1229-1232 [doi]
• Keyword recognition using template concatenationAlan L. Higgins, Robert E. Wohlford. 1233-1236 [doi]
• Parallel algorithms for Toeplitz matrix operationsCamille C. Price, Moktar A. Salama. 1237-1240 [doi]
• The block-processing FTF adaptive algorithmJohn M. Cioffi. 1241-1244 [doi]
• Data echo nonlinear cancellationJosé R. Casar Corredera, Mariano García Otero, Aníbal R. Figueiras-Vidal. 1245-1248 [doi]
• Equalization for transmission line channels: A discussion of three IIR adaptive filtering algorithmsEmmanuel Fernández-Gaucherand, J. R. Cruz. 1249-1252 [doi]
• Some experimental and theoretical results using a new adaptive filter structure for noise cancellation in the presence of crosstalkRichard L. Zinser Jr., Gagan Mirchandani, Joseph B. Evans. 1253-1256 [doi]
• Adaptive digital filtering of differentially coded signalsJ. W. Lee, C. K. Un, J.-C. Lee. 1257-1260 [doi]
• Adaptive systems as optimal processorsGuy R. L. Sohie. 1261-1262 [doi]
• Broadband detection of signals with unknown spectraSteven Kay. 1263-1265 [doi]
• Adaptive detection of transient signalsBoaz Porat, Benjamin Friedlander. 1266-1269 [doi]
• Automatic recognition of underwater transient signals - a reviewC.-H. Chen. 1270-1272 [doi]
• Thresholds in frequency estimationAllan O. Steinhardt, Chris Bretherton. 1273-1276 [doi]
• Detection of a sinusoidal signal in the presence of directional interferenceB. J. Erickson, G. W. Johnson, D. E. Ohlms. 1277-1280 [doi]
• Robust scale invariant detection of coherent narrowband signals in nearly Gaussian noiseM. Weiss, S. C. Schwartz. 1281-1284 [doi]
• Two-dimensional arrays with nonlinear elementsRoger F. Dwyer. 1285-1288 [doi]
• Microscopic correlation signalsMichel Bouvet, Eugenio J. Tacconi, Bernard Picinbono. 1289-1292 [doi]
• Sonar detection in Weibull bottom reverberationJean Le Gall. 1293-1296 [doi]
• A fast maximum-likelihood estimation and detection algorithm for Bernoulli-Gaussian processesChong-Yung Chi, John Goutsias, Jerry M. Mendel. 1297-1300 [doi]
• Parallel block realization of 2-D IIR digital filtersHsing-Hsing Chiang, Chrysostomos L. Nikias. 1301-1304 [doi]
• Conjugate transformations preserving the general overflow-stability property of 2-D digital filtersTyseer Aboulnasr, Moustafa M. Fahmy. 1305-1308 [doi]
• Separating Y, I, Q components of NTSC composite signalDavid A. Border, S. C. Kwatra. 1309-1312 [doi]
• Hadamard transform seperation of NTSC component signalsJoseph Barba, Norman Scheinberg, M. Colef, Erlan H. Feria. 1313-1316 [doi]
• Lattice-filter modeling of two-dimensional fieldsHanoch Lev-Ari, Sydney R. Parker. 1317-1320 [doi]
• Application of symmetrical decomposition to 2-D FIR filter designP. Karivaratha Rajan, Harnatha C. Reddy. 1321-1324 [doi]
• Two-dimensional orthogonal median filters and applicationsR. Lynn Kirlin, Becky Cudzilo, Sharon Wilson. 1325-1328 [doi]
• A unification of linear, median, order-statistics and morphological filters under mathematical morphologyPetros Maragos, Ronald W. Schafer. 1329-1332 [doi]
• Frequency domain analysis of two-pass rotation algorithmTran Thong. 1333-1336 [doi]
• Constrained predictionBernard Picinbono, Michel Bouvet. 1337-1340 [doi]
• Rational parametric coherence estimation via convolved correlationsJames A. Cadzow, Otis M. Solomon Jr., Samuel D. Stearns. 1341-1344 [doi]
• Spectrum model-order determination via significant reflection coefficientsPaul F. Fougere. 1345-1347 [doi]
• Estimation of sequences in a signal class determined from the dataSergio D. Cabrera, Thomas W. Parks. 1348-1351 [doi]
• Bispectrum estimation for short length dataMysore R. Raghuveer, Chrysostomos L. Nikias. 1352-1355 [doi]
• Explicit formulas for super-resolutionGianni Orlandi, Giuseppe Martinelli, Pietro Burrascano. 1356-1359 [doi]
• A new super-resolution spectral estimation technique using staggered PRFsStanley M. Yuen, Harish M. Subbaram. 1360-1363 [doi]
• A comparison of algorithms for polar-to-cartesian interpolation in spotlight mode SARDavid C. Munson Jr., Jorge L. C. Sanz, W. Kenneth Jenkins, Gary Kakazu, Bruce C. Mather. 1364-1367 [doi]
• A block adaptive approach for clutter suppressionSiew Kok Hui, Y. C. Lim. 1368-1371 [doi]
• Comparing the radar clutter-suppression performances of lattice prediction error filters using three variations of Burg's algorithmOktay Alkin, Ronald C. Houts. 1372-1375 [doi]
• 3architecturesN. Venkateswaran, K. M. M. Prabhu. 1376-1379 [doi]
• A novel architecture design for VLSI implementation of an FIR decimation filterHanafy Meleis, Pierre Le Fur. 1380-1383 [doi]
• Cyclo-static multiprocessor scheduling for the optimal realization of shift-invariant flow graphsD. A. Schwartz, Thomas P. Barnwell III. 1384-1387 [doi]
• The VLSI design of a single chip for the multiplication of integers modulo a fermat numberJaw John Chang, Trieu-Kien Truong, Howard M. Shao, Irving S. Reed, In-Shek Hsu. 1388-1391 [doi]
• VLSI Implementation of a linear systolic arrayJ. Greg Nash, C. Petrozolin. 1392-1395 [doi]
• VLSI Implementation of a fast rank order filtering algorithmRonald G. Harber, Steven C. Bass, G. W. Neudeck. 1396-1399 [doi]
• An oversampling A-to-D converter structure for VLSI digital codec'sAkira Yukawa, Rikio Maruta, Kenji Nakayama. 1400-1403 [doi]
• A VLSI design of a pipeline Reed-Solomon decoderHoward M. Shao, Trieu-Kien Truong, Leslie J. Deutsch, Joseph H. Yuen, Irving S. Reed. 1404-1407 [doi]
• 2-D Array processor having a controlled pipelined architecture for elliptical sparse matricesC. E. Goutis, J. S. Sheblee, G. Russell. 1408-1411 [doi]
• The application of multi-dimensional access memories to ultra high performance signal processing systemsDavid Hayes, Bill Strawhorne. 1412-1415 [doi]
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• 32kbs ADPCM/PCM transcoder using TI-320 DSP microprocessorR. Hangartner, Vijay K. Jain. 1421-1424 [doi]
• A CCITT standard 32 kbps ADPCM LSI codecTakao Nishitani, Ichiro Kuroda, Masao Satoh, Tadaharu Katoh, Reiichi Fukuda, Yasushi Aoki. 1425-1428 [doi]
• A multi-pulse LPC speech codec using digital signal processorsY. Wake, S. Tanaka, Kazunori Ozawa, Takashi Araseki. 1429-1432 [doi]
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• Real-time speech compression with a VLSI vector quantization processorGrant A. Davidson, Terry Stanhope, R. Aravind, Allen Gersho. 1437-1440 [doi]
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• An extension of the fast algorithm for linear phase system identificationS. Lawrence Marple Jr.. 1509-1510 [doi]
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• Fast 2-D discrete cosine transformMartin Vetterli. 1538-1541 [doi]
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• Distributed arithmetic implementation of nonlinear echo cancellersGiovanni L. Sicuranza, Giovanni Ramponi. 1617-1620 [doi]
• High throughput signal processing systemGary Albert, David Fronczak, Jay McKinney. 1621-1624 [doi]
• TMDigital signal processor provides 20 MHz analog bandwidthLouis Schirm IV, Richard de Koeyer. 1625-1628 [doi]
• Effective multifrequency receiver designG. P. Klowak, S. Cohn-Sfetcu, W. Steenaart. 1629-1632 [doi]
• Microprocessor based 9600 bps modemRoger W. Cain. 1633-1636 [doi]
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• A resonant section filter design method for optimized coefficients in add-and-shift architecturesStevan Eidson. 1668-1671 [doi]
• Subband coding of speech using backward adaptive prediction and bit allocationFrank K. Soong, Richard V. Cox, Nikil S. Jayant. 1672-1675 [doi]
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• Speech and speaker independent codebook design in VQ coding schemesHerbert Reininger, Dietrich Wolf. 1700-1702 [doi]
• Embedded coding of speech: A vector quantization approachAmine Haoui, David G. Messerschmitt. 1703-1706 [doi]
• Contour vector quantization and waveform codingThomas R. Fischer, Kevin T. Malone. 1707-1710 [doi]
• Error analysis of linear recursions in floating pointErik I. Verriest. 1711-1714 [doi]
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• Roundoff noise in estimation of the signal random parameterPero J. Radonja. 1723-1726 [doi]
• Multiplier-and memory-based digital FIR filters for nyquist channels: An assessment of quantization noiseCurtis A. Siller Jr.. 1727-1730 [doi]
• ∞|h(n)| for second order digital filtersDong-guang He, Shu-kun Han. 1731-1734 [doi]
• Quantization noise effects in the complex LMS adaptive algorithm-linearization using ditherDouglas T. Sherwood, Neil J. Bershad. 1735-1738 [doi]
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• Adaptive filter improvement using randomly quantized coefficientsHervé Dedieu, Francis Castanie. 1743-1746 [doi]
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• Adaptive beam forming in correlated interference environmentB. B. Madan, S. R. Parker. 1792-1795 [doi]
• Minimum rejection response array filters in the presence of white noiseMagdy T. Hanna, Marwan A. Simaan. 1796-1799 [doi]
• A new set of linear constraints for broadband time domain element space processorsMeng Hwa Er, Antonio Cantoni. 1800-1803 [doi]
• Analysis of constrained LMS algorithm with application to adaptive beamforming using perturbation sequencesLal C. Godara, Antonio Cantoni. 1804-1807 [doi]
• Error analysis of eigenvector preprocessors used in adaptive beamformingLeon H. Sibul, Susan E. Burke. 1808-1811 [doi]
• Finite word length effects on array processing algorithmsE. M. Long, J. M. Schumpert. 1812-1815 [doi]
• A new approach to array geometry for improved spatial spectrum estimationS. Unnikrishna Pillai, Fred Haber, Yeheskel Bar-Ness. 1816-1819 [doi]
• Sensitivity constrained optimum endfire array gainHenry Cox, Robert M. Zeskind, Theo Kooij. 1820-1823 [doi]
• Exact maximum likelihood estimation of superimposed exponential signals in noiseYoram Bresler, Albert Macovski. 1824-1827 [doi]
• Designing filters for interpolation beamformingMartin L. Cohen. 1828-1831 [doi]
• Cepstral analysis of electroacoustic transducersPaul D. Bauman, Stanley P. Lipshitz, John Vanderkooy. 1832-1835 [doi]
• Generalized spline signal interpolation and smoothing with simultaneous optimization in time and frequency domainsRui J. P. de Figueiredo. 1836-1837 [doi]
• Word boundary detection and speech recognition of noisy speech by means of iterative noise cancellation techniquesMin-In Chung, William Kushner, John Damoulakis. 1838 [doi]
• A new syntactic/Semantic approach to 3D-surface reconstructionL. Ray Simar, Rui J. P. de Figueiredo. 1839-1842 [doi]
• Three dimensional shape estimation from two dimensional imagesSally L. Wood. 1843-1846 [doi]
• Custom data-flow machines for speech recognitionThomas S. Anantharaman, Roberto Bisiani. 1847-1850 [doi]
• A control point theory for boundary representation and matchingDavid W. Paglieroni, Anil K. Jain 0002. 1851-1854 [doi]
• Tree structures for implementation of a vector quantized speech coding systemS. Kaul, M. Shridhar. 1855-1857 [doi]
• A new look on the parallel implementation of the Shur algorithm for the solution of Toeplitz equationsGeorge Carayannis, Elias Koukoutsis, Dimitris Manolakis 0001, Cristos C. Halkias. 1858-1861 [doi]