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ORCIDICASSP 1982: Paris, France
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '82, Paris, France, May 3-5, 1982. IEEE 1982
Plenary: Impact of Digital Signal Processing on our Society
- Maurizio Dècina:
CCITT Activity on signal processing for integrated services digital networks. 5-10 - Gösta H. Granlund, Hans Knutsson:
Hierarchical processing of structural information in artificial intelligence. 11-16 - Richard A. Guedj:
Human-machine interaction and digital signal processing. 17-19
Digital Signal Processing
Fourier and Polynomial Transforms
- Howard W. Johnson, C. Sidney Burrus:
The design of optimal DFT algorithms using dynamic programming. 20-23 - Christos Caraiscos, Bede Liu:
Two dimensional DFT using mixed time and frequency decimations. 24-27 - V. V. Cizek:
Recursive calculation of Fourier transform of discrete signal. 28-31 - V. Ralph Algazi, Bernard J. Fino:
Performance and computation ranking of fast unitary transforms in applications. 32-35 - Henri J. Nussbaumer:
A polynomial transform approach to transmultiplexing. 36-39 - G. Robert Redinbo, Dean O. Carhoun, Bruce L. Johnson:
Fast algorithms for signal processing using finite field operations. 40-43 - O. M. Makarov:
On the computational complexity of bilinear forms evaluation over a body of quaternions. 44-47
Quantization Effects
- Giovanni L. Sicuranza:
On the accuracy of 2-D digital filter realizations using logarithmic number systems. 48-51 - Anna Z. Baraniecki, Graham A. Jullien:
Quantization error and limit cycles analysis in residue number system coded recursive filters. 52-55 - A. S. Ramnarayan, Fred J. Taylor:
Analysis of errors in residue number system (RNS) based IIR digital filters. 56-59 - W. Kenneth Jenkins:
Failure resistant digital filters based on residue number system product codes. 60-63 - Francis M. Boland, James O. Normile:
Quantization and truncation effects in the design of adaptive digital filters. 64-68 - Ganapati Panda, Ranendra N. Pal, B. Chatterjee:
Fixed-point error analysis of rectangular transform. 69-72 - Charles M. Rader:
The application of dynamic programming to the optimal ordering of digital filter sections. 73-76
Audio
Digital Audio
- David V. James, Noah Mendelsohn, David R. Fuchs:
Digital audio mixer: A VLSI approach. 77-80 - Piet J. Berkhout, Ludwig D. J. Eggermont:
Some design issues in digital signal processing for digital-audio systems. 81-84 - James Anderson Moorer:
The Lucasfilm audio signal processor. 85-88 - Scott Foster, A. Joseph Rockmore:
Signal processing for the analysis of musical sound. 89-92 - Roger Lagadec, Daniele Pelloni, Daniel Weiss:
A 2-channel, 16-bit digital sampling frequency converter for professional digital audio. 93-96 - S. Fuchs, M. Seguin, A. Weisser:
Digital parametric filters for studio mixing desk. 97-100 - Tor A. Ramstad:
Sample-rate conversion by arbitrary ratios. 101-104
Underwater Acoustics
Medium Effects
- U. E. Rupe:
Modulation of acoustic signals in a shallow water using a normal-mode model. 105-108 - Junhua Xu, Geng Chen:
A corrected match for the coherent part of a time-variant channel. 109-112 - Geneviève Jourdain, George Tziritas:
Communication in a fluctuating channel models and use of explicit or implicit diversity. 113-116 - William J. Vetter:
Impulse response for the one-dimensional inhomogeneous medium with an approximation for attenuation and dispersion. 117-120 - Russell P. Kraft, John F. McDonald, J. Erkes:
Homomorphic signal dereverberation for a phased array imaging system. 121-124
Image Processing
Multidimensional Spectral Analysis
- Stephen W. Lang, James H. McClellan:
The extension of Pisarenko's method to multiple dimensions. 125-128 - Naveed A. Malik, Jae S. Lim, Michelle J. Glaser:
Properties of two dimensional maximum entropy power spectrum estimates. 129-132 - Thomas L. Marzetta:
The algebraic inversion of 2-D autoregressive power spectra with applications to spectral estimation. 133-135 - Hayri Korezlioglu, Philippe Loubaton:
On 2-D spectral factorization. 136-139 - Bir Bhanu
:
Computation of two-dimensional complex cepstrum. 140-143 - P. Karivaratha Rajan, Harnatha C. Reddy, M. N. Shanmukha Swamy:
Further results on 4-fold rotational symmetry in 2-D functions. 144-147
Speech Systems
Speech Enhancement and Noise Reduction
- Douglas M. Chabries, Richard W. Christiansen, Robert H. Brey, Martin S. Robinette:
Application of the LMS adaptive filter to improve speech communication in the presence of noise. 148-151 - M. S. Ahmed:
Estimating the parameters of a noisy AR-process by using a bootstrap estimator. 152-155 - Thomas Langhans, Hans Werner Strube:
Speech enhancement by nonlinear multiband envelope filtering. 156-159 - David Malah, Richard V. Cox:
A generalized comb filtering technique for speech enhancement. 160-163 - P. Jeffrey Bloom, Gerald D. Cain:
Evaluation of two-input speech dereverberation techniques. 164-167 - Gerhard Doblinger:
"Optimum" filter for speech enhancement using integrated digital signal processors. 168-171
Speech Synthesis and Recognition
Pitch Detection
- Bruce G. Secrest, George R. Doddington:
Postprocessing techniques for voice pitch trackers. 172-175 - Jordan Cohen:
A pitch measurement algorithm for speech. 176-179 - Philippe Martin:
Comparison of pitch detection by cepstrum and spectral comb analysis. 180-183 - Geoff J. Bristow, Frank Fallside:
An autocorrelation pitch detector with error correction. 184-187 - Robert J. Sluyter, H. J. Kotmans, Theo A. C. M. Claasen:
Improvements of the harmonic-sieve pitch extraction scheme and an appropriate method for voiced-unvoiced detection. 188-191 - John Laver, Steven M. Hiller, R. J. Hanson:
Comparative performance of pitch detection algorithms on dysphonic voices. 192-195
Speech Coding
Medium Band Coding I
- Barbara J. McDermott, Carlo Scagliola:
The perception of spectrally shaped additive noise in speech. 196-198 - James L. Melsa, Arun Pande:
Mediumband speech encoding using time-domain harmonic scaling and adaptive residual coding for noisy channels. 199-202 - Tor A. Ramstad:
Sub-band coder with a simple adaptive bit-allocation algorithm a possible candidate for digital mobile telephony? 203-207 - Ronald S. Cheung:
Real-time implementation of a 9600 bps subband coder with time-domain harmonic scaling. 208-211 - Maurizio Copperi:
A variable rate embedded-code speech waveform coder. 212-215 - Chong Kwan Un, Jong Rak Lee:
On spectral flattening techniques in residual-excited linear prediction vocoding. 216-219 - Claude R. Galand, K. Daulasim, Daniel J. Esteban:
Adaptive predictive coding of base-band speech signals. 220-223
Digital Signal Processing
Rational Model Identification
- Kalle-J. Bry, Joël Le Roux:
Comparison of some algorithms for identifying autoregressive signals in the presence of observation noise. 224-227 - Donald F. Gingras:
Estimation of the autoregressive parameters from observations of a noise corrupted autoregressive time series. 228-231 - Vijay K. Jain, Tapan K. Sarkar, Donald D. Weiner:
Noise correction approach for pole-zero modeling by pencil-of-functions method. 232-235 - J. Bee Bednar, B. J. Roberts:
The R and S arrays and the AIC in ARMA modeling and filter design. 236-239 - Stephen P. Bruzzone, Mostafa Kaveh:
Statistical efficiency of the sample autocorrelation function in ARMA parameter estimation. 240-243 - Bruce R. Musicus:
An iterative algorithm for finding stable solutions to the covariance or modified covariance autoregressive modeling methods. 244-247 - Benjamin Friedlander:
Instrumental variable methods for ARMA spectral estimation. 248-251 - Marc Prevosto, Albert Benveniste, Bruno Barnouin:
Identification of vibrating structures subject to non stationary excitation : A non stationary stochastic realization problem. 252-255 - James A. Cadzow, Behshad Baseghi:
Data adaptive ARMA modeling of time series. 256-261 - Hans-Eberhard Schurk, Ulrich Appel, Werner Wolf:
Parallel identifiers for parameter estimation of strongly disturbed ARMA-processes. 262-265 - W. J. Shanahan:
Circuit models for prediction, Wiener filtering, Levinson and Kalman filters for ARMA time series. 266-269
Filter Design I
- Richard R. Kurth:
Design of FIR filters to complex frequency response specifications. 270-273 - Federico Bonzanigo:
Some improvements to the design programs for equiripple FIR filters. 274-277 - Tapio Saramäki:
Narrowband linear-phase FIR filters requiring a small number of multipliers. 278-281 - Yong Ching Lim, Sydney R. Parker:
Digital lattice filter design using a frequency domain modeling approach. 282-285 - Hon Keung Kwan:
Design of passive second-order digital filters. 286-289 - Hans Wilhelm Schüßler, P. Möhringer, Peter Steffen:
On optimal equalization of an analog antialiasing filter with a nonrecursive digital system. 290-293 - Ezio Biglieri:
Theory of volterra processors and some applications. 294-297 - Bernard C. Picinbono:
Quadratic filters. 298-301 - Frederick L. Kitson, Lloyd J. Griffiths:
The design of time-varying digital filters which employ binary valued coefficients. 302-305 - Andres C. Salazar, Victor B. Lawrence:
Design and implementation of transmitter and receiver filters with periodic coefficient nulls for digital systems. 306-310 - Daniel D. Rivers, Robert A. Rosen:
Efficient formation of filter banks with frequency dependent resolution. 311-314 - Thomas G. Marshall Jr.:
Structures for digital filter banks. 315-318 - Greg C. Copeland:
Transmultiplexers used as adaptive frequency sampling filters. 319-322
Digital Signal Processing Applications
Signal Processing Applications and Radar
- J. Douglas Birdwell, T. J. Paulus, C. J. Mazzola, L. Czapla:
On the generation of accurate high frequency acoustic pulses using modern control theory. 323-326 - Susan K. Numrich, Laurence J. Frank, Louis R. Dragonette:
Acoustic classification of submerged targets. 327-330 - Manell Zakharia, Jean-Pierre Sessarego:
Sonar target classification using a coherent echo processing. 331-334 - J. Terry Ginn:
Time varying autoregressive signal models, with an application to chirped signals. 335-338 - James H. Hesson, James F. Kaiser:
On external properties satisfied by the Io-sinh window. 339-342 - Gyula Hermann, László Horváth, László Monostori:
Real-time monitoring of machine tools via Walsh-Hadamard tranform. 343-346 - Norberto F. Ezquerra, Linda Harkness:
Application of pattern recognition techniques to the processing of radar signals. 347-350 - K. Y. Liu:
A modular fast two-dimensional cyclic convolver and its application to real-time synthetic aperture radar processing. 351-354 - William J. Steinway, Charles M. Luke, Jim D. Echard:
Locating voids beneath pavement using a pulsed radar. 355-358 - D. Baumgarten:
Optimum detection and receiver performance for multistatic radar configurations. 359-362 - William A. Holm, Jim D. Echard:
FFT Signal processing for non-coherent radar systems. 363-366 - J. L. Pourailly, J. De Reffye, Claude Legendre:
Spatial digital processing: Application to radar antennas. 367-370
Underwater Acoustics and Digital Signal Processing Applications
Time Delay Estimation
- Kent Scarbrough, Nasir Ahmed, Dae Hee Youn, G. Clifford Carter:
On the scot and roth algorithms for time delay estimation. 371-374 - J. P. Ianniello:
Threshold effects in time delay estimation using narrowband signals. 375-378 - Jean-Melaine Favennec, B. Georgel, J. Masson:
Time delay estimation: Application to flow rate measurement of cooling fluid in nuclear power plants. 379-382 - Heinrich Meyr, Gerhard Spies, Jörg Bohmann:
Real-time estimation of moving time delay. 383-386 - A. F. Hassan, E. K. Al-Hussainy, M. Bakry:
Nonparametric detectors for signal detection and time delay estimation. 387-390 - Rui J. P. de Figueiredo, Andreas Gerber:
Separation of superimposed signals by a cross-correlation method. 391-394 - J. Pearson, C. J. Macleod, Tariq S. Durrani:
Mode and time delay estimation for non-destructive evaluation systems. 395-398 - Thomas W. Parks, Charles F. Morris, John D. Ingram:
Velocity estimation from short-time temporal and spatial frequency estimates. 399-402 - José M. F. Moura:
Recursive techniques for passive source location. 403-406 - Philippe Bolon:
Speed measurement by cross-correlation - theoretical and practical aspects. 407-410 - Michael J. Coker, E. Ferrara:
A new method for multiple source location. 411-415 - Benjamin Friedlander, Boaz Porat:
A parametric technique for time delay estimation. 416-419 - Zi-Qiang Hou, Zhen-Dong Wu:
A new method for high resolution estimation of time delay. 420-423 - Luís F. Rocha:
Adaptive delay tracking with a delay-lock estimator. 424-427
Image Processing
Image Coding
- Allen Gersho, Bhaskar Ramamurthi:
Image coding using vector quantization. 428-431 - Roland Wilson, Hans Knutsson, Gösta H. Granlund:
Image coding using a predictor controlled by image content. 432-435 - M. Kocher, M. Kunt:
A contour-texture approach to picture coding. 436-439 - P. Fäh, M. Kunt:
Efficient coding of high resolution typographic characters. 440-443 - F. J. Schmitt:
Color texture reconstruction using a bidimensional Markov model. 444-447 - M. Götze, G. Ocylok:
An adaptive interframe transform coding system for images. 448-451 - Nikolaos G. Bourbakis, Nikitas A. Alexandridis:
An efficient, real-time, method for transmitting Walsh-Hadamard transformed pictures. 452-455 - Alberto Sanz, Carlos Muñoz, Narciso García:
On the use of splines in hierarchical image transmission. 456-459 - Claude Labit, Albert Benveniste:
Motion of edges and motion estimation in a sequence of T.V. pictures. 460-463 - Thomas S. Huang, Y. P. Hsu, Roger Y. Tsai:
Interframe coding with general two-dimensional motion compensation. 464-466 - Mario Guglielmo, R. Marion, A. Sciarappa:
Subjective quality evaluation of different intraframe adaptive coding schemes, based on orthogonal transformations. 467-470 - B. Cochrane, K. P. Dawson, Michael A. Fiddy, Trevor J. Hall:
Sampling and interpolation in two dimensions. 471-474
Speech Systems
Digital Signal Processing and Speech System Implementations
- J. Vignes, P. Bois:
Analysis of the numerical stability of algorithms. 475-478 - J. P. Tressières, Francis Castanie:
Optimization of random quantization. 479-483 - Kenji Nakayama:
A discrete optimization method for high-order FIR filters with finite wordlength coefficients. 484-487 - David C. Munson Jr., Emily C. Martin:
Sampling rates for linear shift-variant discrete-time systems. 488-491 - M. Balakrishnan, A. V. S. M. Rao, Rajendar Bahl:
A multi-channel microprogrammed FFT processor. 492-497 - C. A. Wambergue, Richard A. Roberts:
Block processing structures for fixed point digital filtering. 498-501 - M. R. Jarmasz, Gert O. Martens:
A simple design for a fast sliding DFT computer. 502-505 - Hani Mahdi:
A novel structure for implementing DFT-filter banks. 506-509 - Jean-Sylvain Liénard, J. Y. Jourdain, P. Lambert:
Digital modular technology: Application to matched filtering. 510-513 - Pierre Badin, Daniel Degryse:
Speech communication hardware. 514-516