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ORCIDICASSP 1987: Dallas, Texas, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '87, Dallas, Texas, USA, April 6-9, 1987. IEEE 1987
- Haiyan Yé, Denis Tuffelli:
Deterministic characteristics of LPC distances: An inconsistency with perceptual evidence. 1-4 - Herman J. M. Steeneken:
Diagnostic information from subjective and objective intelligibility tests. 5-8 - H. Y. Wu, Pierre Badin, Yan Ming Cheng, Bernard Guérin:
Vocal tract simulation: Implementation of continuous variations of the length in a Kelly-Lochbaum model, effects of area function spatial sampling. 9-12 - Gunnar Ahlbom, Frédéric Bimbot, Gérard Chollet:
Modeling spectral speech transitions using temporal decomposition techniques. 13-16 - L. Thomas Ramsey, David Gribble:
Information-theoretic compressibility of speech data. 17-20 - Yuval Bistritz, Hanoch Lev-Ari, Thomas Kailath:
Complexity reduced lattice filters for digital speech processing. 21-24 - Yariv Ephraim, Amir Dembo, Lawrence R. Rabiner:
A minimum discrimination information approach for hidden Markov modeling. 25-28 - Mariano García Otero
, José R. Casar Corredera:
On the distributions of small-sample estimates of second-order AR process parcor coefficients. 29-32 - Hsien-sen Hung, Mostafa Kaveh:
On the statistical sufficiency of the coherently averaged covariance matrix for the estimation of the parameters of wideband sources. 33-36 - Don H. Johnson, P. Srinivasa Rao:
Properties and generation of non-Gaussian time series. 37-40 - Christophe P. Rialan, Louis L. Scharf:
Fast algorithms for QR and Cholesky factors of Toeplitz operators. 41-44 - Steven Kay, Debasis Sengupta:
Statistically/Computationally efficient estimation of non-Gaussian autoregressive processes. 45-48 - Trung T. Pham, Rui J. P. deFigueiredo:
Maximum likelihood estimation of a class of non-Gaussian densities with application to deconvolution. 49-52 - Kevin M. Buckley:
Incorporated robustness in narrow-band signal subspace spatial spectral estimators. 53-56 - Ronald R. Mohler, Z. Tang:
On identification of non-Gaussian time series. 57-60 - Georgios B. Giannakis, Jerry M. Mendel, W. Wang:
ARMA Modeling using cumulant and autocorrelation statistics. 61-64 - Hengjie Ma:
The four tones recognition of continuous Chinese speech. 65-68 - Hermann Ney:
Dynamic programming speech recognition using a context-free grammar. 69-72 - Salim E. Roucos, Mari O. Dunham:
A stochastic segment model for phoneme-based continuous speech recognition. 73-76 - Torbjørn Svendsen, Frank K. Soong:
On the automatic segmentation of speech signals. 77-80 - Paolo D'Orta, Marco Ferretti, Stefano Scarci:
Phoneme classification for real time speech recognition of Italian. 81-84 - Aaron E. Rosenberg, Anna Maria Colla:
A connected speech recognition system based on spotting diphone-like segments-Preliminary results. 85-88 - Yen-Lu Chow, Mari O. Dunham, Owen Kimball, Michael A. Krasner, Francis Kubala, John Makhoul, Patti J. Price, Salim E. Roucos, Richard M. Schwartz:
BYBLOS: The BBN continuous speech recognition system. 89-92 - Stephen E. Levinson:
Continuous speech recognition by means of acoustic/ Phonetic classification obtained from a hidden Markov model. 93-96 - Harvey F. Silverman, David P. Morgan, Susan H. Miller:
An early-decision, real-time, connected-speech recognizer. 97-100 - Lawrence R. Rabiner, Jay G. Wilpon, Biing-Hwang Juang:
A performance evaluation of a connected digit recognizer. 101-104 - Lloyd J. Griffiths, Michael J. Rude:
Adaptive filtering without a desired signal. 105-108 - Neil J. Bershad, Lian Zuo Qu:
"On the probability density function of the LMS adaptive filter weights". 109-112 - Miguel Angel Lagunas, Francesc Vallverdú, M. E. Santamaria:
Non-linear adaptive signal processor. 113-116 - Ari Nieminen, Pekka Heinonen, Yrjö Neuvo:
Suppression and detection of impulse type interference using adaptive median hybrid filters. 117-120 - Michael P. Beddoes, L. Panych, Juan Qian, Juhn A. Wada:
A criticism of the parametric EEG spike detector. 121-124 - Guy R. L. Sohie, Saud A. Alshibani:
Misadjustment expressions for infinite impulse response adaptive filters. 125-128 - Kun Tang, Charles E. Rohrs:
The use of large adaptation gains to remove the SPR condition from recursive adaptive algorithms. 129-132 - William A. Sethares, C. Richard Johnson Jr.:
Exciting conditions for quantized state adaptive algorithms. 133-136 - Ganapati Panda, C. F. N. Cowan, Peter M. Grant:
Assessment of finite precision limitations in LMS and BLMS adaptive algorithms. 137-140 - Karlheinz Brandenburg:
OCF-A new coding algorithm for high quality sound signals. 141-144 - Philip Arthur Nelson, J. K. Hammond, Stephen J. Elliott:
Causal constraints on the active control of sound. 145-148 - Shigeaki Aoki, Nobuo Koizumi:
Expansion of listening area with good localization in audio conferencing. 149-152 - John T. Lynch:
"A methodology for evaluating the performance of dynamic range control algorithms for speech enhancement". 153-156 - Antoine J. Chaigne:
Spectral distribution and damping factors measurements of musical strings using FFT techniques. 157-160 - Thierry Hervé, Jean Marc Dolmazon, Jacques Demongeot:
Neural network in the auditory system: Influence of the temporal context on the response represented by a random field. 161-164 - Stuart J. Flockton:
Cancellation of acoustic noise in a pipe using digital adaptive filters. 165-168 - Jeffrey J. Rodriguez, Jae S. Lim, Elliot Singer:
Adaptive noise reduction in aircraft communication systems. 169-172 - G. A. Powell, P. Darlington, P. D. Wheeler:
Practical adaptive noise reduction in the aircraft cockpit environment. 173-176 - Kuldip K. Paliwal, Anjan Basu:
A speech enhancement method based on Kalman filtering. 177-180 - D. G. Childers, C. K. Lee:
Co-Channel speech separation. 181-184 - H. Liang, N. Malik:
Reducing cocktail party noise by adaptive array filtering. 185-188 - John H. L. Hansen, Mark A. Clements:
Iterative speech enhancement with spectral constraints. 189-192 - Dale E. Veeneman, Baruch Mazor:
Enhancement of block-coded speech. 193-196 - David L. Thomson:
A multivariate voicing decision rule adapts to noise, distortion, and spectral shaping. 197-200 - Meir Feder, Alan V. Oppenheim, Ehud Weinstein:
Methods for noise cancellation based on the EM algorithm. 201-204 - J. A. Naylor, Steven F. Boll:
Techniques for suppression of an interfering talker in co-channel speech. 205-208 - Kazuo Toraichi, Iwao Sekita, Ryoichi Mori:
An algorithm of signal approximation by hybrid spline. 209-212 - Iwao Sekita, Kazuo Toraichi, Masaru Kamada, Ryoichi Mori, Kazuhiko Yamamoto, Hiromitsu Yamada:
Feature extraction by spline function for relaxation matching. 213-216 - Lalit Gupta, Mandyam D. Srinath:
Invariant planar shape recognition using dynamic alignment. 217-220 - Licia Capodiferro, Roberto Cusani, Giovanni Jacovitti, M. Vascotto:
A correlation based technique for shift, scale, and rotation independent object identification. 221-224 - Hans J. Dohse, Jorge L. C. Sanz, Anil K. Jain:
Object classification and registration by Radon transform based invariants. 225-228 - Jianming Song, Paul Gaillard:
Automatic recognition of isolated or occluded planar objects by a two steps processing. 229-232 - Eric B. Hinkle, Jorge L. C. Sanz:
Fast image segmentation for some machine vision applications. 233-236 - Amlan Kundu:
Lemniscate transform: A new efficient technique for shape coding and representation. 237-240 - Petros Maragos:
Pattern spectrum of images and morphological shape-size complexity. 241-244 - T. R. Esselman, Jacques G. Verly:
Some applications of mathematical morphology to range imagery. 245-248 - J. E. Bevington, Russell M. Mersereau:
Differential operator based edge and line detection. 249-252 - J. A. Reimer, P. D. Lawrence:
Characterizing ∇2G filtered images by their zero crossings. 253-256 - A. Ikonomopoulos:
Aspects of directional filtering and applications in image processing. 257-260 - Yu-Ning Dong, Zhen-Ya He:
A method of acquiring 3-D data of an object from stereo images. 261-264 - B. K. Miller, R. A. Jones:
A concavity based algorithm for the recognition of partially occulded 3-dimensional objects. 265-268 - David L. Wang:
Knowledge based object detection using SAR images. 269-272 - Makoto Sato, Toshikazu Wada, Hiroshi Kawarada:
A hierarchical representation of random waveforms by scale-space filtering. 273-276 - Scott W. Shaw, Rui J. P. deFigueiredo:
Structural processing of waveforms as trees. 277-280 - N. A. Chalabi, Tariq S. Durrani:
A new algorithm for estimating optic flow for low-level vision systems. 281-284 - Joseph S. P. Shu:
A new heuristic edge extraction technique. 285-288 - Chin-Hui Lee:
Robust linear prediction for speech analysis. 289-292 - D. G. Childers, Ke Wu, D. M. Hicks:
Factors in voice quality: Acoustic features related to gender. 293-296 - Kuldip K. Paliwal:
Estimation of noise variance from the noisy AR signal and its application in speech enhancement. 297-300 - B. Yegnanarayana, S. Tanveer Fathima, Hema A. Murthy:
Reconstruction from Fourier transform phase with applications to speech analysis. 301-304 - A. A. Wrench, C. F. N. Cowan:
A new approach to noise-robust LPC. 305-307 - Juergen Schroeter, Jerry N. Larar, M. Mohan Sondhi:
Speech parameter estimation using a vocal tract/Cord model. 308-311 - Tetsuya Harada, Hiroshi Kawarada:
High resolution frequency analysis of voices-Feature extraction of nasal consonants. 312-315 - Harprit S. Chhatwal, Anthony G. Constantinides:
Speech spectral segmentation for spectral estimation and formant modelling. 316-319 - Amro El-Jaroudi, John Makhoul:
Discrete all-pole modeling for voiced speech. 320-323 - Jitendra K. Tugnait:
Fitting noncausal autoregressive signal plus noise models to noisy non-Gaussian linear processes. 324-327 - Shiping Li, Bradley W. Dickinson:
Jump detection and fast parameter tracking for piecewise AR processes using adaptive lattice filters. 328-331 - Ahmed S. Abutaleb:
Adaptive line enhancement using a random AR model. 332-335 - S. Thomas Alexander, Zong M. Rhee:
An analysis of finite precision effects for the autocorrelation method and Burg's method of linear prediction. 336-339 - William J. Vetter, Milton J. Porsani:
Extended matrix formulation for the Marple algorithm. 340-343 - Sverre Holm:
Objective methods for comparing autoregressive order-determining criteria. 344-347 - Bernard Picinbono, Jean-Marc Kerilis:
Some properties of prediction and interpolation errors. 348-351 - Paul F. Fougere:
On the extreme accuracy of maximum entropy spectrum estimation from an error-free autocorrelation function. 352-355 - Vijay K. Jain, B. L. Xu:
Autocorrelation distortion function for improved AR modeling. 356-359 - Anne-Marie Derouault:
Context-dependent phonetic Markov models for large vocabulary speech recognition. 360-363 - Bernard Mérialdo:
Speech recognition with very large size dictionary. 364-367 - Pierre Alinat, Evelyne Gallais, Jean-Paul Haton, Jean-Marie Pierrel, Pascal Richard:
A continuous speech dialog system for the oral control of a sonar console. 368-371 - Diane Kewley-Port, Charles S. Watson, Daniel Maki, Daniel Reed:
Speaker-dependent speech recognition as the basis for a speech training aid. 372-375 - Alexander I. Rudnicky, Lynn K. Baumeister, Kevin H. DeGraaf, Eric Lehmann:
The lexical access component of the CMU continuous speech recognition system. 376-379 - Richard M. Stern, Wayne H. Ward, Alexander G. Hauptmann, Juan Leon:
Sentence parsing with weak grammatical constraints. 380-383 - Christian J. Wellekens:
Explicit time correlation in hidden Markov models for speech recognition. 384-386 - Thérèse Martelli, Laurent Miclet, Jean-Pierre Tubach:
REMORA A software architecture for the collaboration of different knowledge sources in phonetic decoding of continuous speech. 387-390 - Otto Schmidbauer:
Syllable-based segment-hypotheses generation in fluently spoken speech using gross articulatory features. 391-394 - Hanoch Lev-Ari, K.-F. Chiang, Thomas Kailath:
On the stability of adaptive lattice filters. 395-398 - P. Darlington, Stephen J. Elliott:
Stability of adaptively controlled systems-A graphical approach. 399-402 - Jean-Luc Botto:
Stabilization of fast recursive least-squares transversal filters for adaptive filtering. 403-406 - John M. Cioffi:
A fast QR/Frequency-domain RLS adaptive filter. 407-410 - Jae C. Lee, Sanjit K. Mitra:
On frequency-domain least squares adaptive algorithms. 411-414 - Hanoch Lev-Ari, John M. Cioffi, Thomas Kailath:
Continuous-time least-squares fast transversal filters. 415-418 - Dirk T. M. Slock, John M. Cioffi, Thomas Kailath:
A fast transversal filter for adaptive line enhancement. 419-422 - S. Thomas Alexander, C. T. Pan, Robert J. Plemmons:
Numerical properties of a hyperbolic rotation method for windowed RLS filtering. 423-426 - José Manuel Páez-Borrallo, Aníbal R. Figueiras-Vidal, Luis Vergara-Dominguez:
Analysis of LK frequency-adaptive transversal filters in plant identification. 427-430 - Guozhu Long, Fuyun Ling, John G. Proakis:
Adaptive transversal filters with delayed coefficient adaptation. 431-434 - Carl J. Wenk, Harold F. Jarvis Jr.:
Design of continuous gain adaptive αβ trackers for passive sonar application. 435-438 - G. W. Johnson, E. J. Modugno:
True angle estimation from a line array using time-delay estimates over a known rotation. 439-442 - Alfred O. Hero III:
Applications of error intensity measures to bearing estimation. 443-446 - John P. Ianniello:
High resolution multipath time delay estimation for broadband random signals. 447-450 - Ivars P. Kirsteins:
High resolution time delay estimation. 451-454 - M. A. Pallas, Nadine Martin, J. Martin:
Time delay estimation by autoregressive modelization. 455-458 - Miriam Hamilton, Peter M. Schultheiss:
Passive ranging in a multipath dominated environment. 459-462 - Amnon Shefi, Charles W. Therrien, Donald E. Kirk, Rigoberto Saez, Benjamin Friedlander:
Passive multipath target tracking in inhomogeneous acoustic medium. 463-466 - Roger J. Tremblay, G. Clifford Carter, Dean W. Lytle:
A practical approach to the estimation of amplitude and time delay parameters of a composite signal in non-white Gaussian noise. 467-470 - Jonathan S. Abel, Julius O. Smith III:
The spherical interpolation method for closed-form passive source localization using range difference measurements. 471-474 - Magdy A. Bayoumi:
A quadratic residue processor for complex DSP applications. 475-478 - James D. Beatty, Richard D. Calder Jr., Perry Farazi, Daniel P. Kelly, James L. Melsa:
Custom VLSI design of a single chip multi-channel ADPCM processor. 479-482 - Paul M. Chau, Kay-Cheung Chew, Walter H. Ku:
A bit-serial floating-point complex multiplier-accumulator for fault-tolerant digital signal processing arrays. 483-486 - Y. S. Cheung, S. C. Leung:
A second generation Silicon compiler for bit-serial signal processing architecture. 487-490 - Brian C. Richards, Alex Sherstinsky, Robert W. Brodersen:
A parameterized VLSI video-rate histogram processor. 491-494 - Steven G. Smith, Peter B. Denyer, David S. Renshaw, K. Asada, K. P. Coplan, M. Keightley, Javeed I. Mhar:
Full-span structural compilation of DSP hardware. 495-498 - Jun-ichi Takahashi, Takashi Kimura, Shigetatsu Hamaguchi, Naotaka Omiya:
A flexible linear array oriented VLSI processor for continuous speech recognition. 499-502 - Fred J. Taylor:
A reconfigurable binary/RNS/LNS architecture for DSP. 503-506 - Fathy F. Yassa, Steven G. Kratzer:
A VLSI implementation of the partial rank algorithm for adaptive signal processing. 507-510 - Luis Bonet, Tim A. Williams:
A split control store VLSI for 32 kbps ADPCM transcoding. 511-514 - David E. Borth, Ira A. Gerson, John R. Haug:
A cascadable adaptive FIR filter VLSI IC. 515-518 - Stephen C. Glinski, T. Mariano Lalumia, Daniel R. Cassiday, Taiho Koh, Christine M. Gerveshi, Gene A. Wilson, Jitendra Kumar:
The graph search machine (GSM): A programmable processor for connected word speech recognition and other applications. 519-522 - Kevin L. Kloker:
The architecture and applications of the motorola DSP56000 digital signal processor family. 523-526 - Robert E. Owen:
A 15 nanosecond complex multiplier-accumulator for FFTS. 527-530 - David M. Taylor, Rafi Retter:
A novel VLSI digital signal processor architecture for high-speed vector and transform operations. 531-534 - Ray Simar Jr., Tony Leigh, Peter Koeppen, Jerald Leach, Jim Potts, Denise Blalock:
A 40 MFLOPS digital signal processor: The first supercomputer on a chip. 535-538 - Steven G. Smith, Peter B. Denyer:
Serial/Parallel architectures for area-efficient vector multiplication. 539-542 - Steven G. Smith, M. S. McGregor, Peter B. Denyer:
Techniques to increase the computational throughput of bit-serial architectures. 543-546 - Nicolas Demassieux, Gilles Concordel, Jean-Pierre Durandeau, Francis Jutand:
An optimized VLSI architecture for a multiformat discrete cosine transform. 547-550 - M. El-Gabali, Malayappan Shridhar, Majid Ahmadi:
Segmentation of noisy images modelled by Markov random fields with Gibbs distribution. 551-554 - Yi-Tong Zhou, Rama Chellappa:
Linear feature extraction based on an AR model edge detector. 555-558 - Keith D. Hartt, Patrick A. Kelly, Haluk Derin:
The modeling and segmentation of speckled images. 559-562 - Chee Sun Won, Haluk Derin:
Segmentation of noisy textured images using simulated annealing. 563-566 - John Goutsias, Jerry M. Mendel:
Semi-Markov random field models for image segmentation. 567-570 - Marianna Clark, Alan C. Bovik, Wilson S. Geisler:
Texture segmentation using a class of narrowband filters. 571-574 - Hidefumi Kobatake, Toshihiro Watanabe:
Texture recognition using final-prediction-error criterion. 575-578 - Dong-Seok Jeong, Paul M. Lapsa:
Unified approach for low level image analysis. 579-582 - Mohammad V. Malakooti, Keith A. Teague:
CARMA Model method of two-dimensional shape classification: An eigensystem approach vs. the LP norm. 583-586 - Peter L. Chu:
Efficient CFAR detection of line segments in a 2-D image. 587-590 - Boaz Porat, Benjamin Friedlander:
A frequency domain approach to multiframe detection and estimation of dim targets. 591-594 - Gong Wei, Zhen-Ya He:
A new algorithm for maximum entropy image reconstruction. 595-597 - Tariq S. Durrani, A. Rauf, K. Boyle, Franco Lotti, Stefano Baronti:
Thermal imaging techniques for the non destructive inspection of composite materials in real time. 598-601 - Hong Jeong, Bruce R. Musicus:
Mask extraction from optical images of VLSI circuits. 602-605 - Mark W. Merritt:
Spatial and temporal analysis of weather radar reflectivity images. 606-609 - Hidefumi Kobatake, Kunio Oh'ishi, Juichi Miyamichi:
Automatic diagnosis of pneumoconiosis by texture analysis of chest X-ray images. 610-613 - Hidefumi Kobatake, Yoshiaki Inoue, Tatsuro Namai, Nobuhiro Hamba:
Measurement of two-dimensional movement of traffic by image processing. 614-617 - Chen Su-xian, Li Xiao-song:
The automatic classification of the welding defects. 618-621 - Amira M. Badreldin:
Real-time analysis of fuel spray images. 622-624 - Frank K. Soong, M. Mohan Sondhi:
A frequency-weighted Itakura spectral distortion measure and its application to speech recognition in noise. 625-628 - Roman Kuc, Hee Han:
Errors in determining vocal tract shape from the acoustic signal. 629-632 - Richard M. Schwartz, Yen-Lu Chow, Francis Kubala:
Rapid speaker adaptation using a probabilistic spectral mapping. 633-636 - Hiroya Fujisaki, Mats Ljungqvist:
Estimation of voice source and vocal tract parameters based on ARMA analysis and a model for the Glottal source waveform. 637-640 - Ho-Ping Tseng, Michael J. Sabin, Edward A. Lee:
Fuzzy vector quantazation applied to hidden Markov modeling. 641-644 - Sarangarajan Parthasarathy, Donald W. Tufts:
Signal modeling by exponential segments and application in voiced speech analysis. 645-648 - Thomas F. Quatieri, Robert J. McAulay:
Mixed-phase deconvolution of speech based on a sine-wave model. 649-652 - John R. Deller Jr., T. C. Luk:
Set-membership theory applied to linear prediction analysis of speech. 653-656 - Michael D. Riley:
Beyond quasi-stationarity: Designing time-frequency representations for speech signals. 657-660 - Taikang Ning, Chrysostomos L. Nikias:
Power spectrum estimation with uncertainty in the sample location of correlation measurements. 661-664 - Benjamin Friedlander, Boaz Porat:
An accuracy analysis of the Kumaresan-Tufts method for estimating complex damped exponentials. 665-668 - Guy R. L. Sohie, Arun Mirchandani:
Accurate estimation of closely spaced, real, decaying exponentials in noise. 669-672 - James A. Cadzow:
Signal enhancement using canonical projection operators. 673-676 - M. J. E. Salami, S. T. Nichols, Michael R. Smith:
A SVD-based transient error method for analyzing noisy multicomponent exponential signals. 677-680 - Douglas L. Jones, Thomas W. Parks:
A high resolution data-adaptive time-frequency representation. 681-684 - Franz Hlawatsch, Werner Krattenthaler:
Time-frequency signal synthesis on signal subspaces. 685-688 - Yih-Chyun Jenq:
Digital spectra of non-uniformly sampled signals with applications to digitally synthesized sinusoids. 689-692 - Frank K. Soong:
A training procedure for a segment-based-network approach to isolated word recognition. 693-696 - Vishwa Gupta, Matthew Lennig, Paul Mermelstein:
Integration of acoustic information in a large vocabulary word recognizer. 697-700 - Amir Averbuch, Lalit R. Bahl, Raimo Bakis, Peter F. Brown, Gregg Daggett, S. Das, Ken Davies, Steven V. De Gennaro, Peter V. de Souza, E. Epstein, D. Fraleigh, Frederick Jelinek, B. Lewis, Robert L. Mercer, J. Moorhead, Arthur Nádas, David Nahamoo, Michael Picheny, G. Shichman, P. Spinelli, Dirk Van Compernolle, H. Wilkens:
Experiments with the Tangora 20, 000 word speech recognizer. 701-704 - Richard P. Lippmann, Edward A. Martin, Douglas B. Paul:
Multi-style training for robust isolated-word speech recognition. 705-708 - Edward A. Martin, Richard P. Lippmann, Douglas B. Paul:
Two-stage discriminant analysis for improved isolated-word recognition. 709-712 - Douglas B. Paul:
A speaker-stress resistant HMM isolated word recognizer. 713-716 - Yeunung Chen:
Cepstral domain stress compensation for robust speech recogniton. 717-720 - A. Maheswaran, Robert E. Bogner:
A new time-scale warping algorithm and associated modules for single dimensional and multidimensional speech parameter contours. 721-724 - William Equitz:
Fast algorithms for vector quantization picture coding. 725-728 - Mary E. Blain, Thomas R. Fischer:
Optimum rate allocation in pyramid vector quantizer transform coding of imagery. 729-732 - V. John Mathews, Randall W. Waite, Thao Duy Tran:
Image compression using vector quantization of linear (one- step) prediction errors. 733-736 - Chi Hau Chen:
Laplacian pyramid image data compression. 737-739 - J. Barrilleaux, R. Hinkle, S. Wells:
Efficient vector quantization for color image encoding. 740-743 - Anh Tran, Kwun-Min Liu, Kou-Hu Tzou, Eileen B. Vogel:
An efficient pyramid image coding system. 744-747 - K. S. Thyagarajan, Mahesh Viswanathan:
Low bit-rate image coding techniques. 748-751 - K. S. Thyagarajan, Helge Bohlmann, Hüseyin Abut:
Image coding based on segmentation using region growing. 752-755 - Hüseyin Abut, Bertram P. M. Tao, Jack L. Smith:
Vector quantizer architectures for speech and image coding. 756-759 - Richard L. Baker, Hsiao-hui Shen:
A finite-state vector quantizer for low-rate image sequence coding. 760-763 - Satoshi Horiike, Shogo Nishida, Toshiaki Sakaguchi:
A design method of systolic arrays under the constraint of the number of the processors. 764-767 - Uwe Schwiegelshohn, Lothar Thiele:
A systolic algorithm for cyclic-by-rows SVD. 768-770 - Majid Taheri, Graham A. Jullien, William C. Miller:
Systolic ROM arrays for implementing RNS FIR filters. 771-774 - Teiji Emori, Masatoshi Tachibana:
A speech feature extraction system with a linear processor array. 775-778 - Ali H. Abdallah, Yu Hen Hu:
Parallel VLSI computing array implementation for signal subspace updating algorithm. 779-782 - George M. Papadourakis, Fred J. Taylor:
Implementation of Kalman filters using systolic arrays. 783-786 - P. A. Ramamoorthy, Brahmaji Potu:
Bit serial systolic chip set for real-time image coding. 787-790 - Uwe Schwiegelshohn, Lothar Thiele:
One- and two-dimensional systolic arrays for least-squares problems. 791-794 - Nagaraja Srinivasa, Kasi Rajgopal, K. R. Ramakrishnan:
On a programmable signal processor for VLSI. 795-796 - Xi-Xian Chen, Changnian Cai, Peng Guo, Ying Sun:
A hidden Markov model applied to Chinese four-tone recognition. 797-800 - Susan M. Miller, David P. Morgan, Harvey F. Silverman, Michael N. Karam, N. Rex Dixon:
A real-time evaluation system for a real-time connected-speech recognizer. 801-804 - Toshiaki Tsuboi, Akihiro Tomihisa, Noboru Sugamura:
Japanese linguistic processing for continuous speech recognition. 805-808 - Pietro Laface, Giorgio Micca, Roberto Pieraccini:
Experimental results on a large lexicon access task. 809-812 - Melvyn J. Hunt, Claude Lefèbvre:
Speech recognition using an auditory model with pitch-synchronous analysis. 813-816 - David M. Lubensky:
Continuous digit recognition using coarse phonetic segmentation. 817-820 - Jay G. Wilpon, Biing-Hwang Juang, Lawrence R. Rabiner:
An investigation on the use of acoustic sub-word units for automatic speech recognition. 821-824 - Shinta Kimura, Yasuhiro Nara:
Extraction of phonemic variation rules in continuous speech spoken by multiple speakers. 825-828 - Sei-ichi Nakagawa:
Spoken sentence recognition by time-synchronous parsing algorithm of context-free grammar. 829-832 - Hermann Ney, Dieter Mergel, Andreas Noll, Annedore Paeseler:
A data-driven organization of the dynamic programming beam search for continuous speech recognition. 833-836 - Hy Murveit, Mitchel Weintraub, Michael Cohen, Jared Bernstein, Alex Rudnicky:
Lexical access with lattice input. 837-840 - Paolo D'Orta, Marco Ferretti, Alex Martelli, Sergio Melecrinis, Stefano Scarci, Giampiero Volpi:
A speech recognition system for the Italian language. 841-843 - Dieter Mergel, Annedore Paeseler:
Construction of language models for spoken database queries. 844-847 - Kaichiro Hatazaki, Takao Watanabe:
Large vocabulary word detection by searching in a tree-structural word dictionary. 848-851 - Owen Kimball, Lynn Cosell, Richard M. Schwartz, Michael A. Krasner:
Efficient implementation of continuous speech recognition on a large scale parallel processor. 852-855 - Alex Waibel:
Prosodic knowledge sources for word hypothesization in a continuous speech recognition system. 856-859 - Meg Withgott, Steven C. Bagley, Richard F. Lyon, Marcia A. Bush:
Acoustic-phonetic segment classification and scale-space filtering. 860-863 - Sarah K. Yoder, Leah H. Jamieson:
Speaker-independent recognition of stop consonants. 864-867 - Jean-Paul Haton, Noëlle Carbonell, Dominique Fohr, Jean-François Mari, Abdelaziz Kriouille:
Interaction between stochastic modeling and knowledge-based techniques in acoustic-phonetic decoding of speech. 868-871 - Klaus P. Preuss:
A novel approach for complex Chebyshev-approximation with FIR filters using the Remez exchange algorithm. 872-875 - Hans Wilhelm Schüßler, J. Weith:
On the design of recursive Hilbert-transformers. 876-879 - Tamal Bose, David P. Brown:
On the stability of linear shift variant digital filters. 880-883 - Paul P. N. Yang, Moon S. Song, M. J. Narasimha:
On the design of optimal narrowband linear and minimum phase FIR filters. 884-887 - Alfred T. Johnson Jr.:
Optimal window-transforms for FIR digital filter design. 888-891 - Ramasamy Krishnan, Graham A. Jullien, William C. Miller:
Implementation of the generalized FIR filter structure using the residue arithmetic. 892-895 - Mark F. Pyfer, Rashid Ansari:
The design and application of optimal FIR fractional-slope phase filters. 896-899 - Vic Hansen:
Design of a multistage decimation-interpolation filter. 900-903 - Erich Auer:
Digital filter structures free of limit cycles. 904-907 - Sridha Sridharan:
Implementation of state-space digital filter structures using block floating-point arithmetic. 908-911 - Federico Kuhlmann, James A. Bucklew:
Piecewise uniform vector quantizers. 912-915 - David C. Farden, Jerome R. Bellegarda:
A fast algorithm for the recursive design of linear phase filters. 916-919 - K. S. Arun, M. Reuter:
Hankel approximation methods of IIR filter design. 920-923 - Dennis R. Morgan:
Finite limiting effects for a band-limited Gaussian random process with applications to A/D conversion. 924-927 - Frederick A. Williams:
Phase noise in quantized sine waves. 928-931 - A. A. (Louis) Beex, Victor E. DeBrunner:
Direct form sensitivity reduction by order increase. 932-935 - K. P. Prasad, P. Sathyanarayana:
A new scaling procedure for cascade digital filters. 936-939 - Susanna Ragazzini, Lucio Prina Ricotti, Gianni Orlandi, Giuseppe Martinelli:
Elimination of the pitch bias in the non-stationary characterization of speech. 940-943 - Shankar S. Narayan, John Parker Burg:
Spectral estimation of quasi-periodic data. 944-947 - Jean-Sylvain Liénard:
Speech analysis and reconstruction using short-time, elementary waveforms. 948-951 - Nobuhiro Miki, Kunitoshi Motoki, Nobuo Nagai:
A lattice filter model with accurate lip impedance for dynamic articuratory movement. 952-955 - Pekka Heinonen, Yrjö Neuvo:
Median type filters with linear predictive substructures. 956-959 - K. Field, Alan Derr, Lynn Cosell, C. Henry, Michael A. Krasner, J. Tiao:
A single board multirate APC speech coding terminal. 960-963 - Richard V. Cox:
A family of ADPCM coders implemented on real-time hardware. 964-967 - A. Fukui, K. Shibagaki:
Implementation of a multi-pulse speech codec with pitch prediction on a single chip floating-point signal processor. 968-971 - L. Robert Morris, Peter Barszczewski, Jonathan Bosloy:
Algorithm selection and software time/Space optimization for a DSP micro-based speech processor for a multi-electrode cochlear implant. 972-975 - Allen L. Gorin, Richard R. Shively:
The ASPEN parallel computer, speech recognition and parallel dynamic programming. 976-979 - Chrysostomos L. Nikias, Renlong Pan:
Nonminimum phase system identification via cepstrum modeling of higher-order cumulants. 980-983 - Erlendur Karlsson, Monson H. Hayes:
Performance analysis of new least squares ARMA lattice modeling algorithms. 984-987 - Neil J. Grossbard, John M. Retterer:
Estimating decay rates of single-frequency causal AR filters using a "Decimation" method. 988-990 - Surendra Prasad, Shiv Dutt Joshi:
Exact recursive least squares algorithms for ARMA modeling. 991-994 - John Makhoul, Allan O. Steinhardt:
On matching correlation sequences by parametric spectral models. 995-998 - Kambiz Heidarian, Yu Hen Hu:
A principal component approach for adaptive ARMA model identification. 999-1002 - Arye Nehorai, Petre Stoica:
Adaptive algorithms for constrained ARMA signals in the presence of noise. 1003-1006 - Yih-Fang Huang, Ashok K. Rao:
Application of a recursive estimation algorithm with information-dependent updating to ARMAX models and ARMA models with unknown inputs. 1007-1010 - L. Vergara-Domínguez, Aníbal R. Figueiras-Vidal:
A new class of high-order Yule-Walker estimates. 1011-1014 - Gagan Mirchandani, Gerald A. McGuire:
Performance characteristics and speed-up rates of the NEC mPD7281 data flow processor in parallel processing. 1015-1018 - Alastair D. McAulay, Eric A. Parsons:
Performance of Schur's algorithm on an optically connected multiprocessor. 1019-1022 - Gordon L. DeMuth:
Multirate process scheduling and synchronization in distributed signal processors. 1023-1026 - Korina Kassapoglou, Martin Vetterli:
Computer aided implementation of complex algorithms on DSP's using automatic scaling. 1027-1030 - JoEllen Wilbur, Fred J. Taylor:
High-speed Wigner processing based on a single modulus quadratic residue numbering system. 1031-1034 - Mohsin M. Jamali, M. M. Hussain, Graham A. Jullien:
A signal processing cell architecture. 1035-1038 - B. R. Upadhyaya, O. Glöckler, F. P. Wolvaardt:
Combined dynamic data analysis and process variable prediction approach for system fault detection. 1039-1042 - Rivka Shenhav:
The decomposition of long FFT's for high throughput implementation. 1043-1046 - Gregory Y. Tang, Brian K. Lien:
A multiple microprocessor system for general DSP operation. 1047-1050 - Jacques Vaisey, Allen Gersho:
Variable block-size image coding. 1051-1054 - Bernard Hammer, Achim von Brandt, M. Schielein:
Hierarchical encoding of image sequences using multistage vector quantization. 1055-1058 - Chia-Lung Yeh:
Color image-sequence compression using adaptive binary-tree vector quantization with codebook replenishment. 1059-1062 - Atul Puri, H.-M. Hang, Donald L. Schilling:
An efficient block-matching algorithm for motion-compensated coding. 1063-1066 - Hyung Hwa Ko, Choong Woong Lee:
Real time implementation of block truncation coding for picture data compression. 1067-1070 - Michael Maragoudakis, Jerry D. Gibson:
Experiments on video teleconferencing algorithms at 56 kilobits/sec. 1071-1074 - Bodo Braun:
Luminance adaptive chrominance coding. 1075-1078 - Giovanni Ramponi, Giovanni L. Sicuranza, Silvio Cucchi:
2- and 3-D Nonlinear predictors. 1079-1082 - Caspar Horne, Ed F. Deprettere:
Multi-pulse and regular-pulse LP coding of images. 1083-1086 - Steven Kay, Debasis Sengupta:
Optimal detection in colored non-Gaussian noise with unknown parameters. 1087-1090 - Michel Bouvet, Bernard Picinbono:
Minimax robust receiver in coloured noise for local deflection. 1091-1094 - G. W. Johnson, W. A. Bradford:
Thresholds in combined detection and source motion estimation. 1095-1098 - Hua Lee, Thomas S. Huang:
Three-dimensional motion estimation by synthetic aperture underwater acoustic systems. 1099-1102 - Gregory E. Bottomley:
The effects of cross-correlated noise and multi-channel signal on ORing loss. 1103-1106 - Hans P. Widmer, John C. Stapleton, Pierre Lafrance:
Optimal sequences for detection using a matched filter binary integrator. 1107-1110 - Q. T. Zhang, Patrick C. Yip:
On the use of X2-test in signal detection. 1111-1114 - Georgios B. Giannakis, Jerry M. Mendel, Xiaofeng Zhao:
A fast prediction-error detector for estimating sparse-spike sequences. 1115-1118 - Surendra Prasad, Ronald T. Williams, A. K. Mahalanabis, Leon H. Sibul:
A transform based covariance differencing approach to bearing estimation. 1119-1122 - Enrico Bocchieri, George R. Doddington:
Statistical features versus word templates for speaker independent digit recognition over long distance telephone connections. 1123-1126 - Sadaoki Furui:
A VQ-based preprocessor using cepstral dynamic features for large vocabulary word recognition. 1127-1130 - Eric P. Loeb, Richard F. Lyon:
Experiments in isolated digit recognition with a cochlear model. 1131-1134 - Y. S. Cheung, S. T. Leung:
Speaker-independent isolated word recognition using word-based vector quantization and hidden Markov models. 1135-1138 - David B. Roe:
Speech recognition with a noise-adapting codebook. 1139-1142 - Dirk Van Compernolle:
Increased noise immunity in large vocabulary speech recognition with the aid of spectral subtraction. 1143-1146 - Mark A. Clements, Sungjae Lim:
Hidden Markov model speech recognition based on Kalman filtering. 1147-1150 - Basavaraj I. Pawate, M. L. McMahan, Richard H. Wiggins, George R. Doddington, Periagaram K. Rajasekaran:
Connected word recognizer on a multiprocessor system. 1151-1154 - Ted H. Applebaum, Brian A. Hanson, Hisashi Wakita:
Weighted cepstral distance measures in vector quantization based speech recognizers. 1155-1158 - Hynek Hermansky:
An efficient speaker-independent automatic speech recognition by simulation of some properties of human auditory perception. 1159-1162 - Masafumi Nishimura, Koichi Toshioka:
HMM-Based speech recognition using multi-dimensional multi-labeling. 1163-1166 - Xue-Dong Huang, Lian-Hong Cai, Ditang Fang, Bian-Jin Ci, Li Zhou, Li Jian:
A large-vocabulary Chinese speech recognition system. 1167-1170 - Hen-Geul Yeh:
Adaptive noise cancellation for speech with a TMS32020. 1171-1174 - Tatsuya Kimura, Katsuyuki Niyada, Shoji Hiraoka, Shuji Morii, Taisuke Watanabe:
A telephone speech recognition system using word spotting technique based on statistical measure. 1175-1178 - Aggelos K. Katsaggelos:
Multiple input adaptive iterative image restoration algorithms. 1179-1182 - Reginald L. Lagendijk, Russell M. Mersereau, Jan Biemond:
On increasing the convergence rate of regularized iterative image restoration algorithms. 1183-1186 - Tsuneo Saito, Hiroyuki Kudo:
An image reconstruction from limited view angle projection data. 1187-1190 - Wang Yenping, Li Han, Zhu Wenchun:
A new iterative method with histogram equalization constraint for reconstructing image from phase. 1191-1194 - Thomas T. Huang, Jorge L. C. Sanz, W. E. Blanz:
Image reconstruction from one-bit Fourier phase: Theory, sampling, and coherent image model. 1195-1198 - Thomas S. Huang, K. A. Rinaldi, Hua Lee:
Comparison of phase retrieval algorithms. 1199-1200 - Cheng-Tie Chen, M. Ibrahim Sezan, A. Murat Tekalp:
Effects of constraints, initialization, and finite-word length in blind deblurring of images by convex projections. 1201-1204 - H. Joel Trussell, Hatice Örün Öztürk, M. Reha Civanlar:
Error bounds for iterative reprojection methods in computerized tomography. 1205-1208 - H. Joel Trussell, Patrick L. Combettes:
Considerations for the restoration of stochastic degradations. 1209-1212 - H. Joel Trussell, M. Reha Civanlar:
Consistency of the minimum mean square error estimate. 1213-1216 - Stephen A. Laico, Barry J. Sullivan:
ECT Image enhancement. 1217-1220 - Mehrdad Soumekh:
Reconstruction of bandlimited signals from their unevenly-spaced sampled data. 1221-1224 - Zhi-Dong Bai, Paruchuri R. Krishnaiah, C. R. Rao, P. S. Reddy, Yung-Nien Sun, Lin-Cheng Zhao:
A new approach for reconstruction of the left ventricle from biplane angiocardiograms. 1225-1228 - Denise L. Angwin, Howard Kaufman:
Effects of modeling domains on recursive color image restoration. 1229-1231 - Jiang Min, Chen Su Xiang:
A new approach to 2-D Kalman filtering. 1232-1235 - Hao Jinchi, Tal Simchony, Rama Chellappa:
Stochastic relaxation for MAP restoration of gray level images with multiplicative noise. 1236-1239 - Lee C. Potter, K. S. Arun:
Extrapolation of multi-dimensional bandlimited sequences using energy concentration information. 1240-1243 - Nikolas P. Galatsanos, Roland T. Chin:
Digital restoration of multi-channel images. 1244-1247 - Adly T. Fam, Yong Hoon Lee, Sung-Jea Ko:
Order statistic last output reference filters. 1248-1251 - Mark K. Cook, Richard A. Jones:
A minimum-risk quantizer for noisy sources. 1252-1255 - Fumitada Itakura, Taizo Umezaki:
Distance measure for speech recognition based on the smoothed group delay spectrum. 1257-1260 - Kiyohiro Shikano:
Improvement of word recognition results by trigram model. 1261-1264 - Richard F. Lyon:
Speech recognition in scale space. 1265-1268 - M. Codogno, Luciano Fissore:
Duration modelling in finite state automata for speech recognition and fast speaker adaptation. 1269-1272 - A. Jarre, Roberto Pieraccini:
Some experiments on HMM speaker adaptation. 1273-1276 - Andreas Noll, Hermann Ney:
Training of phoneme models in a sentence recognition system. 1277-1280 - Anna Maria Colla, Aaron E. Rosenberg:
Unsupervised bootstrapping of diphone-like templates for connected speech recognition. 1281-1284 - Arthur Nádas, David Nahamoo:
Automatic speech recognition via pseudo-independent marginal mixtures. 1285-1287 - Nikil S. Jayant:
ADPCM Coding of speech with backward-adaptive algorithms for noise feedback and postfiltering. 1288-1291 - Henry M. Dante:
Effect of signal bandwidth on the accuracy of adaptive interpolation of discrete-time signals. 1292-1295 - Jean-Jacques Fuchs:
Estimating the number of sinusoids in additive white-noise. 1296-1299 - Uwe Franke:
Selective deconvolution: A new approach to extrapolation and spectral analysis of discrete signals. 1300-1303 - Richard J. Vaccaro, Fu Li:
A state-space approach to positive sequences. 1304-1307 - Zhi-Dong Bai, Paruchuri R. Krishnaiah, Lin-Cheng Zhao:
On estimation of the number of signals and frequencies of multiple sinusoids. 1308-1311 - Tie-Jun Shan:
On predictive least squares filtering. 1312-1315 - Carlos E. Davila, Ashley J. Welch, H. Grady Rylander III:
A Kalman filter algorithm for estimating sinusoids in colored noise. 1316-1319 - Larry Pearlstein, Bede Liu:
Convergence rate of adaptive line enhancer with tap failures. 1320-1323 - Yariv Ephraim, Jay G. Wilpon, Lawrence R. Rabiner:
A linear predictive front-end processor for speech recognition in noisy environments. 1324-1327 - D. O. Anderton, Craig K. Rushforth:
Waveform coding of voiceband data signals at 16 kb/s. 1328-1331 - Kiyoshi Mizui, Masafumi Hagiwara, Masao Nakagawa:
Modulo-PCM with multi-quantizer. 1332-1335 - Khalid Sayood, David C. Farden:
A bound on predictor misadjustment in ADPCM. 1336-1339 - Tomohiko Taniguchi, Shigeyuki Unagami, Kohei Iseda, Yukou Mochida, Syozi Tominaga:
A 16 kbps ADPCM with multi-quantizer (ADPCM-MQ) codec and its implementation by digital signal processor. 1340-1343 - T. C. Chen:
A fast algorithm for uniform vector quantization. 1344-1347 - J. F. Lynch Jr., J. G. Josenhans, Ronald E. Crochiere:
Speech/Silence segmentation for real-time coding via rule based adaptive endpoint detection. 1348-1351 - Ping Zheng, Hong-ji Zhang:
A new idea of code book design in vector quantization of speech. 1352-1353 - Daniel Lin:
Speech coding using efficient pseudo-stochastic block codes. 1354-1357 - Ramin A. Nobakht, Sarah A. Rajala:
An image coding technique using a human visual system model and image analysis criteria. 1358-1361 - Sarah A. Rajala, M. Reha Civanlar, Wonrae M. Lee:
A second generation image coding technique using human visual system based segmentation. 1362-1365 - Luis Torres-Urgell, R. Lynn Kirlin:
A new adaptive method for image compression using Karhunen-Loeve transform. 1366-1369 - Tony Gioutsos, Susan A. Werness:
Transform coding of synthetic aperture radar (SAR) images. 1370-1373 - Michael Gilge:
Adaptive transform coding of four-color printed images. 1374-1377 - Peter H. Westerink, Jan Biemond, Dick E. Boekee:
Sub-band coding of images using predictive vector quantization. 1378-1381 - Mark J. T. Smith, Steven L. Eddins:
Subband coding of images with octave band tree structures. 1382-1385 - Manohar Das, S. Y. Tan, Nan K. Loh:
Adaptive predictive coding of images based upon multiplicative time series modelling. 1386-1389 - Tao Li, Brent E. Nelson, J. Kelly Flanagan, Christopher J. Read:
A multiprogrammed parallel architecture for digital signal processing. 1390-1393 - Yuang Lou, Chrysostomos L. Nikias, Anastasios N. Venetsanopoulos:
VLSI Array processing structures of quadratic digital filters with LMS algorithm. 1394-1397 - Teresa H.-Y. Meng, Edward A. Lee, David G. Messerschmitt:
Least squares computation at arbitrarily high speeds. 1398-1401 - Qadeer A. Qureshi, Thomas R. Fischer:
A hardware pyramid vector quantizer. 1402-1405 - Helmut Forren, D. A. Schwartz:
Transforming periodic synchronous multiprocessor programs. 1406-1409 - Ramasamy Krishnan, Graham A. Jullien, William C. Miller:
VLSI Modular architectures for complex digital signal processing applications. 1410-1413 - A. P. Shenoy, Ramdas Kumaresan:
An accurate scaling technique in improved residue number system arithmetic. 1414-1417 - Sin-Horng Chen, Min-Tau Lin:
On the use of pitch contour of Mandarin speech in text-independent speaker identification. 1418-1421 - Prem C. Pandey, Hans Kunov, Sharon M. Abel:
A speech processor providing fricative and low-frequency periodicity information for single channel cochlear prosthesis. 1422-1425 - Konrad Lukaszewicz, Matti Karjalainen:
Microphonemic method of speech synthesis. 1426-1429 - Douglas D. O'Shaughnessy:
Specifying intonation in a text-to-speech system using only a small dictionary. 1430-1433 - Hélène Bonneau-Maynard, Jean-Luc Gauvain:
Vector quantization for speaker adaptation. 1434-1437 - Takeshi Kawabata, Masaki Kohda:
Word spotting method based on top-down phoneme verification. 1438-1441 - Joseph Picone, George R. Doddington, Bruce G. Secrest:
Robust pitch detection in a noisy telephone environment. 1442-1445 - Hany Selim, Taghrid Anbar:
A phonetic transcription system of Arabic text. 1446-1449 - Gerhard Rigoll:
The dectalk system for German: A study of the modification of a text-to-speech converter for a foreign language. 1450-1453 - Jörg Höhne, Paul W. Schönle, Bastian Conrad, G. Hong, N. Sandner, H. Veldscholten, C. Appel, Peter Wenig:
Direct registration of articulatory movements versus acoustic analysis for speech production modelling and the treatment of speech motor disorders. 1454-1456 - A. Federico, G. Ibba, Andrea Paoloni:
A new automated method for reliable speaker identification and verification over telephone channels. 1457-1460 - John R. Deller Jr., D. Hsu, L. J. Ferrier:
Recognition of Cerebral Palsy speech: Technical method and a study of vowel consistency. 1461-1464 - Mark A. Jasiuk, Vladimir Goncharoff, John Damoulakis:
Improved speech modification method. 1465-1468 - Kung-Pu Li:
Separating phonetic and speaker features of vowels in formant space. 1469-1472 - S. Eady, B. Craig Dickson, S. C. Urbanczyk, Jocelyn Clayards, A. Wynrib:
Pitch assignment rules for speech synthesis by word concatenation. 1473-1476 - G. B. Rossi, R. W. Mayne:
Identification of a vibration isolation system including results based on nonlinear programming. 1477-1480 - Albert A. Gerlach, K. D. Flowers, E. L. Kunz, W. L. Anderson:
A dynamic spectral transform and its statistical characteristics. 1481-1484 - Theagenis J. Abatzoglou, Jerry M. Mendel:
Constrained total least squares. 1485-1488 - Hamid M. Faridani:
Decentralized filtering with compressed measurements. 1489-1492 - Michael T. Manry, C. T. Huddleston:
Parameter estimation using the autocorrelation of the discrete Fourier transform. 1493-1496 - Gérard Favier:
Computationally efficient adaptive identification algorithms. 1497-1500 - Sung-won Park, J. T. Cordaro:
Maximum likelihood estimation of poles from impulse response data in noise. 1501-1504 - Sophocles J. Orfanidis:
Pole retrieval by eigenvector methods. 1505-1508 - P. A. Ramamoorthy, V. K. Iyer, Y. Ploysongsang:
Autoregressive modeling of the Wigner spectrum. 1509-1512 - Kai-Bor Yu:
Signal representation and processing in the mixed time-frequency domain. 1513-1516 - Bhaskar D. Rao:
Sensitivity analysis of state space methods in spectrum estimation. 1517-1520 - H. Garudadri, Michael P. Beddoes, A.-P. Benguerel, J. H. V. Gilbert:
On computing the smoothed Wigner distribution. 1521-1524 - Fuminori Kobayashi, Hiroshi Suzuki:
Time-varying signals analysis using squared analytic signals. 1525-1528 - Moeness G. Amin:
Time and lag window selection in Wigner-Ville distribution. 1529-1532 - Nagaraja Srinivasa, K. R. Ramakrishnan, Kasi Rajgopal:
Two dimensional spectral estimation: A radon transform approach. 1533-1536 - J. R. Cruz, Zoran B. Banjanin:
Fast computation of high resolution frequency estimates. 1537-1540 - Panos Georgiou Adamopoulos, J. K. Hammond:
The use of the Wigner-Ville spectrum as a method of identifying/Characterising nonlinearities in systems. 1541-1544 - Juin-Hwey Chen, Allen Gersho:
Covariance and autocorrelation methods for vector linear prediction. 1545-1548 - Alex C. Kot, Sarangarajan Parthasarathy, Donald W. Tufts, Richard J. Vaccaro:
The statistical performance of state-variable balancing and Prony's method in parameter estimation. 1549-1552 - Craig E. Morris, Mark A. Richards, Monson H. Hayes:
A generalized fast iterative deconvolution algorithm. 1553-1556 - Chrysostomos L. Nikias, Hsing-Hsing Chiang:
Non-causal autoregressive bispectrum estimation and deconvolution. 1557-1560 - Barry J. Sullivan:
The Lanczos method and signal extrapolation. 1561-1564 - M. Ibrahim Sezan, A. Murat Tekalp, Cheng-Tie Chen:
Regularized signal restoration using the theory of convex projections. 1565-1568 - Hua Lee, Douglas P. Sullivan, Thomas S. Huang:
Improvement of discrete band-limited signal extrapolation by iterative subspace modification. 1569-1572 - Hua Lee, Behzad Noorbehesht:
NMR Spectral parameter estimation by deconvolution. 1573-1576 - Shawn R. McCaslin, Thomas W. Parks, Kenneth Steiglitz:
Finite-record filtering for bandlimited signals. 1577-1580 - Zhongze Wu, Yanda Li, Tong Chang:
Discrete signal reconstruction from its autocorrelation function and one sample. 1581-1584 - Chien-Chung Yeh, Hails M. Bayri:
High resolution bearing estimations by covariance matrix approximation. 1585-1588 - David M. Thomas, Monson H. Hayes III:
A novel data-adaptive power spectrum estimation technique. 1589-1592 - Jar-Fen Yang, Mostafa Kaveh:
Adaptive signal-subspace algorithms for frequency estimation and tracking. 1593-1596 - James A. Cadzow, Young S. Kim, Dong-Chang Shiue, Y. Sun, G. Xu:
Resolution of coherent signals using a linear array. 1597-1600 - Robert A. Muir, Wynn C. Stirling:
A novel approach to time-varying spectral probability estimation. 1601-1604 - Richard R. Hansen Jr., Rama Chellappa:
2-D Spectrum estimation for imperfectly observed lattice data. 1605-1608 - Yu Hen Hu, Pin-Kuan Chou, Ali Hussein Abdallah:
Subspace approximation based algorithms for adaptive high resolution spectrum estimate. 1609-1612 - Richard J. Vaccaro, Alex C. Kot:
A perturbation theory for the analysis of SVD-based algorithms. 1613-1616 - Bhaskar D. Rao, Rong Peng:
Tracking analysis of an ARMA parameter estimation algorithm using weak convergence theory. 1617-1620 - J. Sérvule Rodrigues, Luis B. Almeida:
Harmonic coding at 8 kbits/sec. 1621-1624 - Yong Ching Lim, S. N. Koh, Chi Chung Ko:
Complementary filtering technique for subband speech coder design. 1625-1628 - Takehiro Moriya, Masaaki Honda:
Transform coding of speech with weighted vector quantization. 1629-1632 - Sharad Singhal:
On encoding filter parameters for stochastic coders. 1633-1636 - Richard C. Rose, Thomas P. Barnwell III:
Quality comparison of low complexity 4800 bps self excited and code excited vocoders. 1637-1640 - E. Bryan George, Mark J. T. Smith:
A new speech coding model based on a least-squares sinusoidal representation. 1641-1644 - Robert J. McAulay, Thomas F. Quatieri:
"Multirate sinusoidal transform coding at rates from 2.4 kbps to 8 kbps". 1645-1648 - Peter Kroon, Bishnu S. Atal:
Quantization procedures for the excitation in CELP coders. 1649-1652 - Joseph Picone, George R. Doddington:
Low rate speech coding using contour quantization. 1653-1656 - John W. Woods, David J. Potter, Howard Kaufman:
Parallel realizations of 2-D recursive Kalman filters. 1657-1660 - George A. Lampropoulos:
Systolic array realization of digital filters. 1661-1664 - John E. Diamessis, Charles W. Therrien, William J. Rozwod:
Design of 2-D FIR filters with nonuniform frequency samples. 1665-1668 - Soo-Chang Pei, Sy-Been Jaw:
Efficient design of 2D multiplierless FIR filters by transformation. 1669-1672 - Chwen-Jye Ju, Winser E. Alexander:
Derivation and stability analysis of multidimensional IIR block digital filters. 1673-1676 - Mark J. Paulik, Manohar Das, Nan K. Loh:
A projection based constrained optimization technique for one shot optimal design of stable 1-D and separable 2-D IIR filters. 1677-1680 - George A. Lampropoulos, T. J. Lawson, Y. T. Chan:
On zero phase design of IIR filters. 1681-1684 - Ali Zilouchian, Robert L. Carroll:
Optimal realization of multidimensional digital filters. 1685-1688 - Hari C. Reddy, P. K. Rajan:
Continued fraction expansion of 1-D complex discrete reactance functions with application to 2-D stability testing. 1689-1691 - Benjamin Friedlander, Boaz Porat:
Detection of transient signals by the Gabor representation. 1692-1695 - Leon H. Sibul, John A. Tague:
A canonical representation approach to signal detection and estimation in adaptive array processors. 1696-1699 - Johannes G. Lourens, M. Wynand Coetzer:
Detection of mechanical ship features from underwater acoustic sound. 1700-1703 - Philippe M. Cassereau, Jules S. Jaffe:
Frequency hopping patterns for simultaneous multiple-beam sonar imaging. 1704-1707 - John Dunlop, Manal Jamil Al-Kindi, L. E. Virr:
Application of adaptive noise cancelling to diver voice communications. 1708-1711 - Andrew E. Yagle:
A fast algorithm for linear estimation of three-dimensional homogeneous anisotropic random fields. 1712-1715 - Steven W. Patton:
Robust estimation of the acoustic attenuation parameter. 1716-1719 - Ziad S. Haddad:
A full wave solution for propagation in horizontally stratified elastic media with range variation. 1720-1723 - Jackie S. C. Fung, Anastasios N. Venetsanopoulos:
Design of quadratic filters based on the D norm for seismic deconvolution. 1724-1727 - Yingbo Hua, Tapan K. Sarkar:
Analysis of three high resolution techniques for radio direction estimation. 1728-1731 - Abdalla S. A. Mohamed, T. Prasad, A.-H. Rashwan, Mohamed Emad Mousa Rasmy:
Modelling the neuromuscular system using non-invasive experimental methods. 1732-1735 - Chung H. Lu:
Speech power estimation with a truncated normal distribution. 1736-1739 - Przemyslav Dymarski:
Predictors of speech signal with adaptive delays. 1740-1743 - N. Härle, Johann F. Böhme:
Detection of knocking for spark ignition engines based on structural vibrations. 1744-1747 - David H. Friedman:
Formulation of a vector distance measure for the instantaneous-frequency distribution (IFD) of speech. 1748-1751 - J. S. Lee, Joe K. Hammond:
Estimation of the directionality pattern of a moving acoustic source. 1752-1755 - C. C. Li, Hong Wang:
Short-time high-resolution pulse-Doppler processing with frequency diversity signaling. 1756-1759 - E. Hesham Attia:
Efficient computation of the music algorithm as applied to a low-angle elevation estimation problem in a severe multipath environment. 1760-1763 - Neil J. Grossbard:
Estimating frequencies of interferometer type data with decimated auto-regressive techniques. 1764-1766 - Ziad S. Haddad, Bowen E. Parkins:
Output spectra of non-linear systems. 1767-1769 - Abdulmagid Omar, Abdussalam Addeeb, Charles Slivinsky, Richard DuBroff:
Estimation approach to locating buried geological interfaces. 1770-1773 - Yi-Tong Zhou, Rama Chellappa, George A. Bekey, Ernest L. Bontrager:
Estimation of filtering properties of living tissue for inverse filtering of surface EMG signals. 1774-1777 - Jelisaveta Kesler, Stanislav B. Kesler:
Experiments in joint Doppler and elevation estimation in the near field. 1778-1781 - K. V. S. Prakash, K. M. M. Prabhu, V. Srinivasan:
Simulation and evaluation of an experimental radar clutter model. 1782-1785 - Huang Zhen-xing, Wan Zheng:
Range ambiguity resolution in multiple PRF pulse Doppler radars. 1786-1789 - Chan F. Lam, David Kamins, Kuno Peter Zimmermann:
Signature recognition through spectral analysis. 1790-1792 - K. M. M. Prabhu, R. D. Shenoy:
Some results on time and lag weighting for spectral estimation. 1793-1796 - Yoav Medan, Eyal Yair:
Discrete spectral analysis of periodic time functions. 1797-1800 - Mark A. Richards:
On the efficient implementation of the split-radix FFT. 1801-1804 - Pierre Duhamel, Hedi H'Mida:
New 2nDCT algorithms suitable for VLSI implementation. 1805-1808 - C. Sidney Burrus:
Bit reverse unscrambling for a radix-2MFFT. 1809-1810 - Kwang-Shik Min, J. Carlisle, B. Doughty, C. Jones, Charles H. Rogers:
A fast triangular transform and its applications. 1811-1814 - Michael Unser:
A family of discrete Fourier transforms with pseudo-cyclic convolution properties. 1815-1818 - H. Babic, J. Baumgartner, S. K. Mitra:
An efficient method for computing a very high resolution DFT of a short sequence. 1819-1822 - J. Davidson, Hagit Messer, H. Ur:
Implementation of SAW complex cepstrum and its applications. 1823-1826 - Gloria Faye Boudreaux-Bartels, Thomas W. Parks:
Discrete Fourier transform using summation by parts. 1827-1830 - Henrik V. Sorensen, Douglas L. Jones, C. Sidney Burrus:
Real-valued algorithms for the FFT. 1831-1834 - Tapan K. Sarkar, Xiapu Yang:
Accurate and efficient solution of Hankel matrix systems by FFT and the conjugate gradient methods. 1835-1838 - Bal Krishna, Salvatore D. Morgera, Hari Krishna:
Generalized two-term recurrences and fast algorithms for Hermitian Toeplitz matrices. 1839-1842 - Okan K. Ersoy, Neng-Chung Hu:
A unified approach to the fast computation of all discrete trigonometric transforms. 1843-1846 - Mike Griffin, Fred J. Taylor:
Narrowband reduced complexity transform domain adaptive filter. 1847-1850 - Cédric Demeure, Louis L. Scharf:
Vector algorithms for computing QR and Cholesky factors of close-to-Toeplitz matrices. 1851-1854 - Keshab K. Parhi, David G. Messerschmitt:
Look-ahead computation: Improving iteration bound in linear recursions. 1855-1858 - Ronald D. DeGroat, Richard A. Roberts:
An improved, highly parallel rank-one eigenvector update method with signal processing applications. 1859-1862 - H. K. Kwan, M. T. Tsim:
High speed 1-D FIR digital filtering architectures using polynomial convolution. 1863-1866 - Marc A. Zissman, Gerald C. O'Leary, Don H. Johnson:
A block diagram compiler for a digital signal processing MIMD computer. 1867-1870 - Adrian Raper, Joe K. Hammond:
An expert system for transient data analysis using a model based architecture developed with poplog. 1871-1874 - Yu Hen Hu, Ali Hussein Abdallah:
Knowledge-based adaptive signal processing. 1875-1878 - C. David Covington, G. E. Carter, D. W. Summers:
Graphic oriented signal processing language-GOSPL. 1879-1882 - Edward N. Horn, Benjamin Monderer, Aurel A. Lazar:
SPEED: A distributed software environment for multi-process communications and control. 1883-1886 - Richard Lepage:
Interactive software package for digital signal processing. 1887-1890 - Patrick M. Peterson, Joseph A. Frisbie:
An interactive environment for signal processing on a VAX computer. 1891-1894 - Alberto Ciaramella, Giovanni Venuti:
Vector quantization firmware for an acoustical front-end using the TMS32020. 1895-1898 - Stephen J. A. McGrath, Thomas P. Barnwell III, D. A. Schwartz:
A WE-DSP32 based, low-cost, high-performance, synchronous multiprocessor for cyclo-static implementations. 1899-1902 - James O. Normile:
A video composite to component decoder using a V.L.S.I. digital F.I.R. filter. 1903-1906 - Cheng-Wen Wu, Peter R. Cappello:
Computer-aided design of VLSI second-order sections. 1907-1910 - Amine Haoui, Hui-Hung Lu, David Hedberg:
An all-digital timing recovery scheme for voiceband data modems. 1911-1914 - Hui-Hung Lu, David Hedberg, Bernard Fraenkel:
Implementation of high-speed voiceband data modems using the TMS320C25. 1915-1918 - Thomas G. Marshall Jr.:
Simulating distributed signal processing systems in modula-2. 1919-1921 - Ichiro Kuroda, Takao Nishitani, Teiji Takeuchi, Hitoshi Koyama, Junko Sunaga, Shuji Matsukawa:
Blockdiagram programming system for 32 bit floating point signal processor. 1922-1925 - A. H. Crossman, Frank Fallside:
Multipulse-excited channel vocoder. 1926-1929 - Shan Shan Huang, Robert M. Gray:
Conditional histogram vector quantization for spellmode recognizer. 1930-1933 - Claude R. Galand, C. Arnaud, Jean E. Menez:
High-frequency regeneration of base-band vocoders by multi-pulse excitation. 1934-1937 - S. Adlersberg, Vladimir Cuperman:
Transform domain vector quantization for speech signals. 1938-1941 - Maurizio Copperi, Daniele Sereno:
Feature extraction and product codes in vector excited coders. 1942-1945 - Mohammad Reza Soleymani, Salvatore D. Morgera:
A high-speed search algorithm for vector quantization. 1946-1948 - Salim E. Roucos, Alexander MacLeod Wilgus, William Russell:
A segment vocoder algorithm for real-time implementation. 1949-1952 - Jean-Pierre Adoul, Claude Lamblin:
A comparison of some algebraic structures for CELP coding of speech. 1953-1956 - Jean-Pierre Adoul, Philippe Mabilleau, M. Delprat, Sarto Morissette:
Fast CELP coding based on algebraic codes. 1957-1960 - Hidenobu Harasaki, Ichiro Tamitani, Yukio Endo, Takao Nishitani, Masakatsu Yamashina, Tadayoshi Enomoto, Norio Suzuki:
Realtime video signal processor module. 1961-1964 - Günter Schamel:
Multidimensional interpolation of progressive frames from spatio-temporally subsampled HDTV fields. 1965-1968 - Thomas Reuter:
Multi-dimensional adaptive sampling rate conversion. 1969-1972 - Robert A. Cohen, John W. Woods, Makoto Sanya, John F. McDonald:
A video rate architecture for a fully recursive two-dimensional filter. 1973-1976 - An-Loong Kok, Dimitris G. Manolakis, Vinay K. Ingle:
Efficient algorithms for 1-D and 2-D noncausal autoregressive system modelings. 1977-1980 - Yiannis S. Boutalis, Stefanos D. Kollias, George Carayannis:
A fast multichannel approach to adaptive estimation and filtering of two dimensional images. 1981-1984 - Nirmal Kumar Bose, Yun Q. Shi:
Iterative schemes for two-dimensional spectral factorization. 1985-1986 - Soo-Chang Pei, Ja-Ling Wu:
Split vector radix 2D fast Fourier transform. 1987-1990 - Alexander Skavantzos, Mike Griffin, Fred J. Taylor:
On the multidimensional RNS and its applications to the design of fast digital systems. 1991-1994 - Kevin M. Buckley, Lloyd J. Griffiths:
Design of deterministic beamformers for arbitrarily configured arrays. 1995-1998 - Lloyd J. Griffiths:
A new approach to partially adaptive arrays. 1999-2002 - John W. Fay, Peter M. Schultheiss:
Detection in a flow-noise dominated environment. 2003-2006 - D. E. Ohlms:
Beam output interference cancellation for line arrays. 2007-2010 - Jar-Fen Yang, Mostafa Kaveh:
Wideband adaptive arrays based on the coherent signal-subspace transformation. 2011-2014 - U. Sandkühler, Johann F. Böhme:
Accuracy of maximum-likelihood estimates for array processing. 2015-2018 - Allan O. Steinhardt:
Reconstructing a finite length sequence from several of its correlation lags. 2019-2022 - Robert S. Walker, David N. Swingler:
Beamforming with aperture extrapolation (APEX): Performance in practice. 2023-2026 - William S. Hodgkiss, D. Almagor:
An eigenvalue/Eigenvector interpretation of surface reverberation rejection performance. 2027-2030 - B. W. Dahanayake:
Bearings estimation by QZ and VZ decomposition. 2031-2034 - Miguel Angel Lagunas, Mateo Amengual:
Non-linear spectral estimation. 2035-2038 - Leland B. Jackson, Jianguo Huang, Kevin P. Richards:
AR, ARMA, and AR-in-noise modeling by fitting windowed correlation data. 2039-2042 - Xiao-Hu Yu, Zhen-Ya He:
A modified Yule-Walker equations method for harmonic analysis in unknown colored noise. 2043-2046 - Jack McCready:
Error probability in spectral analysis using DFT or FFT analyzers. 2047-2049 - Shubhada Gadre, J. Chandrasekhar, M. M. Kulkarni:
High resolution spectral estimation. 2050-2053 - Huili Wang, Gregory H. Wakefield:
Signal-subspace approximation for line spectrum estimation. 2054-2057 - Miguel A. Mayorga, Lonnie C. Ludeman:
Improved spectral estimation based on extrapolated and smoothed data records. 2058-2061 - Roberto Cusani, Giovanni Jacovitti:
Estimation of sinusoids and chirps from interrupted time intervals. 2062-2065 - P. J. Tourtier, Louis L. Scharf:
Maximum likelihood identification of correlation matrices for estimation of power spectra at arbitrary resolutions. 2066-2069 - Yong Bin Chen, Yu Qing Gao:
The estimation of evolutionary spectrum by square-root filtering algorithm. 2070-2073 - Zhen-Ya He, Jin-Ling Ni:
Average detection performance of adaptive maximum entropy spectrum estimator. 2074-2076 - Randolph L. Moses, Peter Stoica, Benjamin Friedlander, Torsten Söderström:
An efficient linear method for ARMA spectral estimation. 2077-2080 - M. Isabel Ribeiro, José M. F. Moura:
Dual algorithm for ARMA spectrum estimation. 2081-2084 - Amrane Houacine, Guy Demoment:
Fast adaptive spectrum estimation: Bayesian approach and long AR models. 2085-2088 - Mati Wax:
Order selection for AR models by predictive least-squares. 2089-2092 - You Xu, Xian-Ci Xiao:
A high performance spectral estimation method: The AMW algorithm. 2093-2096 - Lei Xu, Pingfan Yan, Tong Chang:
Almost unique specification of discrete finite length signal: From its end point and Fourier transform magnitude. 2097-2100 - Giri Boray:
Design of digital filters for communication systems. 2101-2104 - Stephen McLaughlin, Bernard Mulgrew, Colin F. N. Cowan:
Performance comparison of least squares and least mean squares algorithms as HF channel estimators. 2105-2108 - Arye Nehorai, David Starer:
An adaptive SSB carrier frequency estimator. 2109-2112 - Gary J. Saulnier, Kiho Yum, Pankaj K. Das:
Narrow-band jammer suppression using an adaptive lattice filter. 2113-2116 - A. B. Sesay, Kon Max Wong, Patrick C. Yip:
On signal design and detection in a multi-user channel. 2117-2120 - Donald W. Tufts, Melbourne Barton, Adam J. Efron:
Robust, adaptive filtering for data transmission. 2121-2124 - Hiroshi Yasukawa, Shoji Shimada, Isao Furukawa:
Acoustic echo canceller with high speech quality. 2125-2128 - Sasan H. Ardalan, John J. Paulos:
An analysis of sinusoidally excited delta-sigma modulators. 2129-2132 - Philip C. Yip, Delores M. Etter:
An adaptive technique for multiple echo cancelation in telephone networks. 2133-2136 - Sergios Theodoridis, Nicholas Kalouptsidis, John G. Proakis:
LS FIR Smoothers and application to interference rejection in PN spread spectrum systems. 2137-2140 - André Gilloire:
Experiments with sub-band acoustic echo cancellers for teleconferencing. 2141-2144 - S. Thomas Alexander, D. H. Kim:
Analytical rate versus distortion characteristics for LMS adaptive source coding. 2145-2148 - Maurice G. Bellanger:
Engineering aspects of fast least squares algorithms in transversal adaptive filters. 2149-2152 - Manal Jamil Al-Kindi, John Dunlop:
A low distortion adaptive noise cancellation structure for real time applications. 2153-2156 - Michael G. Larimore, Sally L. Wood, John R. Treichler:
Tracking speed requirements for time-varying adaptive channel equalizers. 2157-2160 - John P. Princen, A. W. Johnson, Alan Bernard Bradley:
Subband/Transform coding using filter bank designs based on time domain aliasing cancellation. 2161-2164 - Bruno Paillard, Joël Soumagne, Philippe Mabilleau, Sarto Morissette:
Filters for subband coding analytical approach. 2165-2168 - P. P. Vaidyanathan, Phuong-Quan Hoang:
The perfect-reconstruction QMF bank: New architectures, solutions, and optimization strategies. 2169-2172 - Chia-Chuan Hsiao:
Polyphase filter matrix for rational sampling rate conversions. 2173-2176 - Eric Viscito, Jan P. Allebach:
A fast algorithm for the design of narrow-band multirate digital filters. 2177-2180 - Yair Shoham:
Vector predictive quantization of the spectral parameters for low rate speech coding. 2181-2184 - Juin-Hwey Chen, Allen Gersho:
Real-time vector APC speech coding at 4800 bps with adaptive postfiltering. 2185-2188 - Grant A. Davidson, Mei Yong, Allen Gersho:
Real-time vector excitation coding of speech at 4800 bps. 2189-2192 - Philip A. La Follette, James T. Sims, John D. Tardelli:
A parallel implementation of canonical coordinate speech compression. 2193-2196 - Ramón García-Gómez, Francisco Javier Casajús-Quirós, Luis A. Hernández Gómez:
Vector quantized multipulse-LPC. 2197-2200 - Yasuo Matsuyama:
Variable region vector quantization, space warping and speech/Image compression. 2201-2204 - A. Lowry, Sqama Hossain, W. Millar:
Binary search trees for vector quantisation. 2205-2208 - Raymond Toy, William A. Pearlman:
Backward adaptation for transform trellis coding of speech. 2209-2212 - Edward C. Bronson, Douglas A. Carlone, W. Bastiaan Kleijn, Kevin M. O'Dell, Joseph Picone, David L. Thomson:
Harmonic coding of speech at 4.8 kb/s. 2213-2216 - David B. Harris:
A master event strategy for location with seismic array data. 2217-2220 - Donald K. Mitchell, Rangaraj M. Rangayyan:
Restoration of limited-data seismic tomography images. 2221-2224 - Farid U. Dowla, David B. Harris:
Direction estimation of vector-planewave fields. 2225-2228 - Tamar Peli:
Evaluation of a wideband direction estimation algorithm for acoustic arrays. 2229-2232 - R. Foka:
Properties of Toeplitz approximation method (TAM) for direction finding problems. 2233-2236 - Gloria Faye Boudreaux-Bartels, P. J. Wiseman:
Wigner distribution analysis of acoustic well logs. 2237-2240 - Charles W. Therrien, Hamdy Taha El-Shaer:
Methods for multichannel 2-D spectrum analysis: Description and comparison. 2241-2244 - David C. Munson Jr.:
An introduction to strip-mapping synthetic aperture radar. 2245-2248 - J. H. Justice, S. M. Dougherty:
Generalized linear inversion applied to seismic data in one and two dimensions. 2249-2251 - Hong-Bin Chen, Jianrong Chen:
Removing part of the origin of the non-minimum-phase behavior of seismic data sequence in chirp-excited seismic exploration systems. 2252-2255 - Ioannis Pitas, Anastasios N. Venetsanopoulos:
AGIS: An expert system for automated geophysical interpretation of seismic images. 2256-2259 - James Ting-Ho Lo, Stanley Lawrence Marple Jr.:
Eigenstructure methods for array sensor localization. 2260-2263 - L. P. H. K. Seymour, C. F. N. Cowan, Peter M. Grant:
Bearing estimation in the presence of sensor positioning errors. 2264-2267 - Philippe Forster, Georges Vezzosi:
Application of spheroidal sequences to array processing. 2268-2271 - Douglas B. Williams, Don H. Johnson:
Modifying the sphericity test for improved source detection with narrowband passive arrays. 2272-2275 - Hong Wang, C. C. Li, J. X. Zhu:
High-resolution direction finding in the presence of multipath: A frequency-domain smoothing approach. 2276-2279 - Ilan Ziskind, Mati Wax:
Maximum likelihood estimation via the alternating projection maximization algorithm. 2280-2283 - Darel A. Linebarger, Don H. Johnson:
A parametric direction finding technique. 2284-2287 - Tie-Jun Shan, Thomas Kailath:
Directional signal separation by adaptive arrays with a root-tracking algorithm. 2288-2291 - Yoram Bresler, Vellenki U. Reddi, Thomas Kailath:
A polynomial approach to optimum beamforming for correlated or coherent signal and interference. 2292-2295 - Arnab K. Shaw, Ramdas Kumaresan:
Estimation of angles of arrivals of broadband signals. 2296-2299 - B. W. Dahanayake:
A new method of array processing. 2300-2303 - D. J. Jeffries, David R. Farrier:
Prior information and eigenvector rotation. 2304-2307 - A. Enis Çetin, Rashid Ansari:
A procedure for antenna array pattern synthesis. 2308-2311 - Barry D. Van Veen, Richard Roberts:
Analytic design of broadband partially adaptive beamformers. 2312-2315 - Michael D. Zoltowski:
Solving the semi-definite generalized eigenvalue problem with application to ESPRIT. 2316-2319 - Varaz Shahmirian, Stanislav B. Kesler:
Bias and resolution of the vector space methods in the presence of coherent planewaves. 2320-2323 - Qihu Li:
The performance of the optimum array filter for sensor arrays. 2324-2327 - R. Rastogi, Prabhat Kumar Gupta, Ramdas Kumaresan:
Array signal processing with interconnected Neuron-like elements. 2328-2331 - Lihe Zou, Lin Yin:
Spatio-temporal spectral analysis by SVD of signal matrix. 2332-2335 - Charles H. Knapp:
Signal detectors for arrays with randomly perturbed sensor locations. 2336-2339 - Charles L. Byrne, Alan K. Steele:
Sector-focused stability for high resolution array processing. 2340-2343 - Richard H. Roy III, Arogyaswami Paulraj, Thomas Kailath:
Comparative performance of ESPRIT and MUSIC for direction-of-arrival estimation. 2344-2347 - Magdy T. Hanna, Marwan A. Simaan:
Minimally sensitive digital filters for array data processing. 2348-2351 - Ronald T. Williams, Surendra Prasad, A. K. Mahalanabis, Leon H. Sibul:
Localization of coherent sources using a modified spatial smoothing technique. 2352-2355 - Donald F. Gingras, Stephen L. Hobbs:
Asymptotic statistics for wavenumber estimation. 2356-2359 - Walter M. X. Zimmer:
Logarithmic least mean square sound-field estimation. 2360-2363 - Patrick M. Peterson:
Using linearly-constrained adaptive beamforming to reduce interference in hearing aids from competing talkers in reverberant rooms. 2364-2367 - Biing-Hwang Juang, Lawrence R. Rabiner:
Signal restoration by spectral mapping. 2368-2371 - Oded Ghitza:
Robustness against noise: The role of timing-synchrony measurement. 2372-2375 - Martin J. Russell, Anneliese E. Cook:
Experimental evaluation of duration modelling techniques for automatic speech recognition. 2376-2379 - Tony Rohlev, Charles M. Loeffler:
Invertible periodically time-varying digital filters. 2380-2383 - Hamid Gharavi, A. Tabatabai:
Application of quadrature mirror filtering to the coding of monochrome and color images. 2384-2387 - Barbara Caspers, Bishnu S. Atal:
Role of multi-pulse excitation in synthesis of natural-sounding voiced speech. 2388-2391 - Jayant M. Naik, George R. Doddington:
Evaluation of a high performance speaker verification system for access control. 2392-2395 - Nacer K. M'Sirdi, I. D. Landau:
Adaptive evolutionary spectrum analysis for narrow band signals. 2396-2399 - Cory S. Myers:
Symbolic representation and manipulation of signals. 2400-2403 - Bishnu S. Atal:
Stochastic Gaussian model for low-bit rate coding of LPC area parameters. 2404-2407 - Geeta Jayasumana, Charles M. Loeffler:
Searching for the best Cooley-Tukey FFT algorithms. 2408-2411 - V. Ramamoorthy, T. Raj Natarajan:
Transmission quality of digital audio teleconferencing bridge. 2412-2415 - Yousif A. El-Imam:
Speech synthesis by concatenating sub-syllabic sound units. 2416-2417 - Jane Critchley:
Analysis and design of periodically time varying digital filters. 2418-2421 - Hiroya Fujisaki, Keikichi Hirose, Keisuke Shimizu:
A new system for reliable pitch extraction of speech. 2422-2425
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