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ORCIDICASSP 1985: Tampa, Florida, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '85, Tampa, Florida, USA, March 26-29, 1985. IEEE 1985
Isolated Word Recognition
- Kazuhide Sugawara, Masafumi Nishimura, Koichi Toshioka, Masaaki Okochi, Toyohisa Kaneko:
Isolated word recognition using hidden Markov models. 1-4 - Martin J. Russell, Roger K. Moore
:
Explicit modelling of state occupancy in hidden Markov models for automatic speech recognition. 5-8 - Biing-Hwang Juang, Lawrence R. Rabiner, Stephen E. Levinson, M. Mohan Sondhi:
Recent developments in the application of hidden Markov models to speaker-independent isolated word recognition. 9-12 - Douglas B. Paul:
Training of HMM recognizers by simulated annealing. 13-16 - Matthew Lennig:
Experiments on speaker-independent recognition of hand-segmented French vowels. 17-20 - Kathy L. Brown, V. Ralph Algazi:
Discrete utterance speech recognition without time alignment. 21-24 - N. Nocerino, Frank K. Soong, Lawrence R. Rabiner, Dennis H. Klatt:
Comparative study of several distortion measures for speech recognition. 25-28 - David K. Burton:
Applying matrix quantization to isolated word recognition. 29-32 - Kiyoaki Aikawa, Masahide Sugiyama, Kiyohiro Shikano:
Spoken word recognition based on top-down phoneme segmentation. 33-36 - Zhao Guo-Tian:
On associative recognition of isolated Chinese word. 37-40
Digital Filter Design
- P. P. Vaidyanathan:
New design methods for FIR filters with equiripple stopbands and prescribed degrees of passband flatness. 41-44 - Wonyong Sung, Sanjit K. Mitra:
Efficient FIR filter design using differential coding of filter coefficients. 45-48 - Pekka Heinonen, Yrjö Neuvo:
Smoothed median filters with FIR substructures. 49-52 - Darrell Williamson, S. Sridharan:
Residue feedback in ladder and lattice filter structures. 53-56 - Z. S. Li, P. C. Hutchinson:
A new S-Z replacement design technique of IIR digital filters. 57-60 - Hans Wilhelm Schüßler, Peter Steffen:
An approach for designing systems with prescribed behaviour at distinct frequencies regarding additional constraints. 61-64 - Tapio Saramäki, Kari-Pekka Estola:
Design of linear-phase partly digital anti-aliasing filters. 65-68 - Gernot Kubin:
Wave digital filters: Voltage, current, or power waves? 69-72 - Guenter Wackersreuther:
A novel approach to the design of filters for filter banks. 73-76
Spectrum Analysis
- Miguel Angel Lagunas, M. E. Santamaria, Antoni Gasull, Asunción Moreno:
Cross spectrum ML estimate. 77-80 - Huanqun Chen, Tapan K. Sarkar, Soheil A. Dianat, John D. Brule:
Adaptive spectral estimation by the conjugate gradient method. 81-84 - Francesco Palmieri, L. P. Bolgiano Jr.:
Window functions obtained from B-S. 85-88 - Henry M. Dante:
Spectrum estimation of time series with missing data. 89-92 - Donald F. Gingras:
Spectral estimation statistics for noise corrupted autoregressive series-First-order case. 93-96 - Allan O. Steinhardt, Robert K. Goodrich, Richard A. Roberts:
Spectral estimation via minimum energy correlation extension. 97-100 - Shu-Hung Leung, Wel-Yau Horng:
Superresolution autoregressive spectral estimation technique using multiple step prediction. 101-104 - Yu Hen Hu:
Adaptive methods for real time Pisarenko spectrum estimate. 105-108 - Akira Sano, Koh-ichi Hashimoto:
Adaptive recursive scheme for spectral analysis of sinusoids in signals with unknown colored spectrum. 109-112
Single Frame Image Coding
- Alberto Sanz, Carlos Muñoz, Narciso García:
Hierarchical predictive approach to image coding. 113-116 - Shoji Mizuno, Kazumoto Iinuma:
Ordering predictive coding of digital image. 117-120 - Yasuo Yoshida, Akira Nakamura, Hisanao Ogura:
A hybrid image coding technique using a noncausal stochastic model. 121-124 - Nasser M. Nasrabadi:
Use of vector quantizers in image coding. 125-128 - Scott E. Budge, Richard L. Baker:
Compression of color digital images using vector quantization in product codes. 129-132 - Allen Gersho, Mitsuharu Yano:
Adaptive vector quantization by progressive codevector replacement. 133-136 - H. Bheda, K. S. Thyagarajan, Hüseyin Abut:
A fast matrix quantizer for image encoding. 137-140 - K. S. Thyagarajan, S. Parthasarathy, Hüseyin Abut:
A matrix quantizer incorporating the human visual model. 141-144 - William A. Pearlman, May M. Leung, Priyadarshan Jakatdar:
Adaptive transform tree coding of images. 145-148 - Charles F. Hall:
A hybrid image compression technique. 149-152
Topics in Time-Delay Estimation and Geophysical Signal Processing
- Joseph C. Hassab:
Space-time processing of multisensor time delay data in bearing estimation. 153-156 - R. Lynn Kirlin, Alireza Moghaddamjoo:
Robust adaptive Kalman filtering for systems with unknown step inputs and non-Gaussian measurement errors. 157-160 - Mehdi Hatamian, David J. Anderson:
Time delay estimation in presence of jitter. 161-164 - Thomas E. McCannon:
Application of robust filtering techniques to time delay estimation in the Arctic environment. 165-167 - William E. Ryan, J. M. Schumpert:
Time delay estimation for spread spectrum signals using generalized cross correlation. 168-171 - Keith A. Struckman:
Time-delay-of-arrival resolution of multipath communication signals. 172-175 - Kou-Yuan Huang, King-sun Fu:
1-D and 2-D syntactic pattern recognition for the detection of bright spots. 176-179 - Alastair D. McAulay:
Predictive deconvolution of seismic array data for inversion. 180-183 - Lawrence J. Ziomek, Jan Vos:
Linear time-invariant space-variant filters and the WKB approximation. 184-187 - Edmund A. Quincy, Daniel J. Tomich:
Image enhancement using coherence processing with applications to seismograms. 188-191
VLSI Signal Processors
- Magdy A. Bayoumi, Graham A. Jullien, William C. Miller:
A VLSI implementation of an FFT/NTT computational unit. 192-195 - James H. Hesson:
A 32 bit 15M flop floating point programmable signal processor architecture for VLSI implementation. 196-199 - Ed F. Deprettere, Kishan Jainandunsing:
Design and VLSI implementation of a concurrent solver for N coupled least-squares fitting problems. 200-203 - Hironori Yamauchi, Takao Kaneko, Tsutomu Kobayashi, Atsushi Iwata, Sadayasu Ono:
An 18-bit floating-point signal processor VLSI with an on-chip 512W dual-port RAM. 204-207 - Paul M. Toldalagi:
Very fast sequencing and data addressing made easy with new CMOS VLSI components. 208-211 - Robert Fine, Ted Dintersmith:
New CMOS chip family facilitates design of high speed DSP hardware. 212-215 - George G. Ricker, Manuel F. Richey:
Application of an advanced signal processing engine to adaptive techniques. 216-219 - W. K. Jenkins, Edward S. Davidson, D. F. Paul:
A custom-designed integrated circuit for the realization of residue number digital filters. 220-223 - Amnon Aliphas, William F. Ganong III, Peter Stonestrom, Dan Perkins:
High resolution digital filter chip. 224-227 - Cole Erskine, Surendar Magar, Edward R. Caudel, Daniel Essig:
Architecture and applications of a second-generation digital signal processor. 228-231 - Emmanuel A. Arnould, H. T. Kung, Onat Menzilcioglu, Ken Sarocky:
A systolic array computer. 232-235
Narrowband Speech Coding
- Salim E. Roucos, Alexander MacLeod Wilgus:
The waveform segment vocoder: A new approach for very-low-rate speech coding. 236-239 - Joel Crosmer, Thomas P. Barnwell III:
A low bit rate segment vocoder based on line spectrum pairs. 240-243 - George S. Kang, Lawrence J. Fransen:
Application of line-spectrum pairs to low-bit-rate speech encoders. 244-247 - Joseph Rothweiler, Jack Carmody:
A multiple rate low rate voice codec. 248-251 - Maurizio Copperi, Daniele Sereno:
Vector quantization and perceptual criteria for low-rate coding of speech. 252-255 - Jean-Pierre Adoul, F. Didelot, Philippe Mabilleau, Sarto Morissette:
Generalization of the multipulse coding for low bit rate coding purposes: The generalized decimation. 256-259 - Isabel Trancoso, Luís B. Almeida, José M. Tribolet:
Pole-zero multipulse speech representation using harmonic modelling in the frequency domain. 260-263 - H. Joel Trussell, M. Reha Civanlar:
Optimal initial conditions and pulse values for multipulse speech coding. 264-267 - Panos E. Papamichalis:
Bit rate reduction by Markov-Huffman coding of speech parameters. 268-271
VLSI Algorithms and Systolic Arays
- Kenneth Steiglitz, Ronald R. Morita:
A multi-processor cellular automaton chip. 272-275 - Peter R. Cappello:
Towards an FIR filter tissue. 276-279 - Thanos Stouraitis, S. Natarajan, Fred J. Taylor:
A reconfigurable systolic primitive processor for signal processing. 280-283 - Yu Hen Hu:
VLSI Architecture for solving covariance eigen system. 284-287 - Sun-Yuan Kung, Jurgen Annevelink, Patrick M. Dewilde, S. C. Lo:
Hierarchical flowgraph integration for VLSI array processors. 288-291 - Roy Chapman, Tariq S. Durrani, T. Willey:
Design strategies for implementing systolic and wavefront arrays using OCCAM. 292-295 - David S. Broomhead, J. G. Harp, John G. McWhirter, K. J. Palmer, J. B. G. Roberts:
A practical comparison of the systolic and wavefront array processing architectures. 296-299 - José A. B. Fortes, King-Sun Fu, Benjamin W. Wah:
Systematic approaches to the design of algorithmically specified systolic arrays. 300-303
Parameter Estimation
- Aníbal R. Figueiras-Vidal, José R. Casar Corredera, Ramón García-Gómez, José Manuel Páez-Borrallo:
L1-Norm versus L2-Norm minimization in parametric spectral analysis: A general discussion. 304-307 - Xian-Ci Xiao:
An adaptive orthogonal maximum likelihood algorithm for parameter estimation. 308-311 - Stephen W. Lang:
Solving a class of nonlinear least squares problems. 312-315 - Boris Golubev, Steven R. Rogers:
Non-recursive frequency estimation for closely spaced sinusoids. 316-319 - Donald W. Tufts, Costas D. Melissinos:
Simple, effective computation of principal eigenvectors and their eigenvalues and application to high-resolution estimation of frequencies. 320-323 - Leland B. Jackson:
Approximate factorization of unfactorable spectral models. 324-326 - A. A. (Louis) Beex:
Efficient generation of ARMA cross covariance sequences. 327-330 - John W. Ketchum, David Herrick:
Signal detection using autoregressive parameters. 331-334 - Arye Nehorai, Boaz Porat:
Adaptive comb filtering for harmonic signal enhancement. 335-338
Image Sequence Coding
- H. F. Sun, M. Goldberg:
Adaptive vector quantization for image sequence coding. 339-342 - M. Reha Civanlar, P. Santago:
An improved transform coder for image sequences using attributes of difference pictures. 343-346 - Robert J. Moorhead II, Sarah A. Rajala:
Motion-compensated interframe coding. 347-350 - A. Farid Faryar, Sarah A. Rajala:
Transform/Time domain coding using the method of projection onto convex sets. 351-354 - Richard A. Jones, Carl D. Bowling:
A minimum risk quantizer for motion compensated image coding. 355-358 - Kou-Hu Tzou, To R. Hsing, Nancy A. Daly:
Block-recursive matching algorithm (BRMA) for displacement estimation of video images. 359-362 - Toshio Koga, A. Hirano, Yukihiko Iijima, Kazumoto Iinuma:
Motion-compensated adaptive intra-interframe predictive coding algorithm. 363-366 - Staffan Ericsson:
Motion-compensated hybrid coding at 50 kb/s. 367-370 - R. H. J. M. Plompen, B. F. Schuurink, Jan Biemond:
A new motion-compensated transform coding scheme. 371-374 - Dennis Martinez, Jae S. Lim:
Implicit motion compensated noise reduction of motion video scenes. 375-378
Speaker Recognition, and Pitch Extraction
- Herbert Gish, Kenneth F. Karnofsky, Michael A. Krasner, Salim E. Roucos, Richard M. Schwartz, Jared J. Wolf:
Investigation of text-independent speaker indentification over telephone channels. 379-382 - Stephanie S. Everett:
Automatic speaker recognition using vocoded speech. 383-386 - Frank K. Soong, Aaron E. Rosenberg, Lawrence R. Rabiner, Biing-Hwang Juang:
A vector quantization approach to speaker recognition. 387-390 - Joseph T. Buck, David K. Burton, John E. Shore:
Text-dependent speaker recognition using vector quantization. 391-394 - John Amuedo:
Periodicity estimation by hypothesis-directed search. 395-398 - Wolfgang Feix, M. DeGeorge:
A speaker verification system for access-control. 399-402 - Soon Young Kwon, Aaron J. Goldberg, D. Ng, K. Ouellette:
A robust realtime pitch extraction from the ACF of LPC residual error signals. 403-406 - Sanguoon Chung, V. Ralph Algazi:
Improved pitch detection algorithm for noisy speech. 407-410 - Vishu R. Viswanathan, William Russell:
New objective measures for the evaluation of pitch extractors. 411-414 - Helge Indefrey, Wolfgang Hess, Günter Seeser:
Design and evaluation of double-transform pitch determination algorithms with nonlinear distortion in the frequency domain-preliminary results. 415-418 - Schuyler R. Quackenbush, Thomas P. Barnwell III:
Objective estimation of perceptually specific subjective qualities. 419-422 - S. Feijóo, C. Hernández:
Quantification of dysphony with allowance for inter-utterance variation. 423-426 - Dirk Van Compernolle:
A computational model of the cochlea used with cochlear prosthesis patients. 427-429 - James T. Sims, John D. Tardelli:
The effects of controlled speech level input on the intelligibility testing of speech compression algorithms. 430-433
Deconvolution and Bandlimited Extrapolation
- Mark A. Richards:
Iterative deconvolution in noncoherent systems. 434-437 - B. W. Dahanayake, Kon Max Wong:
Deconvolution in the sequency domain. 438-441 - Gérard Thomas, C. Dussaussois, F. Buret, Ph. Auriol:
Deconvolution without system model or a new blind deconvolution. 442-444 - Tapan K. Sarkar, Fung I. Tseng, Soheil A. Dianat, Bruce Z. Hollmann:
Deconvolution by the conjugate gradient method. 445-448 - Peter M. Clarkson, Joe K. Hammond:
Time and frequency selective deconvolution using optimal control. 449-452 - J. Bee Bednar, Rao K. Yarlagadda, Terry L. Watt:
L1 deconvolution and its application to seismic signal processing. 453-456 - M. Reha Civanlar, H. Joel Trussell:
Signal deconvolution using fuzzy sets. 457-460 - Jesús M. Alcázar-Fernández, José R. Casar Corredera, Aníbal R. Figueiras-Vidal:
On absolute value minimization approaches to tauberian modelling. 461-464 - Hua Lee, Thomas S. Huang:
On descrete band-limited signal extrapolation. 465-468 - Barry J. Sullivan:
A comparison of three methods for discrete-time signal extrapolation. 469-472
Speech Analysis and Reconstruction
- Vijay K. Jain, Bishnu S. Atal:
Robust LPC analysis of speech by extended correlation matching. 473-476 - H. A. Hawkins, D. Mitchell Wilkes, Mark A. Clements, Monson H. Hayes III:
Perceptual weightings and optimal pulse positioning in multipulse LPC speech coding. 477-480 - Richard C. Rose, Mark A. Clements:
All-pole speech modeling with a maximally pulse-like residual. 481-484 - Richard J. Mammone, Kent Wang, Steven Gay:
LPC Speech analysis using the L1 norm. 485-488 - Thomas F. Quatieri, Robert J. McAulay:
Speech transformations based on a sinusoidal representation. 489-492 - Salim E. Roucos, Alexander MacLeod Wilgus:
High quality time-scale modification for speech. 493-496 - C. S. Chen, Kaliappan Gopalan, Pallabi Mitra:
Speech signal analysis and synthesis via Fourier-Bessel representation. 497-500 - Marie-Christine Omnes-Chevalier, C. Chollet, Yves Grenier:
Speech analysis and restitution using time-depedent autoregressive models. 501-504 - Oded Ghitza:
A measure of in-synchrony regions in the auditory nerve firing patterns as a basis for speech vocoding. 505-508 - Hynek Hermansky, Brian A. Hanson, Hisashi Wakita:
Perceptually based linear predictive analysis of speech. 509-512 - Daniel W. Griffin, Jae S. Lim:
A new model-based speech analysis/Synthesis system. 513-516
Time-Frequency Analysis and Reconstruction
- Amir Dembo, David Malah:
Statistical design of analysis/Synthesis systems with quantization. 517-520 - Mark J. T. Smith, Thomas P. Barnwell III:
A unifying framework for analysis/Synthesis systems based on maximally decimated filter banks. 521-524 - Claude R. Galand, Henri J. Nussbaumer:
Quadrature mirror filters with perfect reconstruction and reduced computational complexity. 525-528 - Sanjit K. Mitra, Yrjö Neuvo, P. P. Vaidyanathan:
Complementary IIR digital filter banks. 529-532 - James C. Anderson:
A filter/Detector interpretation of the short-time Fourier transform magnitude. 533-536 - Z. Shpiro, David Malah:
Design of filters for discrete short-time Fourier transform synthesis. 537-540 - J. Masson, Z. Picel:
Flexible design of computationaly efficient nearly perfect QMF filter banks. 541-544 - Kamal Jabbour, Jose Fernando Vega-Riveros:
Real-time telephone channel simulation. 545-547 - Leon Cohen:
Properties of the positive time-frequency distribution functions. 548-551
Array Processing and Beamforming I
- Don H. Johnson:
Properties of eigenanalysis methods for bearing estimation algorithms. 552-555 - Mati Wax, Thomas Kailath:
Extending the threshold of the eigenstructure methods. 556-559 - Henri Clergeot, Abdelaziz Ouamri, Sara Tressens:
High resolution spectral method for spatial discrimination of closely spaced correlated sources. 560-563 - Arogyaswami Paulraj, Thomas Kailath:
On beamforming in presence of multipath. 564-567 - Even Borten Lunde:
Normal mode propagation and high resolution methods. 568-571 - Ivars P. Kirsteins, Donald W. Tufts:
On the probability density of signal-to-noise ratio in an improved adaptive detector. 572-575 - Ramdas Kumaresan, Arnab K. Shaw:
High resolution bearing estimation without eigen decomposition. 576-579
Audio and Electroacoustics
- Hikaru Date, Kimitoshi Fukudome, Masakazu Oda, Setsuya Tokuriki:
Observation of impulse response by two relatively prime pseudorandom sequences. 580-583 - J. Robert Ashley, J. Matthew Ashley:
Catholic electroacoustical difficulties. 584-587 - Said E. El-Khamy, Onsy A. Abdel-Alim:
Omnidirectional coded planar loudspeaker arrays. 588-591 - Werner Rosenkranz:
Design and optimization of a digital FM receiver using DPLL techniques. 592-595 - A. Maynard Engebretson, Michael P. O'Connell:
Implementation of a real-time, digital vocoder for tactile hearing prostheses. 596-599 - David G. Meyer:
Signal processing architecture for loudspeaker array directivity control. 600-603 - I. L. Veach, J. J. Narraway:
Reconstruction of tonal sequences. 604-607 - Matti Karjalainen:
A new auditory model for the evaluation of sound quality of audio systems. 608-611 - John Charles Cox:
The maximum tolerable delay of speech and music. 612-615
Estimation and Performance
- Benjamin Friedlander, Boaz Porat:
Bounds for ARMA spectral analysis based on sample covariances. 616-619 - T. Hediger, A. Passamante:
The role of spectral decomposition in the pattern recognition of narrowband signals. 620-623 - D. V. Bhaskar Rao:
Perturbation analysis of a SVD based method for the harmonic retrieval problem. 624-627 - Hideaki Sakai, Hidekatsu Tokumaru:
Statistical properties of coherence and power contribution ratios via multivariate autoregressive modeling. 628-631 - Anders Johansson, L. Gunnar Ahlbom, Lars-Henning Zetterberg:
Event detection using recursively updated lattice filters. 632-635 - Hong Wang, Mos Kaveh:
Sensitivity and performance analysis of coherent signal-subspace processing for multiple wideband sources. 636-639 - Arogyaswami Paulraj, Thomas Kailath:
Direction of arrival estimation by eigenstructure methods with unknown sensor gain and phase. 640-643 - M. K. Ibrahim, Costas E. Goutis:
Detection algorithms based on prediction error - additive noise - data orthogonality. 644-647 - Ravi Kumar:
On nonlinear estimation in presence of non-Gaussian Noise. 648-651
Image Estimation and Restoration
- Rama Chellappa, Hao Jinchi:
A nonrecursive filter for edge preserving image restoration. 652-655 - A. Murat Tekalp, Howard Kaufman, John W. Woods:
Identification of image and blur parameters for the restoration of noncausal blurs. 656-659 - Jan Biemond, F. G. van der Putten:
Image restoration using a parallel indentification and filtering procedure. 660-663 - Richard M. Leahy, Costas E. Goutis:
Optimal techniques for constraint based signal restoration and image reconstruction. 664-667 - Regis J. Crinon:
The Wilcoxon filter: A robust filtering scheme. 668-671 - Steven R. Peterson, Saleem A. Kassam:
Edge preserving signal enhancement using generalizations of order statistic filtering. 672-675 - Chia-Lung Yeh, Roland T. Chin:
Constrained optimization for image restoration using nonlinear programming. 676-679 - Yu-Shan Fong, A. Saha, Carlos A. Pomalaza-Raez:
Post-reconstruction correction in quantitative single photon ECT. 680-683 - K. Mike Tao, Fred M. Weinhaus:
Adaptive noise cancelling applied to image restoration. 684-687 - John W. Woods, Fure-Ching Jeng:
Hierarchical approach to image estimation. 688-691 - John W. Woods, Jan Biemond, A. Murat Tekalp:
Boundary value problem in image restoration. 692-695 - Aggelos K. Katsaggelos, Jan Biemond, Russell M. Mersereau, Ronald W. Schafer:
Nonstationary iterative image restoration. 696-699 - Aggelos K. Katsaggelos, Jan Biemond, Russell M. Mersereau, Ronald W. Schafer:
A general formulation of constrained iterative restoration algorithms. 700-703 - Victor T. Tom, Mark J. Carlotto:
Adaptive least-squares technique for multi-band image enhancement. 704-707 - Levent Onural, Peter D. Scott:
A digital filtering system for decoding in-line holograms. 708-711
Speech Enhancement and Synthesis
- Vishu R. Viswanathan, Claudia M. Henry, Alan G. Derr:
Noise-immune speech transduction using multiple sensors. 712-715 - P. Darlington, P. D. Wheeler, G. A. Powell:
Adaptive noise reduction in aircraft communication systems. 716-719 - Joy A. Thomas, B. Yegnanarayana, Raghuram Karinthi, V. Venkateswar:
Processing of noisy speech using group delay functions. 720-723 - Les E. Atlas, Leo M. Hengky:
Cross-channel correlation for the enhancement of noisy speech. 724-727 - S. Thomas Alexander:
Adaptive reduction of interfering speaker noise using the least mean squares algorithm. 728-731 - James L. Flanagan:
Bandwidth design for speech-seeking microphone arrays. 732-735 - Xavier Rodet, Philippe Depalle:
Synthesis by rule: LPC diphones and calculation of formant trajectories. 736-739 - M. G. Stella, Francis Charpentier:
Diphone synthesis using multipulse coding and a phase vocoder. 740-743 - Georg Dorffner, Markus Kommenda, Gernot Kubin:
GRAPHON-the Vienna speech systhesis system for arbitrary German text. 744-747 - D. G. Childers, B. Yegnanarayana, Ke Wu:
Voice conversion: Factors responsible for quality. 748-751 - Helmut Dettweiler, Wolfgang Hess:
Concatenation rules for demisyllable speech synthesis. 752-755
Fast Transforms
- K. M. M. Prabhu:
Optimum binary windows for discrete Fourier transform. 756-759 - Soo-Chang Pei, Ja-Ling Wu:
Exact fast digital convolution by using P-adic numbers and polynomial transformations. 760-763 - Ramasamy Krishnan, Graham A. Jullien, William C. Miller:
Complex digital signal processing using quadratic residue number systems. 764-767 - Ramdas Kumaresan:
Efficient architectures for implementing the prime-factor Fourier transform modules. 768-771 - Paul P. N. Yang, M. J. Narasimha:
Prime factor decomposition of the discrete cosine transform and its hardware realization. 772-775 - Patrick C. Yip, Kamisetty Ramamohan Rao:
DIF Algorithms for DCT and DST. 776-779 - Michael T. Heideman, C. Sidney Burrus:
Multiply/Add tradeoffs in length-2nFFT algorithms. 780-783 - Pierre Duhamel, Henk D. L. Hollmann:
Implementation of "Split-radix" FFT algorithms for complex, real, and real symmetric data. 784-787 - G. Robert Redinbo, Kotesh K. Rao:
The chord property speeds finite field FFTs. 788-791
Multi-Channel and Multi-Dimensional Spectral Analysis
- Chrysostomos L. Nikias, Anastasios N. Venetsanopoulos:
Sufficient condition for extendibility and two-dimensional power spectrum estimation. 792-795 - José R. Casar Corredera, Jesús M. Alcázar-Fernández, Luis Alfonso Hernández Gómez:
On 2-D prony methods. 796-799 - Taikang Ning, Chrysostomos L. Nikias:
The optimum approach to multichannel AR spectrum estimation. 800-803 - Govind Sharma, Rama Chellappa:
Simultaneous confidence bands for a class of 2-D spectral estimates. 804-807 - Gregory H. Wakefield, M. Kaveh:
Frequency-wavenumber spectral estimation of non-planar random fields. 808-811 - Sverre Holm:
Phase errors in the cross spectrum estimate due to misalignment. 812-815 - Hanoch Lev-Ari:
Multidimensional maximum-entropy covariance extension. 816-819 - Farid U. Dowla, Jae S. Lim:
Resolution property of the improved maximum likelihood method. 820-822
Geophysical DSP and Medium Models
- William S. Hodgkiss, Dimitrios Alexandrou:
Under-ice reverberation rejection. 823-825 - Jules S. Jaffe, Lisa Strong:
Application of a super resolution algorithm to acoustic tomography multipath data. 826-829 - Ioannis Pitas, Anastasios N. Venetsanopoulos:
Bayesian estimation of medium properties in wavefield inversion techniques. 830-833 - David J. Scheibner, Thomas W. Parks:
Inversion of modified beamformed array data. 834-837 - Y. T. Chan, L. G. Gadbois, P. Yansouni:
Identification of the modulation type of a signal. 838-841
Isolated Word Recognition II
- Hirosi Iizuka:
Speaker independent telephone speech recognition. 842-845 - Marcia A. Bush, Gary E. Kopec:
Evaluation of a network-based isolated digit recognizer using the TI multi-dialect database. 846-849 - Louis C. Sauter:
Isolated word recognition using a segmental approach. 850-853 - Dieter Mergel, Hermann Ney:
Phonetically guided clustering for isolated word recognition. 854-857 - Frederick Jelinek:
A real-time, isolated-word, speech recognition system for dictation transcription. 858-861 - Helmut Lagger, Alex Waibel:
A coarse phonetic knowledge source for template independent large vocabulary word recognition. 862-865 - Shuji Morii, Katsuyuki Niyada, Satoru Fujii, Masakatsu Hoshimi:
Large vocabulary speaker-independent Japanese speech recognition system. 866-869 - Aaron E. Rosenberg:
Recognition error measurements from parameterized distance distributions. 870-873 - Kuk-Chin Pan, Frank K. Soong, Lawrence R. Rabiner, A. F. Bergh:
An efficient vector-quantization preprocessor for speaker independent isolated word recognition. 874-877 - Wissam W. Ahmed, George A. Bekey:
A new algorithm for isolated word recognition with large within-class variability. 878-881 - Periagaram K. Rajasekaran, George R. Doddington:
Speech recognition in the F-16 cockpit using principal spectral components. 882-885 - Martin S. Glassman:
Hierarchical DP for word recognition. 886-889
Feature Extraction, Segmentation and Scene Analysis
- Harley R. Myler, Wiley E. Thompson, Gerald M. Flachs:
Knowledge base enhancement of visual tracking. 890-892 - Alan C. Bovik, David C. Munson Jr.:
Boundary detection in speckle images. 893-896 - Fernand Cohen:
Adaptive hierarchical algorithm for accurate image segmentation. 897-900 - Roberto Cristi, Malayappan Shridhar, M. V. Prasadarao:
Segmentation of multilevel images using Gibbs distribution. 901-904 - Jorge L. C. Sanz:
Fast object segmentation in textured backgrouds. 905-908 - D. B. Sharman, Tariq S. Durrani:
Knowledge based object detection. 909-912 - Haluk Derin, Howard Elliott, J. Kuang:
A new approach to parameter estimation for Gibbs random fields. 913-916 - David Cyganski, John A. Orr:
3-D Motion parameters from contours using a canonic differential. 917-920 - John A. Orr, David Cyganski, Richard F. Vaz:
Determination of affine transforms from object contours with no point correspondence information. 921-924 - Fernand S. Cohen, Raymond D. Rimey, Jean-François Cayula:
Segmentation of three dimensional range data. 925-928 - Xinhua Zhuang, Robert M. Haralick:
Two view motion analysis under a small perturbation. 929-932 - Patrick L. Van Hove, Jacques G. Verly:
A silhouette-slice theorem for opaque 3-D objects. 933-936
Medium Band Speech Coding I
- Manfred R. Schroeder, Bishnu S. Atal:
Code-excited linear prediction(CELP): High-quality speech at very low bit rates. 937-940 - Guen Oyama:
A stochastic model excitation source for linear prediction speech analysis-synthesis. 941-944 - Robert J. McAulay, Thomas F. Quatieri:
Mid-rate coding based on a sinusoidal representation of speech. 945-948 - Amir Dembo, David Malah:
A new approach to multipulse LPC coder design. 949-952 - Ramón García-Gómez, Jesús M. Alcázar-Fernández:
A linear programming approach to multipulse speech coding. 953-956 - Jean-Paul Lefèvre, Olivier Passien:
Efficient algorithms for obtaining multipulse excitation for LPC coders. 957-960 - Akira Ichikawa, Shoichi Takeda, Yoshiaki Asakawa:
A speech coding method using thinned-out residual. 961-964 - Ed F. Deprettere, Peter Kroon:
Regular excitation reduction for effective and efficient LP-coding of speech. 965-968 - Richard L. Zinser:
An efficient, pitch-aligned high-frequency regeneration technique for RELP vocoders. 969-972 - P. J. Wilson, J. M. Puetz, M. D. Dankberg:
The transmission of in-band signaling for medium band voice codec implementations. 973-976
VLSI for Speech and Image Processing
- Seth Z. Kalson, Kung Yao:
A systolic array for linearly constrained least squares filtering. 977-980 - Richard F. Lyon, Niels Lauritzen:
Processing speech with the multi-serial signal processor. 981-984 - Robert E. Owen:
A VLSI dynamic time warp processor for connected and isolated word speech recognition. 985-988 - Z. A. Putnins, Gene A. Wilson, Jitendra Kumar, R. D. Trupp:
A multi-pulse LPC synthesizer for telecommunications use. 989-992 - John A. Eldon, Zoltan Stroll, Earl E. Swartzlander Jr.:
Image processing address generator chip. 993-996 - Amlan Kundu, Gururaj Singh, Steven Butner:
VLSI Implementation of two-dimensional generalized mean filter. 997-1000 - Nicolas Demassieux, Francis Jutand, Marc Saint-Paul, Michel Dana:
VLSI Architecture for a one chip video median filter. 1001-1004 - S. Lee, Anastasios N. Venetsanopoulos:
A two-dimensional digital filter chip set for modular two-dimensional filter implementation. 1005-1008 - P. A. Ramamoorthy, Tau Chen:
Systolic architectures based on barrel shifters for real-time signal and image processing. 1009-1012
Modeling of Time-Varing Signals
- Raymond N. J. Veldhuis, A. J. E. M. Janssen, Lodewijk B. Vries:
Adaptive restoration of unknown samples in certain time-discrete signals. 1013-1016 - Steven Kay, Gloria Faye Boudreaux-Bartels:
On the optimality of the Wigner distribution for detection. 1017-1020 - N. M. Marinovic, G. Eichmann:
An expansion of Wigner distribution and its applications. 1021-1024 - Joe K. Hammond, R. F. Harrison:
Wigner-Ville and evolutionary spectra for covariance equivalent nonstationary random processes. 1025-1028 - Franz Hlawatsch:
Transformation, inversion and conversion of bilinear signal representations. 1029-1032 - Leon Cohen, Theodore E. Posch:
Generalized ambiguity functions. 1033-1036 - Kai-Bor Yu, Siuling Cheng:
Signal synthesis from Wigner distribution. 1037-1040 - Dean J. Schmidlin:
A generalized delay operator for shift-variant systems. 1041-1044 - David Chester, JoEllen Wilbur:
Time and spatial varying CAM and AI signal analysis using the Wigner distribution. 1045-1048 - Marie-Christine Omnes-Chevalier, Yves Grenier:
Autoregressive models with time-dependent log area ratios. 1049-1052
Reconstruction and Computerized Tomography
- Mehrdad Soumekh:
A method of image reconstruction in fan beam tomography. 1053-1056 - Susan R. Curtis, Alan V. Oppenheim, Jae S. Lim:
Reconstruction of two-dimensional signals from threshold crossings. 1057-1060 - M. Ibrahim Sezan, Henry Stark:
Incorporation of a priori moment constraints into signal recovery and synthesis problems via the method of convex projections. 1061-1064 - Jorge L. C. Sanz, Thomas S. Huang:
On the stability and sensitivity of multidimensional signal reconstruction from Fourier transform magnitude. 1065-1068 - W. Kenneth Jenkins, Bruce C. Mather, David C. Munson Jr.:
Nearest neighbor and generalized inverse distance interpolation for Fourier domain image reconstruction. 1069-1072 - Barry P. Medoff:
An inner product framework for image reconstruction. 1073-1076 - R. S. Acharya, P. B. Heffernan, Richard A. Robb:
Image reconstruction of the heart using a priori information and spatiotemporal estimation. 1077-1080 - Tsuneo Saito, Takayoshi Chiba:
An image reconstruction from incomplete observation by constrained spectrum extrapolation. 1081-1084
Vocal Tract and Speech Analysis
- Bert Cranen, Louis Boves:
Aerodynamic aspects of voicing: Glottal pulse skewing revisited. 1085-1088 - Jerry N. Larar, Yacoub A. Alsaka, D. G. Childers:
Variability in closed phase analysis of speech. 1089-1092 - A. S. Ananth, D. G. Childers, B. Yegnanarayana:
Measuring source-tract interaction from speech. 1093-1096 - Francis Charpentier:
Analysis of vocal tract lip reflectance by linear prediction. 1097-1100 - Roman Kuc, Franz B. Tuteur, J. Rimas Vaisnys:
Determining vocal tract shape by applying dynamic constraints. 1101-1104 - Trevor Thomas, Frank Fallside:
A new articulatory model for speech production. 1105-1108 - Karen L. Payton:
Speech processing by a model of the auditory periphery. 1109-1112 - Gary E. Kopec:
Formant tracking using hidden Markov models. 1113-1116 - Melvyn J. Hunt:
A robust formant-based speech spectrum comparison measure. 1117-1120 - David H. Friedman:
Instantaneous-frequency distribution vs. time: An interpretation of the phase structure of speech. 1121-1124 - C. David Covington:
Cubic spline modeling of speech spectra. 1125-1128 - G. A. Mack, Vijay K. Jain:
A compensated-Kalman speech parameter estimator. 1129-1132 - Nobuhiro Miki, Yoshikazu Miyanaga, Sato Saga, Nobuo Nagai:
Spectrum and pitch estimation of speech using a time-varying ARMA estimation algorithm. 1133-1136 - G. A. Mack, Vijay K. Jain:
Bayesian deconvolution of speech containing pulsed excitation. 1137-1140 - Per Lunde:
Acoustic transmission-line analysis of formants in hyperbaric Helium speech. 1141-1144
Adaptive Filtering I
- Evangelos Eleftheriou, David D. Falconer:
Steady-state behavior of RLS adaptive algorithms. 1145-1148 - Philippe Fabre, Claude Guéguen:
Fast recursive least-squares algorithms: Preventing divergence. 1149-1152 - Maurice G. Bellanger, Cumhur Cengiz Evci:
An efficient step size adaptation technique for LMS adaptive filters. 1153-1156 - John R. Treichler, Michael G. Larimore:
Convergence rates for the constant modulus algorithm with sinusoidal inputs. 1157-1160 - Julius O. Smith III, Benjamin Friedlander:
Global convergence of the constant modulus algorithm. 1161-1164 - Michael G. Larimore, John R. Treichler:
Noise capture properties of the constant modulus algorithm. 1165-1168 - Delores M. Etter:
Identification of sparse impulse response systems using an adaptive delay filter. 1169-1172 - T. Gardiner, John G. McWhirter, T. J. Shepherd:
"Noise cancellation studies using a least-squares lattice filter". 1173-1176 - Fathy F. Yassa:
A generalized filter structure for IIR adaptive filters. 1177-1180 - Dae Hee Youn, V. John Mathews, Sung Ho Cho:
An efficient algorithm for lattice filter/Predictor. 1181-1184 - Arye Nehorai:
A minimal parameter adaptive notch filter with constrained poles and zeros. 1185-1188 - David C. Swanson, Frank W. Symons Jr.:
The unbiased least-squares lattice. 1189-1192 - Don R. Hush, Nasir Ahmed:
Detection and identification of sinusoids in broadband noise via a parallel recursive ALE. 1193-1196
Continuous Speech Recognition
- Marcia A. Bush, Gary E. Kopec:
Network-based connected digit recognition using vector quantization. 1197-1200 - A. Richard Smith, J. N. Denenberg, T. B. Slack, C. C. Tan, Robert E. Wohlford:
Application of a sequential pattern learning system to connected speech recognition. 1201-1204 - Richard M. Schwartz, Yen-Lu Chow, Owen Kimball, Salim E. Roucos, Michael A. Krasner, John Makhoul:
Context-dependent modeling for acoustic-phonetic recognition of continuous speech. 1205-1208 - Hermann Ney:
A script-guided algorithm for the automatic segmentation of continuous speech. 1209-1212 - Hervé Bourlard, Yves G. Kamp, Christian Wellekens:
Speaker dependent connected speech recognition via phonetic Markov models. 1213-1216 - Jean-Paul Brassard:
Integration of segmenting and nonsegmenting approaches in continuous speech recognition. 1217-1220 - Ryu-ichi Oka:
Continuous speech recognition on the bases of vector field model for segmentation and feature extraction, and continuous dynamic programming for pattern matching. 1221-1224 - Renato De Mori, Giorgio Rossi, Jianli Sun:
Multi-speaker computer recognition of ten connectedly spoken letters. 1225-1228 - Anna Maria Colla, Carlo Scagliola, Donatella Sciarra:
A connected speech recognition system using a diphone-based language model. 1229-1232 - Alan L. Higgins, Robert E. Wohlford:
Keyword recognition using template concatenation. 1233-1236
Adaptive Filtering II
- Camille C. Price, Moktar A. Salama:
Parallel algorithms for Toeplitz matrix operations. 1237-1240 - John M. Cioffi:
The block-processing FTF adaptive algorithm. 1241-1244 - José R. Casar Corredera, Mariano García Otero, Aníbal R. Figueiras-Vidal:
Data echo nonlinear cancellation. 1245-1248 - Emmanuel Fernández-Gaucherand, J. R. Cruz:
Equalization for transmission line channels: A discussion of three IIR adaptive filtering algorithms. 1249-1252 - Richard L. Zinser Jr., Gagan Mirchandani, Joseph B. Evans:
Some experimental and theoretical results using a new adaptive filter structure for noise cancellation in the presence of crosstalk. 1253-1256 - J. W. Lee, C. K. Un, J. C. Lee:
Adaptive digital filtering of differentially coded signals. 1257-1260 - Guy R. L. Sohie:
Adaptive systems as optimal processors. 1261-1262
Detection and Estimation
- Steven Kay:
Broadband detection of signals with unknown spectra. 1263-1265 - Boaz Porat, Benjamin Friedlander:
Adaptive detection of transient signals. 1266-1269 - Chi Hau Chen:
Automatic recognition of underwater transient signals - a review. 1270-1272 - Allan O. Steinhardt, Chris Bretherton:
Thresholds in frequency estimation. 1273-1276 - B. J. Erickson, G. W. Johnson, D. E. Ohlms:
Detection of a sinusoidal signal in the presence of directional interference. 1277-1280 - M. Weiss, Stuart C. Schwartz:
Robust scale invariant detection of coherent narrowband signals in nearly Gaussian noise. 1281-1284 - Roger F. Dwyer:
Two-dimensional arrays with nonlinear elements. 1285-1288 - Michel Bouvet, Eugenio J. Tacconi, Bernard Picinbono:
Microscopic correlation signals. 1289-1292 - Jean Le Gall:
Sonar detection in Weibull bottom reverberation. 1293-1296 - Chong-Yung Chi, John Goutsias, Jerry M. Mendel:
A fast maximum-likelihood estimation and detection algorithm for Bernoulli-Gaussian processes. 1297-1300
Multidimensional Filtering
- Hsing-Hsing Chiang, Chrysostomos L. Nikias:
Parallel block realization of 2-D IIR digital filters. 1301-1304 - Tyseer Aboulnasr, Moustafa M. Fahmy:
Conjugate transformations preserving the general overflow-stability property of 2-D digital filters. 1305-1308 - David A. Border, Subhash C. Kwatra:
Separating Y, I, Q components of NTSC composite signal. 1309-1312 - Joseph Barba, Norman Scheinberg, M. Colef, Erlan H. Feria:
Hadamard transform seperation of NTSC component signals. 1313-1316 - Hanoch Lev-Ari, Sydney R. Parker:
Lattice-filter modeling of two-dimensional fields. 1317-1320 - P. Karivaratha Rajan, Harnatha C. Reddy:
Application of symmetrical decomposition to 2-D FIR filter design. 1321-1324 - R. Lynn Kirlin, Becky Cudzilo, Sharon Wilson:
Two-dimensional orthogonal median filters and applications. 1325-1328 - Petros Maragos, Ronald W. Schafer:
A unification of linear, median, order-statistics and morphological filters under mathematical morphology. 1329-1332 - Tran Thong:
Frequency domain analysis of two-pass rotation algorithm. 1333-1336
Topics in Estimation and Radar
- Bernard Picinbono, Michel Bouvet:
Constrained prediction. 1337-1340 - James A. Cadzow, Otis M. Solomon Jr., Samuel D. Stearns:
Rational parametric coherence estimation via convolved correlations. 1341-1344 - Paul F. Fougere:
Spectrum model-order determination via significant reflection coefficients. 1345-1347 - Sergio D. Cabrera, Thomas W. Parks:
Estimation of sequences in a signal class determined from the data. 1348-1351 - Mysore R. Raghuveer, Chrysostomos L. Nikias:
Bispectrum estimation for short length data. 1352-1355 - Gianni Orlandi, Giuseppe Martinelli, Pietro Burrascano:
Explicit formulas for super-resolution. 1356-1359 - Stanley M. Yuen, Harish M. Subbaram:
A new super-resolution spectral estimation technique using staggered PRFs. 1360-1363 - David C. Munson Jr., Jorge L. C. Sanz, W. Kenneth Jenkins, Gary Kakazu, Bruce C. Mather:
A comparison of algorithms for polar-to-cartesian interpolation in spotlight mode SAR. 1364-1367 - Siew Kok Hui, Yong Ching Lim:
A block adaptive approach for clutter suppression. 1368-1371 - Oktay Alkin, Ronald C. Houts:
Comparing the radar clutter-suppression performances of lattice prediction error filters using three variations of Burg's algorithm. 1372-1375
VLSI Architecture
- N. Venkateswaran, K. M. M. Prabhu:
Programming techniques for PA3architectures. 1376-1379 - Hanafy Meleis, Pierre Le Fur:
A novel architecture design for VLSI implementation of an FIR decimation filter. 1380-1383 - D. A. Schwartz, Thomas P. Barnwell III:
Cyclo-static multiprocessor scheduling for the optimal realization of shift-invariant flow graphs. 1384-1387 - Jaw John Chang, Trieu-Kien Truong, Howard M. Shao, Irving S. Reed, In-Shek Hsu:
The VLSI design of a single chip for the multiplication of integers modulo a fermat number. 1388-1391 - J. Greg Nash, C. Petrozolin:
VLSI Implementation of a linear systolic array. 1392-1395 - Ronald G. Harber, Steven C. Bass, Gerold W. Neudeck:
VLSI Implementation of a fast rank order filtering algorithm. 1396-1399 - Akira Yukawa, Rikio Maruta, Kenji Nakayama:
An oversampling A-to-D converter structure for VLSI digital codec's. 1400-1403 - Howard M. Shao, Trieu-Kien Truong, Leslie J. Deutsch, Joseph H. Yuen, Irving S. Reed:
A VLSI design of a pipeline Reed-Solomon decoder. 1404-1407 - Costas E. Goutis, J. S. Sheblee, G. Russell:
2-D Array processor having a controlled pipelined architecture for elliptical sparse matrices. 1408-1411 - David Hayes, Bill Strawhorne:
The application of multi-dimensional access memories to ultra high performance signal processing systems. 1412-1415
Speech Processing Hardware Implementation
- David Y. Wong, David A. Russo, Carl D. Bergman, C. H. Lee, David M. Lindsay:
Signal processing software for a voice messaging system using the TMS32010 processor. 1417-1420 - R. Hangartner, Vijay K. Jain:
32kbs ADPCM/PCM transcoder using TI-320 DSP microprocessor. 1421-1424 - Takao Nishitani, Ichiro Kuroda, Masao Satoh, Tadaharu Katoh, Reiichi Fukuda, Yasushi Aoki:
A CCITT standard 32 kbps ADPCM LSI codec. 1425-1428 - Y. Wake, S. Tanaka, Kazunori Ozawa, Takashi Araseki:
A multi-pulse LPC speech codec using digital signal processors. 1429-1432 - Marc B. DonVito, Brian W. Schoenherr:
Subband coding with silence detection. 1433-1436 - Grant A. Davidson, Terry Stanhope, R. Aravind, Allen Gersho:
Real-time speech compression with a VLSI vector quantization processor. 1437-1440 - Jing Yuan, Jing Zheng Ou-Yang:
On a delta modulation based real time autocorrelator. 1441-1444 - John G. Ackenhusen, Young-Hwan Oh:
Single chip implementation of feature measurement for LPC-based speech recognition. 1445-1448 - Mark A. Yoder, Leah H. Jamieson:
Simulation of a highly parallel system for word recognition. 1449-1452
VLSI Design Methodology and Transform Architecture
- Kazuo Iwano, Kenneth Steiglitz:
Time-power-area tradeoffs for the nMOS VLSI full-adder. 1453-1456 - Magdy A. Bayoumi, Graham A. Jullien, William C. Miller:
An efficient VLSI adder for DSP architectures based on RNS. 1457-1460 - Charles E. Hauck, Cyrus Bamji, Jonathan Allen:
The systematic exploration of pipelined array multiplier performance. 1461-1464 - Rajeev Jain, Gert Goossens, Luc J. M. Claesen, Joos Vandewalle, Hugo De Man, L. Gazsi, Alfred Fettweis:
CAD Tools for the optimized design of custom VLSI wave digital filters. 1465-1468 - J. S. Ward, John V. McCanny, John G. McWhirter:
A systolic implementation of the Winograd Fourier transform algorithm. 1469-1472 - D. J. Spreadbury, T. M. Rees-Roberts:
VLSI Gate array prime radix Fourier transform processor. 1473-1476 - David Bondurant, Roger Cox, Grant Deming, David Wick:
FFT Drithmetic element built on VLSI high performance gate array. 1477-1480 - Kostas O. Siomalas, B. Archie Bowen:
A proposal for very high performance FFT processor architectures. 1481-1484
System Identification
- Joe K. Hammond, P. Davies:
Bounds on the envelopes of the response of systems to bandlimited inputs. 1485-1488 - Konstantinos Konstantinides, Kung Yao:
Confidence regions for perturbed singular values in system identification. 1489-1492 - Hanoch Lev-Ari:
Orthogonality of oblique projections and lattice-form models. 1493-1496 - K. A. Stewart, Tariq S. Durrani, J. B. Abbiss:
The effects of bandwidth MIS-estimation in bandlimited signal extrapolation. 1497-1500 - Ken C. Sharman, Tariq S. Durrani:
Resolving power of signal subspace methods for finite data lengths. 1501-1504 - Wasfy B. Mikhael, Andreas S. Spanias, Frank H. Wu:
A two stage approach for adaptive prediction of ARMA processes. 1505-1508 - S. Lawrence Marple Jr.:
An extension of the fast algorithm for linear phase system identification. 1509-1510
Image Processing Hardware and Multi-Dimensional Transforms
- T. A. Nodes, J. L. Smith, R. Hecht-Nielsen:
A fuzzy associative memory module and its application to signal processing. 1511-1514 - Jorge L. C. Sanz, Eric B. Hinkle, Its'hak Dinstein:
On the computation and use of projections of digital images in general purpose image processing pipeline architectures. 1515-1518 - James A. Roskind:
A fast sort-selection filter chip with effectively linear hardware complexity. 1519-1522 - M. Del Sordo, Tony Kasvand:
A near-neighbor processor for line thinning. 1523-1526 - Kich Man Ty, Anastasios N. Venetsanopoulos:
Two-dimensional digital filters with minimum cycle time. 1527-1530 - Ramdas Kumaresan, Prabhat Kumar Gupta:
Vector-radix algorithm for a 2-D discrete Hartley transform. 1531-1534 - Abderrezak Guessoum, Russell M. Mersereau:
Solution to the indexing problem of multidimensional DFT's on arbitrary sampling lattices. 1535-1537 - Martin Vetterli:
Fast 2-D discrete cosine transform. 1538-1541 - Bruce W. Suter, Stanley R. Deans:
A Hankel transform algorithm for uniformly sampled data. 1542-1545 - Kaliappan Gopalan, C. S. Chen:
Computation of the two-dimensional Fourier transform of circularly symmetric functions. 1546-1548
Phonetic Analysis and Data Base
- Ann Marie Aull, Victor W. Zue:
Lexical stress determination and its application to large vocabulary speech recognition. 1549-1552 - Jared Bernstein, Margaret Kahn, Tito Poza:
Speaker sampling for enhanced diversity. 1553-1556 - Maria Domenica Di Benedetto, Maria-Gabriella Di Benedetto:
A study on vowel behaviour and its description by a statistical model. 1557-1560 - Seppo Haltsonen, Pekka Ruusunen:
Collection of phoneme samples using time alignment and spectral stationarity of speech signals. 1561-1564 - Jean-Paul Haton, Jean-Paul Damestoy:
A frame language for the control of phonetic decoding in continuous speech recognition. 1565-1568 - James R. Glass, Victor W. Zue:
Detection of nasalized vowels in American English. 1569-1572 - James L. Hieronymus, William J. Majurski:
A reference speech recognition algorithm for benchmarking and speech data base analysis. 1573-1576 - Anne-Marie Derouault, Bernard Mérialdo:
Probabilistic grammar for phonetic to French transcription. 1577-1580 - David B. Pisoni, Robert H. Bernacki, Howard C. Nusbaum, Moshe Yuchtman:
Some acoustic-phonetic correlates of speech produced in noise. 1581-1584 - Takao Watanabe:
Speaker-independent connected Japanese digit recognition based on phonetic approach. 1585-1588 - Benjamin B. Wells:
Voiced/Unvoiced decision based on the bispectrum. 1589-1592 - D. Y. Yeung, C. Chan:
Identification of unaspirated plosives using integrated temporal and spectral features in dynamic representation as acoustic cues. 1593-1596 - Yutaka Kobayashi, Yasuhisa Niimi:
Matching algorithms between a phonetic lattice and two types of templates - Lattice and graph. 1597-1600
Signal Processing Hardware and Software
- David P. Morgan, Harvey F. Silverman:
An investigation into the efficiency of a parallel TMS320 architecture: DFT and speech filterbank applications. 1601-1604 - Thomas K. Miller III, S. Thomas Alexander:
An implementation of the LMS adaptive filter using an SIMD multiprocessor ring architecture. 1605-1608 - S. Ganesan, M. Omair Ahmad, M. N. S. Swamy:
A multimicroprocessor system with distributed common memory for real-time digital correlation and spectrum analysis. 1609-1612 - D. Degryse, F. Druilhe, André Gilloire:
A multi-processor structure for signal processing application to acoustic echo cancellation. 1613-1616 - Giovanni L. Sicuranza, Giovanni Ramponi:
Distributed arithmetic implementation of nonlinear echo cancellers. 1617-1620 - Gary Albert, David Fronczak, Jay McKinney:
High throughput signal processing system. 1621-1624 - Louis Schirm IV, Richard de Koeyer:
GOPSTMDigital signal processor provides 20 MHz analog bandwidth. 1625-1628 - G. P. Klowak, S. Cohn-Sfetcu, Willem J. D. Steenaart:
Effective multifrequency receiver design. 1629-1632 - Roger W. Cain:
Microprocessor based 9600 bps modem. 1633-1636 - T. A. Lanfear, M. G. X. Fernando, Les J. Wu:
Simulation of signal flow graphs for signal processing systems. 1637-1640 - Mohsin M. Jamali, Graham A. Jullien, William C. Miller, S. I. Ahmad:
Software techniques for programming a general purpose data flow signal processor. 1641-1644 - Charles H. Rogers, Kwang-Shik Min, Steven Speier, John Whitson:
A transportable TMS32010 signal processing system. 1645-1647 - L. Robert Morris:
Structural considerations for large FFT programs on the TI TMS 32010 DSP microchip. 1648-1651 - Glenda S. Poston, Robert J. Fornaro:
Concurrent process structured software models for time domain harmonic scaling of speech. 1652-1655 - Ali Vaghar, Veljko Milutinovic:
An analysis of algorithms for microprocessor implementation of high-speed data modems. 1656-1659 - Evangelos E. Milios, S. Hamid Nawab:
Interpretation-guided signal processing via protocol analysis. 1660-1663 - S. H. Lee, C. J. M. Hodges, Thomas P. Barnwell III:
An SSIMD compiler for the implementation of linear shift-invariant flow graphs. 1664-1667 - Stevan Eidson:
A resonant section filter design method for optimized coefficients in add-and-shift architectures. 1668-1671
Medium Band Speech Coding II
- Frank K. Soong, Richard V. Cox, Nikil S. Jayant:
Subband coding of speech using backward adaptive prediction and bit allocation. 1672-1675 - P. Hamel, J. Soumagne, Alain Le Guyader:
A new dynamic bit allocation scheme for sub-band coding. 1676-1679 - Jeffrey H. Derby, Claude Galand:
Multirate sub-band coding applied to digital speech interpolation. 1680-1683 - Mamoru Nakatsui, Kazuo Nakata:
Dual adaptive delta modulation for mobile voice channel and its DSP implementation. 1684-1687 - Kuldip K. Paliwal, Torbjørn Svendsen:
A study of three coders (sub-band, RELP and MPE) for speech with additive white noise. 1688-1691 - Michael Dellomo, JoAnn B. Hoyt, George M. Shuttic:
Critical point coding: Design and real-time implementation at 16 kbps. 1692-1695 - Yair Shoham, Allen Gersho:
Efficient codebook allocation for an arbitrary set of vector quantizers. 1696-1699 - Herbert Reininger, Dietrich Wolf:
Speech and speaker independent codebook design in VQ coding schemes. 1700-1702 - Amine Haoui, David G. Messerschmitt:
Embedded coding of speech: A vector quantization approach. 1703-1706 - Thomas R. Fischer, Kevin T. Malone:
Contour vector quantization and waveform coding. 1707-1710
Quantization and Nonlinear Systems
- Erik I. Verriest:
Error analysis of linear recursions in floating point. 1711-1714 - D. V. Bhaskar Rao:
A study of coefficient quantization errors in state space digital filters. 1715-1718 - Edward Ashford Lee, David G. Messerschmitt:
On quantization effects in state-variable filter implementations. 1719-1722 - Pero J. Radonja:
Roundoff noise in estimation of the signal random parameter. 1723-1726 - Curtis A. Siller Jr.:
Multiplier-and memory-based digital FIR filters for nyquist channels: An assessment of quantization noise. 1727-1730 - Dong-guang He, Shu-kun Han:
Novel approximations to0Σ∞|h(n)| for second order digital filters. 1731-1734 - Douglas T. Sherwood, Neil J. Bershad:
Quantization noise effects in the complex LMS adaptive algorithm-linearization using dither. 1735-1738 - Fuyun Ling, Dimitris Manolakis, John G. Proakis:
New forms of LS lattice algorithms and an analysis of their round-off error characteristics. 1739-1742 - Hervé Dedieu, Francis Castanie:
Adaptive filter improvement using randomly quantized coefficients. 1743-1746
Time Delay Estimation and Localization
- Norman L. Owsley:
Maximum a posteriori array signal processing. 1747-1749 - Isabel M. G. Lourtie, José M. F. Moura:
Time delay determination: Maximum likelihood and Kalman-Bucy type structures. 1750-1753 - John P. Ianniello:
Lower bounds on the worst case probability of large error for two channel time delay estimation. 1754-1757 - Ashok Erramilli, Peter M. Schultheiss:
Source location with arrays subject to travelling wave perturbations. 1758-1761 - Meir Feder, Ehud Weinstein:
Optimal multiple source location estimation via the EM algorithm. 1762-1765 - Jeffrey L. Krolik, Moshe Eizenman, Subbarayan Pasupathy:
Application of the LMS adaptive line enhancer in time delay estimation. 1766-1769 - E. J. Modugno, G. W. Johnson, A. O. Cohen:
A hybrid parallel-serial approach to nonlinear filtering. 1770-1772 - Theagenis J. Abatzoglou:
A fast local maximum likelihood estimator for time delay estimation. 1773-1776
Array Processingand Beamforming II
- Adam J. Efron, Donald W. Tufts:
Estimation of frequencies of multiple two-dimensional sinusoids: Improved methods of linear prediction. 1777-1779 - Walter M. X. Zimmer:
Multiple beamformer performance analysis of the coherent mode. 1780-1783 - Robert S. Walker:
Bearing accuracy and resolution bounds of high-resolution beamformers. 1784-1787 - David R. Farrier, D. J. Jeffries:
Bearing estimation in the presence of unknown correlated noise. 1788-1791 - B. B. Madan, S. R. Parker:
Adaptive beam forming in correlated interference environment. 1792-1795 - Magdy T. Hanna, Marwan A. Simaan:
Minimum rejection response array filters in the presence of white noise. 1796-1799 - Meng Hwa Er, Antonio Cantoni:
A new set of linear constraints for broadband time domain element space processors. 1800-1803 - Lal C. Godara, Antonio Cantoni:
Analysis of constrained LMS algorithm with application to adaptive beamforming using perturbation sequences. 1804-1807 - Leon H. Sibul, Susan E. Burke:
Error analysis of eigenvector preprocessors used in adaptive beamforming. 1808-1811 - E. M. Long, J. M. Schumpert:
Finite word length effects on array processing algorithms. 1812-1815 - S. Unnikrishna Pillai, Fred Haber, Yeheskel Bar-Ness:
A new approach to array geometry for improved spatial spectrum estimation. 1816-1819 - Henry Cox, Robert M. Zeskind, Theo Kooij:
Sensitivity constrained optimum endfire array gain. 1820-1823
Late Papers
- Yoram Bresler, Albert Macovski:
Exact maximum likelihood estimation of superimposed exponential signals in noise. 1824-1827 - Martin L. Cohen:
Designing filters for interpolation beamforming. 1828-1831 - Paul D. Bauman, Stanley P. Lipshitz, John Vanderkooy:
Cepstral analysis of electroacoustic transducers. 1832-1835 - Rui J. P. de Figueiredo:
Generalized spline signal interpolation and smoothing with simultaneous optimization in time and frequency domains. 1836-1837 - Min-In Chung, William M. Kushner, John N. Damoulakis:
Word boundary detection and speech recognition of noisy speech by means of iterative noise cancellation techniques. 1838 - L. Ray Simar, Rui J. P. de Figueiredo:
A new syntactic/Semantic approach to 3D-surface reconstruction. 1839-1842 - Sally L. Wood:
Three dimensional shape estimation from two dimensional images. 1843-1846 - Thomas S. Anantharaman, Roberto Bisiani:
Custom data-flow machines for speech recognition. 1847-1850 - David W. Paglieroni, Anil K. Jain:
A control point theory for boundary representation and matching. 1851-1854 - S. Kaul, Malayappan Shridhar:
Tree structures for implementation of a vector quantized speech coding system. 1855-1857 - George Carayannis, Elias Koukoutsis, Dimitris Manolakis, Cristos C. Halkias:
A new look on the parallel implementation of the Shur algorithm for the solution of Toeplitz equations. 1858-1861
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