Table of Contents

• Speaker-independent connected digit recognitionNobuo Hataoka, Yoshiaki Asakawa, Akio Komatsu, Akira Ichikawa. 1-4 [doi]
• A vector quantizer incorporating both LPC shape and energyLawrence R. Rabiner, M. Mohan Sondhi, Stephen E. Levinson. 1-4 [doi]
• Speaker normalization algorithms for very-low-rate speech codingSalim E. Roucos, Alexander MacLeod Wilgus. 1-4 [doi]
• Low bit rate speech coding by concatenation of sound units and prosody codingGérard Benbassat, Xavier Delon. 5-8 [doi]
• Automatic diphone bootstrapping for speaker-adaptive continuous speech recognitionAnna Maria Colla, Donatella Sciarra. 5-8 [doi]
• Short-cut algorithms for the learning subspace methodHeikki Riittinen. 5-8 [doi]
• On the use of transient information in speech recognitionJean-Sylvain Liénard, Frank K. Soong. 9-12 [doi]
• Improving performance of multi-pulse LPC coders at low bit ratesSharad Singhal, Bishnu S. Atal. 9-12 [doi]
• Application of allophonic and lexical constraints in continuous digit recognitionFrancine Chen, Victor W. Zue. 9-12 [doi]
• Efficient algorithm for multi-pulse LPC analysis of speechVijay K. Jain, R. Hangartner. 13-16 [doi]
• Special feature vector coding and appropriate distance definition developed for a speech recognition systemBernhard R. Kämmerer, Wolfgang A. Küpper, Helmut Lagger. 13-16 [doi]
• A directory listing retrieval system based on connected letter recognitionLawrence R. Rabiner, Jay G. Wilpon, Sandra G. Terrace. 13-16 [doi]
• Syntax driven recognition of connected words by Markov modelsM. Cravero, Luciano Fissore, Roberto Pieraccini, Carlo Scagliola. 17-20 [doi]
• Low bit rate speech enhancement using a new method of multiple impulse excitationA. Parker, S. Thomas Alexander, Henry J. Trussell. 17-20 [doi]
• Performance of isolated word recognition system for confusable vocabularyS. Raman, B. Yegnanarayana. 17-20 [doi]
• Improved hidden Markov modeling of phonemes for continuous speech recognitionRichard M. Schwartz, Yen-Lu Chow, Salim E. Roucos, Michael A. Krasner, John I. Makhoul. 21-24 [doi]
• A glottal LPC-vocoderPer Hedelin. 21-24 [doi]
• Unsupervised adaptation to new speakers in feature-based letter recognitionMoshe J. Lasry, Richard M. Stern. 21-24 [doi]
• Speech recognition under additive noiseChin-Hui Lee, Kalyan Genesan. 25-28 [doi]
• Improvement of the narrowband LPC synthesisGeorge S. Kang, Stephanie Everett. 25-28 [doi]
• Feature-based speaker-independent word recognition without oral learningPatrick Fonsale. 25-28 [doi]
• One-pass syntax-directed connected-word recognition in a time-sharing environmentDenis Jouvet, R. Schwartz. 29-32 [doi]
• A speaker independent word recognition system based on phoneme recognition for a large size (212 words) vocabularyShozo Makino, Ken'iti Kido. 29-32 [doi]
• Optimal Lloyd-Max quantization of LPC speech parametersMatthew I. Noah. 29-32 [doi]
• Use of computational psychoacoustical models in speech processing: Coding and objective performance evaluationJ. Koljonen, Matti Karjalainen. 33-36 [doi]
• The CDTWP: A programmable processor for connected word recognitionJohn G. Ackenhusen. 33-36 [doi]
• Auditory models in isolated word recognitionMats Blomberg, Rolf Carlson, Kjell Elenius, Björn Granström. 33-36 [doi]
• Comparison of a model of the peripheral auditory system and L.P.C. analysis in a speech recognition systemIoannis Dologlou, Jean Marc Dolmazon. 37-40 [doi]
• Evaluation of time compression for connected word recognitionJean-Luc Gauvain, Joseph Mariani. 37-40 [doi]
• Line spectrum pair (LSP) and speech data compressionFrank K. Soong, Biing-Hwang Juang. 37-40 [doi]
• Computational models of neural auditory processingRichard F. Lyon. 41-44 [doi]
• Segmentation in isolated word recognition using vector quantizationMarcia A. Bush, Gary E. Kopec, Niels Lauritzen. 41-44 [doi]
• A multirate root LPC speech synthesizerChia-Chuan Hsiao, Robert W. Brodersen. 41-44 [doi]
• Speech segmentation and recognition using adaptive linear prediction algorithmTakeshi Fukabayashi, Chiu-Kuang Chuang. 45-48 [doi]
• Baseband speech coding at 2400 bps using "Spherical vector quantization"Jean-Pierre Adoul, Claude Lamblin, Alain Le Guyader. 45-48 [doi]
• Pitch and spectral estimation of speech based on auditory synchrony modelStephanie Seneff. 45-48 [doi]
• A nonlinear spectrum processing technique for speech enhancementThomas E. Eger, James Su, L. William Varner. 49-52 [doi]
• A parametric representation of short-time power spectra based on the acoustic properties of the earHarald Höge. 49-51 [doi]
• A sample selective linear prediction analysis of speechOsamu Kakusho, Masuzo Yanagida, Riichiro Mizoguchi. 49-52 [doi]
• Modelling of the laryngeal acoustic source by labile nonlinear oscillatorsB. L. Bardakjian. 52-55 [doi]
• Spectral envelope sampling and interpolation in linear predictive analysis of speechHynek Hermansky, Hiroya Fujisaki, Yasuo Sato. 53-56 [doi]
• Optimal estimators for spectral restoration of noisy speechJack E. Porter, Steven F. Boll. 53-56 [doi]
• Automatic glottal inverse filteringDale E. Veeneman, Spencer L. BeMent. 56-59 [doi]
• An algebraic approach to discrete short-time Fourier transform analysis and synthesisZ. Shpiro, David Malah. 57-60 [doi]
• Multisensor speech input for enhanced immunity to acoustic background noiseVishu Viswanathan, Kenneth F. Karnofsky, Kenneth N. Stevens, Michael N. Alakel. 57-60 [doi]
• Two channel (speech and egg) analysis for formant and glottal inverse filteringA. K. Krishnamurthy. 60-63 [doi]
• Speech synthesis from short-time Fourier transform magnitude and its application to speech processingDaniel W. Griffin, Douglas S. Deadrick, Jae S. Lim. 61-64 [doi]
• Adaptive noise cancellation in a fighter cockpit environmentWilliam Harrison, Jae S. Lim, Elliot Singer. 61-64 [doi]
• Acoustic tube analysis of formant bandwidths and frequencies in helium speechMark A. Richards, Ronald W. Schafer. 64-67 [doi]
• The harmonic magnitude suppression (EMS) technique for intelligibility enhancement in the presence of interfering speechBrian A. Hanson, David Y. Wong. 65-68 [doi]
• Time alignment of natural speech to synthetic speechMelvyn J. Hunt, C. P. Swail. 65-68 [doi]
• An all-zero model for higher pole correctionUnto K. Laine. 68-71 [doi]
• Use of dynamic programming for automatic synchronization of two similar speech signalsP. Jeffrey Bloom. 69-72 [doi]
• The GRASP sound separation systemMitchel Weintraub. 69-72 [doi]
• Comparative evaluation of a new procedure for adaptive estimation of noisy imagesZhang Zhao Pu, Howard Kaufman. 72-75 [doi]
• Accurate pitch determination of speech signals by means of a laryngographWolfgang J. Hess, Helge Indefrey. 73-76 [doi]
• A procedure for automatic alignment of phonetic transcriptions with continuous speechHong C. Leung, Victor W. Zue. 73-76 [doi]
• An iterative method for restoring noisy blurred imagesAggelos K. Katsaggelos, Jan Biemond, Russell M. Mersereau, Ronald W. Schafer. 76-79 [doi]
• Synthesis by rule of english intonation patternsMark Anderson, Janet B. Pierrehumbert, Mark Liberman. 77-80 [doi]
• A powerful post-processing algorithm for time-domain pitch trackersPhilippe Specker. 77-80 [doi]
• One-dimensional processing for adaptive image restorationPhilip Chan, Jae S. Lim. 80-83 [doi]
• A pitch detection algorithm with hypothesis and test strategy by means of fast surface AMDFD. Tuffelli. 81-84 [doi]
• A text-to-speech system for italianRodolfo Delmonte, Gian Antonio Mian, Graziano Tisato. 81-84 [doi]
• Nonstationary 2-D recursive restoration of images with signal-dependent noiseDarwin T. Kuan, Alexander A. Sawchuk, Timothy C. Strand, Pierre Chavel. 84-87 [doi]
• A performance comparison of pitch extraction algorithms for noisy speechKyong-Ae Oh, Chong Kwan Un. 85-88 [doi]
• Real time text-to-speech conversion system for spanishJuan Carlos Olabe, A. Santos, R. Martinez, Elias Munoz-Merino, M. Martinez, A. Quilis, Jared Bernstein. 85-87 [doi]
• Image restoration by the method of generalized projections with application to restoration from magnitudeAharon Levi, Henry Stark. 88-91 [doi]
• A system for converting teletext into speechRoger M. Meli, Frank Fallside. 88-90 [doi]
• Investigation of text-independent speaker identification techniques under conditions of variable dataMichael A. Krasner, Jared J. Wolf, Kenneth F. Karnofsky, Richard M. Schwartz, Salim E. Roucos, Herbert Gish. 89-92 [doi]
• Towards increasing the commercial success of speech synthesizersM. Remmel, T. Tago. 91-93 [doi]
• Singular value decomposition, singular vectors, and the discrete prolate spheroidal sequencesY. Zhou, Craig K. Rushforth, Richard L. Frost. 92-95 [doi]
• A speaker recognizability testPanos Papamichalis, George R. Doddington. 93-96 [doi]
• Fast frequency tracking using an adaptive lattice filter for a vortex flowmeter signalD. Marginedes. 94-97 [doi]
• Experiments on the use of local statistics for adaptive image processingRobin N. Strickland. 96-99 [doi]
• An endpoint detector for LPC speech using residual error look-ahead for vector quantization applicationsChieh Tsao, Robert M. Gray. 97-100 [doi]
• A real-arithmetic implementation of the constant modulus algorithmJohn R. Treichler, Michael G. Larimore. 98-101 [doi]
• Homomorphic restoration of images degraded by light cloud coverTamar Peli, Thomas F. Quatieri. 100-103 [doi]
• Analog interface chips for audio band digital signal processingScott Robertson, William Feger, Richard K. Hester. 101-104 [doi]
• Convergence conditions for an optimal solution in adaptive recursive filtersJohn Y. Cheung. 102-105 [doi]
• Iterative generalized inverse image restorationDidier Saint-Félix, Ali Mohammad-Djafari, Guy Demoment. 104-107 [doi]
• Experimental hardware for real-time wideband speech codingJames H. Snyder. 105-107 [doi]
• Extension facilities and performance of an LSI adaptive filterC. R. South, A. V. Lewis. 106-109 [doi]
• Testing of wideband digital codersRichard V. Cox, Jeffrey Snyder, Ronald E. Crochiere, D. Bock, James D. Johnston. 108-111 [doi]
• A two-dimensional adaptive image deblurring filterS. Cooke, Tariq S. Durrani. 108-111 [doi]
• On the probability density functions of the weight for the complex scalar LMS adaptive algorithmNeil J. Bershad, Lian Zuo Qu. 110-113 [doi]
• Recursive Wiener filtering for image restorationM. S. Ahmed, Khaldoun K. Tahboub. 112-115 [doi]
• A flexible sampling-rate conversion methodJulius O. Smith, Phil Gossett. 112-115 [doi]
• Time correlation statistics of the LMS adaptive algorithm weightsNeil J. Bershad, Yuh-Huu Chang. 114-117 [doi]
• A new digital voice summing technique for teleconferencingTo Russell Hsing. 116-119 [doi]
• An improved signal restoration method using frequency domain informationKai-Bor Yu. 116-119 [doi]
• Nonstationary learning characteristics of least squares adaptive estimation algorithmsFuyun Ling, John G. Proakis. 118-121 [doi]
• A parametric approach to bispectrum estimationMysore R. Raghuveer, Chrysostomos L. Nikias. 120-123 [doi]
• Automatic gain controlGeorge S. Kang, Mark L. Lidd. 120-123 [doi]
• Stable time- and order-recursive algorithm for the adaptive lattice filterYan Chen. 122-125 [doi]
• ARMA model order/Data length tradeoff for specified frequency resolutionT. Srinivasan, David C. Swanson, Frank W. Symons Jr.. 124-127 [doi]
• Modelling musical instruments in the digital domainS. G. Smith. 124-127 [doi]
• Characteristics of adaptive filters with leakageCumhur Cengiz Evci, Maurice G. Bellanger. 126-129 [doi]
• Computationally efficient estimation of the mean frequency for real-valued signalsS. Sjoberg. 128-131 [doi]
• A speech direction finderDavid R. Fischell, Cecil H. Coker. 128-131 [doi]
• Adaptive echo cancellation with nonlinear digital filtersGiovanni L. Sicuranza, Andrea Bucconi, Paolo Mitri. 130-133 [doi]
• Invariant detection of transient ARMA signals with unknown initial conditionsSteven M. Kay, Louis L. Scharf. 132-135 [doi]
• Sound quality measurements of audio systems based on models of auditory perceptionMatti Karjalainen. 132-135 [doi]
• On realizations and related algorithms for adaptive linear phase filteringDae Hee Youn, S. Prakash. 134-137 [doi]
• Explicit formulas for estimating the frequencies of sine waves in noise with few samplesG. Martinelli, Gianni Orlandi. 136-138 [doi]
• The minimum detectable delay of speech and musicJohn Charles Cox. 136-139 [doi]
• Two-dimensional spectral estimation with autoregressive lattice parametersAhmet H. Kayran, Sydney R. Parker, D. J. Klich. 138-141 [doi]
• A quantitative comparison of generalized correlator window functions in presence of strong spectral peak for spatially separated sourcesRonald E. Boucher, Joseph C. Hassab. 139-142 [doi]
• Application of a simple hearing model to the design of audio filtersMichael J. Ready, V. Ralph Algazi. 140-143 [doi]
• Application of SVD to 2-D spectral estimationNan Miao, Zong-Zhi Chen. 142-145 [doi]
• The complete characterization of a multiple sinusoid signalMark Horwedel, Sathyanarsyan S. Rao. 143-146 [doi]
• Requirements for loudspeaker crossover networksJ. Robert Ashley. 144-147 [doi]
• Tomographic and spectral analysis using noise statisticsRichard M. Leahy, Constantinos E. Goutis, S. Drossos. 146-149 [doi]
• A technique for spectral component location within a FFT resolution cellS. S. Ng. 147-149 [doi]
• Initial condition transient suppression for two dimensional recursive digital filtersWinser E. Alexander. 148-151 [doi]
• An iterative algorithm for robust 2-D spectrum estimationGovind Sharma 0002, Ramalingam Chellappa. 150-153 [doi]
• Scale-space filtering: A new approach to multi-scale descriptionAndrew P. Witkin. 150-153 [doi]
• On 2-D FIR and IIR filter designGeorge A. Lampropoulos, Mamdouh F. Fahmy. 152-155 [doi]
• The signal data base system SDBGary E. Kopec. 154-157 [doi]
• A model based approach for 2-D MEPS analysisRamalingam Chellappa, Govind Sharma 0002. 154-157 [doi]
• Circularly symmetric filter design using 2D prolate spheroidal sequencesRoy Chapman, Tariq S. Durrani. 156-159 [doi]
• Signal-to-symbol transformation: Reasoning in the HASP/SIAP programH. Penny Nii. 158-161 [doi]
• The importance of random phase for image reconstruction from frequency offset Fourier dataDavid C. Munson Jr., Jorge L. C. Sanz. 158-161 [doi]
• Realization of two-dimensional digital filters by LU decomposition of their transfer functionAnastasios N. Venetsanopoulos, Chrysostomos L. Nikias. 160-163 [doi]
• Improvement of resolution and reduction of computation in 2D spectral estimation using decimationLi-He Zou, Bede Liu. 162-165 [doi]
• Knowledge based speech analysis and enhancementCory S. Myers, Alan V. Oppenheim, Randall Davis, Webster P. Dove. 162-165 [doi]
• Uncertainty, eigenvalue problems and filter designRoland Wilson. 164-167 [doi]
• Short-space Fourier transform image processingBrian L. Hinman, Jared Bernstein, David H. Staelin. 166-169 [doi]
• "Ignorance-based" systemsAlan S. Gevins, Nelson Morgan. 166-169 [doi]
• Computation of the threshold of stability for N-dimensional digital filtersM. N. S. Swamy, Leonid M. Roytman, Eugene I. Plotkin. 168-170 [doi]
• 2D Discrete cosine transform computation by fast polynomial transform algorithmsSoo-Chang Pei, Eng-Fong Huang. 170-173 [doi]
• Spectral estimation using level-crossing dataGene Hostetter. 170-173 [doi]
• Two-dimensional passive state-space digital filter designHon Keung Kwan. 171-174 [doi]
• A state space approach for the 2-D harmonic retrieval problemD. V. Bhaskar Rao, Sun-Yuan Kung. 174-176 [doi]
• Expert systems as automated decision aidsHerbert E. Rauch. 174-177 [doi]
• Structural pseudolosslessness and structural pseudopassivity of 1-D and 2-D digital filtersMarek Domanski. 175-178 [doi]
• A high speed two-dimensional FFT processorNuthalapati U. Chowdary, Willem J. D. Steenaart. 177-180 [doi]
• Automated detection in multiple-target environments using the censored mean-level detectorJoe A. Presley Jr.. 178-181 [doi]
• Two stability tests for two-dimensional digital filtersM. N. S. Swamy, Leonid M. Roytman, Eugene I. Plotkin. 179-182 [doi]
• A new method for wideband sensor array processingS. Hamid Nawab, Farid U. Dowla, R. T. Lacoss. 181-184 [doi]
• High-resolution techniques for two-dimensional estimation of angle-of-arrival for planar arraysJames M. Alsup. 182-185 [doi]
• Transfer function realization of a class of doubly-terminated two-variable lossless networks and their application in linear-phase 2-dimensional digital filter designM. Omair Ahmad, Majid Ahmadi, Venkat Ramachandran. 183-186 [doi]
• Accurate pole estimation by modified linear predictionVijay K. Jain. 185-188 [doi]
• The effect of an auxiliary source on the performance of a randomly perturbed arrayAshok Erramilli, Peter M. Schultheiss. 186-189 [doi]
• Median filters: Analysis for 2 dimensional recursively filtered signalsGonzalo R. Arce, Regis J. Crinon. 187-190 [doi]
• Comparison of three auto-regressive modeling methodsA. R. Gondeck, Vijay K. Jain. 189-192 [doi]
• Joint source and sensor location estimationNorman L. Owsley. 190-193 [doi]
• Adaptive signal processing for adaptive controlBernard Widrow, Eugene Walach. 191-194 [doi]
• Relationship between maximum-likelihood-method and autoregressive modeling in multidimensional power spectrum estimationFarid U. Dowla, Jae S. Lim. 193-196 [doi]
• A point mechanical model for the dynamics of towed arraysW. Brandenburg. 194-197 [doi]
• Detection of multiple sinusoids using a parallel ALER. A. David. 195-198 [doi]
• Parameter estimation algorithms utilizing implicit phaseMichael T. Manry. 197-199 [doi]
• Optimal passive localization from a single sensor using multiple linear hypothesesG. W. Johnson, A. O. Cohen, E. J. Modugno, C. W. Shier. 198-201 [doi]
• Detection of multiple sinusoids using an adaptive cascaded structureNasir Ahmed, Don R. Hush, G. R. Elliott, R. Fogler. 199-202 [doi]
• Adaptive least-squares for parametric spectral estimation and its application to pulse estimation and deconvolution of seismic dataHossny El-Sherief. 200-203 [doi]
• An all-digital realization of a baseband DLL implemented as dynamical state estimatorJ. Bohmann, H. Meyer. 202-205 [doi]
• Adaptive algorithms that restore signal propertiesJohn R. Treichler. 203-206 [doi]
• Signal subspace analysis and improvement of spectral estimation algorithmsGuaning Su. 204-207 [doi]
• A convergence analysis of an adaptive underwater passive tracking systemRichard L. Moose, Mauro J. Caputi. 206-209 [doi]
• A parametric method for computing magnitude squared coherenceOtis M. Soloman, James A. Cadzow, Samuel D. Stearns. 207-210 [doi]
• Applications of singular value decomposition to system modeling in signal processingKonstantinos Konstantinides, Kung Yao. 208-211 [doi]
• A new approach to decentralized array processingMati Wax, Thomas Kailath. 210-213 [doi]
• Suppression of FM adjacent channel interferenceMichael G. Larimore, John R. Treichler. 211-214 [doi]
• The analysis of non-oscillating transients using the covariance method with many different ordersFinley R. Shapiro, Stephen D. Senturia, David Adler. 212-215 [doi]
• Optimal linear arrays for narrow-band beamformingStuart R. DeGraaf, Don H. Johnson. 214-217 [doi]
• The implementation of digital filters using a modified Widrow-Hoff algorithm for the adaptive cancellation of acoustic noiseL. A. Poole, G. E. Warnaka, R. C. Cutter. 215-218 [doi]
• A new spectrum analysis approach using autocorrelation technique and MEMS. K. Hui, Y. C. Lim. 216-219 [doi]
• Composite complex sinusoidal modeling for the estimation of directions and spectra of incident plane wavesMasato Abe, Ken'iti Kido. 218-221 [doi]
• A fast least squares linear phase adaptive filterS. Lawrence Marple Jr.. 219-222 [doi]
• Topological aspects of the caratheodory problemTryphon T. Georgiou. 220-223 [doi]
• Modal decomposition for detection of plane waves with a line array receiverR. S. Walker, A. T. Ashley. 222-225 [doi]
• A new realization scheme of periodically time-varying digital filtersChrysostomos L. Nikias. 223-226 [doi]
• On adaptive implementations of Pisarenko's harmonic retrieval methodRichard J. Vaccaro. 224-227 [doi]
• A new class of broadband time domain element space antenna array processorsAntonio Cantoni, Meng Hwa Er. 226-229 [doi]
• Time-varying autoregressive modeling of a class of nonstationary signalsKen C. Sharman, Benjamin Friedlander. 227-230 [doi]
• Constrained lattice structures for harmonic retrievalYu Hen Hu, Yao-Cheng Ling. 228-231 [doi]
• A bit serial linear array DFTGregory H. Allen, Peter B. Denyer, David S. Renshaw. 230-233 [doi]
• Signal estimation using modified Wigner distributionsGloria Faye Boudreaux-Bartels, Thomas W. Parks. 231-234 [doi]
• Determining the number of signals by information theoretic criteriaMati Wax, Thomas Kailath. 232-235 [doi]
• Hardware realization of Mersenne number transforms for fast digital convolutionWan-Chi Siu, Anthony G. Constantinides. 234-237 [doi]
• The design of well-conditioned discrete imaging systemsY. Zhou, Richard L. Frost, Craig K. Rushforth. 235-238 [doi]
• Square root normalized feedback ladder algorithm for the identification of moving average systemsCarlos H. Muravchik, Martin Morf. 236-239 [doi]
• A bit serial LDI recursive digital filterLaurence E. Turner, Peter B. Denyer, David S. Renshaw. 238-241 [doi]
• Recognition of time-varying signals in the time-frequency domain by means of the Wigner distributionBoualem Bouachache, Francisco Rodriguez. 239-242 [doi]
• Fitting sinusoids to sampled data using a hybrid S/Z plane moment methodJ. E. Hudson. 240-243 [doi]
• A residue to mixed radix converter and error checker for a five-moduli residue number systemM. J. Bell Jr., W. Kenneth Jenkins. 242-245 [doi]
• Fast sequential least-squares processingEvangelos E. Milios. 243-246 [doi]
• Autoregressive spectral estimation in noise with application to speech analysisRobert D. Preuss, Rao Yarlagadda. 244-247 [doi]
• Parallel processing of digital convolution using finite polynomial ringsI. Defee. 246-249 [doi]
• A framework for the evaluation of spectral analysis techniquesManuel Duarte Ortigueira, José M. Tribolet. 247-250 [doi]
• A new harmonical algorithm for digital signal processingH. Madala. 248-251 [doi]
• Pipelined cordic architectures for fast VLSI filtering and array processingEd F. Deprettere, Patrick Dewilde, R. Udo. 250-253 [doi]
• Applications of the median filter to digital radiographic imagesE. Russell Ritenour, T. R. Nelson, U. Raff. 251-254 [doi]
• Non-uniform sampling for high resolution spectrum analysisNeil J. Malloy. 252-255 [doi]
• Distributions in signal theoryLeon Cohen. 254-257 [doi]
• Statistical analysis of two dimensional median filtered imagesT. A. Nodes, G. Y. Liao, Neal C. Gallagher Jr.. 255-258 [doi]
• Sampling a broad band sparse spectrum without anti-aliassing filtersFred D. Powell. 256-258 [doi]
• Gabor representation and Wigner distribution of signalsAugustus J. E. M. Janssen. 258-261 [doi]
• A comparison of image filtering algorithmsSubrahmanyam Dravida, John W. Woods, W. Shen. 259-262 [doi]
• On-line trend detection based on ARI modelingK. Hashimoto, A. Sano. 259-262 [doi]
• Measuring the degree of non-stationarity by using the Wigner-Ville spectrumW. Martin. 262-265 [doi]
• A new N.A.R power estimation for adaptive detectionEugenio J. Tacconi, Michel Bouvet, Bernard C. Picinbono. 263-266 [doi]
• Algorithms for the simulation of geometric distortion in a satellite imaging systemJ. V. Aanstoos, W. Howard Ruedger, B. D. Meredith, M. E. Beatty III. 263-266 [doi]
• Some features of time-frequency representations of multicomponent signalsPatrick Flandrin. 266-269 [doi]
• Solving for 3D motion parameters of a rigid body by a vector geometrical approach: Uniqueness and numerical resultsB. L. Yen, Thomas S. Huang. 267-270 [doi]
• Multichannel adaptive array processing for optimal detectionS. L. Earp, Loren W. Nolte. 267-270 [doi]
• Time-frequency analysis using time-dependent ARMA modelsYves Grenier. 270-273 [doi]
• Calculation of displacement fields by means of the motion detection transformMargie H. Groves, Sarah A. Rajala, Wesley E. Snyder. 271-274 [doi]
• Estimation of source parameters by maximum likelihood and nonlinear regressionJohann F. Böhme. 271-274 [doi]
• Joint representations (JR) in signal theory (ST) and hilbertian analysis: A powerful tool for signal analysisBernard Escudié, Jean Grea. 274-277 [doi]
• Separation of sinusoids using the constrained adaptive line enhancerRatnam V. Raja Kumar, Ranendra N. Pal. 275-278 [doi]
• Filtering and correlation of time-sequentially sampled spatiotemporal signalsJan P. Allebach. 275-278 [doi]
• On the time-frequency discrimination of energy distributions: Can they look sharper than Heisenberg ?Theo A. C. M. Claasen, Wolfgang F. G. Mecklenbräuker. 278-281 [doi]
• Estimation of angles-of-arrival for wideband sourcesH. Wang, Mostafa Kaveh. 279-282 [doi]
• Object identification and orientation determination in 3-space with no point correspondence informationDavid Cyganski, John A. Orr. 279-282 [doi]
• Spectrograms and generalized spectrograms for classification of random processesRichard A. Altes. 282-285 [doi]
• Performance characteristics of biased estimatorsAlbert A. Gerlach. 283-286 [doi]
• A method for finding straight line and plane correspondences in stereopair imagesSerge Castan, Jun Shen. 283-286 [doi]
• A range and azimuth estimator based on forming the spatial Wigner distributionBen R. Breed, Theodore E. Posch. 286-287 [doi]
• Track likelihood statistic for the I Q locked loopD. E. Ohlms, M. L. Hampton. 287-289 [doi]
• Adaptive image smoothing algorithms for edge and texture preservationK. M. Tao. 287-290 [doi]
• Voiceless stop consonant identification using LPC spectraGary E. Kopec. 288-291 [doi]
• Kalman estimation of frequency-domain interference reduction coefficients for angle-modulated signalsFrank Cornett. 290-293 [doi]
• New results in seismic deconvolution using lattice processorsTariq S. Durrani, J. L. Bowie. 291-294 [doi]
• Recognition of consonants using an ARMA model of the speech signalA. Kumar, George A. Bekey. 292-295 [doi]
• Essential limitations to signal detection and estimation: An application to the arctic under ice environmental noise problemRoger F. Dwyer. 294-297 [doi]
• Beam-steered vertical seismic arrays for wave classificationA. M. Bisbee, E. A. Quincy, D. J. Tomich. 295-298 [doi]
• Properties of consonant sequences within words and across word boundariesLori Faith Lamel, Victor W. Zue. 296-299 [doi]
• The integrated signal processing system ISPGary E. Kopec. 298-301 [doi]
• Slowness aliasing in the Radon transformDavid J. Scheibner, Thomas W. Parks. 299-302 [doi]
• How to avoid vowel normalization in identification of vowels in continuous speechMaria-Gabriella Di Benedetto, Armando Lanaro. 300-303 [doi]
• A graphical notation for describing data flow in digital filtersS. N. Terepin. 302-305 [doi]
• A model for sonar bottom sounding processingC. Sherwin, W. V. McCollough, D. D. McCrady, S. Johnson. 303-306 [doi]
• An information theoretic approach to the automatic determination of phonemic baseformsJohn M. Lucassen, Robert L. Mercer. 304-307 [doi]
• A graph theoretic technique for the generation of systolic implementations for shift-invariant flow graphsD. A. Schwartz, Thomas P. Barnwell III. 306-309 [doi]
• Coherent recombination of sediment borne and water path acoustic signalsWilliam S. Hodgkiss, Richard K. Brienzo. 307-310 [doi]
• Modeling of english speech for the design of a distributed speech understanding systemEdward C. Bronson, Edward J. Coyle, Leah J. Siegel. 308-311 [doi]
• On hardware description from block diagramsHosagrahar V. Jagadish, Thomas Kailath, John A. Newkirk, Robert G. Mathews. 310-313 [doi]
• The influence of Doppler effect on information transmission in sound channelChangxue Ren, Chengqi Xu. 311-314 [doi]
• Algorithms for acoustic prosodic analysisWayne A. Lea, Frantz Clermont. 312-315 [doi]
• Design framework for systolic-type arraysJ. M. Jover, Thomas Kailath. 314-317 [doi]
• Spatial properties of pulsed-Doppler current profiling systemsKenneth B. Theriault. 315-318 [doi]
• An expert system for the automatic reading of French spectrogramsNoëlle Carbonell, Dominique Fohr, Jean-Paul Haton, François Lonchamp, Jean-Marie Pierrel. 316-319 [doi]
• Signal processing software toolsDon H. Johnson. 318-320 [doi]
• A frequency domain method for time-shift estimation and alignment of discrete time signalsMarwan A. Simaan. 319-322 [doi]
• Integration of acoustic, phonetic, prosodic and lexical knowledge in an expert system for speech understandingRenato de Mori, Michel Gilloux, Guy Mercier, M. Simon, C. Tarridec, Jacqueline Vaissière, D. Gillet, M. Gerard. 320-323 [doi]
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• Highly parallel architectures and algorithms for speech analysisLeah J. Siegel. 331-334 [doi]
• Linear complexity fast algorithms for a class of linear equationsHari Krishna, Salvatore D. Morgera. 332-335 [doi]
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• Systolic architectures for deconvolution algorithmsR. Yoho, Douglas Preis. 339-342 [doi]
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• Parameter selection for isolated word recognition using vector quantizationDavid K. Burton, Joseph T. Buck, John E. Shore. 344-347 [doi]
• Fast transform processor implementationEarl E. Swartzlander Jr., George Hallnor. 347-350 [doi]
• Efficient algorithms and structures for lagged least squares (LS) FIR filters in the case of prewindowed signalsGeorge Carayannis, Dimitris Manolakis, Nicholas Kalouptsidis. 347-350 [doi]
• Evaluation of an isolated word recognizer in talker-dependent and talker-independent modes using a large telephone-band data baseAaron E. Rosenberg, Kathleen L. Shipley. 348-351 [doi]
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• A dynamic time warp IC for a one thousand word recognition systemRobert Kavaler, Robert W. Brodersen, Tobias G. Noll, Menahem Lowy, Hy Murveit. 375-378 [doi]
• On the problem of dimensionality and sample size in multi-stage pattern classifiersH. M. Dante. 376-379 [doi]
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• Suprasegmentals in very large vocabulary isolated word recognitionAlex Waibel. 387-390 [doi]
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• A logarithmic arithmetic unit for signal processingFred J. Taylor. 398-401 [doi]
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• One approach to microprocessor implementation of 4800 b/s data modem for telephone channelsVeljko Milutinovic. 402-405 [doi]
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• Implementation of a full duplex 2.4 kbps LPC vocoder on a single TMS-320 microprocessor chipBrian Bryden, Hisham Hassanein. 406-409 [doi]
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• Phrase speech recognition of large vocabulary using feature in articulatory domainKatsuhiko Shirai, Tetsunori Kobayashi. 409-412 [doi]
• A single-processor LPC vocoderAndrew Holck, Wallace W. Anderson. 410-413 [doi]
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• Connected digit recognition using vector quantizationHervé Bourlard, Christian Wellekens, Hermann Ney. 413-416 [doi]
• A DFT-based front-end for word recognition systemsR. Geppert, P. Schwartau. 414-417 [doi]
• Hierarchical vector quantization of speech with dynamic codebook allocationAllen Gersho, Yair Shoham. 416-419 [doi]
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• 16 kbps APC with hybrid quantizationMichael D. Cowing, Jesse W. Fussell. 425-428 [doi]
• System identification techniques for noise reduction in evoked potential processingH. Rauner, Werner Wolf, Ulrich Appel. 426-429 [doi]
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• Efficient nonlinear system identificationDavid Mansour. 432-435 [doi]
• Performance analysis of DPCM codecs operating on adaptively frequency mapped 7.6 khz speechCengiz Evci, P. J. Patrick. 433-436 [doi]
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• Peak-and-valley impulse excited LPC voiceY. Arai, Toshio Yagi, T. Miyagawa. 445-448 [doi]
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• An efficient Doppler compensated correlation and low doppler interference removalE. J. Modugno, M. L. Hampton, G. W. Johnson, D. E. Ohlms. 632-635 [doi]
• Envelope modelling of impulse response functions of systems having a high modal densityP. Davies, Joe K. Hammond. 634-637 [doi]
• Bias free estimation of the variance of time of arrival differencesP. A. Yansouni. 636-639 [doi]
• Band-limited signal extrapolation in the presence of noiseJorge L. C. Sanz, Thomas S. Huang. 638-641 [doi]
• A new method for the passive locating of unknown source by using a vertical dipole receiverChao-Huan Hou. 640-643 [doi]
• Spectra using data distribution and covariance modellingCostas E. Goutis, Richard M. Leahy, P. G. Cassidy. 642-645 [doi]
• Adaptive multipath delay estimationJulius O. Smith, Benjamin Friedlander. 644-647 [doi]
• Nonlinear filters for reducing spiky noise: 2-dimensionLong-bin Ling, Rui Yin, Xinghua Wang. 646-649 [doi]
• Adaptive realization of the phase transform for time delay estimationDae Hee Youn, Shen-Neng Chiou, V. John Mathews. 648-651 [doi]
• Experiments with extrapolation of band-limited signalM.-J. Tsai, D. A. O'Connor. 650-653 [doi]
• A comparative study of the LMS adaptive filter versus generalized correlation method for time delay estimationJeffrey L. Krolik, M. Joy, Subbarayan Pasupathy, Moshe Eizenman. 652-655 [doi]
• Relationship between Paley-Wiener theorem and the stationary phase method?M. E. Zakharia. 654-657 [doi]
• Motion parameter estimation of a fast moving sound source using the retardation effectJ. Schiller. 656-659 [doi]
• Difference operation for Pre/Post-emphasis in linear prediction analysisMasuzo Yanagida, Osamu Kakusho. 658-661 [doi]
• Digital signal processing capabilities of CUSP, a high performance bit-serial VLSI processorRichard W. Linderman, Peter P. Reusens, Paul M. Chau, Walter H. Ku. 660-663 [doi]
• A one-source photometric method for N-order specular surfacesSerge Castan, Jun Shen. 662-665 [doi]
• The SP16 signal processorGottfried Ungerboeck, Dietrich Maiwald, Hans-Peter Kaeser, Pierre R. Chevillat. 664-667 [doi]
• Two dimensional stochastic fractional AR models with strong periodicitiesRangasami L. Kashyap, Paul M. Lapsa. 666-669 [doi]
• SASP-A digital signal processor system for speech processing applicationsManfred Immendörfer, Dieter Kopp, Gebhard Thierer. 668-671 [doi]
• Image segmentation using spatial linear predictionM. D. Richard, Charles W. Therrien, Jae S. Lim. 670-673 [doi]
• A real time general purpose signal processorM. M. Jamali, Graham A. Jullien, William C. Miller, S. I. Ahmad. 672-675 [doi]
• Image segmentation using maximum entropy techniquesJay B. Jordan, Lonnie C. Ludeman. 674-677 [doi]
• A VLSI digital filter bankMatthew Yuschik, Hideaki Kobayashi. 676-679 [doi]
• Application of the Gibbs distribution to image segmentationHoward Elliott, Haluk Derin, Roberto Cristi, Donald Geman. 678-681 [doi]
• A high-speech 32 bit IEEE floating-point chip set for digital signal processingBob Woo, Lyon Lin, F. A. Ware. 680-683 [doi]
• Bayes smoothing algorithms for segmentation of images modeled by Markov random fieldsHaluk Derin, Howard Elliott, Roberto Cristi, Donald Geman. 682-685 [doi]
• A 16-bit three port arithmetic logic and shift unitJ. Nuttall, J. Oxxal. 684-687 [doi]
• A maximum-likelihood approach to image segmentation by textureJ. E. Bevington, Russell M. Mersereau. 686-689 [doi]
• VLSI Architecture for signal processing with alternate low-level primitive structures (ALPS)T. E. Curtis, A. G. Constantinides, Y. S. Wu. 688-691 [doi]
• Multi-level thresholding and its application to feature extraction in machine partsN. Rajendran, Maher A. Sid-Ahmed, James J. Soltis. 690-693 [doi]
• Adder-based digital signal processor architecture for 80 NS cycle timeAlois Rainer, Walter Ulbrich, L. Gazsi. 692-695 [doi]
• Classification of textures using Markov random field modelsRama Chellappa, Sangit Chatterjee. 694-697 [doi]
• Image vector quantization with a perceptually-based cell classifierBhaskar Ramamurthi, Allen Gersho. 698-701 [doi]
• Machine recognition of partial planar shapes using feature vectorsRichard A. Jones, Yogendra J. Tejwani. 702-705 [doi]
• Segmentation of computerized tomography images for three dimensional analysisSally L. Wood. 706-709 [doi]
• Bearing estimation in a shallow water environment using the eigenvector decomposition methodBodo Scholz, Wolfgang Kroll. 710-713 [doi]
• Decreasing high resolution method sensitivity by conventional beamformer preprocessingGeorges Bienvenu, Laurent Kopp. 714-717 [doi]
• An improved eigenvector beamformerTodd K. Citron, Thomas Kailath. 718-721 [doi]
• Matrix decomposition and multiple source locationAng-Zhao Di, Li-Sheng Tian. 722-725 [doi]
• New adaptive processor for coherent signals and interferenceTie-Jun Shan, Thomas Kailath. 726-729 [doi]
• Resolution of incoherent and coherent sources by autoregressive beamformingStanislav Kesler, Samson Boodaghians, Jelisaveta Kesler. 730-733 [doi]
• Sea surface reverberation rejectionWilliam S. Hodgkiss, Dimitrios Alexandrou. 734-737 [doi]
• Structure of covariance matrix and eigenvalues of broadband tapped delay line adaptive arrayKo C. Chung. 738-741 [doi]
• The effect of interference extent on LMS spatial cancellationFrances A. Reed, Curtis M. Flynn, Paul L. Feintuch. 742-745 [doi]
• Application of the maximum entropy beamformer to a shallow water line arrayWolfgang Kroll, Bodo Scholz. 746-749 [doi]
• Application of singular value decomposition to adaptive beamformingLeon H. Sibul. 750-753 [doi]
• Synthetic aperture sonar: An analysis of beamforming and system designEzio G. Pusone, Lewis J. Lloyd. 754-757 [doi]
• Degradation of angular resolution for eigenvector-eigenvalue (EVEV) high-resolution processors with inadequate estimation of noise coherenceG. E. Martin. 758-761 [doi]
• Systolic array technique applied to symmetric FIR filtersB. R. Mercy. 762-765 [doi]
• Concurrent array processor for fast eigenvalue computationsPing Ang, Martin Morf. 766-769 [doi]
• The application of a systolic least squares processing array to adaptive beamformingJohn G. McWhirter, C. R. Ward, Andrew J. Robson, P. J. Hargrave. 770-773 [doi]
• An FFT systolic processor and its applicationsT. Willey, Tariq S. Durrani, Roy Chapman. 774-777 [doi]
• VLSI architectures for dynamic time warping using systolic arraysFrancis Jutand, Nicolas Demassieux, D. Vicard, Gérard Chollet. 778-781 [doi]
• A three dimensional systolic array architecture for fast matrix multiplicationRichard W. Linderman, Walter H. Ku. 782-785 [doi]
• A manpack portable LPC 10 vocoderBruce Fette, Chaz Rimpo, Joseph Kish. 786-789 [doi]
• The VLSI design of a sub-band coderS. A. Townes, Trieu-Kien Truong. 790-793 [doi]
• Real time implimentation of 16kbs APC with Hybrid QuantizationJim W. Burgett, John S. Collura. 794-797 [doi]
• A speech input and processing board for a personal computerBertrand Denoix. 798-801 [doi]
• Architecture for a VLSI implementation of an LPC-based, isolated-word recognition systemB. P. Tao, M. Oijala. 802-805 [doi]