Abstract is missing.
- CCITT Activity on signal processing for integrated services digital networksMaurizio Dècina. 5-10 [doi]
- Hierarchical processing of structural information in artificial intelligenceGösta H. Granlund, Hans Knutsson. 11-16 [doi]
- Human-machine interaction and digital signal processingRichard A. Guedj. 17-19 [doi]
- The design of optimal DFT algorithms using dynamic programmingHoward W. Johnson, C. Sidney Burrus. 20-23 [doi]
- Two dimensional DFT using mixed time and frequency decimationsChristos Caraiscos, Bede Liu. 24-27 [doi]
- Recursive calculation of Fourier transform of discrete signalV. V. Cizek. 28-31 [doi]
- Performance and computation ranking of fast unitary transforms in applicationsV. Ralph Algazi, Bernard J. Fino. 32-35 [doi]
- A polynomial transform approach to transmultiplexingHenri J. Nussbaumer. 36-39 [doi]
- Fast algorithms for signal processing using finite field operationsG. Robert Redinbo, D. O. Carhoun, B. L. Johnson. 40-43 [doi]
- On the computational complexity of bilinear forms evaluation over a body of quaternionsO. M. Makarov. 44-47 [doi]
- On the accuracy of 2-D digital filter realizations using logarithmic number systemsGiovanni L. Sicuranza. 48-51 [doi]
- Quantization error and limit cycles analysis in residue number system coded recursive filtersA. Z. Baraniecki, Graham A. Jullien. 52-55 [doi]
- Analysis of errors in residue number system (RNS) based IIR digital filtersA. S. Ramnarayan, Fred J. Taylor. 56-59 [doi]
- Failure resistant digital filters based on residue number system product codesW. Kenneth Jenkins. 60-63 [doi]
- Quantization and truncation effects in the design of adaptive digital filtersF. M. Boland, J. O. Normile. 64-68 [doi]
- Fixed-point error analysis of rectangular transformGanapati Panda, Ranendal N. Pal, B. Chatterjee. 69-72 [doi]
- The application of dynamic programming to the optimal ordering of digital filter sectionsCharles M. Rader. 73-76 [doi]
- Digital audio mixer: A VLSI approachDavid V. James, Noah Mendelsohn, David R. Fuchs. 77-80 [doi]
- Some design issues in digital signal processing for digital-audio systemsPiet J. Berkhout, Ludwig D. J. Eggermont. 81-84 [doi]
- The Lucasfilm audio signal processorJames A. Moorer. 85-88 [doi]
- Signal processing for the analysis of musical soundScott Foster, A. Joseph Rockmore. 89-92 [doi]
- A 2-channel, 16-bit digital sampling frequency converter for professional digital audioRoger Lagadec, Daniele Pelloni, Daniel Weiss. 93-96 [doi]
- Digital parametric filters for studio mixing deskS. Fuchs, M. Seguin, A. Weisser. 97-100 [doi]
- Sample-rate conversion by arbitrary ratiosTor A. Ramstad. 101-104 [doi]
- Modulation of acoustic signals in a shallow water using a normal-mode modelU. E. Rupe. 105-108 [doi]
- A corrected match for the coherent part of a time-variant channelJunhua Xu, Geng Chen. 109-112 [doi]
- Communication in a fluctuating channel models and use of explicit or implicit diversityGeneviève Jourdain, George Tziritas. 113-116 [doi]
- Impulse response for the one-dimensional inhomogeneous medium with an approximation for attenuation and dispersionWilliam J. Vetter. 117-120 [doi]
- Homomorphic signal dereverberation for a phased array imaging systemRussell P. Kraft, John F. McDonald, J. Erkes. 121-124 [doi]
- The extension of Pisarenko's method to multiple dimensionsStephen W. Lang, James H. McClellan. 125-128 [doi]
- Properties of two dimensional maximum entropy power spectrum estimatesNaveed A. Malik, Jae S. Lim, Michelle J. Glaser. 129-132 [doi]
- The algebraic inversion of 2-D autoregressive power spectra with applications to spectral estimationThomas L. Marzetta. 133-135 [doi]
- On 2-D spectral factorizationHayri Korezlioglu, Philippe Loubaton. 136-139 [doi]
- Computation of two-dimensional complex cepstrumBir Bhanu. 140-143 [doi]
- Further results on 4-fold rotational symmetry in 2-D functionsP. Karivaratha Rajan, Hanartha C. Reddy, M. N. Shanmukha Swamy. 144-147 [doi]
- Application of the LMS adaptive filter to improve speech communication in the presence of noiseDouglas M. Chabries, Richard W. Christiansen, Robert H. Brey, Martin S. Robinette. 148-151 [doi]
- Estimating the parameters of a noisy AR-process by using a bootstrap estimatorM. S. Ahmed. 152-155 [doi]
- Speech enhancement by nonlinear multiband envelope filteringThomas Langhans, Hans Werner Strube. 156-159 [doi]
- A generalized comb filtering technique for speech enhancementDavid Malah, Richard V. Cox. 160-163 [doi]
- Evaluation of two-input speech dereverberation techniquesP. Jeffrey Bloom, G. D. Cain. 164-167 [doi]
- "Optimum" filter for speech enhancement using integrated digital signal processorsGerhard Doblinger. 168-171 [doi]
- Postprocessing techniques for voice pitch trackersBruce G. Secrest, George R. Doddington. 172-175 [doi]
- A pitch measurement algorithm for speechJordan Cohen. 176-179 [doi]
- Comparison of pitch detection by cepstrum and spectral comb analysisPhilippe Martin. 180-183 [doi]
- An autocorrelation pitch detector with error correctionGeoff J. Bristow, Frank Fallside. 184-187 [doi]
- Improvements of the harmonic-sieve pitch extraction scheme and an appropriate method for voiced-unvoiced detectionRobert J. Sluyter, H. J. Kotmans, Theo A. C. M. Claasen. 188-191 [doi]
- Comparative performance of pitch detection algorithms on dysphonic voicesJohn Laver, Steven M. Hiller, R. J. Hanson. 192-195 [doi]
- The perception of spectrally shaped additive noise in speechBarbara J. McDermott, Carlo Scagliola. 196-198 [doi]
- Mediumband speech encoding using time-domain harmonic scaling and adaptive residual coding for noisy channelsJames L. Melsa, Arun Pande. 199-202 [doi]
- Sub-band coder with a simple adaptive bit-allocation algorithm a possible candidate for digital mobile telephony?Tor A. Ramstad. 203-207 [doi]
- Real-time implementation of a 9600 bps subband coder with time-domain harmonic scalingRonald S. Cheung. 208-211 [doi]
- A variable rate embedded-code speech waveform coderMaurizio Copperi. 212-215 [doi]
- On spectral flattening techniques in residual-excited linear prediction vocodingChong Kwan Un, Jong Rak Lee. 216-219 [doi]
- Adaptive predictive coding of base-band speech signalsClaude R. Galand, K. Daulasim, Daniel J. Esteban. 220-223 [doi]
- Comparison of some algorithms for identifying autoregressive signals in the presence of observation noiseKalle-J. Bry, Joël Le Roux. 224-227 [doi]
- Estimation of the autoregressive parameters from observations of a noise corrupted autoregressive time seriesDonald F. Gingras. 228-231 [doi]
- Noise correction approach for pole-zero modeling by pencil-of-functions methodVijay K. Jain, Tapan K. Sarkar, Donald D. Weiner. 232-235 [doi]
- The R and S arrays and the AIC in ARMA modeling and filter designJ. Bee Bednar, B. J. Roberts. 236-239 [doi]
- Statistical efficiency of the sample autocorrelation function in ARMA parameter estimationS. P. Bruzzone, Mostafa Kaveh. 240-243 [doi]
- An iterative algorithm for finding stable solutions to the covariance or modified covariance autoregressive modeling methodsBruce R. Musicus. 244-247 [doi]
- Instrumental variable methods for ARMA spectral estimationBenjamin Friedlander. 248-251 [doi]
- Identification of vibrating structures subject to non stationary excitation : A non stationary stochastic realization problemMarc Prevosto, Albert Benveniste, Bruno Barnouin. 252-255 [doi]
- Data adaptive ARMA modeling of time seriesJames A. Cadzow, Behshad Baseghi. 256-261 [doi]
- Parallel identifiers for parameter estimation of strongly disturbed ARMA-processesHans-Eberhard Schurk, Ulrich Appel, Werner Wolf. 262-265 [doi]
- Circuit models for prediction, Wiener filtering, Levinson and Kalman filters for ARMA time seriesW. J. Shanahan. 266-269 [doi]
- Design of FIR filters to complex frequency response specificationsRichard R. Kurth. 270-273 [doi]
- Some improvements to the design programs for equiripple FIR filtersFederico Bonzanigo. 274-277 [doi]
- Narrowband linear-phase FIR filters requiring a small number of multipliersTapio Saramäki. 278-281 [doi]
- Digital lattice filter design using a frequency domain modeling approachYong Ching Lim, Sydney R. Parker. 282-285 [doi]
- Design of passive second-order digital filtersHon Keung Kwan. 286-289 [doi]
- On optimal equalization of an analog antialiasing filter with a nonrecursive digital systemHans Wilhelm Schüßler, P. Möhringer, P. Steffen. 290-293 [doi]
- Theory of volterra processors and some applicationsEzio Biglieri. 294-297 [doi]
- Quadratic filtersBernard C. Picinbono. 298-301 [doi]
- The design of time-varying digital filters which employ binary valued coefficientsFred Kitson, Lloyd J. Griffiths. 302-305 [doi]
- Design and implementation of transmitter and receiver filters with periodic coefficient nulls for digital systemsAndres C. Salazar, Victor B. Lawrence. 306-310 [doi]
- Efficient formation of filter banks with frequency dependent resolutionDaniel D. Rivers, Robert A. Rosen. 311-314 [doi]
- Structures for digital filter banksThomas G. Marshall Jr.. 315-318 [doi]
- Transmultiplexers used as adaptive frequency sampling filtersGreg C. Copeland. 319-322 [doi]
- On the generation of accurate high frequency acoustic pulses using modern control theoryJ. Douglas Birdwell, T. J. Paulus, C. J. Mazzola, L. Czapla. 323-326 [doi]
- Acoustic classification of submerged targetsSusan K. Numrich, Laurence J. Frank, Louis R. Dragonette. 327-330 [doi]
- Sonar target classification using a coherent echo processingManell Zakharia, Jean-Pierre Sessarego. 331-334 [doi]
- Time varying autoregressive signal models, with an application to chirped signalsJ. Terry Ginn. 335-338 [doi]
- On external properties satisfied by the Io-sinh windowJ. H. Hesson, James F. Kaiser. 339-342 [doi]
- Real-time monitoring of machine tools via Walsh-Hadamard tranformGyula Hermann, László Horváth, László Monostori. 343-346 [doi]
- Application of pattern recognition techniques to the processing of radar signalsNorberto F. Ezquerra, Linda Harkness. 347-350 [doi]
- A modular fast two-dimensional cyclic convolver and its application to real-time synthetic aperture radar processingK. Y. Liu. 351-354 [doi]
- Locating voids beneath pavement using a pulsed radarWilliam J. Steinway, Charles M. Luke, Jim D. Echard. 355-358 [doi]
- Optimum detection and receiver performance for multistatic radar configurationsD. Baumgarten. 359-362 [doi]
- FFT Signal processing for non-coherent radar systemsWilliam A. Holm, Jim D. Echard. 363-366 [doi]
- Spatial digital processing: Application to radar antennasJ. L. Pourailly, J. De Reffye, C. Legendre. 367-370 [doi]
- On the scot and roth algorithms for time delay estimationKent Scarbrough, Nasir Ahmed, Dae Hee Youn, G. Clifford Carter. 371-374 [doi]
- Threshold effects in time delay estimation using narrowband signalsJ. P. Ianniello. 375-378 [doi]
- Time delay estimation: Application to flow rate measurement of cooling fluid in nuclear power plantsJean-Melaine Favennec, B. Georgel, J. Masson. 379-382 [doi]
- Real-time estimation of moving time delayHeinrich Meyr, Gerhard Spies, Jörg Bohmann. 383-386 [doi]
- Nonparametric detectors for signal detection and time delay estimationA. F. Hassan, E. K. Al-Hussainy, M. Bakry. 387-390 [doi]
- Separation of superimposed signals by a cross-correlation methodRui J. P. de Figueiredo, Andreas Gerber. 391-394 [doi]
- Mode and time delay estimation for non-destructive evaluation systemsJ. Pearson, C. J. Macleod, Tariq S. Durrani. 395-398 [doi]
- Velocity estimation from short-time temporal and spatial frequency estimatesThomas W. Parks, Charles F. Morris, John D. Ingram. 399-402 [doi]
- Recursive techniques for passive source locationJosé M. F. Moura. 403-406 [doi]
- Speed measurement by cross-correlation - theoretical and practical aspectsPhilippe Bolon. 407-410 [doi]
- A new method for multiple source locationMichael J. Coker, E. R. Ferrara. 411-415 [doi]
- A parametric technique for time delay estimationBenjamin Friedlander, Boaz Porat. 416-419 [doi]
- A new method for high resolution estimation of time delayZi-Qiang Hou, Zhen-Dong Wu. 420-423 [doi]
- Adaptive delay tracking with a delay-lock estimatorLuis F. Rocha. 424-427 [doi]
- Image coding using vector quantizationAllen Gersho, Bhaskar Ramamurthi. 428-431 [doi]
- Image coding using a predictor controlled by image contentRoland Wilson, Hans Knutsson, Gösta H. Granlund. 432-435 [doi]
- A contour-texture approach to picture codingM. Kocher, M. Kunt. 436-439 [doi]
- Efficient coding of high resolution typographic charactersP. Fäh, M. Kunt. 440-443 [doi]
- Color texture reconstruction using a bidimensional Markov modelF. J. Schmitt. 444-447 [doi]
- An adaptive interframe transform coding system for imagesM. Götze, G. Ocylok. 448-451 [doi]
- An efficient, real-time, method for transmitting Walsh-Hadamard transformed picturesNikolaos G. Bourbakis, Nikitas A. Alexandridis. 452-455 [doi]
- On the use of splines in hierarchical image transmissionAlberto Sanz, Carlos Muñoz, Narciso García. 456-459 [doi]
- Motion of edges and motion estimation in a sequence of T.V. picturesClaude Labit, Albert Benveniste. 460-463 [doi]
- Interframe coding with general two-dimensional motion compensationThomas S. Huang, Y.-P. Hsu, Roger Y. Tsai. 464-466 [doi]
- Subjective quality evaluation of different intraframe adaptive coding schemes, based on orthogonal transformationsM. Guglielmo, R. Marion, A. Sciarappa. 467-470 [doi]
- Sampling and interpolation in two dimensionsB. Cochrane, K. P. Dawson, M. A. Fiddy, Trevor J. Hall. 471-474 [doi]
- Analysis of the numerical stability of algorithmsJ. Vignes, P. Bois. 475-478 [doi]
- Optimization of random quantizationJ. P. Tressières, Francis Castanie. 479-483 [doi]
- A discrete optimization method for high-order FIR filters with finite wordlength coefficientsKenji Nakayama. 484-487 [doi]
- Sampling rates for linear shift-variant discrete-time systemsDavid C. Munson Jr., Emily C. Martin. 488-491 [doi]
- A multi-channel microprogrammed FFT processorM. Balakrishnan, A. V. S. M. Rao, Rajendar Bahl. 492-497 [doi]
- Block processing structures for fixed point digital filteringC. A. Wambergue, Richard A. Roberts. 498-501 [doi]
- A simple design for a fast sliding DFT computerM. R. Jarmasz, G. O. Martens. 502-505 [doi]
- A novel structure for implementing DFT-filter banksHani Mahdi. 506-509 [doi]
- Digital modular technology: Application to matched filteringJ. S. Liénard, J. Y. Jourdain, P. Lambert. 510-513 [doi]
- Speech communication hardwarePierre Badin, Daniel Degryse. 514-516 [doi]
- Speech processing using programmable VLSIMichael McMahan, David Cox, Michael S. Wengrovitz. 517-520 [doi]
- Custom LST chip-set for speech analysisP. Zuidweg, Jef L. van Meerbergen, M. L. van der Meulen. 521-524 [doi]
- A single chip speech periodicity detectorRichard V. Cox, Ronald E. Crochiere. 525-528 [doi]
- Articulatory description of speech signal in isolated word recognizerW. W. Wiezlak, Ryszard Gubrynowicz. 529-534 [doi]
- Effects of emphasizing transitional or stationary parts of the speech signal in a discrete utterance recognition systemKjell Elenius, Mats Blomberg. 535-538 [doi]
- Discrete utterance recognition based upon source coding techniquesAndres Buzo, Horacio G. Martinez, Carlos Rivera. 539-542 [doi]
- Application of recursive exact least square ladder estimation algorithm for speech recognitionJ. M. Turner. 543-545 [doi]
- Properties of large lexicons: Implications for advanced isolated word recognition systemsDavid W. Shipman, Victor W. Zue. 546-549 [doi]
- Automatic recognition of syllabic speech segments using spectral and temporal featuresGünther Ruske. 550-553 [doi]
- A comparison of learning techniques in speech recognitionGary L. Bradshaw, Ronald A. Cole, Zongge Li. 554-557 [doi]
- Performance improvement in a dynamic-programming-based isolated word recognition system for the alpha-digit taskLori Faith Lamel, Victor W. Zue. 558-561 [doi]
- Hypothesizing of words for isolated and connected word recognition systems based on phonem preclassificationE. Schulze. 562-565 [doi]
- Toward speaker-independent recognition-by-synthesisRaimo Bakis, N. Rex Dixon. 566-569 [doi]
- Performance trade-offs in search techniques for isolated word speech recognitionRoberto Bisiani, A. Waibel. 570-573 [doi]
- Vector quantization and Markov source models applied to speech recognitionRoberto Billi. 574-577 [doi]
- Some general, user-oriented concepts for discrete utterance recognition (DUR) applicationHarvey F. Silverman, N. Rex Dixon. 578-581 [doi]
- A variable-order Markov chain for coding of speech spectraSalim E. Roucos, John I. Makhoul, Richard M. Schwartz. 582-585 [doi]
- Optimized frame selection for variable frame rate synthesisJosef Heiler. 586-588 [doi]
- Time encoding of LPC rootsPanos Papamichalis, George R. Doddington. 589-592 [doi]
- Full search and tree searched vector quantization of speech waveformsRobert M. Gray, Hüseyin Abut. 593-596 [doi]
- Multiple stage vector quantization for speech codingBiing-Hwang Juang, Augustine H. Gray Jr.. 597-600 [doi]
- 4800 Bps RELP Vocoder using vector quantization for both filter and residual representationsJean-Pierre Adoul, Philippe Mabilleau. 601 [doi]
- An 800 Bps real-time voice coding system based on efficient encoding techniquesThomas E. Carter, Duncan M. Dlugos, D. C. LeDoux. 602-605 [doi]
- Voice coding at 800 BPS and lower data rates with LPC vector quantizationDavid Y. Wong, Biing-Hwang Juang. 606-609 [doi]
- A harmonic deviations linear prediction vocoder for improved narrowband speech transmissionV. Viswanathan, Michael G. Berouti, A. Higgins, William Russell. 610-613 [doi]
- A new model of LPC excitation for producing natural-sounding speech at low bit ratesBishnu S. Atal, Joel R. Remde. 614-617 [doi]
- A vocoder scheme for very low bit rates (quality evaluation)Arild Lacroix, Bela Makai. 618-621 [doi]
- Cepstral residual vocoder for improved quality speech transmission at 4.8 kbpsDavid Malah. 622-625 [doi]
- Recursive identification techniquesLennart Ljung. 627-630 [doi]
- Thinned impulse responses for adaptive FIR filtersJohn R. Treichler, Michael G. Larimore. 631-634 [doi]
- Recursive adaptive filter design using an adaptive genetic algorithmDelores M. Etter, M. J. Hicks, K.-H. Cho. 635-638 [doi]
- Applications of output error recursive estimation algorithms for adaptive signal processingI. D. Landau, L. Dugard, S. Cabrera. 639-642 [doi]
- What does parameter mean in adaptive lattice algorithmsMiguel Angel Lagunas Hernandez, Enrique Masgrau-Gomez. 643-646 [doi]
- Recursive lattice algorithms with finite-duration windowsUlrich Appel, Achim V. Brandt. 647-650 [doi]
- On a covariance-lattice algorithm for linear predictionA. Cumani. 651-654 [doi]
- Fast approximate whitening ladder filterJ. M. Turner. 655-658 [doi]
- Real time estimation of amplitudes phases and frequencies of a sampled signal plus noise by nulling processingLuis F. Rocha. 659-662 [doi]
- Analysis and implementation of the adaptive notch filter for frequency estimationSun-Yuan Kung, D. V. Bhaskar Rao. 663-666 [doi]
- Multichannel adaptive filtering with a feedback convergence functionC. Y. Chang. 667-670 [doi]
- Non-stationary learning characteristics of adaptive lattice filtersCarey Gibson, Simon Haykin. 671-674 [doi]
- Efficient multiprocessor architecture for digital signal processingMichel Auguin, Fernand Boéri. 675-678 [doi]
- Optimum implementation of single time index signal flow graphs on synceronous multiprocessor machinesThomas P. Barnwell III, C. J. M. Hodges, Mark A. Randolph. 679-682 [doi]
- High order notation and automated program generation for realtime signal processingGordon L. DeMuth. 683-686 [doi]
- Hardware-software configuration for high performance digital filtering in real-timeJ. R. Trinder. 687-690 [doi]
- Novel structure of a user-programmable integrated digital signal processorR. Geppert. 691-694 [doi]
- Performance of an experimental data flow architecture for signal processingKlaus Kronlöf, Jorma Skyttä, Iiro Hartimo, Olli Simula. 695-698 [doi]
- Bit parallel-serial implementation of combinatorial filter in canonical formShigeyoshi Kawarai. 699-702 [doi]
- MOS Implementations of TTL architectures: A case studyWilliam Hall Evans, Jonathan Allen. 703-706 [doi]
- N-Port arithmetic unit for DSPLajos Gazsi. 707-710 [doi]
- Knuth's complex number arithmetic revisitedTich T. Dao. 711-716 [doi]
- A floating point format for signal processingJohn A. Eldon, Craig Robertson. 717-720 [doi]
- A Kalman filter procedure for the processing of the electroencephalogramFurio Bartoli, Sergio Cerutti. 721-724 [doi]
- Edge detection of the medullary canal of the femur: A Bayesian approachBernard Chalmond. 725-728 [doi]
- A new robust 2-D spectral estimation method and its application in cardiac data analysisChrysostomos L. Nikias, Peter D. Scott, John H. Siegel. 729-732 [doi]
- Time delay estimator for EEG analysis based on information theoryNicolaas J. I. Mars. 733-735 [doi]
- Digital processing techniques of breath sounds for objective assistance of asthma diagnosisGérard Charbonneau, Jean-Louis Racineux, M. Sudraud, E. Tuchais. 736-738 [doi]
- Phonetic recognition to assist lip-reading for deaf childrenMaria Domenica Di Benedetto, Francis Destombes, Bernard Mérialdo, Jean-Pierre Tubach. 739-742 [doi]
- A speech training aid for the deaf with display of voicing, frication and silenceE. M. Bate, Frank Fallside, E. Gulian, P. Hinds, C. Keiller. 743-746 [doi]
- Bliss communication with speech or text outputRolf Carlson, Björn Granström, Sheri Hunnicutt. 747-750 [doi]
- Predicting word-expressions to increase output rates of speech prostheses used in communication disordersKenneth Mark Colby, Daniel Christinaz, Roger C. Parkison, Mark Tiedemann. 751-754 [doi]
- A speech display computer for use in schools for the deafA. King, A. Parker, M. Spanner, R. D. Wright. 755-758 [doi]
- Evaluation of laryngeal dysfunction based on features of an accurate estimate of the glottal waveformJohn R. Deller Jr.. 759-762 [doi]
- Vocal tract shape analysis for childrenJosé M. Pardo. 763-766 [doi]
- An improved beam-forming algorithm for adaptive arraysShalhav Zohar. 767-770 [doi]
- Optimum array processing in presence of randomly distorded wavefrontsJean-Pierre Le Cadre, J.-L. Lambla. 771-774 [doi]
- Dominant mode power spectrum estimationNorman L. Owsley, J. W. Law. 775-778 [doi]
- New approach to source detection in passive listeningLaurent Kopp, Georges Bienvenu, Marc Aiach. 779-782 [doi]
- Bearing and ranging simultaneous measurements of an acoustic source using a parametric method: Some resultsJean-Paul Pignon. 783-786 [doi]
- On the sensitivity of orthogonal beamformingJohann F. Böhme. 787-790 [doi]
- ARMA Techniques for the location of multiple sources from linear array dataTariq S. Durrani, Ken C. Sharman. 791-794 [doi]
- Antenna array processing by multichannel ARMA modelsMénad Sidahmed. 795-798 [doi]
- Implementation of an adaptive space-time processor by an unconstrained multichannel latticeLeon H. Sibul, Guy R. L. Sohie. 799-802 [doi]
- Digital beamsteering with recursive multichannel filtersRobert A. Gabel, Richard R. Kurth. 803-806 [doi]
- Direct adaptive motion compensation for towed arraysL. Meier. 807-810 [doi]
- Virtual beams from an FFT beamformer and their use to assess the quality of a towed-array systemJean-Louis Berrou, Ronald A. Wagstaff. 811-814 [doi]
- Digital computer simulation study of an ultrasonic 3-D imaging system using frequency sweep and synthetic aperture techniquesYoh-Han Pao, Ahmed El Sherbini, Victor C. Chen. 815-817 [doi]
- Feature extraction as a tool for computer inputNimish Mehta, Kenneth C. Smith, F. E. Holmes. 818-820 [doi]
- Real time image registration based on the Cauchy-Schwarz inequalityRichard S. Schlunt, Hans-Peter Schmid. 821-824 [doi]
- A recursive method to apply the Hough transform to a set of moving objectsC. A. Darmon. 825-829 [doi]
- Water current determination by picture processingLiu Jian, Francis J. Schmitt. 830-833 [doi]
- Uniqueness and estimation of three-dimensional motion parameters of a rigid planar patch from three perspective viewsRoger Y. Tsai, Thomas S. Huang. 834-838 [doi]
- A procedure for classifying patternsE. R. Barnes. 839-842 [doi]
- Digital morphology in the 3-D spaceJean Serra. 843-845 [doi]
- A structural method of pattern recognition and its application to on line recognition of Chinese ideographsSerge Castan, Jun Shen. 846-849 [doi]
- Omnifont OCR with a structural modelN. Tripon, Philippe Coueignoux. 850-854 [doi]
- Application of image analysis techniques to seismic dataN. Keskes, A. Boulanouar, Olivier D. Faugeras. 855-858 [doi]
- Dynamic scenes and object descriptionsWorthy N. Martin, Jake K. Aggarwal. 859-862 [doi]
- LOGOS - A real time hardware continuous speech recognition systemJ. Peckham, J. Green, J. Canning, P. Stephens. 863-866 [doi]
- How humans perform on a connected-digits data baseLouis C. W. Pols. 867-870 [doi]
- Development of Japanese voice-activated word processor using isolated monosyllable recognitionTsuneo Nitta, T. Murata, H. Tsuboi, K. Takeda, T. Kawada, Sadakazu Watanabe. 871-874 [doi]
- The on-line version of the Otaniemi speech recognition systemErkki Reuhkala, Heikki Riittinen, Seppo Haltsonen, Olli Ventä, Kai Makisara, Teuvo Kohonen. 875-878 [doi]
- A simple and efficient isolated words recognition systemMichel Baudry, B. Dupeyrat. 879-882 [doi]
- Phoneme recognition in continuous speechAkio Komatsu, Akira Ichikawa, Kazuo Nakata, Yoshiaki Asakawa, Hiroko Matsuzaka. 883-886 [doi]
- Seraphine: A connected word speech recognition systemChristian Gagnoulet, Marc Couvrat. 887-890 [doi]
- A method for connected word recognition and word spotting on a microprocessorJean-Luc Gauvain, Joseph J. Mariani. 891-894 [doi]
- A system of speech communication with computer through noiseG. S. Ramishvili, V. D. Serdiukov. 895-898 [doi]
- An algorithm for connected word recognitionJohn S. Bridle, Michael D. Brown, Richard M. Chamberlain. 899-902 [doi]
- Multiprocessor architecture for real-time speech recognition systemsCesare Vicenzi, Carlo Scagliola. 903-906 [doi]
- Discrete utterance speech recognition without time normalizationJohn E. Shore, David K. Burton. 907-910 [doi]
- The role of the sinus cavities in the production of nasal vowelsShinji Maeda. 911-914 [doi]
- On the acoustic theory of coarticulation and reductionB. M. Lobanov. 915-918 [doi]
- A polynomial vocal tract model for speech synthesisS. N. Terepin, Frank Fallside. 919-922 [doi]
- Time-varying wave digital filters and vocal-tract modelsHans Werner Strube. 923-926 [doi]
- Estimation of vocal tract shape from input/Output measurementsN. Rao Vemula, A. Maynard Engebretson, David L. Elliott. 927-931 [doi]
- A composite model of speech productionElizabeth Allwood, Celia Scully. 932-935 [doi]
- Speech synthesis in the time domain from textE. Grossmann. 936-939 [doi]
- PARCAS, A new terminal analog model for speech synthesisUnto K. Laine. 940-943 [doi]
- Perceptually-based coding for root LPC synthesisNelson Morgan. 944-946 [doi]
- The modelling of F0 contoursRoger M. Meli, Frank Fallside. 947-949 [doi]
- Analysis and synthesis of voice fundamental frequency contours of spoken sentencesKeikichi Hirose, Hiroya Fujisaki. 950-953 [doi]
- Speech coding activities within CCITT: Status and trendsXavier Maître, T. Aoyama. 954-959 [doi]
- A 32 kb/s toll quality ADPCM codec using a single chip signal processorTakao Nishitani, Shinichi Aikoh, Takashi Araseki, Kazunori Ozawa, Rikio Maruta. 960-963 [doi]
- A 32-kbit/sec ADPCM coder robust to channel errorsD. Cointot. 964-967 [doi]
- A 32 kbit/s PCM to ADPCM converterJean-Marie Raulin, Jean-Louis Jeandot. 968-971 [doi]
- Adaptive predictive coding of speech and voiceband data signalsPaul Mermelstein, D. Millar. 972-975 [doi]
- An 32-kbps ADPCM encoding with a variable initially large leakage and adaptive dual loop predictorsYohtaro Yatsuzuka, Henri G. Suyderhoud. 976-979 [doi]
- A high quality ADM LSI codec at 32 kbit/s for digital speech communicationsNaohisa Ohta, Kazunari Irie, Takehiko Uno, Atsushi Iwata, Tomonori Aoyama. 980-983 [doi]
- Subjective quality of the same speech transmission conditions in seven different countriesDavid J. Goodman, Randy D. Nash. 984-987 [doi]
- Quality evaluation of 32 kbit/s coded speech by means of degradation category ratingsP. Combescure, Alain Le Guyader, André Gilloire. 988-991 [doi]
- Pay-off between quantizing distortion and injected circuit noiseD. L. Richards, G. J. Barnes. 992-995 [doi]
- An analysis of objectively computable measures for speech quality testingThomas P. Barnwell III, Schuyler R. Quackenbush. 996-999 [doi]
- Comparison of objective speech quality measures for voiceband CODECsNobuhiko Kitawaki, Kenzo Itoh, Masaaki Honda, Kazuhiko Kakehi. 1000-1003 [doi]
- Measurement of intrinsic deficiency in transmitted speech: The diagnostic discrimination test (DDT)William D. Voiers. 1004-1007 [doi]
- A generalized window method for spectral estimationMiquel Bertran-Salvans. 1008-1011 [doi]
- Applications of the short time Fourier transform to speech processing and spectral analysisJont B. Allen. 1012-1015 [doi]
- Linear transformations and parametric spectrum analysisLouis L. Scharf, Claude Guéguen, Jean-Pierre Dugré, Nicolas Moreau. 1016-1020 [doi]
- Windows associated with parametric spectral estimatorsTariq S. Durrani, Avedis S. Arslanian. 1021-1024 [doi]
- Spectral window in a MEM based spectrum analyzerTran Thong, Arif Kareem. 1025-1027 [doi]
- On the Fougere's maximum entropy spectral analysis methodC.-H. Chen. 1028-1029 [doi]
- AR Spectrum analysis based on a noisy autocovariance sequenceHideaki Sakai, K. Orita, N. Iwama. 1030-1033 [doi]
- Close frequencies resolution by maximum entropy spectral estimatorsJean-Louis Lacoume, Mohamed Gharbi, Claudine Latombe. 1034-1037 [doi]
- Auto-regressive spectral estimation of noisy sinusoidsP. L. Sharma, C. S. Chen. 1038-1041 [doi]
- Statistical analysis of frequency modulated signalsT. Subba Rao, M. Yar. 1042-1045 [doi]
- Signal reconstruction from the short-time Fourier transform magnitudeS. Hamid Nawab, Thomas F. Quatieri, Jae S. Lim. 1046-1048 [doi]
- The architecture of the real-time signal processorFred Mintzer, Abraham Peled. 1049-1052 [doi]
- A high performance VLSI CMOS arithmetic processor chipG. Culler, E. Greenwood, Dave Harrison. 1053-1056 [doi]
- A VLSI I/O chip for multiple signal processor architecturesA. Frey Jr.. 1057-1060 [doi]
- VLSI Building blocks for digital signal processingDavid Karlin, R. E. Owen. 1061-1064 [doi]
- A chip set for audio frequency digital signal processingE. R. Caudel, R. K. Hester, Khen-Sang Tan. 1065-1068 [doi]
- An integrated processor for adaptive and parallel algorithmsM. Cand, P. Le Scan, A. Roset. 1069-1072 [doi]
- An LSI digital signal processorM. Yano, K. Inoue, T. Senba. 1073-1076 [doi]
- An expandable single-IC digital filter/CorrelatorFrederick A. Williams. 1077-1080 [doi]
- A 32 point monolithic FFT processor chipG. D. Covert. 1081-1083 [doi]
- Bipolar VLSI facilitates Fourier transformationBernard New, David Brain. 1084-1087 [doi]
- Integrated floating point signal processorK. Böttcher, Arild Lacroix, Maati Talmi, Dieter Wesseling. 1088-1091 [doi]
- Detection of jumps in mean and adaptive filteringMichèle Basseville. 1092-1095 [doi]
- Estimation of coherence via ARMA modellingY. T. Chan, D. Parks. 1096-1099 [doi]
- Estimation of magnitude-squared coherence function: An adaptive approachDae Hee Youn, Nasir Ahmed, G. Clifford Carter. 1100-1103 [doi]
- Direct coherence estimation via a constrained least-squares linear-predictive fast algorithmA. H. Nuttal. 1104-1107 [doi]
- The impact of signal overcontainment on cross-correlation detection performanceJoseph R. Lapointe Jr.. 1108-1111 [doi]
- Detection performance of an operator using lofarMagnus Moll. 1112-1115 [doi]
- Measurement of azimuth and distance using arbitrary sensor configuration with unknown coordinatesW. Brandenburg. 1120-1123 [doi]
- Detection and bearing angle estimation of low flying aircraft by acoustical meansJ. Schiller. 1124-1127 [doi]
- Optimum demodulation of multiple signals having overlapped spectraOtis L. Frost. 1128-1131 [doi]
- Bat's sonar signals and acceleration toleranceMalik Mamode, Yvon Biraud, Bernard Escudié. 1132-1135 [doi]
- An adaptive Kalman window filter to restore degraded imagesSudhir S. Dikshit. 1136-1141 [doi]
- Noise reduction in images using statistical filteringS. Patil, Maher A. Sid-Ahmed, Malayappan Shridhar. 1142-1145 [doi]
- A fast Kalman filter for images degraded by both blur and noiseJan Biemond. 1146-1149 [doi]
- Two-dimensional recursive estimation for ARMA signal modelsJohn W. Woods, Subrahmanyam Dravida. 1150-1153 [doi]
- An estimator for image desmearing using a Bernoulli-Gaussian modelM. J. D. Bishop, Tariq S. Durrani. 1154-1157 [doi]
- Efficient MVE image reconstruction for arbitrary measurement geometriesSally L. Wood. 1158-1161 [doi]
- Image understanding and graph matchingOlivier D. Faugeras. 1162-1165 [doi]
- Recursive region segmentation by analysis of histogramsSteven A. Shafer, Takeo Kanade. 1166-1171 [doi]
- Edge detection : A tuttorial reviewM. Kunt. 1172-1175 [doi]
- Image segmentation or image understanding ?Jean Serra. 1176-1178 [doi]
- Recent advances in motion interpretation based on image sequencesHans-Hellmut Nagel. 1179-1186 [doi]
- Hardware for image processing and analysis: The PICAP approachBjörn Kruse, Björn Gudmundsson, Dan Antonsson, Tomas Hedblom, Arne Linge, Peter Lord, Tomas Ohlsson. 1187-1190 [doi]
- Nonlinear local image transforms with a new type of pipelined processorH. Keller, A. Favre, A. Comazzi. 1191-1194 [doi]
- Predite, a real time processor for bandwith compression in TV transmissionFinn Jørgensen, G. Michel, Charles Wagner. 1195-1198 [doi]
- A real-time relational processorBenkt Linnander, Lars-Erik Nordell, Björn Kruse. 1199-1202 [doi]
- Memory architecture of a video-rate image convolverHari K. Nagpal, Graham A. Jullien, William C. Miller. 1203-1206 [doi]
- A hardware digital processor for image bandwidth compressionH.-J. Alker, K. Andreassen. 1207-1210 [doi]
- MIP: A flexible, microprogrammable image processorM. Gonzalez, J. Gonzalez. 1211-1214 [doi]
- Data transmission and error correcting codesF. Delamotte, M. C. Gennero, Alain Poli. 1215-1218 [doi]
- Analysis and simulation of an adaptive image coding system using the LMS algorithmS. Thomas Alexander, Sarah A. Rajala. 1219-1222 [doi]
- Composite source coding techniques for image bandwidth compressionD. K. Mitrakos, George A. Constantinides. 1223-1226 [doi]
- Some experiments in ADPCM coding of imagesPetros Maragos, Russell M. Mersereau, Ronald W. Schafer. 1227-1230 [doi]
- Pre-processing technique for block coding of graphicsVolker Märgner. 1231-1234 [doi]
- Adaptive block truncation coding scheme using an edge following algorithmJoseph Ronsin, J. Dewitte. 1235-1238 [doi]
- Trapezoidal DP matching with time reversibilityM. Okochi, T. Sakai. 1239-1242 [doi]
- A systolic algorithm for connected word recognitionJean-Pierre Banâtre, Patrice Frison, Patrice Quinton. 1243-1246 [doi]
- Relative timing measures of acoustic segments aid automatic word recognitionHollis L. Fitch. 1247-1250 [doi]
- Acoustic pattern matching and beam searchingK. L. Greer, Bruce T. Lowerre, L. D. Wilcox. 1251-1254 [doi]
- Dynamic time warping for isolated word recognition based on ordered graph searching techniquesMichael K. Brown, Lawrence R. Rabiner. 1255-1258 [doi]
- A modification over Sakoe and Chiba's dynamic time warping algorithm for isolated word recognitionKuldip K. Paliwal, Anant Agarwal, Sarvajit S. Sinha. 1259-1261 [doi]
- Segmentation for data reduction in isolated word recognitionR. W. Brown. 1262-1265 [doi]
- Large-vocabulary spoken word recognition using simplified time-warping patternsYasuhiro Nara, K. Iwata, Y. Kijima, A. Kobayashi, Shinta Kimura, S. Sasaki, J. Tanahashi. 1266-1269 [doi]
- Locally constrained dynamic programming in automatic speech recognitionRoger K. Moore, Martin J. Russell, M. J. Tomlinson. 1270-1273 [doi]
- Dynamic time warping algorithms for SIMD machines and VLSI processor arraysMark A. Yoder, Leah J. Siegel. 1274-1277 [doi]
- Prediction of perceived phonetic distance from critical-band spectra: A first stepDennis H. Klatt. 1278-1281 [doi]
- A computational model of filtering, detection, and compression in the cochleaRichard F. Lyon. 1282-1285 [doi]
- Finite word length effects of the Leroux-Gueguen algorithm in computing reflection coefficientsPeriagaram K. Rajasekaran, J. C. Hansen. 1286-1290 [doi]
- Linear predictive hidden Markov models and the speech signalA. B. Poritz. 1291-1294 [doi]
- Hierarchical AR model for time varying speech signalsOsamu Kakusho, Masuzo Yanagida. 1295-1298 [doi]
- Non-steady state speech analysis method with dynamic feature enhancing effectTakayuki Nakajima, Torazo Suzuki, Hiroshi Ohmura. 1299-1302 [doi]
- A spectral model for nonstationary voiced speechLuis B. Almeida, José M. Tribolet. 1303-1306 [doi]
- Speech analysis in the time domain using syntactic technics an attempt to formalize the description of phonemes using acoustical cuesMarc Baudry, Paul Deléglise, J.-C. Friedmann. 1307-1310 [doi]
- Use of attributed grammars in speech signal processingM. Morgenthaler, C. Hansen. 1311-1313 [doi]
- Adaptive estimation of some parameters of speech excitationHarald Höge. 1314-1317 [doi]
- Frequency weighted linear predictionPeter L. Chu, David G. Messerschmitt. 1318-1321 [doi]
- An initial segmentation of the speech signalW. J. Borodziewicz. 1322-1324 [doi]
- Time-frequency analysis of random signalsW. Martin. 1325-1328 [doi]
- Wigner-Ville analysis of time-varying signalsBouatem Bouachache, Patrick Flandrin. 1329-1332 [doi]
- A non-aliased discrete-time Wigner distribution for time-frequency signal analysisDavid S. K. Chan. 1333-1336 [doi]
- Time varying lattices and autoregressive models : Parameter estimationYves Grenier. 1337-1340 [doi]
- A comparsion between time- and frequency-domain techniques for time-varying signal processingNian-Chyi Huang, Jake K. Aggarwal. 1341-1344 [doi]
- Efficient realization of adaptive digital filters in the time and frequency domainsGregory A. Clark, Sydney R. Parker, Sanjit K. Mitra. 1345-1348 [doi]
- An evaluation of time redundant DFT processing of stochastic signals with time varying spectraErnest G. Baxa Jr.. 1349-1352 [doi]
- Whiteness of the measurement noise as a criterion for ARMA modelization of speechJoël Leroux, F. Giannella. 1353-1356 [doi]
- Accurate parameter estimation of noisy speech-like signalsRamdas Kumaresan, Donald W. Tufts. 1357-1361 [doi]
- On the high resolution and unbiased frequency estimates of sinusoids in white noise-A new adaptive approachFrank K. Soong, Allen M. Peterson. 1362-1366 [doi]
- Signal compression model interrelationships in the time, frequency, principal component and canonical coordinate domainsCharlton M. Walter, John D. Tardelli. 1367-1370 [doi]
- The singular case and robust linear predictionClaude Guéguen, Ménad Sidahmed. 1371-1374 [doi]
- Spectral line analysis via a fast Prony algorithmS. Lawrence Marple Jr.. 1375-1378 [doi]
- An algorithm for the stable operation of a digitally-implemented fractionally-spaced adaptive equalizerRichard D. Gitlin, Howard C. Meadors, Stephen B. Weinstein. 1379-1383 [doi]
- Adaptive echo cancellation with dispersion and delay in the adjustment loopM. S. Mueller, J. J. Werner. 1384 [doi]
- Performance evaluation of three adaptive equalization algorithmsH. Sari. 1385-1389 [doi]
- Equalization of rapid selective fadings with unknown and time-varying formsEweda Eweda, Odile Macchi. 1390-1393 [doi]
- "A digital adaptive noise-canceller based on a stabilized version of the widrow L.M.S. algorithm"John G. McWhirter, K. J. Palmer, J. B. G. Roberts. 1394-1397 [doi]
- Fast least-squares (LS) in the voice echo cancellation applicationFrank K. Soong, Allen M. Peterson. 1398-1403 [doi]
- Comparison of some algorithms for tap weight evaluation in adaptive echo cancellersRoberto Montagna, Luciano Nebbia. 1404-1407 [doi]
- The influence of pseudo noise data signals on echo canceller performanceTheo A. C. M. Claasen, N. A. M. Verhoeckx. 1408-1411 [doi]
- Further results of a least squares and gradient adaptive lattice algorithm comparisonRaymond S. Medaugh, Lloyd J. Griffiths. 1412-1415 [doi]
- Theoretical and computer simulation study of the "Correlofiltre" adaptative system for non stationary processesG. Bouthemy, W. Kofman, A. Silvent. 1416-1419 [doi]
- Proposal and experimental evaluation of a combined structure "Correlofilter-adapter" for the continuous estimation of a noisy signal with a reference noisePierre-Yves Arques, Gérard Faucon. 1420-1423 [doi]
- A synchronous adaptive noise canceller for periodic interferenceHidefumi Kobatake. 1424-1427 [doi]
- Signal processing technique for measurement of vented-box loudspeaker system parametersVijay K. Jain, W. Marshall Leach Jr., Ronald W. Schafer. 1428-1431 [doi]
- Circular electret microphones : Theoretical and experimental investigationsRudolf Zahn. 1432-1435 [doi]
- Measurements on non linear microphones by a speech-like signalAdriano Depaoli, Giulio Modena, Aldo Reolon. 1436-1439 [doi]
- Measurement of nonlinear distortions in tape recorders and electro-acoustic transducers applying T.I.M. test signalsP. Skritek. 1440-1443 [doi]
- Application of echo-Cancelling techniques to audioconferenceRodolfo Ceruti, Franco Pira. 1444-1447 [doi]
- A new principle to avoid undesirable oscillations in electro acoustic loops application to the design of a hands free telephone set without voice switchingG. Ferrieu, P. Amstutz. 1448-1451 [doi]
- Evaluation of a physical method for estimating speech intelligibility in auditoriaHerman J. M. Steeneken, Tammo Houtgast. 1452-1454 [doi]
- Listening evaluation of auditoriumsJ. Robert Ashley. 1455-1458 [doi]
- OctophonyHubert Caron, Daniel Laberge. 1459-1461 [doi]
- Timbre of complex tone bursts with time varying spectral envelopeEiichi Miyasaka. 1462-1465 [doi]
- The architecture of a digital sound synthesis systemDaniel Arfib. 1466-1468 [doi]
- VLSI Circuits for a sampling digital acoustic energy meterSalim M. R. Taha, Majid A. H. Abdul-Karim. 1469-1472 [doi]
- Adaptive array processing using predicted coefficients as constrained conditionsChaohuan Hou, Shi-Zun Yan. 1473-1476 [doi]
- Array processing for estimating multiple emitter parametersWei-wen Gu. 1477-1480 [doi]
- Generalized Burg algorithm for beamforming in correlated multipath fieldStanislav Kesler. 1481-1484 [doi]
- Interferometric acoustic imaging, joint representation and image deconvolutionMonique Chiollaz, Bernard Escudié, Yvon Biraud, G. Pachiaudi. 1485-1488 [doi]
- Quantization degradation in superdirective processing of underwater acoustic arraysR. S. Walker, K. L. Déon. 1489-1492 [doi]
- Simulation for testing array responseW. W. Wolfe, M. J. Wilmut. 1493-1496 [doi]
- Optimum signal processing of vertical mode-selective array in shallow waterZi-Qiang Hou, Zhi-Guang Li. 1497-1500 [doi]
- Two channels random acoustics simulation and multichannel data acquisitionP. Monteil, V. Thiebaud. 1501-1504 [doi]
- Fitting polynomials to data in the presence of noiseNorman L. Owsley. 1505-1508 [doi]
- Optimum clutter suppression in airborne phased array radarsRichard Klemm. 1509-1512 [doi]
- Dynamic focussing and ultrasonic imagingMichel Martin, Jean-François Piquard. 1513-1515 [doi]
- P-Time delay measurement of a doublet of microearthquakesGeorges Poupinet, F. Glangeaud, Philippe Côté. 1516-1519 [doi]
- Image restoration and edge extraction based on 2-D stochastic modelsAnil K. Jain 0002, Surendra Ranganath. 1520-1523 [doi]
- Restoration and enhancement of arbitrary finite-energy images from incomplete spatial and spectral informationHenry Stark. 1524-1526 [doi]
- Convergence of iterative restoration methodsHenry J. Trussell. 1527-1530 [doi]
- Decomposition and separated adaptive digital processing of degraded images with an visual quality criterionDominique Barba. 1531-1536 [doi]
- Superresolution using linear system methods a comparisonHenri Maître, Armand J. Levy. 1537-1540 [doi]
- On the inversion of singular operatorsVicente Casares Giner. 1541-1544 [doi]
- The importance of boundary conditions in the phase retrieval problemMonson H. Hayes, Thomas F. Quatieri. 1545-1548 [doi]
- An optimal technique for tomographic image reconstruction from curved ray projectionsConstantinos E. Goutis, Tariq S. Durrani. 1549-1552 [doi]
- Tomographic imaging via wave equation inversionMostafa Kaveh, Mehrad Soumekh, Rolf K. Mueller. 1553-1556 [doi]
- Algorithms and experimental results on image reconstruction from limited dataD. Hayner, S. Renjen, Thomas S. Huang, W. Kenneth Jenkins. 1557-1560 [doi]
- Nonstationary 2-D recursive filter for speckle reductionDarwin T. Kuan, Alexander A. Sawchuk, Timothy C. Strand, Pierre Chavel. 1561-1564 [doi]
- Segment quantization for very-low-rate speech codingSalim E. Roucos, Richard M. Schwartz, John Makhoul. 1565-1568 [doi]
- Impulse analysis of speech : Spotting and preclassifying the impulses in the speech waveFrédéric Manceron, Jean-Sylvain Liénard. 1569-1572 [doi]
- Speech analysis by selective linear prediction in the time domainRiichiro Mizoguchi, Masuzo Yanagida, Osamu Kakusho. 1573-1576 [doi]
- Analytic pole-zero modelling of speech spectraF. J. Owens, R. J. Linggard. 1577-1580 [doi]
- Pole-zero modeling of noisy speech and its application to vocodingKil Ho Song, Chong Kwan Un. 1581-1584 [doi]
- Speech analysis and modelling using a sequential ARMA estimation techniqueR. García Gómez, José M. Tribolet. 1585-1588 [doi]
- The klattalk text-to-speech conversion systemDennis H. Klatt. 1589-1592 [doi]
- Text-to-speech conversion in Spanish a complete rule-based synthesis systemJuan M. Santos, José R. Nombela. 1593-1596 [doi]
- Sparte: A text-to-speech machine using synthesis by diphonesJean-Luc Courbon, Françoise Emerard. 1597-1600 [doi]
- A Chinese text-to-speech synthesis system based on an initial-final modelTai-Yi Huang, Cai-Fei Wang, Yoh-Han Pao. 1601-1603 [doi]
- A multi-language text-to-speech moduleRolf Carlson, Björn Granström, Sheri Hunnicutt. 1604-1607 [doi]
- SYNTEX - A microprocessor based system for automatic conversion of German text to speechHans-Wilhelm Rühl. 1608-1611 [doi]
- On talker-independent word recognition in continuous speechGary M. Kuhn. 1612-1615 [doi]
- Problems in the design and use of a connected speech understanding systemJean-Paul Haton, Jean-Marie Pierrel, Simon Sabbagh. 1616-1620 [doi]
- An embedded word training procedure for connected digit recognitionLawrence R. Rabiner, A. Bergh, Jay G. Wilpon. 1621-1624 [doi]
- A speech recognition systemHenri Meloni, Jacques Guizol. 1625-1628 [doi]
- Partial traceback and dynamic programmingPeter F. Brown, James C. Spohrer, Peter H. Hochschild, James K. Baker. 1629-1632 [doi]
- From speech recognition to speech understanding : A case study of KealDominique Gillet, Patrice Quinton, Jacques Siroux. 1633-1636 [doi]
- The AESOP continuous speech understanding systemJoseph J. Mariani. 1637-1640 [doi]
- Automatic labeling of speechJames C. Spohrer, Peter F. Brown, Robert Roth. 1641-1644 [doi]
- Speaker recognition using a feature weighting techniqueHermann Ney, Rainer Gierloff. 1645-1648 [doi]
- The application of probability density estimation to text-independent speaker identificationRichard M. Schwartz, Salim E. Roucos, Michael G. Berouti. 1649-1652 [doi]
- A composite scheme for text-independent speaker recognitionN. Mohankrishnan, Malayappan Shridhar, Maher A. Sid-Ahmed. 1653-1656 [doi]
- Speaker adaptation by a linear transformation with optimised parametersJohannes Jaschul. 1657-1660 [doi]
- Development of an automatic identification system of spoken languages: Phase IDeidre Cimarusti, Russell B. Ives. 1661-1663 [doi]
- Harmonic coding: A low bit-rate, good-quality speech coding techniqueLuis B. Almeida, José M. Tribolet. 1664-1667 [doi]
- Speech coding using efficient block codesManfred R. Schroeder, Bishnu S. Atal. 1668-1671 [doi]
- Adaptive bit allocation scheme in predictive coding of speechMasaaki Honda, Nobuhiko Kitawaki, Fumitada Itakura. 1672-1675 [doi]
- Comparison of basic and simplified sequential algorithms for the computation of lattice filter predictor coefficients in ADPCM coding of speechAlain Le Guyader, André Gilloire. 1676-1679 [doi]
- A comparative study of DPCM-AQF speech coders for bit rates of 16-32 kb/sCostas S. Xydeas, Cumhur Cengiz Evci. 1680-1683 [doi]
- 16 Kbps sub-band coder incorporating variable overhead informationClaude R. Galand, Daniel J. Esteban. 1684-1687 [doi]
- Performance evaluation of adaptive quantizers for a 16-kbits/s sub-band coderVaikunth Gupta, Krishna Virupaksha. 1688-1691 [doi]
- A 9.6 kb/s speech coder using the Bell laboratories DSP integrated circuitRonald E. Crochiere, Richard V. Cox, James D. Johnston, L. Seltzer. 1692-1695 [doi]
- An adaptive transform coding system with short primary blocklengths and frequency domain quantization using feedback adaptationKadiresan Annamalai, Tore Fjällbrant. 1696-1699 [doi]
- Quality measurements on speech coders for mobile radioJ. Svean. 1700-1703 [doi]
- Signal coding properties of asynchronous delta modulatorBülent Sankur, Hilmi Güngen. 1704-1708 [doi]
- Subjective quality of several 9.6-32 kb/s speech codersW. R. Daumer, J. L. Sullivan. 1709-1712 [doi]
- Adaptive quantization and prediction in speech codingChongxi Feng, Hui-Juan Yao, WeiLi Yang. 1713-1716 [doi]
- On the mathematical foundations of the generalized Levinson algorithmPhilippe Delsarte, Yves V. Genin, Yves G. Kamp. 1717-1720 [doi]
- Fast non-stationary lattice recursions for adaptive modeling and estimationEd F. Deprettere. 1721-1726 [doi]
- Fast Cholesky algorithms and adaptive feedback filtersMartin Morf, Carlos H. Muravchik, Ping Ang, Jean-Marc Delosme. 1727-1731 [doi]
- Constant-gain filters for finite shift-rank processesJean-Marc Delosme, Martin Morf. 1732-1735 [doi]
- How to approximate a spectrum recursively using ARMA modelsPatrick M. Dewilde. 1736-1739 [doi]
- Near to Toeplitz structure and efficient schemes for linear modelingEmmanuel N. Protonotarios, George Carayannis, E. Milios. 1740-1743 [doi]
- On block matrices with elements of special structureNicholas Kalouptsidis, George Carayannis, Dimitris Manolakis. 1744-1747 [doi]
- Generalized CORDIC for digital signal processingDaniel T. L. Lee, Martin Morf. 1748-1751 [doi]
- Fixed point error analysis of the normalized ladder algorithmC. G. Samson, V. Umapathi Reddy. 1752-1755 [doi]
- Identification of multivariable ARMA models by use of fast algorithmsGérard Favier. 1756-1759 [doi]
- A new generalized recursion for the fast computation of the Kalman gain to solve the covariance equationsCristos C. Halkias, George Carayannis, Ioannis Dologlou, D. Emmanoulopoulos. 1760-1763 [doi]
- Generalized least squares lattice algorithm and its application to decision feedback equalizationFuyun Ling, John G. Proakis. 1764-1769 [doi]
- Fast measurements of the attenuation and delay characteristics of data channelsRené Boite, Henri Leich. 1770-1772 [doi]
- Minimax equalizers for digital communicationDouglas Preis, C. Bunks. 1773-1776 [doi]
- Digital signal processing in a LSI 4.8 kbit/s modemLoïc Guidoux, O. Le Riche. 1777-1780 [doi]
- Digits to the customer: How to approach the problemG. Pellegrini, A. Tofanelli. 1781-1784 [doi]
- Digital local network systems: The impact of signal processingM. Vry. 1785-1788 [doi]
- Digital signal processing applied to the subscriber line interfaceA. Cook, A. R. Potter. 1789-1792 [doi]
- Signal processing in the E10 digital switching systemP. Delpit, P. Lavanant, B. Morin. 1793-1796 [doi]
- Effects of hybrid nonlinearity on a full-duplex telephone network with vocoderDavid Mansour, Augustine H. Gray Jr.. 1797-1800 [doi]
- Embedding data in speech using scrambling techniquesRaymond Steele, Diane Vitello. 1801-1804 [doi]
- Generalized tamed frequency modulationKah-Seng Chung, Leo E. Zegers. 1805-1808 [doi]
- Iterative and non iterative techniques for the design of recursive digital filtersA. Hanafy, Joël Le Roux, J. Prado. 1809-1812 [doi]
- The use of multiple criterion optimization for the simultaneous phase and magnitude design of IIR digital filtersGuido M. Cortelazzo, Michael R. Lightner. 1813-1816 [doi]
- Optimal design of minimum phase FIR filtersYves G. Kamp, Christian Wellekens. 1817-1820 [doi]
- A minimum-phase FIR complex filter design for transmultiplexer implementationEnrico Del Re, Pier Luigi Emiliani, Damiano F. Maffucci. 1821-1824 [doi]
- Linear phase recursive digital filters for special applicationsR. Czarnach, Hans Wilhelm Schüßler, G. Rohrlein. 1825-1828 [doi]
- Digital pulse frequency demodulation using state-space filteringPaul Loewenstein. 1829-1832 [doi]
- Effects of noise on signal reconstruction from Fourier transform phaseCarol Y. Espy, Jae S. Lim. 1833-1836 [doi]
- An approximately unbiased recovery of discrete Fourier transforms altered by jittered sampling epochsFrancis Castanie. 1837-1840 [doi]
- Kernel splitting method in support constrained deconvolution for super-resolutionRémy Prost, Robert Goutte. 1841-1844 [doi]
- Constrained iterative deconvolution using a conjugate gradient algorithmR. Marucci, Russell M. Mersereau, Ronald W. Schafer. 1845-1848 [doi]
- Least-squares method for multi-dimensional deconvolutionMasuzo Yanagida, Osamu Kakusho. 1849-1852 [doi]
- Deconvolution in real time of noisy signalsF. M. Boland, T. Doyle. 1853-1857 [doi]
- A comparative study of least-squares and homomorphic techniques for the inversion of mixed phase signalsJohn Mourjopoulos, Peter M. Clarkson, Joe K. Hammond. 1858-1861 [doi]
- Maximum likelihood dereverberation with applications in sonic well loggingAndrew L. Kurkjian. 1862-1865 [doi]
- An application of Kalman filtering in nuclear well loggingGuy Ruckebusch. 1866-1869 [doi]
- Reversible seismic data compressionMeemong Lee, Rao Yarlagadda. 1870-1873 [doi]
- Aspects of digital signal processing in radiometric remote sensing of geophysical variablesPiero Ciotti, Domenico Solimini. 1874-1877 [doi]
- Seismic sensing of extremely-low-frequency sounds in coastal watersA. Barbagelata, E. Michelozzi, Dieter Rauch, Bernd Schmalfeldt. 1878-1881 [doi]
- Improvement of the signal to noise ratio of seismic traces by re-alignment of reverberant energyPeter M. Clarkson, Joe K. Hammond. 1882-1885 [doi]
- Use of multidimentional MEM spectral analysis in geophysicsF. Glangeaud, M. Gharbi, Nadine Martin, Jean-Louis Lacoume. 1886-1889 [doi]
- Signal processing for event detectionL. G. Ahlbom, Anders Forsen, L. H. Zetterberg. 1890-1893 [doi]
- Automatic event detection applied to single channel seismic recordsTorild van Eck, L. G. Ahlbom. 1894-1897 [doi]
- Minimum-variance and maximum-likelihood recursive waveshapingJerry M. Mendel. 1898-1901 [doi]
- Study of climatic series by time-frequency analysisJ. P. Benoist, F. Glangeaud, Nadine Martin, Jean-Louis Lacoume, C. Lorius, A. Ait Ouahman. 1902-1905 [doi]
- Detection of edges using range informationAmar Mitiche, Jake K. Aggarwal. 1906-1911 [doi]
- Image segmentation using simple Markov field modelsHoward Elliott, F. R. Hansen. 1912-1915 [doi]
- Generalized phase planes for feature extractionIts'hak Dinstein, Eugene I. Plotkin, A. M. Zayezdny. 1916-1919 [doi]
- Algorithms for region description and modifications based on chain code transformationsJean-Pierre Gambotto. 1920-1923 [doi]
- An algorithm for isolated object location in digital imagesGérard Rives, J. P. Derutin, Marc Richetin, Joseph Alizon, Jean Gallice. 1924-1927 [doi]
- Watersheds of functions and picture segmentationS. Beucher. 1928-1931 [doi]
- The perceptual graph: A new algorithmF. Meyer. 1932-1935 [doi]
- New performance criteria of edge detection algorithmsW. Geuen, H.-G. Preuth. 1936-1939 [doi]
- A real-time automatic ranging algorithm for vision systemsPeter Bock, Aysel Basci. 1940-1943 [doi]
- On the interpretations of a polyhedral figureEric Térouanne. 1944-1947 [doi]
- Diffuse edge tracing using a predictor-corrector procedure with adaptive scopeAnthony L. Shipman, Robert R. Bitmead, Gregory H. Allen. 1948-1951 [doi]
- DFT-Vocoder using harmonic-sieve pitch extractionGeert J. Bosscha, Robert J. Sluyter. 1952-1955 [doi]
- An analysis-by-synthesis approach for automatic time segmentation of speech signalsYeunung Chen. 1956-1959 [doi]
- A compact digital channel vocoder using commercial devicesJoel A. Feldman. 1960-1963 [doi]
- A hardware implementation of a new narrow to medium band speech codingFumitada Itakura, Takao Kobayashi, Masaaki Honda. 1964-1967 [doi]
- Formant estimation by LPC with a new error criterionVijay K. Jain. 1968-1971 [doi]
- Broadcasting-quality transmission of audio signals at 64 kbpsLuciano Bertorello, Maurizio Copperi, G. Pirani, Fulvio Rusina. 1972-1975 [doi]
- A.D.P.C.M Algorithms applied to wideband speech encoding (64 kbit/s, 0-7 kHz)P. Combescure, Alain Le Guyader, M. Haghiri. 1976-1979 [doi]
- Specification-based design of ΣΔM for A/D and D/A conversionB. P. Agrawal, Kishan Shenoi. 1980-1983 [doi]
- Application of an optimization technique to the inversion of an articulatory speech production modelFrancis Charpentier. 1984-1987 [doi]
- Evaluation of glottal inverse filtering by means of physiological registrationsLou Boves, Bert Cranen. 1988-1991 [doi]
- Modelling of LIP radiation impedance in Z-domainUnto K. Laine. 1992-1995 [doi]
- Automatic recognition of voiced stop consonants in CV and VCV utterancesHiroya Fujisaki, M. Tominaga. 1996-1999 [doi]
- Automatic segmentation, recognition of phonetic units and training in the KEAL speech recognition systemGuy Mercier, A. Callec, Jean Monné, M. Querre, O. Trevarain. 2000-2003 [doi]
- Recognition of semivowels and consonants in continuous speech using articulatory parametersKatsuhiko Shirai, Tetsunori Kobayashi. 2004-2007 [doi]
- Continuous speech recognition via diphone spotting a preliminary implementationCarlo Scagliola, Luciano Marmi. 2008-2011 [doi]
- Segmentation of continuous speech by using multidimensional scaling techniquesGérard Charbonneau, Tarek Moussa. 2012-2014 [doi]
- A syntax-controlled segmentation of speech signal on the basis of dynamic spectraLeszek Kot. 2015-2017 [doi]
- Speaker trained recognition of large vocabularies of isolated wordsAaron E. Rosenberg, Lawrence R. Rabiner, Jay G. Wilpon. 2018-2021 [doi]
- Study of human and machine discrete utterance recognition (DUR)Carolyn A. Vickroy, Harvey F. Silverman, N. Rex Dixon. 2022-2025 [doi]
- On the evaluation of speech recognizers and data bases using a reference systemGérard Chollet, Christian Gagnoulet. 2026-2029 [doi]
- What causes speech recognizers to make mistakes?Wayne A. Lea. 2030-2033 [doi]
- Processing schemes for sampled multi-dimensional signalsGerhard Linnenberg. 2034-2037 [doi]
- State space approach to two dimensional filtersEttore Fornasini, Giovanni Marchesini. 2038-2041 [doi]
- Approximation of 2-D separable denominator recursive filtersBijan Lashgari, Leonard M. Silverman, Jean-François Abramatic. 2042-2045 [doi]
- Design of 2-D recursive filters using singular value decomposition techniquesRachid Deriche, Jean-François Abramatic. 2046-2049 [doi]
- Design of 2-D recursive digital filters with specified magnitude and constant group-delay responses by spectral factorizationN. Nagamuthu, Maher A. Sid-Ahmed, Malayappan Shridhar. 2050-2054 [doi]
- Design of nonsquare 2D FIR filters by transformationsSoo-Chang Pei, Yu-Chang Wu. 2055-2058 [doi]
- Three-dimensional recursive filteringGiovanni Garibotto, G. Piretta. 2059-2062 [doi]
- M-FiltersChristian Lantuéjoul, Jean Serra. 2063-2066 [doi]
- Nonlinear filtering using linear combinations of order statisticsAlan C. Bovik, Thomas S. Huang, David C. Munson Jr.. 2067-2070 [doi]
- A multifactor algorithm for two dimensional convolutionAmal El Fallah, Richard A. Roberts. 2071 [doi]
- Asymmetric half-plane planar least-squares inversesLuis F. Chaparro. 2072-2075 [doi]
- Fast computation of Fourier-Bessel transformsS. M. Candel. 2076-2079 [doi]
- An approach to recursive estimation of time-varying spectraMarina V. Dragosevic, Srdjan S. Stankovic, Miodrag Carapic. 2080-2083 [doi]
- The architecture of a signal processor developed through simulationMiodrag Carapic, Zoran Jovanovic, Zorica Mihajlovic. 2084-2088 [doi]
- Analysis of errors in mixed fast Fourier transform algorithms with decimation in frequency for fixed point arithmeticK. Wolinski. 2089-2093 [doi]
- A comparison of some spectral estimation techniques for short data lengthsT. Citron, Thomas Kailath, V. Umapathi Reddy. 2094-2097 [doi]
- Nonlinear image restoration: A visual quality constrained approachF. Clara, Leonard M. Silverman, Jean-François Abramatic. 2098-2101 [doi]
- The examination of influence of geophysical property of soil on equipotential earthed equipment by means of digital processingWladyslaw Wasiluk, Grzegorz Lukowski. 2102-2104 [doi]