Table of Contents

• IC Technology: Trends and impact on digital signal processingMarcian E. Hoff Jr.. 1-6 [doi]
• Effect of corruption within the recursive estimation of spectral parameters for LPCR. H. Wiggins, J. H. Parry. 7-10 [doi]
• Two-dimensional prediction of area functions for coding of LPC speech parametersJames D. Marr, Thomas P. Barnwell III. 11-14 [doi]
• Speech coding based upon vector quantizationAndres Buzo, Augustine H. Gray Jr., Robert M. Gray, John D. Markel. 15-18 [doi]
• A robust channel vocoder for adverse environmentsNick G. Kingsbury, W. A. Amos. 19-22 [doi]
• Improvements in the classical model for better speech qualitySidhartha Maitra, Charles R. Davis. 23-27 [doi]
• A dynamic programming approach to variable rate speech transmissionPanos Papamichalis, Thomas P. Barnwell III. 28-31 [doi]
• A preliminary design of a phonetic vocoder based on a diphone modelRichard M. Schwartz, John W. Klovstad, John Makhoul, John Sorensen. 32-35 [doi]
• The Karhunen-Loeve transform applied to the log area ratios of a linear predictive speech coderJesse W. Fussell. 36-39 [doi]
• A novel innovation based time domain pitch detectorDaniel T. L. Lee, Martin Morf. 40-44 [doi]
• A novel method for pitch extraction from speech and a hardware model applicable to vocoder systemsRobert J. Sluyter, H. J. Kotmans, A. V. Leeuwaarden. 45-48 [doi]
• Windowing functions for the average magnitude difference function pitch extractorB. A. Fette, R. Gibson, E. Greenwood. 49-52 [doi]
• A decision tree procedure for voiced/Unvoiced/Mixed excitation classification of speechLeah J. Siegel, Alan C. Bessey. 53-56 [doi]
• Voice/Unvoice detection based on a composite-Gaussian source model of speechV. Ramamoorthy. 57-60 [doi]
• Continuous model of the vocal sourceRaymond Descout, Jean-Yves Auloge, Bernard Guérin. 61-64 [doi]
• Automatic parameterization of vocal cord motion from ultra high speed filmsDonald G. Childers, J. S. Mott, G. P. Moore. 65-68 [doi]
• Floating point error bound in the prime factor FFTDavid C. Munson Jr., Bede Liu. 69-72 [doi]
• An algorithm for the design of optimal finite word-length FIR digital filtersDusan M. Kodek. 73-76 [doi]
• Precise pole realization by unobservable digital filtersChrysostomos L. Nikias, Adly T. Fam. 77-80 [doi]
• Block-state recursive digital filters with minimum round-off noiseP. Ananthakrishna, Sanjit K. Mitra. 81-84 [doi]
• A sufficient condition for absence of overflow oscillations in arbitrary digital filters based on the element equationsMasayuki Kawamata, Tatsuo Higuchi. 85-88 [doi]
• Zero sensitivity analysis of the digital lattice filterPeter L. Chu, David G. Messerschmitt. 89-93 [doi]
• Signal processing and complexity of computationShmuel Winograd. 94-101 [doi]
• Generalized Levinson algorithms and Ladder filters for nonstationary signal processingThomas Kailath. 102 [doi]
• A raypath reflection model for layered media with source and receiver in different layersWilliam J. Vetter. 103-106 [doi]
• Seismic imaging of faults in multi-moded coal seamsG. Beresford-Smith, I. M. Mason. 107-110 [doi]
• Digital generation of accurate synthetic seismogramsA. D. McAulay, W. Clay Choate, R. N. Shurtleff. 111-114 [doi]
• Environmental influences on acoustic array design and performance in shallow waterAnthony I. Eller, John F. Miller. 115-119 [doi]
• An experimental study of the MEM applied to array antennas in the presence of multipathJames P. Reilly, Simon Haykin. 120-123 [doi]
• A compound random process for underwater ambient acoustic noiseMagnus Moll. 124-127 [doi]
• Formant representation of parameters for a channel vocoderBernard Gold. 128-130 [doi]
• Characteristics of reflection coefficient estimates based on a Markov chain modelJohn M. Turner, Bradley W. Dickinson, Daniel Lai. 131-134 [doi]
• An experimental comparison of two scalar quantization methodsJohn D. Markel, Augustine H. Gray Jr.. 135-137 [doi]
• A 4.8 KBPS voice excited DFT vocoder with time encoded basebandNils Rydbeck, Per Tjernlund, Jan Uddenfeldt. 138-141 [doi]
• A 4800 bps LPC vocoder with improved excitationChong Kwan Un, Wonyong Sung. 142-145 [doi]
• Observation and modelling of "Formant" transitions using ISASSMichel Chafcouloff, Gérard Chollet, Paul P. Durand, Jacques Guizol, Xavier Rodet. 146-149 [doi]
• Some notes on phase in speech signalsR. C. Cox, David M. Robinson. 150-153 [doi]
• A maximum peakiness criterion for deconvolving speech waveformsSidhartha Maitra, Scott H. Foster, Charles R. Davis. 154-157 [doi]
• Fast spectral estimation of speech signal in analytic formFrank K. Soong, Allen M. Peterson. 158-161 [doi]
• On pole-zero modeling of speechKil Ho Song, Chong Kwan Un. 162-165 [doi]
• Frequency-axis warping to improve automatic word recognitionEdward P. Neuburg. 166-168 [doi]
• State constrained dynamic programming (SCDP) for discrete utterance recognitionHarvey F. Silverman, N. Rex Dixon. 169-172 [doi]
• An investigation of the use of dynamic time warping for word spotting and connected speech recognitionCory S. Myers, Lawrence R. Rabiner, Aaron E. Rosenberg. 173-177 [doi]
• Some experiments in discrete utterance recognitionSubrata K. Das. 178-181 [doi]
• Application of isolated word recognition to a voice controlled repertory dialer systemLawrence R. Rabiner, Jay G. Wilpon, Aaron E. Rosenberg. 182-185 [doi]
• Some aspects of evaluating the performance of a speech recognition system in real applicationsH. Bierfert, M. Kirstein, D. Lance. 186-189 [doi]
• Reduction of minimum word-boundary gap lengths in isolated word recognitionJohn R. Welch, Sheldon C. Oxenberg. 190-193 [doi]
• A connected digit recognizer based on dynamic time warping and isolated digit templatesLawrence R. Rabiner, C. E. Schmidt. 194-198 [doi]
• A speech recognition machine for connected wordsRyohei Nakatsu. 199-202 [doi]
• A conversational mode airline information and reservation system using speech input and outputStephen E. Levinson, Kathleen L. Shipley. 203-208 [doi]
• The enhancement of wordspotting techniquesRobert E. Wohlford, A. Richard Smith, Marvin R. Sambur. 209-212 [doi]
• A limited range discrete Fourier transform algorithmJames W. Cooley, Shmuel Winograd. 213-217 [doi]
• Mathematical background for generalized, partial, and incomplete discrete Fourier transformsDavid P. Maher. 218-221 [doi]
• Rapidly "Bit-reversing" data for the past Fourier transformNorman Brenner. 222-223 [doi]
• A scaling approach for FFT processingGordon L. DeMuth. 224-226 [doi]
• An approach to the diagonalization of the discrete Fourier transformBradley W. Dickinson, Kenneth Steiglitz. 227-230 [doi]
• Computation of the discrete cosine transform via the arcsine transformStephen A. Dyer, Nasir Ahmed, Donald R. Hummels. 231-234 [doi]
• Fast polynomial transform methods for multidimensional DFTsHenri J. Nussbaumer. 235-237 [doi]
• High speed DFT's using residue numbersChao H. Huang, Fred J. Taylor. 238-242 [doi]
• Mersenne numbers rooted on 3 for number theoretic transformsDaniel Minoli, Wendell Nakamine. 243-247 [doi]
• Digital filters with thinned numeratorsG. F. Boudreaux, Thomas W. Parks. 248-251 [doi]
• Design of FIR digital phase networksKenneth Steiglitz. 252-255 [doi]
• On asymmetric FIR interpolators with minimum LperrorRobert A. Gabel. 256-259 [doi]
• Covariance sequence approximation for recursive digital filter designA. A. (Louis) Beex, Louis L. Scharf. 260-263 [doi]
• A new approach to causal filter design by Padé approximantsCharles K. Chui, Andrew K. Chan. 264-267 [doi]
• A multiplicative realization of FIR systems that is logarithmically efficientAdly T. Fam. 268-270 [doi]
• A class of digital filters for decimation and interpolationEugene B. Hogenauer. 271-274 [doi]
• Optimum recursive digital filters with zeros on the unit circleTapio Saramäki. 275-278 [doi]
• Pole-zero decomposition: A new technique for design of digital filtersB. Yegnanarayana. 279-282 [doi]
• Equal ripple amplitude and group delay digital filtersTapio Saramäki, Yrjö Neuvo. 283-286 [doi]
• Order selection for lowpass IIR filtersJ. Bee Bednar, William A. Coberly. 287-290 [doi]
• A filter family designed for use in quadrature mirror filter banksJames D. Johnston. 291-294 [doi]
• On the problem of fixed shading in conjunction with an optimal/Adaptive array processorA. M. Vural. 295-298 [doi]
• Application of ridge regression analysis to optimum array processingBernard J. Repasky, Ben R. Breed. 299-302 [doi]
• Interference removal for random arrays: Beam decoupling approachesAndrew C. Callahan. 303-306 [doi]
• Adaptivity to background noise spatial coherence for high resolution passive methodsGeorges Bienvenu, Laurent Kopp. 307-310 [doi]
• Dynamic beamforming of a random acoustic arrayWilliam S. Hodgkiss. 311-314 [doi]
• Digital encoding of phase shift keying voiceband data signalsJohn B. O'Neal Jr., R. R. Koneru, J. P. Agrawal. 315-318 [doi]
• High quality 16 kb/s voice transmission: The subband coder approachRonald S. Cheung, Raimond L. Winslow. 319-322 [doi]
• A technique for speech coding using dynamic programmingSubrata K. Das. 323-326 [doi]
• The critical band coder-Digital encoding of speech signals based on the perceptual requirements of the auditory systemMichael A. Krasner. 327-331 [doi]
• 16kbps Real time QMF sub-band coding implementationClaude Galand, Daniel Esteban. 332-335 [doi]
• A modified adaptive transform coding scheme with post-processing-enhancementJosé M. Tribolet, Ronald E. Crochiere. 336-339 [doi]
• On the measurement of waveform coder distortion using the log likelihood ratioRonald E. Crochiere, José M. Tribolet, Lawrence R. Rabiner. 340-343 [doi]
• A robust, adaptive transform coder for 9.6 kb/s speech transmissionL. E. Bergeron, Aaron J. Goldberg, Soon-young Kwon, M. Miller. 344-347 [doi]
• Baseband LPC coders for speech transmission over 9.6 kb/s noisy channelsR. Viswanathan, A. Higgins, William Russell, John Makhoul. 348-351 [doi]
• Speech articulation rate change using recursive bandwidth scalingH. Ravindra. 352-355 [doi]
• An embedded-code multirate speech transform coderMichael G. Berouti, John Makhoul. 356-359 [doi]
• Error correction scheme for telephone line transmission of RELP vocoderAspi B. Wadia. 360-363 [doi]
• A single chip speech codec and filterMichael J. McLane, James L. Melsa, David L. Cohn. 364-367 [doi]
• Signal processing architectures with VLSIEarl E. Swartzlander Jr.. 368-371 [doi]
• Bit slice devices for signal processingJohn R. Mick, Bernard New. 372-375 [doi]
• Survey of VLSI for digital signal processingShlomo Waser. 376-379 [doi]
• Floating-point arithmetic for digital signal processingBill Koral, Louis Schirm. 380-382 [doi]
• An LSI digital signal processorJohn S. Thompson, James R. Boddie. 383-385 [doi]
• LSI signal processor development for communications equipmentTakao Nishitani, Yuichi Kawakami, Rikio Maruta, Akira Sawai. 386-389 [doi]
• A single chip NMOS signal processorMatt Townsend, Marcian E. Hoff Jr.. 390-393 [doi]
• A speech/Speaker recognition and response systemGwyn Edwards. 394-397 [doi]
• An integrated circuit for speech synthesisRichard Wiggins. 398-401 [doi]
• A new speech synthesis chip setDennis Morris, David Weinrich. 402-405 [doi]
• Experimental comparison of reduced update Kalman filters and Wiener filters for two-dimensional LMMSE estimationJohn W. Woods, Vinay K. Ingle, R. Hingorani, G. Juskovic. 406-409 [doi]
• Two-dimensional image boundary estimation by use of likelihood maximization and Kalman filteringFernand S. Cohen, David B. Cooper, Howard Elliott, Peter F. Symosek. 410-413 [doi]
• Adaptive nonlinear image restoration by a modified Kalman filtering approachSarah A. Rajala, Rui J. P. de Figueiredo. 414-417 [doi]
• Moving image restoration and registrationRoger Y. Tsai, Thomas S. Huang. 418-421 [doi]
• Estimation of cloud motion from satellite picturesUwe L. Haass, Thomas A. Brubaker. 422-425 [doi]
• 2-D Digital signal processing with an array processorRichard E. Twogood. 426-429 [doi]
• A number theoretic transform approach to image rotation in parallel array processorsThomas A. Kriz, Dale F. Bachman. 430-433 [doi]
• 22D FFT with an FPS AP-120B array processorR. Lynn Kirlin. 434-436 [doi]
• Phase-only signal reconstructionMonson H. Hayes, Jae S. Lim, Alan V. Oppenheim. 437-440 [doi]
• A new non-linear superresolution algorithmRichard L. Frost, Craig K. Rushforth. 441-443 [doi]
• Image restoration by short space spectral subtractionJae S. Lim. 444-448 [doi]
• Artifacts in alpha-rooting of imagesJames H. McClellan. 449-452 [doi]
• Adaptive linear estimation based on time domain orthogonalityStephen D. Huffman, Loren W. Nolte. 453-456 [doi]
• An analysis of multiple correlation cancellation loops with a filter in the auxiliary pathDennis R. Morgan. 457-461 [doi]
• Adaptive multichannel filteringC. Y. Chang. 462-465 [doi]
• Tracking properties of adaptive signal processing algorithmsDavid C. Farden, Khalid Sayood. 466-469 [doi]
• A nonlinear adaptive noise cancellerMichael J. Coker, Donald N. Simkins. 470-473 [doi]
• Miniature CCD-based analog adaptive filtersColin F. N. Cowan, H. Martin Reekie, John Mavor, John W. Arthur, P. B. Denyer. 474-477 [doi]
• On the stability and performance of the adaptive line enhancerArye Nehorai, David Malah. 478-481 [doi]
• ALE Gain performance for narrowband signals in white Gaussian noiseCandace M. Anderson. 482-485 [doi]
• Coherent gain through a frequency domain adaptive LMS algorithmJohn Y. Cheung. 486-489 [doi]
• Measures and perception of phase distortion in electroacoustical systemsDouglas Preis. 490-493 [doi]
• The spatial alignment of loudspeaker drivers on a baffle effects on system amplitude and phase responsesW. Marshall Leach Jr.. 494-497 [doi]
• Conversion of amplitude nonlinearities to phase nonlinearities in feedback audio amplifiersMatti Otala. 498-499 [doi]
• Evaluation of a dereverberation process by normal and impaired listenersP. Jeffrey Bloom. 500-503 [doi]
• Combined time-domain harmonic compression and CVSD for 7.2 kbit/s transmission of speech signalsDavid Malah. 504-507 [doi]
• Experimental comparison of forward and backward adaptive prediction in DPCMJerry D. Gibson, Louis C. Sauter. 508-511 [doi]
• Spectral distance measure applied to the optimum design of DPCM coders with L predictorsJean-Pierre Adoul, Jean-Louis Debray, Daniel Dalle. 512-515 [doi]
• Slope limiting filters for enhancing noisy channel performance of codecsR. Steele, James D. Johnston. 516-519 [doi]
• Speech-quality optimization of 16 kb/s adaptive predictive codersR. Viswanathan, William Russell, A. Higgins, Michael G. Berouti, John Makhoul. 520-525 [doi]
• A split band adaptive predictive coding (SBAPC) speech systemJoel A. Feldman, Robert J. McAulay, Elliot Singer. 526-529 [doi]
• Techniques for improving the robustness of an adaptive predictive coder in the presence of channel errorsElliot Singer. 530-534 [doi]
• Improved quantizer for adaptive predictive coding of speech signals at low bit ratesBishnu S. Atal, Manfred R. Schroeder. 535-538 [doi]
• Optimal noise shaping in adaptive predictive coding of speechNeviano Dal Degan, Carlo Scagliola. 539-542 [doi]
• An adaptively sampled delta modulatorH. F. Vanlandingham, J. F. Bogdanski Jr.. 543-546 [doi]
• Tree and trellis coding of speech and stationary speech-like signalsHeinz G. Fehn, Peter Noll. 547-551 [doi]
• An efficient method for formant to reflection coefficient conversionKeith Blanton. 552-556 [doi]
• Cepstral synthesis of Japanese from CV syllable parametersSatoshi Imai, Yoshiharu Abe. 557-560 [doi]
• Rules for demisyllable synthesis using Lingua, a language interpreterCatherine P. Browman. 561-564 [doi]
• A phonetic dictionary for demisyllabic speech synthesisMarian J. Macchi. 565-567 [doi]
• A scheme for concatenating units for speech synthesisJoseph Olive. 568-571 [doi]
• Perceptual evaluation of MITalk: The MIT unrestricted text-to-speech systemDavid B. Pisoni, Sharon Hunnicutt. 572-575 [doi]
• Unlimited text-to-speech system: Description and evaluation of a microprocessor based deviceJared Bernstein, David B. Pisoni. 576-579 [doi]
• High resolution 2-D spectral analysis at low SNROtis L. Frost. 580-583 [doi]
• High resolution spectral estimates obtained using data extrapolationLloyd J. Griffiths. 584-587 [doi]
• Exponential energy spectral density estimationLarry Marple. 588-591 [doi]
• Improved spectral resolution IIDonald W. Tufts, Ramdas Kumaresan. 592-597 [doi]
• ARMA Spectral estimation: A model equation error procedureJames A. Cadzow. 598-602 [doi]
• Spectrum analysis and resolution enhancement by band limited extrapolationKenneth Abend. 603-606 [doi]
• A note on the measurement of spectral flatness and the calculation of prediction error variancesJean-Pierre Dugré, Louis L. Scharf, A. A. (Louis) Beex. 607-611 [doi]
• Spectral distance measures between Gaussian processesDimitri Kazakos, Panayota Papantoni-Kazakos. 612-613 [doi]
• An iterative procedure for moving average models estimationJoël Le Roux, Yves Grenier. 614-617 [doi]
• Factorial linear modelling, algorithms and applicationsClaude Guéguen, Yves Grenier, F. Giannella. 618-621 [doi]
• An iterative estimation technique for power spectra by an ARMA modelWilliam L. Mills, Clifford T. Mullis, Richard A. Roberts. 622-625 [doi]
• Cascade decimation-A technique for real time estimation of power spectraD. E. Wight, Francis X. Bostick. 626-629 [doi]
• A two-threshold detector for pulses of unknown duration and dopplerJ. B. Plant, Y. T. Chan. 630-633 [doi]
• Comparison of sampling techniques for automatic detection of pulse signals with unknown time of arrivalThomas C. Cantwell, Richard D. Wilmot. 634-637 [doi]
• Detection of partitioned signals by discrete cross-spectrum analysisRoger F. Dwyer. 638-641 [doi]
• Detection performance of an FM correlatorJhong S. Lee, Leonard E. Miller. 642-645 [doi]
• A Wiener filter approach to coherence estimationY. T. Chan, J. M. Riley, J. B. Plant. 646-649 [doi]
• Confidence bounds for signal-to-noise ratios from magnitude-squared coherence estimatesJohn W. Fay. 650-653 [doi]
• Reconstruction of discrete-time signals from a subset of weighted DFT outputsWilliam P. Whyland. 654-657 [doi]
• Detection of a sinusoid in white noise by autoregressive spectrum analysisSteven Kay. 658-661 [doi]
• DFT Interpolation for estimation of tone amplitudes and phasesVijay K. Jain, William L. Collins, David C. Davis. 662-665 [doi]
• Information theory and the concert hall problem: The development of tone, ensemble, and diffusionJames B. Lee. 670-673 [doi]
• University centre - A "Slightly surround" concert hallTheodore J. Schultz. 674-677 [doi]
• Requirements for successful concert hall design: Need for lateral and ensemble reflectionsJerald R. Hyde, A. Harold Marshall. 678-681 [doi]
• Four outdoor classical music sound reinforcement systems comparedDavid L. Klepper. 682-685 [doi]
• Big sound with small things on a medium scaleW. J. Gelow, W. Steven Bussey. 686-689 [doi]
• "Return to forever": A touring sound system for concert hallsWayne R. Lund. 690-691 [doi]
• Adaptive noise cancelling in speech using the short-time transformSteven F. Boll. 692-695 [doi]
• Reducing the effect of background noise for low-bit-rate voice digitizersSidhartha Maitra. 696-698 [doi]
• A real-time noise suppression filter for speech enhancement and robust channel vocodingRobert J. McAulay, Marilyn L. Malpass. 699-702 [doi]
• Interdependencies among measures of speech intelligility and speech "Quality"William D. Voiers. 703-705 [doi]
• Correlation analysis of subjective and objective measures for speech qualityThomas P. Barnwell III. 706-709 [doi]
• A comparison of parametrically different objective speech quality measures using correlation analysis with subjective quality resultsThomas P. Barnwell III. 710-713 [doi]
• An identification method for objective quality measurements on speech waveform codersRoberto Billi, Carlo Scagliola. 714-718 [doi]
• CVSD to LPC conversion using noise tolerant analysisJ. D. Tomcik, James L. Melsa. 719-724 [doi]
• On understanding the quality problems of LPC speechDavid Y. Wong. 725-728 [doi]
• A comparison of hexagonally and rectangularly-sampled two-dimensional FIR digital filtersRussell M. Mersereau, Tae H. Joo, Theresa C. Speake. 729-732 [doi]
• 2-D FIR Filter design via semipolynomial approximationRichard R. Kurth, Michael T. McCallig. 733-736 [doi]
• An efficient ℓp optimization technique for the design of 2-D linear phase FIR digital filtersJohn H. Lodge, Moustafa M. Fahmy. 737-740 [doi]
• An iterative implementation for 2-D digital filtersDan E. Dudgeon. 741-744 [doi]
• A frequency domain approach to synthesizing near optimum edge detection filtersWilliam H. Haas, Claude S. Lindquist. 745-748 [doi]
• Singular value decomposition of 2-D impulse responsesJean-François Abramatic, S. U. Lee. 749-752 [doi]
• Studies on N-dimensional filter transfer functions without second kind singularitiesC. H. Reddy, P. Karivaratha Rajan, M. N. S. Swamy. 753-757 [doi]
• Spatial-domain design of two-dimensional recursive digital filtersSamy A. H. Aly, Moustafa M. Fahmy. 758-761 [doi]
• Comparing 2-D recursive and nonrecursive least-square-error approximation filtersGary A. Shaw, Russell M. Mersereau. 762-765 [doi]
• Computation of correlations in 1-D and 2-D digital signals and systemsS. Y. Hwang. 766-769 [doi]
• Image coding by auto regressive synthesisAmit K. Jain, Surendra Ranganath. 770-773 [doi]
• A video rate two dimensional FFT processorC. S. Joshi, Jack F. McDonald, Randy H. Steinvorth. 774-777 [doi]
• On the development of an integrated circuit for parallel processing of digital filter flow-diagramsReinder Nouta. 778-779 [doi]
• A modular hardware structure for digital filteringMasud Arjmand, Clifford T. Mullis, Richard A. Roberts. 780-783 [doi]
• An ultra-high speed FFT processorRobert A. Collesidis, Todd A. Dutton, Joseph R. Fisher. 784-787 [doi]
• A hardware realization of an NTT convolver using ROM arraysGraham A. Jullien, William C. Miller. 788-791 [doi]
• Large moduli multipliersFred J. Taylor. 792-795 [doi]
• Outline of a fast hardware implementation of Winograd's DFT algorithmShalhav Zohar. 796-799 [doi]
• Multisignal time difference estimator with application to the sound ranging problemLonnie C. Ludeman. 800-803 [doi]
• Improvement of delay measurements from sonar arrays via sequential state estimationR. Lynn Kirlin. 804-806 [doi]
• An experimental comparison of the cross correlation and SCOT techniques for time delay estimationKent Scarbrough, Nasir Ahmed, G. Clifford Carter. 807-810 [doi]
• A structure for the combined reduction of bias and variance in estimating source location and motionJoseph C. Hassab, Brian W. Guimond, Steven C. Nardone. 811-817 [doi]
• Source location from time differences of arrival: Identifiability and estimationJean-Marc Delosme, Martin Morf, Benjamin Friedlander. 818-824 [doi]
• Nonlinear filtering lower bound evaluation of passive tracking systemsJorge I. Galdos, T. Sen Lee. 825-828 [doi]
• Near optimal frequency/Angle of arrival estimates based on maximum entropy spectral techniquesStephen W. Lang. 829-832 [doi]
• A comparative evaluation of several bearings-only tracking filtersBarry L. Clark. 833-847 [doi]
• Event location using recursive least squares signal processingWebster P. Dove, Alan V. Oppenheim. 848-850 [doi]
• Aids for the handicapped based on "Synte 2" speech synthesizerMatti Karjalainen, Unto K. Laine, Raimo Toivonen, Ken R. Haymond, R. Jerome Folmar, Joe Wood. 851-854 [doi]
• The implementation of an all digital speech synthesizer using a multimicroprocessor architectureC. J. M. Hodges, Thomas P. Barnwell III, Daniel McWhorter. 855-858 [doi]
• High performance processor for real-time speech applicationsMichael McLaughlin, Frank Hudziak, Ira A. Gerson, Kevin Kloker. 859-863 [doi]
• The attached processor for speechGuy Hochgesang, Robert V. Lemay, Harvey F. Silverman. 864-867 [doi]
• Programmable synthesis using a new "Speech microprocessor"James L. Caldwell. 868-871 [doi]
• Further results on the recognition of a continuously read natural corpusLalit R. Bahl, Raimo Bakis, Paul S. Cohen, A. G. Cole, Frederick Jelinek, Burn L. Lewis, Robert L. Mercer. 872-875 [doi]
• A parser for segmenting continuous speech into pseudo-syllabic nucleiRenato de Mori, Giovanna Giordano. 876-879 [doi]
• Experiments in syllable-based recognition of continuous speechMelvyn J. Hunt, Matthew Lennig, Paul Mermelstein. 880-883 [doi]
• Statistical models for automatic language identificationK. P. Li, T. J. Edwards. 884-887 [doi]
• Speaker adaptation through canonical correlation analysisYves Grenier. 888-891 [doi]
• Syntactic-Semantic interpretation of sentences in the MYRTILLE II speech understanding systemJean-Marie Pierrel, Jean-Paul Haton. 892-895 [doi]
• A continuous speech recognition system for data base consultationB. Groc, D. Tuffelli. 896-899 [doi]
• Word boundary detection by pitch contours in an artificial languageYutaka Kobayashi, Yasuhisa Niimi. 900-903 [doi]
• A speaker verification algorithm for speech utterances corrupted by noise with unknown statisticsM. Baraniecki, Malayappan Shridhar. 904-907 [doi]
• A comparison of four techniques for automatic speaker recognitionRobert E. Wohlford, Edwin H. Wrench Jr., B. Patrick Landell. 908-911 [doi]
• Attributes of parallel and cascade microprocessor implementations of digital signal processingFred Mintzer. 912-915 [doi]
• Fast multiprocessor realizations of digital filtersMarkku Renfors, Yrjö Neuvo. 916-919 [doi]
• An extensible high level language signal processorNicholas Roethe, Joachim Lenzer. 920-923 [doi]
• Architecture analysis for a communications signal processorElwood L. Seifert, Frank Cornett. 924-926 [doi]
• Why not a high level assembly language?Ken Davies. 927-930 [doi]
• Interactive software systems for digital signal processing applicationsGervasio Prado, R. K. Pearson. 931-934 [doi]
• A facility for interactive digital signal processingRobert B. Fisher III, Allen M. Peterson. 935-938 [doi]
• On design strategies for parallel algorithms in signal processing using graph modelsJoachim Lenzer, Gerald Wieber. 939-942 [doi]
• Minimax time-domain deconvolution for transversal filter equalizersC. Bunks, Douglas Preis. 943-946 [doi]
• Application of the optimal control theory to the deconvolution problemGérard Thomas. 947-949 [doi]
• Deconvolution algorithms based on spline interpolationDietmar Achilles. 950-953 [doi]
• Doubling algorithms for Toeplitz and related equationsMartin Morf. 954-959 [doi]
• Parallel processing algorithms for linear predictive codingLeah J. Siegel. 960-963 [doi]
• Frequency domain data transmission using reduced computational complexity algorithmsAbraham Peled, Antonio Ruiz. 964-967 [doi]
• Signal processing techniques in error control systemsG. Robert Redinbo, W. Y. Cheung. 968-973 [doi]
• Universal coding for quasi-stationary processesRobert J. Fontana. 974-977 [doi]
• A comparison of algorithms for the calculation of adaptive lattice filtersCarey Gibson, Simon Haykin. 978-983 [doi]
• Convergence properties of an adaptive digital lattice filterMichael L. Honig, David G. Messerschmitt. 984-988 [doi]
• Frequency domain considerations of an adaptive escalator predictorDae Hee Youn, Nasir Ahmed. 989-992 [doi]
• Rapid detection of weak signals using adaptive recursive filter weightsLoren R. McMurray. 993-996 [doi]
• On the convergence properties of the simple hyperstable adaptive recursive filter (SHARF)John R. Treichler, Michael G. Larimore, C. Richard Johnson Jr.. 997-1000 [doi]
• A stable family of adaptive IIR filtersC. Richard Johnson Jr.. 1001-1004 [doi]
• Recursive square-root ladder estimation algorithmsDaniel T. L. Lee, Martin Morf. 1005-1017 [doi]
• On using the sequential regression (SER) algorithm for long-term signal processingDavid L. Soldan, Nasir Ahmed, Samuel D. Stearns. 1018-1021 [doi]
• Adaptive algorithms for non-stastical parameter estimation in linear modelsEli Fogel, Y. F. Huang. 1022-1025 [doi]
• Wavenumber acoustics: Passive localization and multipath decompositionRichard B. Lauer. 1026-1029 [doi]
• Group and phase delay requirements for loudspeaker systemsJ. Robert Ashley. 1030-1033 [doi]
• Time delay effects on speech intelligibilityJohn Charles Cox. 1034-1036 [doi]
• On the use of operational amplifiers in loudspeaker analogsBrian Atkinson. 1037-1039 [doi]
• Noise pollution, phase VI: The effects of the onset of industrial noise on the physiological and psyshological functionsMartha E. Hean. 1040 [doi]
• Deductive approach to automatic recognition of Russian spoken sentencesMiron Derkach. 1041-1044 [doi]
• Reduction of broadband noise in speech by spectral weightingM. J. Irwin. 1045-1051 [doi]
• A rapidly converging algorithm for adaptive beam formingBharat B. Madan. 1052-1055 [doi]
• SIPROL: A high level language for digital signal processingHans Gethöffer. 1056-1059 [doi]
• Experimental studies in a new automatic speaker verification system using telephone speechSadaoki Furui, Aaron E. Rosenberg. 1060-1062 [doi]