default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
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 - Michael McMahan, David Cox, Michael S. Wengrovitz:
Speech processing using programmable VLSI. 517-520 - P. Zuidweg, Jef L. van Meerbergen, M. L. van der Meulen:
Custom LST chip-set for speech analysis. 521-524 - Richard V. Cox, Ronald E. Crochiere:
A single chip speech periodicity detector. 525-528
Speech Recognition
Discrete Utterance Recognition
- W. W. Wiezlak, Ryszard Gubrynowicz:
Articulatory description of speech signal in isolated word recognizer. 529-534 - Kjell Elenius, Mats Blomberg:
Effects of emphasizing transitional or stationary parts of the speech signal in a discrete utterance recognition system. 535-538 - Andres Buzo, Horacio G. Martinez, Carlos Rivera:
Discrete utterance recognition based upon source coding techniques. 539-542 - J. M. Turner:
Application of recursive exact least square ladder estimation algorithm for speech recognition. 543-545 - David W. Shipman, Victor W. Zue:
Properties of large lexicons: Implications for advanced isolated word recognition systems. 546-549 - Günther Ruske:
Automatic recognition of syllabic speech segments using spectral and temporal features. 550-553 - Gary L. Bradshaw, Ronald A. Cole, Zongge Li:
A comparison of learning techniques in speech recognition. 554-557 - Lori Faith Lamel, Victor W. Zue:
Performance improvement in a dynamic-programming-based isolated word recognition system for the alpha-digit task. 558-561 - E. Schulze:
Hypothesizing of words for isolated and connected word recognition systems based on phonem preclassification. 562-565 - Raimo Bakis, N. Rex Dixon:
Toward speaker-independent recognition-by-synthesis. 566-569 - Roberto Bisiani, Alex Waibel:
Performance trade-offs in search techniques for isolated word speech recognition. 570-573 - Roberto Billi:
Vector quantization and Markov source models applied to speech recognition. 574-577 - Harvey F. Silverman, N. Rex Dixon:
Some general, user-oriented concepts for discrete utterance recognition (DUR) application. 578-581
Speech Coding
Narrow Band Coding
- Salim E. Roucos, John I. Makhoul, Richard M. Schwartz:
A variable-order Markov chain for coding of speech spectra. 582-585 - Josef Heiler:
Optimized frame selection for variable frame rate synthesis. 586-588 - Panos E. Papamichalis, George R. Doddington:
Time encoding of LPC roots. 589-592 - Robert M. Gray, Hüseyin Abut:
Full search and tree searched vector quantization of speech waveforms. 593-596 - Biing-Hwang Juang, Augustine H. Gray Jr.:
Multiple stage vector quantization for speech coding. 597-600 - Jean-Pierre Adoul, Philippe Mabilleau:
4800 Bps RELP Vocoder using vector quantization for both filter and residual representations. 601 - Thomas E. Carter, Duncan M. Dlugos, D. C. LeDoux:
An 800 Bps real-time voice coding system based on efficient encoding techniques. 602-605 - David Y. Wong, Biing-Hwang Juang:
Voice coding at 800 BPS and lower data rates with LPC vector quantization. 606-609 - V. Viswanathan, Michael G. Berouti, Alan L. Higgins, William Russell:
A harmonic deviations linear prediction vocoder for improved narrowband speech transmission. 610-613 - Bishnu S. Atal, Joel R. Remde:
A new model of LPC excitation for producing natural-sounding speech at low bit rates. 614-617 - Arild Lacroix, Bela Makai:
A vocoder scheme for very low bit rates (quality evaluation). 618-621 - David Malah:
Cepstral residual vocoder for improved quality speech transmission at 4.8 kbps. 622-625
Digital Signal Processing
Adaptive Filtering
- Lennart Ljung:
Recursive identification techniques. 627-630 - John R. Treichler, Michael G. Larimore:
Thinned impulse responses for adaptive FIR filters. 631-634 - Delores M. Etter, M. J. Hicks, K. H. Cho:
Recursive adaptive filter design using an adaptive genetic algorithm. 635-638 - I. D. Landau, Luc Dugard, S. Cabrera:
Applications of output error recursive estimation algorithms for adaptive signal processing. 639-642 - Miguel Angel Lagunas Hernandez, Enrique Masgrau-Gomez:
What does parameter mean in adaptive lattice algorithms. 643-646 - Ulrich Appel, Achim V. Brandt:
Recursive lattice algorithms with finite-duration windows. 647-650 - Aldo Cumani:
On a covariance-lattice algorithm for linear prediction. 651-654 - J. M. Turner:
Fast approximate whitening ladder filter. 655-658 - Luís F. Rocha:
Real time estimation of amplitudes phases and frequencies of a sampled signal plus noise by nulling processing. 659-662 - Sun-Yuan Kung, D. V. Bhaskar Rao:
Analysis and implementation of the adaptive notch filter for frequency estimation. 663-666 - C. Y. Chang:
Multichannel adaptive filtering with a feedback convergence function. 667-670 - Carey Gibson, Simon Haykin:
Non-stationary learning characteristics of adaptive lattice filters. 671-674
Hardware and Software
- Michel Auguin, Fernand Boéri:
Efficient multiprocessor architecture for digital signal processing. 675-678 - Thomas P. Barnwell III, C. J. M. Hodges, Mark A. Randolph:
Optimum implementation of single time index signal flow graphs on synceronous multiprocessor machines. 679-682 - Gordon L. DeMuth:
High order notation and automated program generation for realtime signal processing. 683-686 - J. R. Trinder:
Hardware-software configuration for high performance digital filtering in real-time. 687-690 - R. Geppert:
Novel structure of a user-programmable integrated digital signal processor. 691-694 - Klaus Kronlöf, Jorma Skyttä, Iiro Hartimo, Olli Simula:
Performance of an experimental data flow architecture for signal processing. 695-698 - Shigeyoshi Kawarai:
Bit parallel-serial implementation of combinatorial filter in canonical form. 699-702 - William Hall Evans, Jonathan Allen:
MOS Implementations of TTL architectures: A case study. 703-706 - Lajos Gazsi:
N-Port arithmetic unit for DSP. 707-710 - Tich T. Dao:
Knuth's complex number arithmetic revisited. 711-716 - John A. Eldon, Craig Robertson:
A floating point format for signal processing. 717-720
Digital Signal Processing Applications
Biomedical Signals and Aids for the Handicapped
- Furio Bartoli, Sergio Cerutti:
A Kalman filter procedure for the processing of the electroencephalogram. 721-724 - Bernard Chalmond:
Edge detection of the medullary canal of the femur: A Bayesian approach. 725-728 - Chrysostomos L. Nikias, Peter D. Scott, John H. Siegel:
A new robust 2-D spectral estimation method and its application in cardiac data analysis. 729-732 - Nicolaas J. I. Mars:
Time delay estimator for EEG analysis based on information theory. 733-735 - Gérard Charbonneau, Jean-Louis Racineux, M. Sudraud, E. Tuchais:
Digital processing techniques of breath sounds for objective assistance of asthma diagnosis. 736-738 - Maria Domenica Di Benedetto, Francis Destombes, Bernard Mérialdo, Jean-Pierre Tubach:
Phonetic recognition to assist lip-reading for deaf children. 739-742 - E. M. Bate, Frank Fallside, E. Gulian, P. Hinds, C. Keiller:
A speech training aid for the deaf with display of voicing, frication and silence. 743-746 - Rolf Carlson, Björn Granström, Sheri Hunnicutt:
Bliss communication with speech or text output. 747-750 - Kenneth Mark Colby, Daniel Christinaz, Roger C. Parkison, Mark Tiedemann:
Predicting word-expressions to increase output rates of speech prostheses used in communication disorders. 751-754 - A. King, A. Parker, M. Spanner, R. D. Wright:
A speech display computer for use in schools for the deaf. 755-758 - John R. Deller Jr.:
Evaluation of laryngeal dysfunction based on features of an accurate estimate of the glottal waveform. 759-762 - José M. Pardo:
Vocal tract shape analysis for children. 763-766
Underwater Acoustics and Digital Signal Processing Applications
Array Processing I
- Shalhav Zohar:
An improved beam-forming algorithm for adaptive arrays. 767-770 - Jean-Pierre Le Cadre, J.-L. Lambla:
Optimum array processing in presence of randomly distorded wavefronts. 771-774 - Norman L. Owsley, J. W. Law:
Dominant mode power spectrum estimation. 775-778 - Laurent Kopp, Georges Bienvenu, Marc Aiach:
New approach to source detection in passive listening. 779-782 - Jean-Paul Pignon:
Bearing and ranging simultaneous measurements of an acoustic source using a parametric method: Some results. 783-786 - Johann F. Böhme:
On the sensitivity of orthogonal beamforming. 787-790 - Tariq S. Durrani, Ken C. Sharman:
ARMA Techniques for the location of multiple sources from linear array data. 791-794 - Ménad Sidahmed:
Antenna array processing by multichannel ARMA models. 795-798 - Leon H. Sibul, Guy R. L. Sohie:
Implementation of an adaptive space-time processor by an unconstrained multichannel lattice. 799-802 - Robert A. Gabel, Richard R. Kurth:
Digital beamsteering with recursive multichannel filters. 803-806 - L. Meier:
Direct adaptive motion compensation for towed arrays. 807-810 - Jean-Louis Berrou, Ronald A. Wagstaff:
Virtual beams from an FFT beamformer and their use to assess the quality of a towed-array system. 811-814 - Yoh-Han Pao, Ahmed El Sherbini, Victor C. Chen:
Digital computer simulation study of an ultrasonic 3-D imaging system using frequency sweep and synthetic aperture techniques. 815-817
Image Processing
Image Analysis I
- Nimish Mehta, Kenneth C. Smith, F. E. Holmes:
Feature extraction as a tool for computer input. 818-820 - Richard S. Schlunt, Hans-Peter Schmid:
Real time image registration based on the Cauchy-Schwarz inequality. 821-824 - C. A. Darmon:
A recursive method to apply the Hough transform to a set of moving objects. 825-829 - Liu Jian, Francis Schmitt:
Water current determination by picture processing. 830-833 - Roger Y. Tsai, Thomas S. Huang:
Uniqueness and estimation of three-dimensional motion parameters of a rigid planar patch from three perspective views. 834-838 - Earl R. Barnes:
A procedure for classifying patterns. 839-842 - Jean Serra:
Digital morphology in the 3-D space. 843-845 - Serge Castan, Jun Shen:
A structural method of pattern recognition and its application to on line recognition of Chinese ideographs. 846-849 - Nicolas Tripon, Philippe Coueignoux:
Omnifont OCR with a structural model. 850-854 - Noamen Keskes, A. Boulanouar, Olivier D. Faugeras:
Application of image analysis techniques to seismic data. 855-858 - Worthy N. Martin, Jake K. Aggarwal:
Dynamic scenes and object descriptions. 859-862
Speech Systems
Speech Recognition Systems
- J. Peckham, J. Green, J. Canning, P. Stephens:
LOGOS - A real time hardware continuous speech recognition system. 863-866 - Louis C. W. Pols:
How humans perform on a connected-digits data base. 867-870 - Tsuneo Nitta, T. Murata, Hiroyuki Tsuboi, K. Takeda, T. Kawada, Sadakazu Watanabe:
Development of Japanese voice-activated word processor using isolated monosyllable recognition. 871-874 - Erkki Reuhkala, Heikki Riittinen, Seppo Haltsonen, Olli Ventä, Kai Makisara, Teuvo Kohonen:
The on-line version of the Otaniemi speech recognition system. 875-878 - Michel Baudry, B. Dupeyrat:
A simple and efficient isolated words recognition system. 879-882 - Akio Komatsu, Akira Ichikawa, Kazuo Nakata, Yoshiaki Asakawa, Hiroko Matsuzaka:
Phoneme recognition in continuous speech. 883-886 - Christian Gagnoulet, Marc Couvrat:
Seraphine: A connected word speech recognition system. 887-890 - Jean-Luc Gauvain, Joseph-Jean Mariani:
A method for connected word recognition and word spotting on a microprocessor. 891-894 - G. S. Ramishvili, V. D. Serdiukov:
A system of speech communication with computer through noise. 895-898 - John S. Bridle, Michael D. Brown, Richard M. Chamberlain:
An algorithm for connected word recognition. 899-902 - Cesare Vicenzi, Carlo Scagliola:
Multiprocessor architecture for real-time speech recognition systems. 903-906 - John E. Shore, David K. Burton:
Discrete utterance speech recognition without time normalization. 907-910
Speech Synthesis and Recognition
Vocal Tract and Cord Models
- Shinji Maeda:
The role of the sinus cavities in the production of nasal vowels. 911-914 - B. M. Lobanov:
On the acoustic theory of coarticulation and reduction. 915-918 - S. N. Terepin, Frank Fallside:
A polynomial vocal tract model for speech synthesis. 919-922 - Hans Werner Strube:
Time-varying wave digital filters and vocal-tract models. 923-926 - N. Rao Vemula, A. Maynard Engebretson, David L. Elliott:
Estimation of vocal tract shape from input/Output measurements. 927-931 - Elizabeth Allwood, Celia Scully:
A composite model of speech production. 932-935
Speech Synthesis
- Eberhard Grossmann:
Speech synthesis in the time domain from text. 936-939 - Unto K. Laine:
PARCAS, A new terminal analog model for speech synthesis. 940-943 - Nelson Morgan:
Perceptually-based coding for root LPC synthesis. 944-946 - Roger M. Meli, Frank Fallside:
The modelling of F0 contours. 947-949 - Keikichi Hirose, Hiroya Fujisaki:
Analysis and synthesis of voice fundamental frequency contours of spoken sentences. 950-953
Speech Coding
32 kbit/s Coding (invited)
- Xavier Maitre, T. Aoyama:
Speech coding activities within CCITT: Status and trends. 954-959 - Takao Nishitani, Shinichi Aikoh, Takashi Araseki, Kazunori Ozawa, Rikio Maruta:
A 32 kb/s toll quality ADPCM codec using a single chip signal processor. 960-963 - D. Cointot:
A 32-kbit/sec ADPCM coder robust to channel errors. 964-967 - Jean-Marie Raulin, Jean-Louis Jeandot:
A 32 kbit/s PCM to ADPCM converter. 968-971 - Paul Mermelstein, D. J. Millar:
Adaptive predictive coding of speech and voiceband data signals. 972-975 - Yohtaro Yatsuzuka, Henri G. Suyderhoud:
An 32-kbps ADPCM encoding with a variable initially large leakage and adaptive dual loop predictors. 976-979 - Naohisa Ohta, Kazunari Irie, Takehiko Uno, Atsushi Iwata, Tomonori Aoyama:
A high quality ADM LSI codec at 32 kbit/s for digital speech communications. 980-983
Subjective Quality of Codecs (invited)
- David J. Goodman, Randy D. Nash:
Subjective quality of the same speech transmission conditions in seven different countries. 984-987 - Pierre Combescure, Alain Le Guyader, André Gilloire:
Quality evaluation of 32 kbit/s coded speech by means of degradation category ratings. 988-991 - D. L. Richards, G. J. Barnes:
Pay-off between quantizing distortion and injected circuit noise. 992-995 - Thomas P. Barnwell III, Schuyler R. Quackenbush:
An analysis of objectively computable measures for speech quality testing. 996-999 - Nobuhiko Kitawaki, Kenzo Itoh, Masaaki Honda, Kazuhiko Kakehi:
Comparison of objective speech quality measures for voiceband CODECs. 1000-1003 - William D. Voiers:
Measurement of intrinsic deficiency in transmitted speech: The diagnostic discrimination test (DDT). 1004-1007
Digital Signal Processing
Spectrum Analysis I
- Miquel Bertran-Salvans:
A generalized window method for spectral estimation. 1008-1011 - Jont B. Allen:
Applications of the short time Fourier transform to speech processing and spectral analysis. 1012-1015 - Louis L. Scharf, Claude Guéguen, Jean-Pierre Dugré, Nicolas Moreau:
Linear transformations and parametric spectrum analysis. 1016-1020 - Tariq S. Durrani, Avedis S. Arslanian:
Windows associated with parametric spectral estimators. 1021-1024 - Tran-Thong, Arif Kareem:
Spectral window in a MEM based spectrum analyzer. 1025-1027 - Chi Hau Chen:
On the Fougere's maximum entropy spectral analysis method. 1028-1029 - Hideaki Sakai, K. Orita, N. Iwama:
AR Spectrum analysis based on a noisy autocovariance sequence. 1030-1033 - Jean-Louis Lacoume, Mohamed Gharbi, Claudine Latombe:
Close frequencies resolution by maximum entropy spectral estimators. 1034-1037 - P. L. Sharma, C. S. Chen:
Auto-regressive spectral estimation of noisy sinusoids. 1038-1041 - T. Subba Rao, M. Yar:
Statistical analysis of frequency modulated signals. 1042-1045 - S. Hamid Nawab, Thomas F. Quatieri, Jae S. Lim:
Signal reconstruction from the short-time Fourier transform magnitude. 1046-1048
VLSI in Digital Signal Processing (invited)
- Fred Mintzer, Abraham Peled:
The architecture of the real-time signal processor. 1049-1052 - Glen J. Culler, E. Greenwood, Dave Harrison:
A high performance VLSI CMOS arithmetic processor chip. 1053-1056 - A. Frey Jr.:
A VLSI I/O chip for multiple signal processor architectures. 1057-1060 - David Karlin, R. E. Owen:
VLSI Building blocks for digital signal processing. 1061-1064 - Edward R. Caudel, Richard K. Hester, Khen-Sang Tan:
A chip set for audio frequency digital signal processing. 1065-1068 - M. Cand, P. Le Scan, A. Roset:
An integrated processor for adaptive and parallel algorithms. 1069-1072 - M. Yano, K. Inoue, T. Senba:
An LSI digital signal processor. 1073-1076 - Frederick A. Williams:
An expandable single-IC digital filter/Correlator. 1077-1080 - G. D. Covert:
A 32 point monolithic FFT processor chip. 1081-1083 - Bernard New, David Brain:
Bipolar VLSI facilitates Fourier transformation. 1084-1087 - K. Böttcher, Arild Lacroix, Maati Talmi, Dieter Wesseling:
Integrated floating point signal processor. 1088-1091
Underwater Acoustics
Detection Estimation
- Michèle Basseville:
Detection of jumps in mean and adaptive filtering. 1092-1095 - Yiu Tong Chan, D. Parks:
Estimation of coherence via ARMA modelling. 1096-1099 - Dae Hee Youn, Nasir Ahmed, G. Clifford Carter:
Estimation of magnitude-squared coherence function: An adaptive approach. 1100-1103 - Albert H. Nuttal:
Direct coherence estimation via a constrained least-squares linear-predictive fast algorithm. 1104-1107 - Joseph R. Lapointe Jr.:
The impact of signal overcontainment on cross-correlation detection performance. 1108-1111 - Magnus Moll:
Detection performance of an operator using lofar. 1112-1115 - W. Brandenburg:
Measurement of azimuth and distance using arbitrary sensor configuration with unknown coordinates. 1120-1123 - J. Schiller:
Detection and bearing angle estimation of low flying aircraft by acoustical means. 1124-1127 - Otis L. Frost:
Optimum demodulation of multiple signals having overlapped spectra. 1128-1131 - Malik Mamode, Yvon Biraud, Bernard Escudié:
Bat's sonar signals and acceleration tolerance. 1132-1135
Image Processing
Statistical Image Processing
- Sudhir S. Dikshit:
An adaptive Kalman window filter to restore degraded images. 1136-1141 - Sunil Patil, Maher A. Sid-Ahmed, Malayappan Shridhar:
Noise reduction in images using statistical filtering. 1142-1145 - Jan Biemond:
A fast Kalman filter for images degraded by both blur and noise. 1146-1149 - John W. Woods, Subrahmanyam Dravida:
Two-dimensional recursive estimation for ARMA signal models. 1150-1153 - M. J. D. Bishop, Tariq S. Durrani:
An estimator for image desmearing using a Bernoulli-Gaussian model. 1154-1157 - Sally L. Wood:
Efficient MVE image reconstruction for arbitrary measurement geometries. 1158-1161
Image Understanding (invited)
- Olivier D. Faugeras:
Image understanding and graph matching. 1162-1165 - Steven A. Shafer, Takeo Kanade:
Recursive region segmentation by analysis of histograms. 1166-1171 - M. Kunt:
Edge detection : A tuttorial review. 1172-1175 - Jean Serra:
Image segmentation or image understanding ? 1176-1178 - Hans-Hellmut Nagel:
Recent advances in motion interpretation based on image sequences. 1179-1186 - Björn Kruse, Björn Gudmundsson, Dan Antonsson, Tomas Hedblom, Arne Linge, Peter Lord, Tomas Ohlsson:
Hardware for image processing and analysis: The PICAP approach. 1187-1190
Image Systems
Image Coding and Hardware
- H. Keller, A. Favre, A. Comazzi:
Nonlinear local image transforms with a new type of pipelined processor. 1191-1194 - Finn Jørgensen, G. Michel, Charles Wagner:
Predite, a real time processor for bandwith compression in TV transmission. 1195-1198 - Benkt Linnander, Lars-Erik Nordell, Björn Kruse:
A real-time relational processor. 1199-1202 - Hari K. Nagpal, Graham A. Jullien, William C. Miller:
Memory architecture of a video-rate image convolver. 1203-1206 - Hans-J. Alker, Kjell Andreassen:
A hardware digital processor for image bandwidth compression. 1207-1210 - Manuel Gonzalez, Jorge Gonzalez:
MIP: A flexible, microprogrammable image processor. 1211-1214 - F. Delamotte, M. C. Gennero, Alain Poli:
Data transmission and error correcting codes. 1215-1218 - S. Thomas Alexander, Sarah A. Rajala:
Analysis and simulation of an adaptive image coding system using the LMS algorithm. 1219-1222 - D. K. Mitrakos, George A. Constantinides:
Composite source coding techniques for image bandwidth compression. 1223-1226 - Petros Maragos, Russell M. Mersereau, Ronald W. Schafer:
Some experiments in ADPCM coding of images. 1227-1230 - Volker Märgner:
Pre-processing technique for block coding of graphics. 1231-1234 - Joseph Ronsin, J. Dewitte:
Adaptive block truncation coding scheme using an edge following algorithm. 1235-1238
Speech Recognition
Dynamic Time Warping
- Masaaki Okochi, Toshiyuki Sakai:
Trapezoidal DP matching with time reversibility. 1239-1242 - Jean-Pierre Banâtre, Patrice Frison, Patrice Quinton:
A systolic algorithm for connected word recognition. 1243-1246 - Hollis L. Fitch:
Relative timing measures of acoustic segments aid automatic word recognition. 1247-1250 - K. L. Greer, Bruce T. Lowerre, Lynn D. Wilcox:
Acoustic pattern matching and beam searching. 1251-1254 - Michael K. Brown, Lawrence R. Rabiner:
Dynamic time warping for isolated word recognition based on ordered graph searching techniques. 1255-1258 - Kuldip K. Paliwal, Anant Agarwal, Sarvajit S. Sinha:
A modification over Sakoe and Chiba's dynamic time warping algorithm for isolated word recognition. 1259-1261 - R. W. Brown:
Segmentation for data reduction in isolated word recognition. 1262-1265 - Yasuhiro Nara, K. Iwata, Yuji Kijima, Atsuhito Kobayashi, Shinta Kimura, S. Sasaki, J. Tanahashi:
Large-vocabulary spoken word recognition using simplified time-warping patterns. 1266-1269 - Roger K. Moore, Martin J. Russell, Michael J. Tomlinson:
Locally constrained dynamic programming in automatic speech recognition. 1270-1273 - Mark A. Yoder, Leah J. Siegel:
Dynamic time warping algorithms for SIMD machines and VLSI processor arrays. 1274-1277
Speech Analysis
Speech Analysis
- Dennis H. Klatt:
Prediction of perceived phonetic distance from critical-band spectra: A first step. 1278-1281 - Richard F. Lyon:
A computational model of filtering, detection, and compression in the cochlea. 1282-1285 - Periagaram K. Rajasekaran, J. C. Hansen:
Finite word length effects of the Leroux-Gueguen algorithm in computing reflection coefficients. 1286-1290 - Alan B. Poritz:
Linear predictive hidden Markov models and the speech signal. 1291-1294 - Osamu Kakusho, Masuzo Yanagida:
Hierarchical AR model for time varying speech signals. 1295-1298 - Takayuki Nakajima, Torazo Suzuki, Hiroshi Ohmura:
Non-steady state speech analysis method with dynamic feature enhancing effect. 1299-1302 - Luís B. Almeida, José M. Tribolet:
A spectral model for nonstationary voiced speech. 1303-1306 - Marc Baudry, Paul Deléglise, J.-C. Friedmann:
Speech analysis in the time domain using syntactic technics an attempt to formalize the description of phonemes using acoustical cues. 1307-1310 - M. Morgenthaler, C. Hansen:
Use of attributed grammars in speech signal processing. 1311-1313 - Harald Höge:
Adaptive estimation of some parameters of speech excitation. 1314-1317 - Peter L. Chu, David G. Messerschmitt:
Frequency weighted linear prediction. 1318-1321 - W. J. Borodziewicz:
An initial segmentation of the speech signal. 1322-1324
Digital Signal Processing
Non Stationary Modeling and Processing
- Wolfgang Martin:
Time-frequency analysis of random signals. 1325-1328 - Boualem Bouachache, Patrick Flandrin:
Wigner-Ville analysis of time-varying signals. 1329-1332 - David S. K. Chan:
A non-aliased discrete-time Wigner distribution for time-frequency signal analysis. 1333-1336 - Yves Grenier:
Time varying lattices and autoregressive models : Parameter estimation. 1337-1340 - Nian-Chyi Huang, Jake K. Aggarwal:
A comparsion between time- and frequency-domain techniques for time-varying signal processing. 1341-1344 - Gregory A. Clark, Sydney R. Parker, Sanjit K. Mitra:
Efficient realization of adaptive digital filters in the time and frequency domains. 1345-1348 - Ernest G. Baxa Jr.:
An evaluation of time redundant DFT processing of stochastic signals with time varying spectra. 1349-1352
Spectrum Analysis II
- Joël Le Roux, F. Giannella:
Whiteness of the measurement noise as a criterion for ARMA modelization of speech. 1353-1356 - Ramdas Kumaresan, Donald W. Tufts:
Accurate parameter estimation of noisy speech-like signals. 1357-1361 - Frank K. Soong, Allen M. Peterson:
On the high resolution and unbiased frequency estimates of sinusoids in white noise-A new adaptive approach. 1362-1366 - Charlton M. Walter, John D. Tardelli:
Signal compression model interrelationships in the time, frequency, principal component and canonical coordinate domains. 1367-1370 - Claude Guéguen, Ménad Sidahmed:
The singular case and robust linear prediction. 1371-1374 - S. Lawrence Marple Jr.:
Spectral line analysis via a fast Prony algorithm. 1375-1378
Equalization
- Richard D. Gitlin, Howard C. Meadors, Stephen B. Weinstein:
An algorithm for the stable operation of a digitally-implemented fractionally-spaced adaptive equalizer. 1379-1383 - Markus S. Mueller, Jean-Jacques Werner:
Adaptive echo cancellation with dispersion and delay in the adjustment loop. 1384 - Hikmet Sari:
Performance evaluation of three adaptive equalization algorithms. 1385-1389 - Eweda Eweda, Odile Macchi:
Equalization of rapid selective fadings with unknown and time-varying forms. 1390-1393
Echo Cancellation
- John G. McWhirter, K. J. Palmer, J. B. G. Roberts:
"A digital adaptive noise-canceller based on a stabilized version of the widrow L.M.S. algorithm". 1394-1397 - Frank K. Soong, Allen M. Peterson:
Fast least-squares (LS) in the voice echo cancellation application. 1398-1403 - Roberto Montagna, Luciano Nebbia:
Comparison of some algorithms for tap weight evaluation in adaptive echo cancellers. 1404-1407 - Theo A. C. M. Claasen, Niek A. M. Verhoeckx:
The influence of pseudo noise data signals on echo canceller performance. 1408-1411 - Raymond S. Medaugh, Lloyd J. Griffiths:
Further results of a least squares and gradient adaptive lattice algorithm comparison. 1412-1415 - G. Bouthemy, Wlodek Kofman, André Silvent:
Theoretical and computer simulation study of the "Correlofiltre" adaptative system for non stationary processes. 1416-1419 - Pierre-Yves Arquès, Gérard Faucon:
Proposal and experimental evaluation of a combined structure "Correlofilter-adapter" for the continuous estimation of a noisy signal with a reference noise. 1420-1423 - Hidefumi Kobatake:
A synchronous adaptive noise canceller for periodic interference. 1424-1427
Audio and DSP Applications
Audio and Electro Acoustics
- Vijay K. Jain, W. Marshall Leach Jr., Ronald W. Schafer:
Signal processing technique for measurement of vented-box loudspeaker system parameters. 1428-1431 - Rolf W. Zahn:
Circular electret microphones : Theoretical and experimental investigations. 1432-1435 - Adriano Depaoli, Giulio Modena, Aldo Reolon:
Measurements on non linear microphones by a speech-like signal. 1436-1439 - P. Skritek:
Measurement of nonlinear distortions in tape recorders and electro-acoustic transducers applying T.I.M. test signals. 1440-1443 - Rodolfo Ceruti, Franco Pira:
Application of echo-Cancelling techniques to audioconference. 1444-1447 - G. Ferrieu, P. Amstutz:
A new principle to avoid undesirable oscillations in electro acoustic loops application to the design of a hands free telephone set without voice switching. 1448-1451 - Herman J. M. Steeneken, Tammo Houtgast:
Evaluation of a physical method for estimating speech intelligibility in auditoria. 1452-1454 - J. Robert Ashley:
Listening evaluation of auditoriums. 1455-1458 - Hubert Caron, Daniel Laberge:
Octophony. 1459-1461 - Eiichi Miyasaka:
Timbre of complex tone bursts with time varying spectral envelope. 1462-1465 - Daniel Arfib:
The architecture of a digital sound synthesis system. 1466-1468 - Salim M. R. Taha, Majid A. H. Abdul-Karim:
VLSI Circuits for a sampling digital acoustic energy meter. 1469-1472
Underwater Acoustics
Array Processing II
- Chaohuan Hou, Shi-Zun Yan:
Adaptive array processing using predicted coefficients as constrained conditions. 1473-1476 - Wei-wen Gu:
Array processing for estimating multiple emitter parameters. 1477-1480 - Stanislav B. Kesler:
Generalized Burg algorithm for beamforming in correlated multipath field. 1481-1484 - Monique Chiollaz, Bernard Escudié, Yvon Biraud, G. Pachiaudi:
Interferometric acoustic imaging, joint representation and image deconvolution. 1485-1488 - Robert S. Walker, K. L. Déon:
Quantization degradation in superdirective processing of underwater acoustic arrays. 1489-1492 - W. W. Wolfe, M. J. Wilmut:
Simulation for testing array response. 1493-1496 - Zi-Qiang Hou, Zhi-Guang Li:
Optimum signal processing of vertical mode-selective array in shallow water. 1497-1500 - P. Monteil, V. Thiebaud:
Two channels random acoustics simulation and multichannel data acquisition. 1501-1504 - Norman L. Owsley:
Fitting polynomials to data in the presence of noise. 1505-1508 - Richard Klemm:
Optimum clutter suppression in airborne phased array radars. 1509-1512 - Michel Martin, Jean-François Piquard:
Dynamic focussing and ultrasonic imaging. 1513-1515 - Georges Poupinet, F. Glangeaud, Philippe Côté:
P-Time delay measurement of a doublet of microearthquakes. 1516-1519
Image Processing
Image Restoration and Reconstruction
- Anil K. Jain, Surendra Ranganath:
Image restoration and edge extraction based on 2-D stochastic models. 1520-1523 - Henry Stark:
Restoration and enhancement of arbitrary finite-energy images from incomplete spatial and spectral information. 1524-1526 - Henry J. Trussell:
Convergence of iterative restoration methods. 1527-1530 - Dominique Barba:
Decomposition and separated adaptive digital processing of degraded images with an visual quality criterion. 1531-1536 - Henri Maître, Armand J. Levy:
Superresolution using linear system methods a comparison. 1537-1540 - Vicente Casares Giner:
On the inversion of singular operators. 1541-1544 - Monson H. Hayes, Thomas F. Quatieri:
The importance of boundary conditions in the phase retrieval problem. 1545-1548 - Constantinos E. Goutis, Tariq S. Durrani:
An optimal technique for tomographic image reconstruction from curved ray projections. 1549-1552 - Mostafa Kaveh, Mehrdad Soumekh, Rolf K. Mueller:
Tomographic imaging via wave equation inversion. 1553-1556 - D. Hayner, S. Renjen, Thomas S. Huang, W. Kenneth Jenkins:
Algorithms and experimental results on image reconstruction from limited data. 1557-1560 - Darwin T. Kuan, Alexander A. Sawchuk, Timothy C. Strand, Pierre Chavel:
Nonstationary 2-D recursive filter for speckle reduction. 1561-1564
Speech Systems
Speech Analysis and Synthesis Systems
- Salim E. Roucos, Richard M. Schwartz, John Makhoul:
Segment quantization for very-low-rate speech coding. 1565-1568 - Frédéric Manceron, Jean-Sylvain Liénard:
Impulse analysis of speech : Spotting and preclassifying the impulses in the speech wave. 1569-1572 - Riichiro Mizoguchi, Masuzo Yanagida, Osamu Kakusho:
Speech analysis by selective linear prediction in the time domain. 1573-1576 - F. J. Owens, R. J. Linggard:
Analytic pole-zero modelling of speech spectra. 1577-1580 - Kil Ho Song, Chong Kwan Un:
Pole-zero modeling of noisy speech and its application to vocoding. 1581-1584 - Ramón García-Gómez, José M. Tribolet:
Speech analysis and modelling using a sequential ARMA estimation technique. 1585-1588 - Dennis H. Klatt:
The klattalk text-to-speech conversion system. 1589-1592 - Juan M. Santos, José R. Nombela:
Text-to-speech conversion in Spanish a complete rule-based synthesis system. 1593-1596 - Jean-Luc Courbon, Françoise Emerard:
Sparte: A text-to-speech machine using synthesis by diphones. 1597-1600 - Tai-Yi Huang, Cai-Fei Wang, Yoh-Han Pao:
A Chinese text-to-speech synthesis system based on an initial-final model. 1601-1603 - Rolf Carlson, Björn Granström, Sheri Hunnicutt:
A multi-language text-to-speech module. 1604-1607 - Hans-Wilhelm Rühl:
SYNTEX - A microprocessor based system for automatic conversion of German text to speech. 1608-1611
Speech Recognition
Connected Speech Recognition
- Gary M. Kuhn:
On talker-independent word recognition in continuous speech. 1612-1615 - Jean Paul Haton, Jean-Marie Pierrel, Simon Sabbagh:
Problems in the design and use of a connected speech understanding system. 1616-1620 - Lawrence R. Rabiner, A. F. Bergh, Jay G. Wilpon:
An embedded word training procedure for connected digit recognition. 1621-1624 - Henri Meloni, Jacques Guizol:
A speech recognition system. 1625-1628 - Peter F. Brown, James C. Spohrer, Peter H. Hochschild, James K. Baker:
Partial traceback and dynamic programming. 1629-1632 - Dominique Gillet, Patrice Quinton, Jacques Siroux:
From speech recognition to speech understanding : A case study of Keal. 1633-1636 - Joseph-Jean Mariani:
The AESOP continuous speech understanding system. 1637-1640
Speech Synthesis and Recognition
Speaker Recognition
- James C. Spohrer, Peter F. Brown, Robert Roth:
Automatic labeling of speech. 1641-1644 - Hermann Ney, Rainer Gierloff:
Speaker recognition using a feature weighting technique. 1645-1648 - Richard M. Schwartz, Salim E. Roucos, Michael G. Berouti:
The application of probability density estimation to text-independent speaker identification. 1649-1652 - N. Mohankrishnan, Malayappan Shridhar, Maher A. Sid-Ahmed:
A composite scheme for text-independent speaker recognition. 1653-1656 - Johannes Jaschul:
Speaker adaptation by a linear transformation with optimised parameters. 1657-1660 - Deidre Cimarusti, Russell B. Ives:
Development of an automatic identification system of spoken languages: Phase I. 1661-1663
Speech Coding and Analysis
Medium Band Coding II
- Luís B. Almeida, José M. Tribolet:
Harmonic coding: A low bit-rate, good-quality speech coding technique. 1664-1667 - Manfred R. Schroeder, Bishnu S. Atal:
Speech coding using efficient block codes. 1668-1671 - Masaaki Honda, Nobuhiko Kitawaki, Fumitada Itakura:
Adaptive bit allocation scheme in predictive coding of speech. 1672-1675 - Alain Le Guyader, André Gilloire:
Comparison of basic and simplified sequential algorithms for the computation of lattice filter predictor coefficients in ADPCM coding of speech. 1676-1679 - Costas S. Xydeas, Cumhur Cengiz Evci:
A comparative study of DPCM-AQF speech coders for bit rates of 16-32 kb/s. 1680-1683 - Claude R. Galand, Daniel J. Esteban:
16 Kbps sub-band coder incorporating variable overhead information. 1684-1687 - Vaikunth Gupta, Krishna Virupaksha:
Performance evaluation of adaptive quantizers for a 16-kbits/s sub-band coder. 1688-1691 - Ronald E. Crochiere, Richard V. Cox, James D. Johnston, Linda Seltzer:
A 9.6 kb/s speech coder using the Bell laboratories DSP integrated circuit. 1692-1695 - Kadiresan Annamalai, Tore Fjällbrant:
An adaptive transform coding system with short primary blocklengths and frequency domain quantization using feedback adaptation. 1696-1699 - J. Svean:
Quality measurements on speech coders for mobile radio. 1700-1703 - Bülent Sankur, Hilmi Güngen:
Signal coding properties of asynchronous delta modulator. 1704-1708 - William R. Daumer, J. L. Sullivan:
Subjective quality of several 9.6-32 kb/s speech coders. 1709-1712 - Chongxi Feng, Hui-Juan Yao, Weili Yang:
Adaptive quantization and prediction in speech coding. 1713-1716
Digital Signal Processing
Linear Models and Fast Algorithms
- Philippe Delsarte, Yves V. Genin, Yves G. Kamp:
On the mathematical foundations of the generalized Levinson algorithm. 1717-1720 - Ed F. Deprettere:
Fast non-stationary lattice recursions for adaptive modeling and estimation. 1721-1726 - Martin Morf, Carlos H. Muravchik, Ping Ang, Jean-Marc Delosme:
Fast Cholesky algorithms and adaptive feedback filters. 1727-1731 - Jean-Marc Delosme, Martin Morf:
Constant-gain filters for finite shift-rank processes. 1732-1735 - Patrick M. Dewilde:
How to approximate a spectrum recursively using ARMA models. 1736-1739 - Emmanuel N. Protonotarios, George Carayannis, Evangelos E. Milios:
Near to Toeplitz structure and efficient schemes for linear modeling. 1740-1743 - Nicholas Kalouptsidis, George Carayannis, Dimitris Manolakis:
On block matrices with elements of special structure. 1744-1747 - Daniel T. L. Lee, Martin Morf:
Generalized CORDIC for digital signal processing. 1748-1751 - Claude Samson, V. Umapathi Reddy:
Fixed point error analysis of the normalized ladder algorithm. 1752-1755 - Gérard Favier:
Identification of multivariable ARMA models by use of fast algorithms. 1756-1759 - Cristos C. Halkias, George Carayannis, Ioannis Dologlou, D. Emmanoulopoulos:
A new generalized recursion for the fast computation of the Kalman gain to solve the covariance equations. 1760-1763 - Fuyun Ling, John G. Proakis:
Generalized least squares lattice algorithm and its application to decision feedback equalization. 1764-1769
DSP in Communication Networks (invited)
- René Boite, Henri Leich:
Fast measurements of the attenuation and delay characteristics of data channels. 1770-1772 - Douglas Preis, Carey D. Bunks:
Minimax equalizers for digital communication. 1773-1776 - Loïc Guidoux, O. Le Riche:
Digital signal processing in a LSI 4.8 kbit/s modem. 1777-1780 - G. Pellegrini, A. Tofanelli:
Digits to the customer: How to approach the problem. 1781-1784 - M. Vry:
Digital local network systems: The impact of signal processing. 1785-1788 - A. Cook, A. R. Potter:
Digital signal processing applied to the subscriber line interface. 1789-1792 - P. Delpit, P. Lavanant, B. Morin:
Signal processing in the E10 digital switching system. 1793-1796 - David Mansour, Augustine H. Gray Jr.:
Effects of hybrid nonlinearity on a full-duplex telephone network with vocoder. 1797-1800 - Raymond Steele, Diane Vitello:
Embedding data in speech using scrambling techniques. 1801-1804 - Kah-Seng Chung, Leo E. Zegers:
Generalized tamed frequency modulation. 1805-1808
Filter Design II
- A. Hanafy, Joël Le Roux, J. Prado:
Iterative and non iterative techniques for the design of recursive digital filters. 1809-1812 - Guido M. Cortelazzo, Michael R. Lightner:
The use of multiple criterion optimization for the simultaneous phase and magnitude design of IIR digital filters. 1813-1816 - Yves G. Kamp, Christian Wellekens:
Optimal design of minimum phase FIR filters. 1817-1820 - Enrico Del Re, Pier Luigi Emiliani, Damiano F. Maffucci:
A minimum-phase FIR complex filter design for transmultiplexer implementation. 1821-1824 - R. Czarnach, Hans Wilhelm Schüßler, G. Rohrlein:
Linear phase recursive digital filters for special applications. 1825-1828 - Paul Loewenstein:
Digital pulse frequency demodulation using state-space filtering. 1829-1832
Signal Reconstruction and Deconvolution
- Carol Y. Espy, Jae S. Lim:
Effects of noise on signal reconstruction from Fourier transform phase. 1833-1836 - Francis Castanie:
An approximately unbiased recovery of discrete Fourier transforms altered by jittered sampling epochs. 1837-1840 - Rémy Prost, Robert Goutte:
Kernel splitting method in support constrained deconvolution for super-resolution. 1841-1844 - R. Marucci, Russell M. Mersereau, Ronald W. Schafer:
Constrained iterative deconvolution using a conjugate gradient algorithm. 1845-1848 - Masuzo Yanagida, Osamu Kakusho:
Least-squares method for multi-dimensional deconvolution. 1849-1852 - Francis M. Boland, T. Doyle:
Deconvolution in real time of noisy signals. 1853-1857 - John Mourjopoulos, Peter M. Clarkson, Joe K. Hammond:
A comparative study of least-squares and homomorphic techniques for the inversion of mixed phase signals. 1858-1861
DSP Applications
Signal Processing in Geophysics
- Andrew L. Kurkjian:
Maximum likelihood dereverberation with applications in sonic well logging. 1862-1865 - Guy Ruckebusch:
An application of Kalman filtering in nuclear well logging. 1866-1869 - Meemong M. Lee, Rao K. Yarlagadda:
Reversible seismic data compression. 1870-1873 - Piero Ciotti, Domenico Solimini:
Aspects of digital signal processing in radiometric remote sensing of geophysical variables. 1874-1877 - A. Barbagelata, E. Michelozzi, Dieter Rauch, Bernd Schmalfeldt:
Seismic sensing of extremely-low-frequency sounds in coastal waters. 1878-1881 - Peter M. Clarkson, Joe K. Hammond:
Improvement of the signal to noise ratio of seismic traces by re-alignment of reverberant energy. 1882-1885 - F. Glangeaud, Mohamed Gharbi, Nadine Martin, Jean-Louis Lacoume:
Use of multidimentional MEM spectral analysis in geophysics. 1886-1889 - L. Gunnar Ahlbom, Anders Forsen, Lars-Henning Zetterberg:
Signal processing for event detection. 1890-1893 - Torild van Eck, L. Gunnar Ahlbom:
Automatic event detection applied to single channel seismic records. 1894-1897 - Jerry M. Mendel:
Minimum-variance and maximum-likelihood recursive waveshaping. 1898-1901 - J. P. Benoist, F. Glangeaud, Nadine Martin, Jean-Louis Lacoume, C. Lorius, A. Ait Ouahman:
Study of climatic series by time-frequency analysis. 1902-1905
Image Processing
Image Analysis II
- Amar Mitiche, Jake K. Aggarwal:
Detection of edges using range information. 1906-1911 - Howard Elliott, F. R. Hansen:
Image segmentation using simple Markov field models. 1912-1915 - Its'hak Dinstein, Eugene I. Plotkin, A. M. Zayezdny:
Generalized phase planes for feature extraction. 1916-1919 - Jean-Pierre Gambotto:
Algorithms for region description and modifications based on chain code transformations. 1920-1923 - Gérard Rives, J. P. Derutin, Marc Richetin, Joseph Alizon, Jean Gallice:
An algorithm for isolated object location in digital images. 1924-1927 - S. Beucher:
Watersheds of functions and picture segmentation. 1928-1931 - Fernand Meyer:
The perceptual graph: A new algorithm. 1932-1935 - Wilfried Geuen, H.-G. Preuth:
New performance criteria of edge detection algorithms. 1936-1939 - Peter Bock, Aysel Basci:
A real-time automatic ranging algorithm for vision systems. 1940-1943 - Eric Térouanne:
On the interpretations of a polyhedral figure. 1944-1947
Speech Systems
Speech Coding Systems
- Anthony L. Shipman, Robert R. Bitmead, Gregory H. Allen:
Diffuse edge tracing using a predictor-corrector procedure with adaptive scope. 1948-1951 - Geert J. Bosscha, Robert J. Sluyter:
DFT-Vocoder using harmonic-sieve pitch extraction. 1952-1955 - Yeunung Chen:
An analysis-by-synthesis approach for automatic time segmentation of speech signals. 1956-1959 - Joel A. Feldman:
A compact digital channel vocoder using commercial devices. 1960-1963 - Fumitada Itakura, Takao Kobayashi, Masaaki Honda:
A hardware implementation of a new narrow to medium band speech coding. 1964-1967 - Vijay K. Jain:
Formant estimation by LPC with a new error criterion. 1968-1971 - Luciano Bertorello, Maurizio Copperi, Giancarlo Pirani, Fulvio Rusina:
Broadcasting-quality transmission of audio signals at 64 kbps. 1972-1975 - Pierre Combescure, Alain Le Guyader, M. Haghiri:
A.D.P.C.M Algorithms applied to wideband speech encoding (64 kbit/s, 0-7 kHz). 1976-1979 - Bhagwati Prasad Agrawal, Kishan Shenoi:
Specification-based design of ΣΔM for A/D and D/A conversion. 1980-1983 - Francis Charpentier:
Application of an optimization technique to the inversion of an articulatory speech production model. 1984-1987 - Lou Boves, Bert Cranen:
Evaluation of glottal inverse filtering by means of physiological registrations. 1988-1991 - Unto K. Laine:
Modelling of LIP radiation impedance in Z-domain. 1992-1995 - Hiroya Fujisaki, M. Tominaga:
Automatic recognition of voiced stop consonants in CV and VCV utterances. 1996-1999 - Guy Mercier, A. Callec, Jean Monné, M. Querre, O. Trevarain:
Automatic segmentation, recognition of phonetic units and training in the KEAL speech recognition system. 2000-2003 - Katsuhiko Shirai, Tetsunori Kobayashi:
Recognition of semivowels and consonants in continuous speech using articulatory parameters. 2004-2007 - Carlo Scagliola, Luciano Marmi:
Continuous speech recognition via diphone spotting a preliminary implementation. 2008-2011 - Gérard Charbonneau, Tarek Moussa:
Segmentation of continuous speech by using multidimensional scaling techniques. 2012-2014 - Leszek Kot:
A syntax-controlled segmentation of speech signal on the basis of dynamic spectra. 2015-2017 - Aaron E. Rosenberg, Lawrence R. Rabiner, Jay G. Wilpon:
Speaker trained recognition of large vocabularies of isolated words. 2018-2021 - Carolyn A. Vickroy, Harvey F. Silverman, N. Rex Dixon:
Study of human and machine discrete utterance recognition (DUR). 2022-2025 - Gérard Chollet, Christian Gagnoulet:
On the evaluation of speech recognizers and data bases using a reference system. 2026-2029 - Wayne A. Lea:
What causes speech recognizers to make mistakes? 2030-2033 - Gerhard Linnenberg:
Processing schemes for sampled multi-dimensional signals. 2034-2037 - Ettore Fornasini, Giovanni Marchesini:
State space approach to two dimensional filters. 2038-2041 - Bijan Lashgari, Leonard M. Silverman, Jean-François Abramatic:
Approximation of 2-D separable denominator recursive filters. 2042-2045 - Rachid Deriche, Jean-François Abramatic:
Design of 2-D recursive filters using singular value decomposition techniques. 2046-2049 - N. Nagamuthu, Maher A. Sid-Ahmed, Malayappan Shridhar:
Design of 2-D recursive digital filters with specified magnitude and constant group-delay responses by spectral factorization. 2050-2054 - Soo-Chang Pei, Yu-Chang Wu:
Design of nonsquare 2D FIR filters by transformations. 2055-2058 - Giovanni Garibotto, G. Piretta:
Three-dimensional recursive filtering. 2059-2062 - Christian Lantuéjoul, Jean Serra:
M-Filters. 2063-2066 - Alan C. Bovik, Thomas S. Huang, David C. Munson Jr.:
Nonlinear filtering using linear combinations of order statistics. 2067-2070 - Amal El Fallah, Richard A. Roberts:
A multifactor algorithm for two dimensional convolution. 2071 - Luis F. Chaparro:
Asymmetric half-plane planar least-squares inverses. 2072-2075 - S. M. Candel:
Fast computation of Fourier-Bessel transforms. 2076-2079 - Marina V. Dragosevic, Srdjan S. Stankovic, Miodrag Carapic:
An approach to recursive estimation of time-varying spectra. 2080-2083 - Miodrag Carapic, Zoran Jovanovic, Zorica Mihajlovic:
The architecture of a signal processor developed through simulation. 2084-2088 - K. Wolinski:
Analysis of errors in mixed fast Fourier transform algorithms with decimation in frequency for fixed point arithmetic. 2089-2093 - Todd K. Citron, Thomas Kailath, V. Umapathi Reddy:
A comparison of some spectral estimation techniques for short data lengths. 2094-2097 - F. Clara, Leonard M. Silverman, Jean-François Abramatic:
Nonlinear image restoration: A visual quality constrained approach. 2098-2101 - Wladyslaw Wasiluk, Grzegorz Lukowski:
The examination of influence of geophysical property of soil on equipotential earthed equipment by means of digital processing. 2102-2104
manage site settings
To protect your privacy, all features that rely on external API calls from your browser are turned off by default. You need to opt-in for them to become active. All settings here will be stored as cookies with your web browser. For more information see our F.A.Q.
last updated on 2025-12-12 01:07 CET by the dblp team
all metadata released as open data under CC0 1.0 license
see also: Terms of Use | Privacy Policy | Imprint
dblp was originally created in 1993 at:
since 2018, dblp has been operated and maintained by:
the dblp computer science bibliography is funded and supported by:
