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ORCIDICASSP 1983: Boston, Massachusetts, USA
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '83, Boston, Massachusetts, USA, April 14-16, 1983. IEEE 1983
Techniques and Applications of Adaptive Filtering
- C. Richard Johnson Jr., James P. Lyons Jr., Chris Heegard:
A new adaptive parameter estimation structure applicable to ADPCM. 1-4 - Avni Morgül, Peter M. Grant, Colin F. N. Cowan:
Wideband frequency domain adaptive filter module. 5-8 - Moeness G. Amin, Lloyd J. Griffiths:
Time-varying spectral estimation using symmetric smoothing. 9-12 - Michael G. Larimore, John R. Treichler:
Convergence behavior of the constant modulus algorithm. 13-16 - R. A. David, Samuel D. Stearns, G. R. Elliott, Delores M. Etter:
IIR algorithms for adaptive line enhancement. 17-20 - Malcolm J. Rutter, Peter M. Grant, David Renshaw, Peter B. Denyer:
Design and realisation of adaptive lattice filters. 21-24 - Nasir Ahmed, S. Vijayendra:
An adaptive short-term correlator algorithm. 25-28 - Christos Caraiscos, Bede Liu:
A round-off error analysis of the LMS adaptive algorithm. 29-32 - Edgar H. Satorius, S. C. Larisch, S. C. Lee, Lloyd J. Griffiths:
Fixed-point implementation of adaptive digital filters. 33-36 - Taiho Koh, Edward J. Powers:
An adaptive nonlinear digital filter with lattice orthogonalization. 37-40 - S. Shaffer, C. S. Williams:
The filtered error LMS Algorithm. 41-44 - Maurice G. Bellanger, Cumhur Cengiz Evci:
On computational complexity in adaptive digital filters. 45-48 - Sally G. Smith, Colin F. N. Cowan, Malcolm J. Rutter:
A new structure for adaptive echo cancellation. 49-52 - V. Umapathi Reddy, Tie-Jun Shan, Thomas Kailath:
Application of modified least-squares algorithms to adaptive echo cancellation. 53-56 - E. Andresdottir, Ronald W. Schafer:
Application of adaptive noise cancelling in a noisy reverberant environment. 57-60
Narrowband and Low Rate Speech Coding
- Salim E. Roucos, Richard M. Schwartz, John Makhoul:
A segment vocoder at 150 b/s. 61-64 - David Y. Wong, Biing-Hwang Juang, D. Y. Cheng:
Very low data rate speech compression with LPC vector and matrix quantization. 65-68 - Richard M. Schwartz, Salim E. Roucos:
A comparison of methods for 300-400 b/s vocoders. 69-72 - Douglas B. Paul:
An 800 bps adaptive vector quantization vocoder using a perceptual distance measure. 73-76 - Alexander MacLeod Wilgus, Thomas P. Barnwell III:
Data rate reduction of gain and pitch parameters in an LPC vocoder. 77-80 - Bishnu S. Atal:
Efficient coding of LPC parameters by temporal decomposition. 81-84 - Panos E. Papamichalis, George R. Doddington:
Delta coding of LPC parameters. 85-88 - George S. Kang, Stephanie Everett:
Improvement of the LPC analysis. 89-92 - Satoshi Imai:
Cepstral analysis synthesis on the mel frequency scale. 93-96 - Neviano Dal Degan, V. Di Lago:
Design and test of a real-time floating point LPC vocoder. 97-100 - Bernard Gold, J. Lynch, Joseph Tierney:
Vocoded speech through fading channels. 101-103
Image Analysis
- Petros Maragos, Russell M. Mersereau, Ronald W. Schafer:
Two-dimensional linear predictive analysis of arbitrarily-shaped regions. 104-107 - Thomas F. Quatieri:
Object detection by two-dimensional linear prediction. 108-111 - Lawrence O'Gorman, Arthur C. Sanderson:
The converging squares algorithm: An efficient multidimensional peak picking method. 112-115 - Gerald J. Lemay, Jean-Daniel Dessimoz:
Recovery of gray scaled images from contour processed representations. 116-117 - B. L. Yen, Thomas S. Huang:
Determining 3-D motion and structure of a rigid body using straight line correspondences. 118-121 - Roger Y. Tsai:
Estimating 3-D motion parameters and object surface structures from the image motion of conic arcs, I: Theoretical basis. 122-125 - Ruud M. Bolle, Bruno Cernuschi-Frías, David B. Cooper:
Fast parallel image processing for robot vision for the purpose of extracting information about objects. 126-127
Image Reconstruction from Projections
- David J. Rossi, Alan S. Willsky:
Reconstruction from projections based on detection and estimation of objects. 128-130 - Barry P. Medoff, William R. Brody, Albert Macovski:
The use of a priori information in image reconstruction from limited data. 131-134 - Mehrdad Soumekh, Mostafa Kaveh, Rolf K. Mueller:
Algorithms and experimental results in acoutistic tomography using Rytov's approximation. 135-138 - M. Ibrahim Sezan, Henry Stark:
Image restoration in CT by the method of projection onto convex sets. 139-142 - Constantinos E. Goutis, Richard M. Leahy, S. Drossos:
Reconstruction algorithms for limited view projection data. 143-146 - A. A. (Louis) Beex:
Iterative reconstruction of space-limited scenes from noisy frequency-domain measurements. 147-150
Fast DSP Algorithms
- Kaliappan Gopalan, C. S. Chen:
Numerical evaluation of Fourier-Bessel series expansion. 151-154 - Prabhakara C. Balla, Andreas Antoniou:
Number theoretic transform based on ternary arithmetic. 155-158 - Hendrik Holmann, Pierre Duhamel:
Longer NTT's with 2 as a root of unity. 159-162 - Howard W. Johnson, C. Sidney Burrus:
On the structure of efficient DFT algorithms. 163-165 - Rajendra Kumar:
A fast algorithm for solving Toeplitz system of equations. 166-169 - Mati Wax, Thomas Kailath:
Efficient inversion of doubly block Toeplitz matrix. 170-173 - Basile Dimitriadis:
A fast non-recursive algorithm and a parallel processor architecture for smoothing spline fitting. 174-177 - S. Lawrence Marple Jr.:
Fast algorithms for linear prediction and system identification filters with linear phase. 178-181 - Benjamin Friedlander:
Efficient computation of the covariance sequence of an autoregressive process. 182-185 - George Carayannis, Dimitris Manolakis, Nicholas Kalouptsidis:
Fast Kalman type algorithms for sequential signal processing. 186-189 - Ramesh C. Agarwal:
An in-place and in-order WFTA. 190-193
Digital Filter Design
- Francis Grenez:
Constrained Chebyshev approximation for FIR filters. 194-196 - James F. Kaiser, Kenneth Steiglitz:
Design of FIR filters with flatness constraints. 197-200 - Guido M. Cortelazzo, Michael R. Lightner:
Simultaneous design in both magnitude and group-delay of IIR and FIR filters: Problems and results. 201-204 - Zhongqi Jing, Adly T. Fam:
A new class of narrow transition band digital filters. 205-208 - Ramesh C. Agarwal, R. Sudhakar:
Multiplier-less design of FIR filters. 209-212 - Yong Ching Lim, Sydney R. Parker:
On the synthesis of lattice parameter digital filters. 213-216 - Ed F. Deprettere:
Synthesis and fixed-point implementation of pipelined true orthogonal filters. 217-220 - Henri J. Nussbaumer:
Complex quadrature mirror filters. 221-223 - Claude R. Galand, Daniel J. Esteban:
Design and evaluation of parallel quadrature mirror filters (PQMF). 224-227 - Vijay K. Jain, Ronald E. Crochiere:
A novel approach to the design of analysis/Synthesis filter banks. 228-231 - Zheng-Sing Huang:
A least squares approximation design of complex FIR filters. 232-234
Signal Modeling
- Bruce R. Musicus:
Iterative algorithms for optimal signal reconstruction and parameter identification given noisy and incomplete data. 235-238 - Per Hedelin, Gunnar Hult:
Maximum likelihood parameter estimation with a min/Max criterion. 239-242 - J. V. White:
Stochastic state-space models from empirical data. 243-246 - Sueo Sugimoto, Ikuo Ishizuka:
Identification and estimation algorithms for a Markov chain plus AR process. 247-250 - Francis Castanie, P. Soule:
An improvement of non-stationarity detection by level-crossing analysis of linear prediction error. 251-253 - Lowell Jacobson, Harry Wechsler:
The composite pseudo Wigner distribution (CPWD): A computable and versatile approximation to the Wigner distribution (WD). 254-256 - Cheryl Richmond, Vijay K. Jain:
Systems modeling by digital signal processing. 257-260 - Joe K. Hammond, Y. H. Tsao, R. F. Harrison:
Evolutionary spectral density models for random processes having a frequency modulated structure. 261-264 - Giuseppe Martinelli:
Non-stationary AR model identification by batch estimation. 265-267 - Yves Grenier:
Estimation of non-stationary moving-average models. 268-271 - James A. Cadzow, Thomas P. Bronez:
Time series modeling via general linear estimation theory. 272-275 - M. N. Shanmukha Swamy, Eugene I. Plotkin, Leonid M. Roytman, A. M. Zayezdny:
One approach to simulation of modulated signals. 276-279 - H. Rauner, Ulrich Appel, Werner Wolf:
Application of a cepstral distance measure in evoked potential processing. 280-283 - Hossny El-Sherief, Youssef Lotfy Abdel-Magid:
An efficient on-line load-modeling algorithm for short-term forecasting of interconnected power systems. 284-287
Connected Speech Recognition
- Hermann Ney:
Experiments in connected word recognition. 288-291 - Subrata K. Das:
Some dimensionality reduction studies in continuous speech recognition. 292-295 - Sei-ichi Nakagawa:
A connected spoken word recognition method by O(n) dynamic programming pattern matching algorithm. 296-299 - Dean P. McCullough:
Secondary testing techniques for word recognition in continuous speech. 300-303 - Astrid Brietzmann, Hans-Werner Hein, Heinrich Niemann, Peter Regel:
The Erlangen system for understanding continuous German speech. 304-307 - Nanning Zheng, Guorong Xuan:
A Chinese speech recognition system. 308-311 - Patrick Fonsale:
Connected-word recognition system using speaker-independent phonetic features. 312-315 - Renato De Mori, Attilio Giordana, Pietro Laface:
Phonetic feature hypothesization in continuous speech. 316-319 - Takao Watanabe:
Segmentation-free syllable recognition in continuously spoken Japanese. 320-323 - Katsuhiko Shirai, Tetsunori Kobayashi:
Considerations on articulatory dynamics for continuous speech recognition. 324-327
Sensor Array Processing
- James M. Kates:
A generalized approach to high-resolution array processing. 328-331 - Georges Bienvenu:
Eigensystem properties of the sampled space correlation matrix. 332-335 - Arthur Jay Barabell:
Improving the resolution performance of eigenstructure-based direction-finding algorithms. 336-339 - Guaning Su, Martin Morf:
Modal decomposition signal subspace algorithms. 340-343 - Stuart R. DeGraaf, Don H. Johnson:
Capability of array processing algorithms to estimate source bearings. 344-347 - Ken C. Sharman, Tariq S. Durrani:
A triangular adaptive lattice filter for spatial signal processing. 348-351 - Neil J. Malloy:
Analysis and synthesis of general planar interferometer arrays. 352-355 - Bharat B. Madan, C. V. Kuriyan:
An escalator structure for adaptive beamforming. 356-358 - Harald Schneider:
Evaluation of an orthogonal beamforming procedure using real data. 359-362 - Qi-Hu Li:
Signal separation theory by using adaptive array. 363-366 - Hong Fan, Ezz I. El-Masry, W. Kenneth Jenkins:
Digital extrapolation beamforming. 367-370 - Peter M. Schultheiss, Ashok Erramilli:
Localization with arrays subject to sensor motion. 371-374 - Robert S. Walker, A. T. Ashley, P. F. Kavanagh:
Noise normalization of broadband sonar data in bearing space. 375-378
Multidimensional Filter Design and Implementation
- Ahmed Abo-Taleb, Moustafa M. Fahmy:
Design of FIR two-dimensional digital filters by successive projection. 379-382 - Dieter Fritsch:
Optimal design of two-dimensioal FIR-filters. 383-386 - Robert A. Gabel:
Equiripple two-dimensional interpolation filters. 387-390 - Gilberto Câmara, Nelson D. A. Mascarenhas:
Methods for image interpolation through FIR filter design techniques. 391-394 - M. Omair Ahmad, Majid Ahmadi, Venkat Ramachandran:
Design of low sensitive 2-D analog and recursive digital filters with prescribed magnitude and group delay specifications. 395-398 - Majid Ahmadi, S. Golikeri, Venkat Ramachandran:
A new method for the design of 2-dimensional stable recursive digital filters satisfying prescribed magnitude and group delay response. 399-402 - A. C. Hsueh, Jerry M. Mendel, Bijan Lashgari:
2-D non-causal systems: State space modeling for half-plane support. 403-406 - Thomas A. Nodes, Neal C. Gallagher Jr.:
Image convergence under two dimensional separable median filtering. 407-410 - Yong H. Lee, Saleem A. Kassam:
Some generalizations of median filters. 411-414 - Tyseer Aboulnasr, Moustafa M. Fahmy:
2-D state-space realizations with fewer multipliers and invariant norms. 415-418 - Ju-Hong Lee, John M. Woods:
Sectioning implementation of two-dimensional symmetric half-plane filters. 419-422
DSP Hardware and Architectures
- Roberto Bisiani:
A class of data-flow architectures for speech recognition. 423-426 - Nicholas Roethe:
Architecture and programming of a multirate digital filter. 427-430 - M. J. Knudsen:
MUSEC, a powerful network of signal microprocessors. 431-434 - S. N. Terepin, Paul Loewenstein:
Architecture and instruction set of a programmable LSI digital filter. 435-438 - Gordon L. DeMuth:
A distributed signal processor incorporating VLSI and high order language programming. 439-442 - Yong Ching Lim, Sydney R. Parker:
Efficient FIR filter implementation using microprocessor. 443-446 - Z. Orbach:
Enhancing microcomputer for high-speed vector processing. 447-450 - B. Arambepola:
Algorithm and a new processor architecture for computing the DFT. 451-454 - Hanoch Lev-Ari:
Modular architectures for adaptive multichannel lattice algorithms. 455-458 - Richard H. Jackson, Hughe M. South:
A reconfigurable signal processor for high throughput applications. 459-461 - R. A. Benson:
A multichannel input subsystem. 462-465 - Kazunori Ozawa, Takashi Araseki, Yasuo Itoh:
An adaptive echo canceller using digital signal processor LSI chips. 466-469
Speech Hardware
- Thomas A. Rice, Leah J. Siegel:
Parallel processing for computationally intensive speech analysis operations. 471-474 - Daniel F. Daly, L. E. Bergeron:
A programmable voice digitizer using the T.I. TMS-320 microcomputer. 475-478 - Arvind Arora, James L. Melsa, James D. Mills:
Real time implementation of a speech coding algorithm using time-domain harmonic scaling and adaptive residual coding. 479-482 - Larry P. Lewis, Zwie Amitai, Harvey F. Silverman:
The APS-II processor for speech recognition. 483-486 - Alfred Kaltenmeier:
Implementation of various LPC algorithms using commercial digital signal processors. 487-490 - David B. Chester, Fred J. Taylor, Mike Doyle:
On the Wigner distribution. 491-494 - Chi-Foon Chan, Marcian E. Hoff Jr., Peter Nevard, Meemong Lee:
Architecture and application of a commercially available speech recognition board. 495-498 - Terry Montlick, David Vetter, Klaus Skoge, Paul Ahrens:
A combination speech synthesis and recognition integrated circuit. 499-502 - Hisao Ishizuka, Masao Watari, Hiroaki Sakoe, Seibi Chiba, Toshiki Iwata, Tomoko Matsuki, Yuichi Kawakami:
A microprocessor for speech recognition. 503-506 - J. MacAllister:
Systolic arrays for dynamic programming in speech recognition systems. 507-510 - Joel A. Feldman, Edward J. Beauchemin:
A custom IC for automatic gain control in LPC vocoders. 511-514 - Kenji Nakayama, Yutaka Ishikawa, Yoshiaki Kuraishi:
Design of LSI speech spectrum analyzer using switched capacitor filter techniques. 515-518 - I. Versvik, Ole Berg:
Comparison of high speed hardware and software implementation approaches for a multichannel PCM-Delta converter. 519-522
System Performance Evaluation and Speaker Recognition
- Ann M. Rollins, Jennifer Wiesen:
Speech recognition and noise. 523-526 - Janet M. Baker, David S. Pallett, John S. Bridle:
Speech recognition performance assessments and available databases. 527-530 - Margaret Kahn, Peter Garst:
The effects of five voice characteristics on LPC quality. 531-534 - David B. Pisoni, Howard C. Nusbaum, Paul A. Luce, Eileen C. Schwab:
Perceptual evaluation of synthetic speech: Some considerations of the user/System interface. 535-538 - Joel Crosmer, Thomas P. Barnwell III:
An algorithm for designing optimum quantizers subject to a multiclass distortion criterion. 539-542 - V. Viswanathan, William Russell, A. W. F. Huggins:
Objective speech quality evaluation of mediumband and narrowband real-time speech coders. 543-546 - Schuyler R. Quackenbush, Thomas P. Barnwell III:
The estimation and evaluation of pointwise nonlinearities for improving the performance of objective speech quality measures. 547-550 - Jared J. Wolf, Michael A. Krasner, Kenneth F. Karnofsky, Richard M. Schwartz, Salim E. Roucos:
Further investigation of probabilistic methods for text-independent speaker identification. 551-554 - K. P. Li, Edwin H. Wrench Jr.:
An approach to text-independent speaker recognition with short utterances. 555-558 - Malayappan Shridhar, N. Mohankrishnan, Maher A. Sid-Ahmed:
A comparison of distance measures for text-independent speaker identification. 559-562 - Melvyn J. Hunt:
Further experiments in text-independent speaker recognition over communications channels. 563-566
Detection and Estimation
- Yiu Tong Chan, R. K. Miskowicz:
An ARMA modeling method for estimation of coherence and time delay. 567-570 - Stanislav B. Kesler:
Autoregressive detection by generalized Burg algorithm. 571-574 - Amin G. Jaffer, Ryan L. Stoutenborough, William B. Green:
Improved detection and tracking of dynamic signals by Bayes-Markov techniques. 575-578 - William A. Struzinski:
ORing loss model for implementation in signal processing systems for data display. 579-582 - G. W. Johnson, D. E. Ohlms, M. L. Hampton:
Broadband correlation processing. 583-586 - D. M. Klamer, Joe A. Presley Jr.:
Detection performance of the parametric correlator. 587-590 - Kenneth S. Vastola, John S. Farnbach, Stuart C. Schwartz:
Maximin sonar system design for detection. 591-594 - Frank W. Symons Jr.:
Performance bounds for SNR enhancement of narrowband signals in surface reverberation. 595-598 - William S. Hodgkiss, Dimitrios Alexandrou:
Application of adaptive linear predictor structures to the prewhitening of acoustic reverberation data. 599-602 - Andreas Gerber, Rui J. P. de Figueiredo:
MAP detection and ML estimation of signals overlapping in time and frequency for both Gaussian and Laplacian noise. 603-606 - Roger F. Dwyer:
Detection of non-Gaussian signals by frequency domain Kurtosis estimation. 607-610 - Evriclea Voudouri, Ludwik Kurz:
Sequential robust m-interval polynomial approximation (MIPA) partition detectors. 611-614
Quantization Effects
- P. P. Vaidyanathan, Sanjit K. Mitra:
A new approach for synthesis of low sensitivity digital filter structures based on lossless building blocks. 615-618 - Markku Renfors:
Roundoff noise in error-feedback state-space filters. 619-622 - Denis Henrot, Clifford T. Mullis:
A modular and orthogonal digital filter structure for parallel processing. 623-626 - Mahmood R. Azimi-Sadjadi:
Finite word effects in 2-D block implemented fixed point digital filters. 627-630 - Ulrich Heute, Hans Wilhelm Schüßler:
FFT-accuracy - new insights and a new point-of-view. 631-634 - Iiro Hartimo:
Self-sustained stable oscillations of second order recursive algorithms. 635-638
Deconvolution
- B. Yegnanarayana, A. Dhayalan:
Noniterative techniques for minimum phase signal reconstruction from phase or magnitude. 639-642 - H. Joel Trussell, M. Reha Civanlar:
The initial estimate in constrained iterative restoration. 643-646 - Rémy Prost, Robert Goutte:
In-time non-iterative fast-algorithm for support constrained deconvolution. 647-650 - Gérard Thomas:
A positive optimal deconvolution procedure. 651-654 - Chaw-Bing Chang, Richard B. Holmes:
Application of pseudoinversion to scattering function estimation with discrete measurements. 655-658 - Aggelos K. Katsaggelos, Ronald W. Schafer:
Iterative deconvolution using several different distorted versions of an unknown signal. 659-662
Adaptive Filtering
- Guy R. L. Sohie, Leon H. Sibul:
Stochastic convergence properties of the adaptive gradient lattice. 663-666 - Tien C. Hsia:
Convergence analysis of LMS and NLMS adaptive algorithms. 667-670 - Qitu Zhang, Simon Haykin:
Tracking characteristics of the Kalman filter in a nonstationary environment for adaptive filter applications. 671-674 - Arye Nehorai, Martin Morf:
A new derivation for fast recursive least squares and Levinson algorithms by the conjugate direction method. 675-678 - John M. Cioffi, Thomas Kailath:
Fast, fixed-order, least-squares algorithms for adaptive filtering. 679-682 - Carlos H. Muravchik, Martin Morf:
A new stable feedback ladder algorithm for the identification of moving average processes. 683-686
Radar DSP
- N. Sridhar Reddy, M. N. Shanmukha Swamy:
Time-domain estimation of unambiguous Doppler frequency in low and medium PRF radars. 687-690 - Theagenis J. Abatzoglou:
A fast and accurate method for estimating target Doppler. 691-694 - Simon Haykin, Jelisaveta Kesler, John Litva:
Evaluation of angle of arrival estimators using real multipath data. 695-698 - S. V. Varanasi, S. C. Gupta:
Random phase tracking of non-uniformly sampled signals. 699-702 - Tariq S. Durrani, Avedis S. Arslanian:
Signal detection performance of lattice processors. 703-706 - K. V. S. AnandBabu, S. Prasad:
Some new approaches for the implementation of optimum MTI filters. 707-710 - V. Nagarajan, V. Hanuma Sai, G. K. Chaturvedi:
A new approach to scan to scan correlation and its implementation. 711-714
Speaker-Independent Speech Recognition
- Masao Watari, Makoto Akabane, Yoichiro Sako:
A speaker independent word recognition based on transient matching. 715-718 - Edward C. Bronson:
Syntactic pattern recognition of discrete utterances. 719-722 - Noboru Sugamura, Kiyohiro Shikano, Sadaoki Furui:
Isolated word recognition using phoneme-like templates. 723-726 - H. Scott Hinton, Leah J. Siegel:
Speaker independent isolated word automatic speech recognition using computer generated phonemes. 727-730 - Ronald A. Cole, Richard M. Stern, Michael S. Phillips, Scott M. Brill, Andrew P. Pilant, Philippe Specker:
Feature-based speaker-independent recognition of isolated english letters. 731-733 - Richard M. Stern, Moshé J. Lasry:
Dynamic speaker adaptation for isolated letter recognition using MAP estimation. 734-737 - Shozo Makino, Takeshi Kawabata, Ken'iti Kido:
Recognition of consonant based on the perceptron model. 738-741 - Marcia A. Bush, Gary E. Kopec, Victor W. Zue:
Selecting acoustic features for stop consonant identification. 742-745 - J. Johannsen, J. MacAllister, T. Michalek, S. Ross:
A speech spectrogram expert. 746-749 - Matthew Yuschik:
An adaptive feature extraction expert. 750-752 - Periagaram K. Rajasekaran, George R. Doddington:
Microcomputer implementable low cost speaker-independent word recognition. 753-756 - Yeunung Chen:
Vocabulary selection for high performance speech recognition. 757-760 - Peter F. Brown, Chin-Hui Lee, James C. Spohrer:
Bayesian adaptation in speech recognition. 761-764 - Les T. Niles, Harvey F. Silverman, N. Rex Dixon:
A comparison of three feature vector clustering procedures in a speech recognition paradigm. 765-768 - R. J. Golibersuch:
Automatic prediction of linear frequency warp for speech recognition. 769-772 - C. A. Olano:
An investigation of spectral match statistics using a phonemically marked data base. 773-776
Speech Analysis and Reconstruction
- Hynek Hermansky, Hiroya Fujisaki, Yasuo Sato:
Analysis and synthesis of speech based on spectral transform linear predictive method. 777-780 - Sharad Singhal, Bishnu S. Atal:
Optimizing LPC filter parameters for multi-pulse excitation. 781-784 - G. A. Mack, Vijay K. Jain:
Effect of a time-weighted error criterion on recursive estimation of speech parameters. 785-788 - Philippe Delsarte, Yves V. Genin, Yves G. Kamp, Paul Van Dooren:
On the role of the partial trigonometric moment problem in AR speech modelling. 789-792 - Ing Widya, Patrick M. Dewilde:
Stable modelling of a continuous covariance function with application to continuous speech. 793-796 - Evangelos E. Milios, Alan V. Oppenheim:
The phase-only version of the LPC residual in speech coding. 797-799 - S. Hamid Nawab, Thomas F. Quatieri, Jae S. Lim:
Algorithms for signal reconstruction from short-time Fourier transform magnitude. 800-803 - Daniel W. Griffin, Jae S. Lim:
Signal estimation from modified short-time Fourier transform. 804-807 - Stephen A. Zahorian, Paul E. Gordy:
Finite impulse response (FIR) filters for speech analysis and synthesis. 808-811 - Jay T. Rubinstein, Harvey F. Silverman:
Some comments on the design and implementation of FIR filterbanks for speech recognition. 812-815 - Richard M. Chamberlain, John S. Bridle:
ZIP: A dynamic programming algorithm for time-aligning two indefinitely long utterances. 816-819
Image Restoration and Enhancement
- Yasuo Yoshida, Hisanao Ogura:
Image restoration based on an anisotropic noncausal stochastic model. 820-823 - F. B. Hoogterp, Nan K. Loh:
Noncausal modeling and restoration of noisy images. 824-827 - Alan C. Bovik, Thomas S. Huang, David C. Munson Jr.:
Image restoration using order-constrained least-squares methods. 828-831 - A. Murat Tekalp, John W. Woods, Howard Kaufman:
A multiple model algorithm for the adaptive restoration of images. 832-835 - Irwin Scollar, Thomas S. Huang, Bernd Weidner:
Image enhancement using the median and the interquartile distance. 836-839 - Robert A. King, A. J. Singarajah, Alan S. Kwabwe:
A hybrid technique to restore the Fourier phase and magnitude spectra of noisy linearly degraded images. 840-843 - Thomas L. Marzetta, Stephen W. Lang:
New interpretations for the MLM and DASE spectral estimators. 844-846 - Stephen W. Lang, Thomas L. Marzetta:
A linear programming approach to bounding spectral power. 847-850 - Jae S. Lim, Farid U. Dowla:
Improved maximum likelihood method for two-dimensional spectral estimation. 851-854 - Rama Chellappa, Govind Sharma:
Two-dimensional spectral estimation using spatial autoregressive models. 855-858 - Chrysostomos L. Nikias, Mysore R. Raghuveer:
A new class of high-resolution and robust multi-dimensional spectral estimation algorithms. 859-862 - Tariq S. Durrani, Roy Chapman:
Eigenfilter methods for 2D spectral estimation. 863-866 - Sergio D. Cabrera, Thomas W. Parks:
Deterministic estimation of two-dimensional signals. 867-870
Time Delay Estimation and Source Location
- C. Y. Wu, Allan E. Pearson:
On time delay estimation involving received signals. 871-874 - J. P. Ianniello, Ehud Weinstein, Anthony J. Weiss:
Comparison of the Ziv-Zakai lower bound on time delay estimation with correlator performance. 875-878 - Robert M. Zeskind, Kenneth B. Theriault, Peter Moss:
Accuracy of linearized performance predictions for wavefront curvature ranging systems. 879-882 - Dae Hee Youn, Nasir Ahmed:
Comparison of two adaptive methods for time delay estimation. 883-886 - Benjamin Friedlander:
A parametric technique for estimation of delay and Doppler. 887-890 - Mati Wax, Tie-Jun Shan, Thomas Kailath:
Covariance eigenstructure approach to 2-D harmonic retrievel. 891-894 - R. Lynn Kirlin:
Optimal delay estimation in a multiple sensor array having spatially correlated noise. 895-898 - Wolfgang K. Fischer, Lawrence C. Ng:
Improved time delay estimation in the presence of interference. 899-902 - Harold F. Jarvis Jr., Lawrence C. Ng:
Comparison of error detectors for time delay tracking. 903-906 - G. W. Johnson, A. O. Cohen, D. E. Ohlms, C. W. Shier:
Modified polar coordinates for ranging from Doppler and bearing measurements. 907-910 - P. A. Yansouni, Robert J. Inkol:
The use of linear constraints to reduce the variance of time of arrival difference estimates for source location. 911-914 - Kenneth W. Moreland, Robert J. Inkol:
An effective implementation of a time delay estimator for source location. 915-918
VLSI for DSP
- Benjamin Friedlander, John A. Newkirk:
A comparison of two SAR processing architectures for VLSI implementation. 919-922 - Irving S. Reed, C.-S. Yeh, Trieu-Kien Truong:
A VLSI architecture for digital filters using complex number-theoretic transforms. 923-926 - Jean-Marc Delosme:
VLSI implementation of rotations in pseudo-Euclidean spaces. 927-930 - Noel R. Strader II:
VLSI bit-sequential architectures for digital signal processing. 931-934 - Peter R. Cappello, Andrea S. LaPaugh, Kenneth Steiglitz:
Optimal choice of intermediate latching to maximize throughput in VLSI circuits. 935-938 - Peter B. Denyer, David Renshaw:
Case studies in VLSI signal processing using a silicon compiler. 939-942 - John A. Eldon:
A 22-bit floating point registered arithmetic logic unit. 943-946 - Gary E. Winter, Robert R. Yamashita:
A single board floating point signal processor. 947-950 - Earl E. Swartzlander Jr., Louis S. Lome, George Hallnor:
Digital signal processing with VLSI technology. 951-954 - Toshitaka Tsuda, Kazuo Murano, Shigeyuki Unagami, M. Shimada, H. Kikuchi, S. Sumi, Y. Miwa:
CMOS LSI DSP and its application to voice band signals. 955-958
Underwater Acoustics
- William S. Hodgkiss, D. S. Hansen:
High resolution spectral analysis and the acoustic remote sensing of ocean current velocity. 961-964 - Charles Schmid:
The application of linear prediction techniques to background noise generation for sonar trainers. 965-968 - R. N. Carpenter, Thomas J. Curry, D. Pearson:
Applications of adaptive signal processing to the reduction of flow-induced noise in small underwater vehicles. 969-972 - K. D. Flowers:
Modal dispersion effects on coherent signal processing parameters. 973-976 - Philip G. Harper, Stuart I. Jardine, Andrew J. Quinn, David M. Treherne:
A detector for acoustic directionality. 977-980 - E. Richard Robinson, Azizul H. Quazi:
The impact of refraction on time-delay estimation. 981-984 - Kenneth B. Theriault:
Lower bounds on pulsed-Doppler current profiler accuracy. 985-988 - Marwan A. Simaan:
Optimum array filters for array data signal processing. 989-992
Geophysical DSP
- James E. Gaby, Kenneth R. Anderson:
Primitive extraction for a syntactic pattern recognizer of features in seismic signals. 993-996 - Tong Chang, Yenta Li, Zhongze Wu:
The wavelet reconstructed with phase-only and a study of its uniqueness. 997-1000 - R. Lynn Kirlin, Lois A. Dewey:
Seismic velocity estimators. 1001-1004 - Nicholas Kalouptsidis, George Carayannis, Dimitris Manolakis:
Systems of equations with near-to-Toeplitz or near-to-Hankel parameters and applications to signal processing. 1005-1008 - D. R. Martinez, S. H. Bickel:
Deconvolution of band-limited signals. 1009-1012 - Stefanos D. Kollias, Cristos C. Halkias:
A model reduction algorithm by spline approximation in the deconvolution of seismic signals. 1013-1016 - Achim V. Brandt:
Detecting and estimating parameter jumps using ladder algorithms and likelihood ratio tests. 1017-1020
Isolated-Word Speech Recognition
- David K. Burton, John E. Shore, Joseph T. Buck:
A generalization of isolated word recognition using vector quantization. 1021-1024 - Roberto Pieraccini, Roberto Billi:
Experimental comparison among data compression techniques in isolated word recognition. 1025-1028 - Jean-Luc Gauvain, Joseph Mariani, Jean-Sylvain Liénard:
On the use of time compression for word-based recognition. 1029-1032 - Chiu-Kuang Chuang, S. W. Chan:
Speech recognition using variable frame rate coding. 1033-1036 - Martin J. Russell, Roger K. Moore
, Michael J. Tomlinson:
Some techniques for incorporating local timescale variability information into a dynamic time-warping algorithm for automatic speech recognition. 1037-1040 - Roger K. Moore, Martin J. Russell, Michael J. Tomlinson:
The discriminative network: A mechanism for focusing recognition in whole-word pattern matching. 1041-1044 - Lynn D. Wilcox, Bruce T. Lowerre, M. Kahn:
Use of a priori knowledge of vocabulary for real time discrete utterance recognition. 1045-1048 - Stephen E. Levinson, Lawrence R. Rabiner, Man Mohan Sondhi:
Speaker independent isolated digit recognition using hidden Markov models. 1049-1052 - Rainer Zelinski, Fritz Class:
A learning procedure for speaker-dependent word recognition systems based on sequential processing of input tokens. 1053-1056 - Aaron E. Rosenberg:
Probabilistic model for the performance of speech recognition systems. 1057-1060 - Bruce A. Dautrich, Lawrence R. Rabiner, Thomas B. Martin:
On the use of filter bank features for isolated word recognition. 1061-1064 - Lalit R. Bahl, A. G. Cole, Frederick Jelinek, Robert L. Mercer, Arthur Nádas, David Nahamoo, Michael Picheny:
Recognition of isolated-word sentences from a 5000-word vocabulary office correspondence task. 1065-1067
Parametric Spectral Analysis
- Jean-Pierre Schott, James H. McClellan:
Maximum entropy power spectrum estimation with uncertainty in correlation measurements. 1068-1071 - Peter D. Scott, Chrysostomos L. Nikias:
High-resolution frequency estimation via a weighted forward and backward autoregressive modelling. 1072-1075 - Manuel Duarte Ortigueira, José M. Tribolet:
On the double Levinson recursion formulation of ARMA spectral estimation. 1076-1079 - El-Sayed A. Talkhan, A. W. Hassan, N. A. Amin:
An ARMA model for power spectrum estimation. 1080-1083 - Benjamin Friedlander, Boaz Porat:
A spectral matching technique for ARMA parameter estimation. 1084-1087 - Darcy P. McGinn, Don H. Johnson:
Reduction of all-pole parameter estimator bias by successive autocorrelation. 1088-1091 - Tapan K. Sarkar, Joshua Nebat, Donald D. Weiner, Vijay K. Jain:
Effect of record length on the correlation of complex exponentials. 1092-1094 - Maureen Quirk, Bede Liu:
On the resolution of autoregressive spectral estimation. 1095-1098 - Robert H. Seegal:
On harmonic retrieval. 1099-1101 - Tapan K. Sarkar, Sohail A. Dianat, S. M. Rao:
A finite step adaptive implementation of the Pisarenko's harmonic retrieval method in colored noise. 1102-1105 - K. Ogino:
A fast algorithm for unmodified ARMA spectrum estimation. 1106-1109 - Randolph L. Moses:
Fast recursive AR estimation from an overdetermined system of extended Yule Walker equations. 1110-1113 - K. Ogino, Wlodzimierz Maciej Hipolit Kozak:
Spectrum analysis of surface electromyogram (EMG). 1114-1117
Speech Enhancement and Noise Reduction
- Yariv Ephraim, David Malah:
Speech enhancement using optimal non-linear spectral amplitude estimation. 1118-1121 - Brian A. Hanson, David Y. Wong, Biing-Hwang Juang:
Speech enhancement with harmonic synthesis. 1122-1125 - L. William Varner, Thomas A. Miller, Thomas E. Eger:
A simple adaptive filtering technique for speech enhancement. 1126-1128 - Rodney W. Johnson, John E. Shore, Joseph T. Buck, David K. Burton:
Speech noise reduction by means of multi-signal minimum-cross-entropy spectral analysis. 1129-1132 - L. Hoy, B. Burns, David L. Soldan, Rao K. Yarlagadda:
Noise suppression methods for speech applications. 1133-1136 - Cengiz Esmersoy, Jae S. Lim:
Subjective evaluation of a PCM speech coding system with quantization noise reduction. 1137-1140 - David Cyganski:
Parameter isolating transforms and admissible distortions. 1141-1143 - John Mourjopoulos, Joe K. Hammond:
Modelling and enhancement of reverberant speech using an envelope convolution method. 1144-1147 - Richard F. Lyon:
A computational model of binaural localization and separation. 1148-1151 - Steven F. Boll, Robert E. Wohlford:
Event driven speech enhancement. 1152-1155 - G. Neben, Robert J. McAulay, Clifford J. Weinstein:
Experiments in isolated word recognition using noisy speech. 1156-1159 - Edward O. Belcher, Knut Andersen:
Helium speech enhancement by frequency-domain processing. 1160-1163
Signal Processing Software and Hardware
- J. R. Masse, D. Cante:
General - N Winograd D.F.T. programs with inverse option. 1164-1167 - Gary E. Kopec:
The signal representation language SRL. 1168-1171 - Y. S. Wu:
A common operational software (ACOS) approach to a signal processing development system. 1172-1175 - Martin Morf, Ping Ang, Jean-Marc Delosme:
Development of signal processing algorithms via functional programming. 1176-1179 - Joseph R. Fisher, Martin E. Kaliski, Burton S. Kaliski Jr.:
A new level of signal processing software: Automatic buffer address generation. 1180-1183 - J. M. Glass, J. P. Hepp:
Software structure for a distributed signal processing system. 1184-1187 - Ralf Steinmetz, Renate Gemballa, Joachim Lenzer, Herbert Roth:
Realization of digital filter algorithms by use of a high speed parallel processing architecture. 1188-1191 - Stanley A. White:
A preprocessor architecture to equalize focal-plane detector-array elements. 1192-1195 - Taruna Tjahjadi, Willem J. D. Steenaart:
Non-recursive digital FIR filter implementation using stored square ROM multipliers. 1196-1199 - L. Robert Morris:
A tale of two architectures: TI TMS 320 SPC vs. DEC Micro/J-11. 1200-1203 - S. A. Newton, K. P. Jackson, J. E. Bowers, C. C. Cutler, H. J. Shaw:
Fiber-optic delay line devices for GigaHertz signal processing. 1204-1207 - Yusuf A. Haque:
An adaptive transversal filter. 1208-1211
Image Coding
- Howard C. Reeve III, Jae S. Lim:
Reduction of blocking effect in image coding. 1212-1215 - S. Thomas Alexander, Sarah A. Rajala:
Adaptive compression of teleconference sequences using the LMS Algorithm. 1216-1219 - I. Paul, John W. Woods:
Some experimental results in adaptive prediction DPCM coding of images. 1220-1223 - G. Ocylok:
A comparison of interframe coding techniques. 1224-1227 - Erlendur Karlsson, Russell M. Mersereau:
An objective quality measure for still moncromatic DPCM-coded images. 1228-1231 - J. M. Schumpert, R. J. Jenkins:
A two-component image coding scheme based on two-dimensional interpolation and the discrete cosine transform. 1232-1235 - Edward Angel, L. Don Daigle:
A high speed maximum entropy encoder for images. 1236-1239 - P. L. Poehler, Junho Choi:
Linear predictive coding of imagery for data compression applications. 1240-1243 - D. K. Mitrakos, George A. Constantinides:
Maximum likelihood estimation of composite source models for image coding. 1244-1247
Multidimensional Transform Techniques
- Yoshitaka Morikawa, Hiroshi Hamada:
Implementation for two-dimensional FIR filters using the number theoretic transform. 1248-1251 - B. Arambepola:
Transform algorithm for computing two-dimensional convolutions. 1252-1255 - Zhongde Wang, Bobby R. Hunt:
The discrete cosine transform-A new version. 1256-1259 - Eric W. Hansen:
New algorithms for Abel inversions and Hankel transforms. 1260-1263 - Russell M. Mersereau, E. W. Brown III, Abderrezak Guessoum:
Row-column algorithms for the evaluation of multidimensional DFT'S on arbitrary periodic smapling lattices. 1264-1267 - Soo-Chang Pei, Eng-Fong Huang:
High speed 2D hexagonal convolution by polynomial transform. 1268-1271 - Roger Y. Tsai:
Multiframe image point matching and 3-D surface reconstruction. 1272-1275
Wideband and Mediumband Speech Coding
- C. D. Heron, Ronald E. Crochiere, Richard V. Cox:
A 32-band sub-band/Transform coder incorporating vector quantization for dynamic bit allocation. 1276-1279 - Joseph Rothweiler:
Polyphase quadrature filters-A new subband coding technique. 1280-1283 - Claude R. Galand, Daniel J. Esteban:
Multirate sub-band coder with embedded bit stream: Application to digital TASI. 1284-1287 - V. Ramamoorthy:
Design of a novel 32.8 kbs speech coder using modulo-PCM principles. 1288-1291 - Edward Angel, L. Don Daigle, Michael A. Rodriguez:
A maximum entropy encoder for speech. 1292-1295 - P. J. J. A. Wolters:
Enhancement of differentially coded speech signals by means of quantization error correction. 1296-1299 - Thomas R. Fischer:
On the tandem connection of differential encoding systems: The case of cascaded quantizers. 1300-1303 - P. J. Patrick, Raymond Steele, Costas S. Xydeas:
Frequency compression of 7.6 kHz speed into 3.3 kHz bandwidth. 1304-1307 - J. F. Galliano, Jean E. Menez, Claude R. Galand:
Fixed block rate quantizer using entropy coding. 1308-1311 - Luciano Bertorello, Maurizio Copperi:
Design of a 4.8/9.6 kbps baseband LPC coder using split-band and vector quantization. 1312-1315 - Harald Katterfeldt, E. Behl:
Implementation of a robust RELP speech coder. 1316-1319 - Per Hedelin:
RELP-vocoding with uniform and non-uniform down-sampling. 1320-1323 - Masud Arjmand, George R. Doddington:
Pitch-congruent baseband speech coding. 1324-1327
Speech Synthesis and Analysis
- Gérard Chollet, J. F. Galliano, J.-P. Lefevre, E. Viara:
On the generation and use of a segment dictionary for speech coding, synthesis and recognition. 1328-1331 - J. J. Yea, Ashok Krishnamurthy, Jayant M. Naik, G. P. Moore, Donald G. Childers:
Glottal sensing for speech analysis and synthesis. 1332-1335 - Susan R. Hertz, Mary E. Beckman:
A look at the SRS synthesis rules for Japanese. 1336-1339 - Anders Jonsson, Per Hedelin:
A Swedish text-to-speech system based on an area function model. 1340-1343 - William M. Fisher:
A text-to-speech development system. 1344-1347 - Webster P. Dove, Cory S. Myers, Alan V. Oppenheim, K. R. Davis, Gary E. Kopec:
Knowledge-based pitch detection. 1348-1351 - Bruce G. Secrest, George R. Doddington:
An integrated pitch tracking algorithm for speech systems. 1352-1355 - Wynn C. Stirling, John M. Turner:
Joint estimation of excitation and vocal tract response. 1356-1359 - David W. Shipman:
SpireX: Statistical analysis in the spire acoustic-phonetic workstation. 1360-1363 - James C. Anderson, Campbell L. Searle:
Speech analysis/Synthesis based on perception. 1364-1367 - Richard S. Goldhor:
A speech signal processing system based on a peripheral auditory model. 1368-1371 - Hideki Kasuya, Yasunori Kobayashi, Takao Kobayashi, Satoshi Ebihara:
Characteristics pf pitch period and amplitude perturbations in pathologic voice. 1372-1375 - Mark A. Clements, Louis D. Braida, Nathaniel I. Durlach:
Speech processing for artificial tactile speech displays. 1376-1379
Audio and Electroacoustics
- F. Cahn:
Pitch translation of trumpet tones. 1380-1383 - Gary W. Schwede:
An algorithm and architecture for constant-Q spectrum analysis. 1384-1387 - Douglas Preis, H. Polchlopek:
Restoration of nonlinearly distorted magnetic recordings. 1388-1391 - Hikaru Date, Kimitoshi Fukudome, Keiichiro Mori, Masakazu Oda:
Observation of reverberation curves by two relatively prime pseudorandom sequences. 1392-1395 - P. Jeffrey Bloom, Douglas Preis:
Perceptual identification and discrimination of phase distortions. 1396-1399 - Michael A. Krasner, K. Stevens, D. Green, S. Blumenthal:
Development of a technique for high frequency audiometry. 1400-1403 - George J. Boggs:
A preliminary model of subjective voice quality. 1404-1406 - George J. Frye:
A harmonic distortion standard. 1407-1410 - Said E. El-Khamy, Onsy A. Abdel-Alim:
Omnidirectional coded loudspeaker arrays. 1411-1414 - J. Robert Ashley, Larry J. Fruit:
Complex matrix inversion algorithm for calculating digital system frequency response. 1415-1418
Nonparametric Spectral Analysis and SVD Methods
- S. Lawrence Marple Jr.:
A fast computational algorithm for and performance of the Kumaresan-Prony method of spectrum analysis. 1419-1421 - Yu Hen Hu, Sun-Yuan Kung:
Highly concurrent Toeplitz eigen-system solver for high resolution spectral estimation. 1422-1425 - Leland B. Jackson:
Simple, effective MA and ARMA techniques. 1426-1429 - Allan O. Steinhardt, Richard A. Roberts:
An optimization theoretic framework for spectral estimation. 1430-1433 - Eli Fogel, Michael J. Villalba:
On the sampling rate issue in spectral analysis. 1434-1437 - Gloria Faye Boudreaux, Thomas W. Parks:
Reducing aliasing in the Wigner distribution using implicit spline interpolation. 1438-1441 - Miguel Angel Lagunas:
Cepstrum constraints in ME spectral estimation. 1442-1445 - Monson H. Hayes:
The representation of signals in terms of spectral amplitude. 1446-1449 - Monson H. Hayes, Ronald W. Schafer:
On the bandlimited extrapolation of discrete signals. 1450-1453 - Arthur B. Baggeroer:
Confidence interval determination for spectral estimates using "Tilted densities". 1454-1457 - Aharon Levi, Henry Stark:
Signal restoration from phase by projections onto convex sets. 1458-1460
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