Table of Contents

• Digital audio magnetic recording: Progress in packing density and key technologyK. Odaka, T. T. Doi, H. Nakajima. 1-4 [doi]
• Perpendicular magnetic recording and its application to digital recordingToshiyuki Suzuki, Takeharu Niioka. 5-8 [doi]
• Error correction method for R-DAT and its evaluationKenji Hayashi, Takao Arai, Takaharu Noguchi, Hiroo Okamoto, Masaharu Kobayashi. 9-12 [doi]
• Efficient data reduction for digital audio using a digital filter arrayJohn P. Stautner, David M. Horowitz. 13-16 [doi]
• Digital data equalizing in multitrack digital audio recordingRoger Lagadec. 17-20 [doi]
• A comparative study of the proposed high quality coding schemes for digital musicJoël Soumagne, Philippe Mabilleau, Sarto Morissette, Gerald Chouinard, David Bennett. 21-24 [doi]
• Permissible value of group delay distortion on tone quality due to low-pass filtersYoshiharu Hoshino, Toshiyuki Takegahara. 25-28 [doi]
• A study on the digitization of audio signals for video tape recorderTakao Arai, Takaharu Noguchi, Masaharu Kobayashi, Nobutaka Amada, Yasufumi Yumde, Kuniaki Miura. 29-32 [doi]
• An audio sampling frequency conversion using digital signal processorsYasushi Katsumata, Osamu Hamada. 33-36 [doi]
• An approach to high resolution D/A converter utilizing a linear predictive codingMasao Kasuga. 37-40 [doi]
• Mixture autoregressive hidden Markov models for speaker independent isolated word recognitionBiing-Hwang Juang, Lawrence R. Rabiner. 41-44 [doi]
• Markov modeling of continuous parameters in speech recognitionSerge Soudoplatoff. 45-48 [doi]
• Maximum mutual information estimation of hidden Markov model parameters for speech recognitionLalit R. Bahl, Peter F. Brown, Peter V. de Souza, Robert L. Mercer. 49-52 [doi]
• An IBM PC based large-vocabulary isolated-utterance speech recognizerAmir Averbuch, Lalit R. Bahl, Raimo Bakis, Peter F. Brown, A. G. Cole, G. Daggett, Subrata K. Das, Ken Davies, S. DeGennaro, Peter V. de Souza, E. Epstein, D. Fraleigh, Frederick Jelinek, Slava M. Katz, B. Lewis, Robert L. Mercer, Arthur Nádas, David Nahamoo, Michael Picheny, G. Shichman, P. Spinelli. 53-56 [doi]
• A syllable-based isolated word recognition experimentJean-Luc Gauvain. 57-60 [doi]
• Quantitative knowledge on word structure, from a phonetic corpus, with application to large vocabularies recognition systemsJean-Pierre Tubach, Louis-Jean Boë. 61-64 [doi]
• Word preselection for large vocabulary speech recognitionRoberto Billi, G. Massia, F. Nesti. 65-68 [doi]
• Compact isolated word recognition system for large vocabularyShoji Hiraoka, Shuji Morii, Masakatsu Hoshimi, Katsuyuki Niyada. 69-72 [doi]
• Coarse classification using a hierarchical decision tree and top down parsingLynn Wilcox, Bruce T. Lowerre. 73-76 [doi]
• Incremental network generation in word recognitionKai-Fu Lee. 77-80 [doi]
• A computational model for separating two simultaneous talkersMitchel Weintraub. 81-84 [doi]
• Evaluation of multisensor speech input for speech recognition in high ambient noiseVishu Viswanathan, Claudia M. Henry, Richard M. Schwartz, Salim E. Roucos. 85-88 [doi]
• Enhancement of noisy speech by forward/Backward adaptive digital filteringJ. W. Kim, C. K. Un. 89-92 [doi]
• Performance characteristics of a hardware implementation of the cross-talk resistant adaptive noise cancellerGagan Mirchandani, Richard C. Gaus Jr., L. Kathy Bechtel. 93-96 [doi]
• Acoustic noise reduction by two dimensional spectral smoothing and spectral amplitude transformationYasuo Ariki, K. Kajimoto, Toshiyuki Sakai. 97-100 [doi]
• Speech enhancement using multi-pulse excited linear prediction systemKuldip K. Paliwal. 101-104 [doi]
• Waveform substitution techniques for recovering missing speech segments in packet voice communicationsDavid J. Goodman, O. G. Jaffe, Gordon B. Lockhart, W. C. Wong. 105-108 [doi]
• Fast and accurate pitch detection using pattern recognition and adaptive time-domain analysisDimitrios P. Prezas, Joe Picone, David L. Thomson. 109-112 [doi]
• Pitch detection using the short-term phase spectrumFrancis Charpentier. 113-116 [doi]
• Frequency identification by complex spectrumMasakazu Imai, Seiji Inokuchi. 117-120 [doi]
• An adaptive non-uniform sign clipping preprocessor (ANUSC) for real-time autocorrelative pitch detectionWerner Verhelst, Fernand De Decker, Bart Francq, Oscar Steenhaut. 121-124 [doi]
• A high quality 9.6 kbps speech coding systemDaniel W. Griffin, Jae S. Lim. 125-128 [doi]
• Adaptive discrete cosine transform image coding using gain/Shape vector quantizersTakahiro Saito, Hideya Takeo, Kiyoharu Aizawa, Hiroshi Harashima, Hiroshi Miyakawa. 129-132 [doi]
• Vector quantization of color imagesTokumichi Murakami, Kohtaro Asai, Atsushi Itoh. 133-136 [doi]
• Low-rate image coding with finite-state vector quantizationR. Aravind 0001, Allen Gersho. 137-140 [doi]
• A gain-shape vector quantizer for image codingHo John Lee, Daniel T. L. Lee. 141-144 [doi]
• Vector quantization in transformed image codingJ. P. Marescq, Claude Labit. 145-148 [doi]
• A technique using a one-dimensional mapping for vector quantisation of imagesF. Oliveri, G. Conte, Mario Guglielmo. 149-152 [doi]
• A hybrid coding involving ADM and vector quantization for digital video image compressionP. A. Ramamoorthy, T. Tran. 153-156 [doi]
• The performance of a hybrid videoconferencing coder using displacement estimation in the transform domain R. H. J. M. Plompen, J. G. P. Groenveld, Dick E. Boekee, F. Booman. 157-160 [doi]
• Contour based representation of the displacement field for motion compensated image codingStefan Carlsson, Christian Reillo. 161-164 [doi]
• Hybrid interframe coding of video signals with backward-acting motion detectionSharaf E. Elnahas, Kou-Hu Tzou. 165-168 [doi]
• High-resolution spectral estimation: Rethinking the Fourier transformLloyd J. Griffiths. 169-172 [doi]
• Single vector approaches to eigenstructure analysis for harmonic retrievalGenshen Xu, Yoh-Han Pao. 173-176 [doi]
• Estimation of frequencies of sinusoids in colored noiseHideaki Sakai. 177-180 [doi]
• High resolution spectral estimation of sinusoids in colored noise using a modified Pisarenko decompositionPeter J. Sherman, Arthur E. Frazho. 181-184 [doi]
• Spectral and parameter estimation problems arising in the metrology of high performance mirror surfacesEugene Church, P. Z. Takacs. 185-188 [doi]
• On the use of linear programming for spectral estimationCory Myers. 189-192 [doi]
• A Toeplitz approximation approach to coherent source direction findingS. Y. Kung, C. K. Lo, R. Foka. 193-196 [doi]
• Performance of multichannel autoregressive spectral estimatorsS. Lawrence Marple Jr.. 197-200 [doi]
• p normJim E. Schroeder, Rao K. Yarlagadda. 201-204 [doi]
• Refined ARMA digital lattice filterYoshikazu Miyanaga, Nobuo Nagai, Nobuhiro Miki. 205-208 [doi]
• ARMA parameter estimation based on sample covariances, for missing dataYonina Rosen, Boaz Porat. 209-212 [doi]
• A new FFT approach to the interpolation of discrete-time signalsJohn W. Adams. 213-216 [doi]
• Fast algorithms for the discrete Fourier transform and for other transformsNaoki Suehiro, Mitsutoshi Hatori. 217-220 [doi]
• Performance of fixed-point FFT's: Rounding and scaling considerationsPeter Kabal, B. Sayar. 221-224 [doi]
• A fast Fourier transform algorithm using Hadamard transformC. X. Fan, S. H. Wang. 225-228 [doi]
• Cyclic convolution of real sequences: Hartley versus Fourier and new schemesPierre Duhamel, Martin Vetterli. 229-232 [doi]
• Computation of complex number theoretic transforms using quadratic residue number systemsRamasamy Krishnan, Graham A. Jullien, William C. Miller. 233-236 [doi]
• A hardware efficient realisation of number theoretic convolversWan-Chi Siu, Anthony G. Constantinides. 237-240 [doi]
• Half-Fourier transform and application to radar signalsVijay K. Jain, T. E. McClellan, Tapan K. Sarkar. 241-244 [doi]
• Optimal design of multiplierless DFTS and FFTSWirendre A. Perera, Peter J. W. Rayner. 245-248 [doi]
• An efficient vector implementation of the FFT algorithm on IBM 3090VFRamesh C. Agarwal, James W. Cooley. 249-252 [doi]
• Immitance-domain Levinson algorithmsYuval Bistritz, Hanoch Lev-Ari, Thomas Kailath. 253-256 [doi]
• Efficient multi-processor implementation of recursive digital filtersWonyong Sung, Sanjit K. Mitra. 257-260 [doi]
• A unified derivation of the fast RLS algorithmsJin-Der Wang, H. Joel Trussell. 261-264 [doi]
• A fast codebook search algorithm for nearest-neighbor pattern matchingDe-Yuan Cheng, Allen Gersho. 265-268 [doi]
• The continuous-time limit of the discrete-time stability theoryHiroshi Nagaoka, Yoshimi Monden, Suguru Arimoto. 269-272 [doi]
• Maximum entropy deconvolutionC. AuYeung, Russell M. Mersereau, Ronald W. Schafer. 273-276 [doi]
• L1-norm noisy Tauberian deconvolutionJesús M. Alcázar-Fernández, José R. Casar Corredera, Ramón García Gómez. 277-280 [doi]
• A new homomorphic deconvolution systemZhao-Xiong Wu. 281-284 [doi]
• Generalized Lanczos method for signal smoothingAtsushi Imiya, Toshio Sasaki, Hidemitu Ogawa. 285-288 [doi]
• FFT and convolution algorithms on DSP microprocessorsZhenyu Li, Henrik V. Sorensen, C. Sidney Burrus. 289-292 [doi]
• A unified transform architecturePaul M. Farrelle, S. Srinivasan, Anil K. Jain 0002. 293-296 [doi]
• The array matrix generalization for signal processingWilliam J. Vetter. 297-300 [doi]
• A fast B-spline transform and its applicationsKazuo Toraichi, Kazuki Katagishi, Ryoichi Mori. 301-304 [doi]
• A generalization of Levinson algorithm for solving Toeplitz systemsYasuo Sugiyama. 305-308 [doi]
• A demiphoneme network representation of speech and automatic labeling techniques for speech data base constructionKazuyo Tanaka, Satoru Hayamizu, Kozo Ohta. 309-312 [doi]
• A speech data base at the united states air force academyMichael F. Guyote, Keith A. Lewis, Donald Lijana. 313-316 [doi]
• A PCM/VCR speech database exchange formatDavid S. Pallett. 317-320 [doi]
• A Japanese language speech databaseShuichi Itahashi. 321-324 [doi]
• B.D.L.E.X. : A data and cognition base of spoken FrenchGuy Perennou. 325-328 [doi]
• The development of the MIT Lisp-machine based speech research workstationVictor W. Zue, D. Scott Cyphers, Robert H. Kassel, David H. Kaufman, Hong C. Leung, Mark Randolph, Stephanie Seneff, John E. Unverferth III, Timothy Wilson. 329-332 [doi]
• A microprocessor-based word recognition system for large vocabulariesKari Torkkola, Heikki Riittinen. 333-336 [doi]
• Signal processor application to voice dialing equipmentA. Fukui, Y. Fujihashi, F. Nakagawa. 337-340 [doi]
• The analog voice privacy systemRichard V. Cox, D. Bock, K. B. Bauer, James D. Johnston, Jeffrey Snyder. 341-344 [doi]
• Development of a 16-kb/s speech codec for telephone applicationsN. Morgan, Baruch Mazor. 345-348 [doi]
• 24-channel 32 kb/s ADPCM transcoder using the CCITT recommendation G.721Sami Aly. 349-352 [doi]
• A special architecture for dynamic programmingPh. Missakian, M. Milgram, B. Zavidovique. 353-356 [doi]
• A modular architecture for dynamic programming and maximum likelihood sequence estimationWilliam G. Bliss, J. Girard, J. Avery, M. Lightner, Louis Scharf. 357-360 [doi]
• Speech recognition on the DADO/DSP multiprocessorAllen L. Gorin, J. E. Shoenfelt, R. N. Lewine. 361-364 [doi]
• A new systolic decomposition for the dynamic time warping algorithmEvert Dijkstra, Christian Piguet. 365-368 [doi]
• VDP : A versatile high performance vector distance processorFrancis Jutand, Nicolas Demassieux, Dominique Vicard. 369-372 [doi]
• Design of an efficient dynamic time warping LSIYoshitake Suzuki. 373-376 [doi]
• A single-chip speaker independent voice recognition systemMakoto Morito, Kozo Yamada, Akihiko Fujisawa, Masao Takeuchi. 377-380 [doi]
• VLSI Architecture for a real-time LPC-based feature extractorHenri Barral, Nicolas Moreau. 381-384 [doi]
• On the IC architecture and design of a 2 µm CMOS 8 MIPS digital signal processor with parallel processing capability: The PCB5010/5011Frans J. van Wijk, Frank P. Welten, Jef L. van Meerbergen, Jan Stoter, Jos A. Huisken, Antoine Delaruelle, Karel E. van Eerdewijk, Josef Schmid, Jan H. Wittek. 385-388 [doi]
• VLSI Digital signal processor (PSI)J. L. Laborie, D. F. Martin, J. C. Michalina, A. Picco. 389-392 [doi]
• Architecture and applications of a 100-ns CMOS VLSI digital signal processorS. Abiko, M. Hashizume, Y. Matsushita, K. Shinozaki, T. Takamizawa, Cole Erskine, Surendar Magar. 393-396 [doi]
• A high performance microprocessor for DSP applicationsJohn P. Roesgen. 397-400 [doi]
• A 50ns floating-point signal processor VLSITakao Kaneko, Hironori Yamauchi, Atsushi Iwata. 401-404 [doi]
• Architecture of high-speed 22-bit floating-point digital signal processorYoshikazu Mori, Toshio Jufuku, Masao Iida, Akira Nomura, Noboru Ichiura, Takao Nakamura. 405-408 [doi]
• Advanced single-chip signal processorTakao Nishitani, Ichiro Kuroda, Yuichi Kawakami, H. Tanaka, Tom Nukiyama. 409-412 [doi]
• A next-generation 32-bit VLSI signal processorShingo Tsujimichi, TakaHide Ohkami, Yukihiko Shimazu. 413-416 [doi]
• Architectural considerations for a sub 10 nanosecond DSP building block familyRobert E. Owen, Bruce E. Miller. 417-420 [doi]
• ®DSP32 digital signal processorJames R. Boddie, W. Patrick Hays, James Tow. 421-424 [doi]
• Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3)Yoshihiro Tomita, Shigeyuki Unagami, Tomohiko Taniguchi, Yasuhiko Tada, Masahiro Taka. 425-428 [doi]
• A high performance digital voice echo canceller on a single TMS32020Christopher R. Cole, Amine Haoui, Peter L. Winship. 429-432 [doi]
• Hidden Markov models applied to very low bit rate speech codingEric P. Farges, Mark A. Clements. 433-436 [doi]
• Design and performance of trellis vector quantizers for speech signalsBiing-Hwang Juang. 437-440 [doi]
• Joint time-spectral vector quantization and inverse filter setYasuo Matsuyama. 441-444 [doi]
• Fully vector-quantized multipulse LPC at 4800 bpsHitoshi Koyama, Allen Gersho. 445-448 [doi]
• Low-rate speech encoding using vector quantization and subband codingHüseyin Abut, Siegfried Ergezinger. 449-452 [doi]
• The self excited vocoder - an alternate approach to toll quality at 4800 bpsRichard C. Rose, Thomas P. Barnwell III. 453-456 [doi]
• Low bit rate multi-pulse speech coder with natural speech qualityKazunori Ozawa, Takashi Araseki. 457-460 [doi]
• A novel LPC synthesis model using a binary pulse source excitationDaniel Lin. 461-464 [doi]
• High quality glottal LPC-vocodingPer Hedelin. 465-468 [doi]
• On the behaviour of reduced complexity code-excited linear prediction (CELP)Luis A. Hernández Gómez, Francisco Javier Casajús-Quirós, Aníbal R. Figueiras-Vidal, Ramón García Gómez. 469-472 [doi]
• Voiced/Unvoiced classification of speech with applications to the U.S. government LPC-10E algorithmJoseph P. Campbell Jr., Thomas E. Tremain. 473-476 [doi]
• Optical circuitry and architectures for digital optical computingKarl-Heinz Brenner, Adolf W. Lohmann. 477-480 [doi]
• Optical parallel array logic systemYoshiki Ichioka, Jun Tanida. 481-484 [doi]
• Use of optical signal processing techniques to spectrum analysis of speechKeikichi Hirose, Hiroya Fujisaki, Yasuhiro Kosugi. 485-488 [doi]
• Optical signal processors consisting of collinear acousto-optic channel waveguidesNobuo Goto, Yasumitsu Miyazaki. 489-492 [doi]
• Reproduction of the sounds from old wax phonographic cylinders using the laser-beam reflection methodT. Asakura, T. Iwai, Tohru Ifukube, Toshio Kawashima. 493-496 [doi]
• New cascaded lattice structures for FIR filters having extremely low coefficent sensitivityP. P. Vaidyanathan. 497-500 [doi]
• Synthesis of minimum sensitivity structures in linear systems using controllability and observability measuresMasami Iwatsuki, Masayuki Kawamata, Tatsuo Higuchi 0001. 501-504 [doi]
• Embedded max quantizationKou-Hu Tzou. 505-508 [doi]
• Nonlinear quantization and data communicationK. Pahlavan. 509-512 [doi]
• Floating point error analysis of recursive least squares and least means squares adaptive filtersSasan H. Ardalan. 513-516 [doi]
• Design of limit-cycle-free digital biquad filtersAkinori Nishihara. 517-520 [doi]
• Predictive coding of multi-viewpoint image setsMichael E. Lukacs. 521-524 [doi]
• An architecture for the universal video codecLeonardo Chiariglione, Luigi Corgnier, Mario Guglielmo. 525-528 [doi]
• A fractal based approach to image compressionEugene Walach, Ehud D. Karnin. 529-532 [doi]
• Bit-sliced progressive transmission and reconstruction of transformed imagesKou-Hu Tzou, Sharaf E. Elnahas. 533-536 [doi]
• A binary representation of mixed documents (text/Graphic/Image) that compressesYi-Hsin Chen, Frederick C. Mintzer, Keith S. Pennington. 537-540 [doi]
• A representation method of color pictures by arranging fixed size dots of the primary colorsK. Yamada, Michihiko Minoh, Toshiyuki Sakai. 541-544 [doi]
• Design of 3-D IIR filters via transformations of 2-D circularly symmetric rotated filtersMichael E. Zervakis, Anastasios N. Venetsanopoulos. 545-548 [doi]
• A computer program for designing optimum 2-D FIR linear phase digital filtersChristakis Charalambous, Hana Khreishi. 549-552 [doi]
• An algorithm for the design of optimal finite wordlength 2-D FIR digital filtersAcyl Benslimane, Pierre Siohan. 553-556 [doi]
• Reduction of the number of parameters in the design of 2-D digital filtersTakao Hinamoto, Sadao Maekawa. 557-560 [doi]
• On the stability test for 2-D digital recursive filtersMyung-Ho Pee, B. A. Shenoi, E. Bruce Lee. 561-564 [doi]
• A fast image filtering processor-FIFPNobumoto Yamane, Yoshitaka Morikawa, Hiroshi Hamada. 565-568 [doi]
• Lower bounds in parameter estimation - summary of resultsAnthony J. Weiss, Ehud Weinstein. 569-572 [doi]
• New forms of least squares lattice algorithms and a comparison of their round-off error characteristicsPeter Strobach. 573-576 [doi]
• Effective adaptive Pisarenko spectrum estimateYu Hen Hu, Pin-Kuan Chou. 577-580 [doi]
• Determining MA models as salvos of pulsesClaude Guéguen, Nicolas Moreau. 581-584 [doi]
• On the limiting behavior of estimates based on sample covariancesBoaz Porat, Benjamin Friedlander. 585-588 [doi]
• Performance of narrowband signal-subspace processingHong Wang, Mostafa Kaveh. 589-592 [doi]
• Phase-strip sequence estimation: A phase unwrapping/Tracking algorithmScott A. Merritt. 593-596 [doi]
• x evaluation criterionShin'Ya Kuwahara, Mitsuo Ohta. 597-600 [doi]
• Estimation of anti-resonance frequencies by using an over-determined high-order Yule-Walker equationMasuzo Yanagida, Osamu Kakusho. 601-604 [doi]
• A new methodological trial on state estimation of linear structure vibration model with noisy power observation mechanism of non-Gaussian typeE. Uchino, M. Ohta. 605-608 [doi]
• Asymptotic properties of high-order Yule-Walker estimates of frequencies of multiple sinusoidsPeter Stoica, Benjamin Friedlander, Torsten Söderström. 609-612 [doi]
• Assessment of speech discrimination ability of hearing impaired subjects using FIR digital filterHiroshi Hosoi, H. Abe, Fumihiko Ohta, Satoshi Imaizumi. 613-616 [doi]
• A speech processor with lateral inhibition for an eight channel cochlea implant and its evaluation by subject testingTohru Ifukube, Robert L. White. 617-620 [doi]
• Sound processing for cochlear implantJ. Genin, R. Charachon. 621-624 [doi]
• A wearable, pocket-sized processor for digital hearing aid and other hearing prostheses applicationsA. Maynard Engebretson, Robert E. Morley Jr., Michael P. O'Connell. 625-628 [doi]
• Assessment of electronic aids for the hearing impaired which transmit visible or tactile speechYuichi Ueda, Akira Watanabe. 629-632 [doi]
• Speech training devices for profoundly deaf childrenLynne E. Bernstein, James B. Ferguson III, Moise H. Goldstein Jr.. 633-636 [doi]
• Speech training systems for handicapped children using vocal tract lateral shapesMinoru Shigenaga, Hirotaka Kubo. 637-640 [doi]
• Speech-analysis-based devices for diagnosis and education of speech and hearing impaired peopleSantiago Aguilera, A. Borrajo, José M. Pardo, E. Muñoz. 641-644 [doi]
• Refutation based recognition to help vowel articulationMarc El-Bèze. 645-648 [doi]
• Memory-intensive recognition for word articulation trainingFrancis Destombes. 649-652 [doi]
• Speech control of assistive devices for the physically disabledRobert I. Damper. 653-656 [doi]
• Integrated communication aids for the blindG. Castellini, Pier Luigi Emiliani, Paolo Graziani, A. Tronconi. 657-660 [doi]
• Japanese speech synthesis system in a book reader for the blindYukio Mitome, Katsunobu Fushikida. 661-664 [doi]
• A lexical prediction systemSheri Hunnicutt, Lennart Neovius. 665-668 [doi]
• An adaptive comb filtering method as applied to acoustic analyses of pathological voiceHideki Kasuya, Shigeki Ogawa, Yoshinobu Kikuchi. 669-672 [doi]
• Multidimensional analysis of phonological degeneration in pathological voicesSergio Feijóo, C. Hernández, R. Carmelo. 673-676 [doi]
• Acoustic measurement of pathological voice qualities for medical purposesSatoshi Imaizumi. 677-680 [doi]
• A microcomputer based system for acoustic analysis of voice characteristicsJan Gauffin, Britta Hammarberg, Satoshi Imaizumi. 681-684 [doi]
• Wave-flow index: A measure of vocal efficiencyYuki Kakita. 685-688 [doi]
• Electroglottography: Assessing articulatory function to aid the handicappedD. G. Childers, D. M. Hicks, G. P. Moore, Y. A. Alsaka. 689-692 [doi]
• Analysis and classification of snoring signalsA. Cohen, A. Lieberman. 693-696 [doi]
• Speaker independent isolated Arabic word recognition systemM. Elghonemy, M. Fikri, M. Hashish, El-Sayed A. Talkhan. 697-700 [doi]
• Isolated-word recognition of the complete vocabulary of spoken ChineseMichael Wagner, Wei Wang, Helen Ho, Mary O'Kane. 701-704 [doi]
• On hidden Markov models in isolated word recognitionAlan B. Poritz, Alan G. Richter. 705-708 [doi]
• Large-vocabulary isolated word recognition with fast coarse time alignmentAbdulmesih Aktas, Bernhard R. Kämmerer, Wolfgang A. Küpper, Helmut Lagger. 709-712 [doi]
• A proposal of a knowledge based isolated word recognitionShigeo Morishima, Hiroshi Harashima, Hiroshi Miyakawa. 713-716 [doi]
• Isolated word recognition based on finite-state vector quantizationWon Sik Youn, C. K. Un. 717-720 [doi]
• Speech processing with a Boltzmann machineJ. F. Trehern, Mervyn A. Jack, John Laver. 721-724 [doi]
• Simulating an acoustic recognizerSusan L. Banner. 725-728 [doi]
• Experiments in speech recognition over the telephone networkDaniel Kahn, Anand Gnanadesikan. 729-732 [doi]
• Recognition of speech under stress and in noisePeriagaram K. Rajasekaran, George R. Doddington, Joseph Picone. 733-736 [doi]
• Word recognition using multisensor speech input in high ambient noiseSalim E. Roucos, Vishu Viswanathan, Claudia M. Henry, Richard M. Schwartz. 737-740 [doi]
• Noise compensation for speech recognition using probabilistic modelsJohn N. Holmes, Nigel C. Sedgwick. 741-744 [doi]
• Optimal and suboptimal training strategies for automatic speech recognition in noise, and the effects of adaptation on performanceJanet M. Baker, David F. Pinto. 745-748 [doi]
• Improved speech recognition in noiseB. Patrick Landell, Robert E. Wohlford, Lawrence G. Bahler. 749-752 [doi]
• Clustering acoustic prototypes with self organizing distortion measuresDavid Nahamoo. 753-756 [doi]
• Spectral slope based distortion measures for all-pole models of speechBrian A. Hanson, Hisashi Wakita. 757-760 [doi]
• A weighted cepstral distance measure for speech recognitionYoh'ichi Tohkura. 761-764 [doi]
• On the use of bandpass liftering in speech recognitionBiing-Hwang Juang, Lawrence R. Rabiner, Jay G. Wilpon. 765-768 [doi]
• Comparative study of various spectrum matching measures on noise robustnessHiroshi Matsumoto, Haruyuki Imai. 769-772 [doi]
• An adaptive filtering approach to speaker independent vowel recognitionItsuo Kumazawa, Taizo Iijima. 773-776 [doi]
• Systolic image processor for automatic detection and recognition of traffic violationsJames H. Hesson, K. Andrew Harrington. 777-780 [doi]
• Image processing on linear transputer arraysRoy Chapman, T. Willey, J. G. Bartkowiak, Tariq S. Durrani. 781-784 [doi]
• Data flow chip ImPP and its system for image processingMasao Iwashita, Tsutomu Temma. 785-788 [doi]
• Construction methods and performance evaluation for image processing systems with ISP LSIsTadashi Fukushima, Yoshiki Kobayashi, Seiji Kashioka, Kazuyoshi Asada. 789-792 [doi]
• Architecture of a real-time video signal processorKazumasa Enami, Nobuyuki Yagi, Keinosuke Murakami. 793-796 [doi]
• Video signal processor configuration by multiprocessor approachTakao Nishitani, Ichiro Tamitani, Hidenobu Harasaki, Masakatsu Yamashina, Tadayoshi Enomoto. 797-800 [doi]
• A custom chip set for real-time image processingPeter A. Ruetz, Robert W. Brodersen. 801-804 [doi]
• A single chip video rate 16×16 discrete cosine transformFrancis Jutand, Nicolas Demassieux, Gilles Concordel, Jacques Guichard, Eric Cassimatis. 805-808 [doi]
• Video signal processing LSI and its application to TV CODECShin-ichi Maki, Kiichi Matsuda, Toshitaka Tsuda, Hirokazu Fukui, Hirohisa Gambe. 809-812 [doi]
• A study of VLSI logic design for DPCM codingNaoki Mukawa, Yutaka Suzuki, Hideo Kuroda, Hiroshi Yoshimura. 813-816 [doi]
• CCITT Standardizing activities on speech codingMasahiro Taka, Xavier Maitre. 817-820 [doi]
• An advanced 32 kbit/s ADPCM coding to transmit speech and high-speed voiceband dataAtsushi Fukasawa, Kenichiro Hosoda, Takashi Kanda, Yohtaro Yatsuzuka, Makoto Takahashi. 821-824 [doi]
• A real-time ADPCM encoder using variable order predictionFrederick L. Kitson, Kenneth A. Zeger. 825-828 [doi]
• Adaptive postfiltering of 16 kb/s-ADPCM speechNikil S. Jayant, V. Ramamoorthy. 829-832 [doi]
• Amplitude normalization and its application to speech codingJaswant R. Jain. 833-836 [doi]
• Range extended speech decoder for Modulo-PCMMasafumi Hagiwara, Masao Nakagawa. 837-840 [doi]
• Considerations on quantization and dynamic bit-allocation in subband codersTor A. Ramstad. 841-844 [doi]
• 16 kbit/s Split-band APC coder using vector quantization and dynamic bit allocationMaurizio Copperi, Daniele Sereno, Luciano Bertorello. 845-848 [doi]
• Adaptive predictive coding with dynamic quantization adjustment (APC-DQA) at 16 kbits/sYasuhiko Tada, Masahiro Taka. 849-852 [doi]
• Performance of 16 kbit/s APC codec with maximum likelihood quantization for low-speed voiceband dataShigeru Iizuka, Yohtaro Yatsuzuka, Tomohiro Yamazaki. 853-856 [doi]
• A robust 16 kbits/s vector adaptive predictive coder for mobile communicationsAlain Le Guyader, Pierre Combescure, Claude Lamblin, M. Mouly, Jean-Frédéric Zurcher. 857-860 [doi]
• Language identification using noisy speechJerry T. Foil. 861-864 [doi]
• Methods and experiments for text-independent speaker recognition over telephone channelsHerbert Gish, Michael A. Krasner, William Russell, Jared J. Wolf. 865-868 [doi]
• A new method of text-independent speaker recognitionAlan L. Higgins, Robert E. Wohlford. 869-872 [doi]
• Evaluation of a vector quantization talker recognition system in text independent and text dependent modesAaron E. Rosenberg, Frank K. Soong. 873-876 [doi]
• On the use of instantaneous and transitional spectral information in speaker recognitionFrank K. Soong, Aaron E. Rosenberg. 877-880 [doi]
• High performance speaker verification using principal spectral componentsJayant M. Naik, George R. Doddington. 881-884 [doi]
• The effects of voice disguise upon formant transitionLü Shinan, A. Almeida. 885-888 [doi]
• Contributions of pitch, formant frequency and bandwidth to the perception of voice-personalityTohru Takagi, Hisao Kuwabara. 889-892 [doi]
• The effect of formant trajectory and spectral shape on the tense/Lax distinction in American vowelsC.-B. Huang. 893-896 [doi]
• A pitch perception modelAlain de Cheveigné. 897-900 [doi]
• Comparative evaluation of the speech quality of speech coders and text-to-speech synthesizersLouis C. W. Pols, Gerard Boxelaar. 901-904 [doi]
• Acoustical transformation technique as a practical toolOle Roth. 905-907 [doi]
• Grasp and development of spatial informations in a room by closely located four-point microphone methodKenji Endoh, Yoshio Yamasaki, Takeshi Itow. 909-912 [doi]
• Acoustic noise cancellationGerhardus S. Müller, Christoff K. Pauw. 913-916 [doi]
• Inverse control of room acoustics using multiple loudspeakers and/Or microphonesMasato Miyoshi, Yutaka Kaneda. 917-920 [doi]
• Modelling of reverberators and audioconference roomsOlivier Muron, Jacques Sikorav. 921-924 [doi]
• Extraction of fundamental physical parameters in the sound field by multidimensional analysisTakashi Nishi. 925-928 [doi]
• Impulse response measurement using Golay codesScott Foster. 929-932 [doi]
• A new method to locate sound sources by searching the minimum value of error functionMasato Abe, Yoshifumi Nagata, Ken'iti Kido. 933-936 [doi]
• Headphone responses on real ears and a head and torso simulatorKaoru Okabe, Hareo Hamada, Tanetoshi Miura. 937-940 [doi]
• Pre-howling howlback detection methodSatoru Ibaraki, Hiroki Furukawa, Hiroyuki Naono. 941-944 [doi]
• Octave and fractional octave band digital filtering based on the proposed ANSI standardSteven B. Davis. 945-948 [doi]
• Data equalization based on the constant modulus adaptive filterMichael G. Larimore, John R. Treichler. 949-952 [doi]
• The least-squares CMA: A new technique for rapid correction of constant modulus signalsBrian G. Agee. 953-956 [doi]
• An adaptive filter convergence method for echo cancellation and decision feedback equalizationAkira Kanemasa, Akihiko Sugiyama. 957-960 [doi]
• Convergence of an adaptive echo cancellation system with an augmented predictorAlex C. Orgren, N. R. Malik, Dae H. Youn. 961-964 [doi]
• An adaptive echo canceller using parallel Kalman filtersTakashi Yahagi. 965-968 [doi]
• Adaptive rate conversion filterTomoki Ohsawa, Junji Namiki. 969-972 [doi]
• Interference cancellation for enhanced detection of frequency-hopped signalsRonald A. Iltis. 973-976 [doi]
• Noise cancellation for hearing aidsDan Chazan, Yoav Medan, Uzi Shvadron. 977-980 [doi]
• Adaptive optimization of microphone arrays under a nonlinear constraintMan Mohan Sondhi, Gary W. Elko. 981-984 [doi]
• Adaptive discrete cosine transform coding with vector quantization for color imagesKiyoharu Aizawa, Hiroshi Harashima, Hiroshi Miyakawa. 985-988 [doi]
• Motion compensated vector quantizationRey R. Furner, Richard W. Christiansen, Douglas M. Chabries. 989-992 [doi]
• A novel method for constructing a set of Hadamard matricesK. Veerabhadra Rao, V. Umapathi Reddy. 993-996 [doi]
• A nearly optimal transformationZhongde Wang. 997-1000 [doi]
• Two-dimensional transform domain decimation techniquesKing N. Ngan. 1001-1004 [doi]
• Sub-band coding of imagesJohn W. Woods, Sean D. O'Neil. 1005-1008 [doi]
• Contour coding of imagesStephen Marshall, Roy Chapman, Tariq S. Durrani, Louis L. Scharf. 1009-1012 [doi]
• An adaptive state-prediction digital facsimile coding techniqueShan Rong Dai. 1013-1016 [doi]
• Proposal of line element chain codingEiichirou Murakami, Tatsumi Mashimo, Norikazu Onda, Shinji Ozawa. 1017-1020 [doi]
• Two alternatives for motion compensated coding of composite NTSC video signalChow-Ming Lin, Subhash C. Kwatra, Wayne A. Whyte. 1021-1024 [doi]
• 2-D Multilevel FIR-median hybrid filtersAri Nieminen, Pekka Heinonen, Yrjö Neuvo. 1025-1028 [doi]
• High speed delayed multipath 2-D digital filter structuresHon Keung Kwan, Kotaro Hirano. 1029-1032 [doi]
• Optimized shift-varying filters for image interpolationDavid J. Rossi, Stephen W. Lang. 1033-1036 [doi]
• New interpolatory approximation method with application to the design of multi-dimensional FIR filtersTakuro Kida. 1037-1040 [doi]
• 2D Spectral factorization and stability test for 2D matrix polynomials based on the radon projectionJoël Le Roux. 1041-1044 [doi]
• A unified study on the roundoff noise in 2-D state space digital filtersTao Lin, Masayuki Kawamata, Tatsuo Higuchi 0001. 1045-1048 [doi]
• A generalized state-space model for noncausal 2-D systemsYoichi Uetake, Shigenori Okubo. 1049-1052 [doi]
• On compression of multi-level document imagesSudhir S. Dixit, Robert D. Klein. 1053-1056 [doi]
• 3-D Lattice predictive modeling of random fieldsHon Keung Kwan, Ying Chun Lui. 1057-1060 [doi]
• Theory and realization of M-D nonlinear digital filtersGiovanni L. Sicuranza, G. Ramooni. 1061-1064 [doi]
• A continuous training procedure for connected digit recognitionLawrence R. Rabiner, Jay G. Wilpon, Biing-Hwang Juang. 1065-1068 [doi]
• An efficient algorithm for combining vector quantization and stochastic modeling for speaker-independent speech recognitionKalyan Ganesan, M. Marlot, P. Mehta. 1069-1071 [doi]
• Connected digit speech recognition by using demi-word pair reference pattern (DWPR)Hiromi Fujii, Masao Watari, Hiroaki Sakoe, Seibi Chiba. 1073-1076 [doi]
• A speaker independent recognition algorithm for connected word using word boundary hypothesizerTeruhiko Ukita, Tsuneo Nitta, Sadakazu Watanabe. 1077-1080 [doi]
• Global connected digit recognition using Baum-Welch algorithmChristian Wellekens. 1081-1084 [doi]
• Fast feature-based preclassification of segments in continuous digit recognitionDavid M. Lubensky, Wolfgang Feix. 1085-1088 [doi]
• Lexical access and verification in a broad phonetic approach to continuous digit recognitionFrancine R. Chen. 1089-1092 [doi]
• Speaker-independent recognition of connected Swedish digitsGunnar A. Hult. 1093-1096 [doi]
• Network-based connected digit recognition using explicit acoustic-phonetic modelingMarcia A. Bush, Gary E. Kopec. 1097-1100 [doi]
• Connected word recognition by overlap and split of reference patterns and its performance evaluation testsKoji Tajima, Mitsuo Komura, Yasuo Sato. 1101-1104 [doi]
• Two novel algorithms for variable frame analysis and word matching for connected word recognitionCarlo Scagliola, Donatella Sciarra. 1105-1108 [doi]
• A new network-based speaker-independent connected-word recognition systemDenis Jouvet, Jean Monné, Dominique Dubois. 1109-1112 [doi]
• New DP matching algorithms for connected word recognitionMasao Watari. 1113-1116 [doi]
• Syllable-based connected spoken word recognition by two pass O(n) DP matching and hidden Markov modelsSei-Ichi Nakagawa, Mohammed M. Jilan. 1117-1120 [doi]
• Voice-activated word processor with automatic learning for dynamic optimization of syllable-templatesFumio Togawa, Mitsuhiro Hakaridani, Hiroyuki Iwahashi, Toru Ueda. 1121-1124 [doi]
• Japanese text input system based on continuous speech recognitionNoboru Sugamura, Toshiaki Tsuboi, Ryohei Nakatsu. 1125-1128 [doi]
• A "Bunsetsu" speech recognition system using a top-down processor controlled by bottom-up informationAkio Ando, Kazuhiko Ozeki. 1129-1132 [doi]
• Speech understanding system using knowledge engineering techniquesFumihiro Yato, Tohru Asami, Norio Higuchi. 1133-1136 [doi]
• A linguistic processor for Japanese continuous speech recognitionKaichiro Hatazaki, Takao Watanabe. 1137-1140 [doi]
• A grammatical approach to reducing the statistical sparsity of language models in natural domainsThomas M. English, Lois C. Boggess. 1141-1144 [doi]
• N-gram driven search for sentences in a syntactic networkOlli Ventä. 1145-1148 [doi]
• On the analysis of systolic arraysJ. M. Jover, Thomas Kailath. 1149-1152 [doi]
• Implementations of arbitrarily fast adaptive lattice filters with multiple slow processing elementsTeresa H. Y. Meng, David G. Messerschmitt. 1153-1156 [doi]
• A systolic discrete Fourier transform using residue number systems over the ring of Gaussian integersJohn J. Vaccaro, Bruce L. Johnson, Carol L. Nowacki. 1157-1160 [doi]
• A high speed CMOS/SOS implementation of a bit level systolic correlatorJ. C. White, John V. McCanny, A. McCabe, John G. McWhirter, R. Evans. 1161-1164 [doi]
• Systolic implementation for deconvolution iterative algorithmJuan José Navarro-Guerrero, Vicente Casares Giner. 1165-1168 [doi]
• Doubly pipelined Cordic array for digital signal processing algorithmsTze-Yun Sung, Yu Hen Hu, H. J. Yu. 1169-1172 [doi]
• High speed signal processing, pipelining, and VLSIMehdi Hatamian, Glenn L. Cash. 1173-1176 [doi]
• A dataflow algorithm for digital filteringLeah H. Jamieson, Edward A. Ashcroft. 1177-1180 [doi]
• A round-robin architecture for relaxation operationMasaru Kamada, Kazuo Toraichi, Kazuhiko Yamamoto, Hiromitsu Yamada, Ryoichi Mori. 1181-1184 [doi]
• A parallel architecture for recursive least square identificationK. Hashimoto, H. Kimura. 1185-1188 [doi]
• Fault-tolerant digital filtering structures for wafer scale VLSIG. Robert Redinbo. 1189-1192 [doi]
• An expert system for speech spectrogram readingPaul-Eric Stern, Maxine Eskénazi, Daniel Memmi. 1193-1196 [doi]
• An expert spectrogram reader: A knowledge-based approach to speech recognitionVictor W. Zue, Lori Lamel. 1197-1200 [doi]
• APHODEX, design and implementation of an acoustic-phonetic decoding expert systemNoëlle Carbonell, Jean-Paul Damestoy, Dominique Fohr, Jean-Paul Haton, François Lonchamp. 1201-1204 [doi]
• A representational approach to knowledge-based acoustic-phonetic processing in speech recognitionPhil D. Green, A. R. Wood. 1205-1208 [doi]
• Producing and organizing phonetic knowledge from acoustics facts in multi-level data-informationJean Caelen, Nadine Vigouroux. 1209-1212 [doi]
• A hybrid recogniser for speech patternsMichael Allerhand, Frank Fallside. 1213-1216 [doi]
• Plan refinement in a knowledge-based system for automatic speech recognitionRenato de Mori, Lily Lam. 1217-1220 [doi]
• A continuous speech recognition system based on knowledge engineering techniquesRiichiro Mizoguchi, Katsuhiko Tsujino, Osamu Kakusho. 1221-1224 [doi]
• A family of formant trackers based on hidden Markov modelsGary E. Kopec. 1225-1228 [doi]
• A new algorithm for estimation of formant trajectories directly from the speech signal based on an extended Kalman-filterGerhard Rigoll. 1229-1232 [doi]
• A background for sinusoid based representation of voiced speechJorge S. Marques, Luís B. Almeida. 1233-1236 [doi]
• A speech waveform analysis and reconstruction process based on non-euclidean error minimization and matrix array processing techniquesJohn D. Tardelli, Charlton M. Walter. 1237-1240 [doi]
• Continuously variable duration hidden Markov models for speech analysisStephen E. Levinson. 1241-1244 [doi]
• Analysis of speech signals of short pitch period by the sample-selective linear predictionYoshiaki Miyoshi, Kazuharu Yamato, Masuzo Yanagida, Osamu Kakusho. 1245-1248 [doi]
• Efficient implementation of the multipulse LPC method for speechDimitris Manolakis 0001, George Carayannis. 1249-1252 [doi]
• A speech analysis-synthesis system based on the ARMA model and its evaluationHiroyoshi Morikawa, Hiroya Fujisaki. 1253-1256 [doi]
• Analysis of time fluctuating characteristics of linear predictive coefficientsTaizo Umezaki, Fumitada Itakura. 1257-1260 [doi]
• Duality theory of composite sinusoidal modeling and linear predictionShigeki Sagayama, Fumitada Itakura. 1261-1264 [doi]
• Optimal speech data reduction using a stochastic realization algorithmA. K. Mahalanabis, Takeo Kanai. 1265-1268 [doi]
• Musical sound models for digital synthesisJean-Claude Risset. 1269-1271 [doi]
• A signal source in a digital musical synthesizerRokuya Ishii, Hiroshi Katsumoto, Yoshinori Kihara. 1273-1276 [doi]
• Transcription of sung songTakami Niihara, Seiji Inokuchi. 1277-1280 [doi]
• A study on annoyance of musical signal using LAeq measurement and digital signal processingJun Miura, Yasuhiko Yahata, Kiminori Yamaguchi. 1281-1284 [doi]
• Model analysis of a hammer-string interactionHideo Suzuki. 1285-1288 [doi]
• Source separation and note identification in polyphonic musicChris Chafe, David A. Jaffe. 1289-1292 [doi]
• Piano tone synthesis using digital filters by computer simulationIsao Nakamura, Soichiro Iwaoka. 1293-1296 [doi]
• Digital and analog bows: Hybrid mechanical-electrical systemsGabriel Weinreich, René Caussé. 1297-1299 [doi]
• Hardware implementation of FM multipath distortion cancellerMakoto Itami, Takashi Mochizuki, Mitsutoshi Hatori. 1301-1304 [doi]
• Multipath echo cancellation in FM transmission using adaptive FIR filtersKarl-Dirk Kammeyer. 1305-1308 [doi]
• Echo return loss required for acoustic echo canceller based on subjective assessmentHiroshi Yasukawa, Masakazu Nishino, Kaoru Ishimaru, Hideyo Murakami. 1309-1312 [doi]
• Dynamically-reduced complexity implementation of echo cancelersVijay K. Madisetti, David G. Messerschmitt, Niklas Nordström. 1313-1316 [doi]
• An acoustic echo canceller for teleconference systemsJosé R. Casar Corredera, Jesús M. Alcázar-Fernández. 1317-1320 [doi]
• Stereophonic speech teleconferencingRadamis Botros, Onsy A. Abdel-Alim, Peter Damaske. 1321-1324 [doi]
• A new adaptive summing technique for audio teleconferencingTo R. Hsing. 1325-1328 [doi]
• Stability and insertion loss required for audio conference bridgesShoji Shimada, Tetsurou Fujii. 1329-1332 [doi]
• An integrated voice codec and echo canceller implemented in a single DSP processorP. J. Wilson, J. M. Puetz, Alan McCree, D. T. Wang. 1333-1336 [doi]
• Decision-directed echo cancellation for full-duplex data transmission at 4800bpsThomas F. Quatieri, Gerald C. O'Leary. 1337-1340 [doi]
• A system design for real time signal processingT. D. Hopmann, R. J. Canniff, M. A. Derrenberger, P. A. Stiling. 1341-1344 [doi]
• Spectral analysis using the QD algorithmJ. R. Cruz. 1345-1348 [doi]
• A new series of digital filter windows B-spline windowsKazuo Toraichi, Masaru Kamada, Ryoichi Mori. 1349-1352 [doi]
• Spectral analysis of order statistic filtersAlfredo Restrepo, Alan C. Bovik. 1353-1356 [doi]
• Spectral analysis for speech signals with bounded disturbanceY. F. Huang, Mawlin Yeh. 1357-1360 [doi]
• Analysis of a moving sound source - compensation of the Doppler effectTsuyoshi Usagawa, Seiji Nishimura, Masanao Ebata, Josuke Okda. 1361-1364 [doi]
• Parallel AR computation with a reconfigurable signal processorAlastair D. McAulay. 1365-1368 [doi]
• A noise-compensated long correlation matching method for AR spectral estimation of noisy signalsKuldip K. Paliwal. 1369-1372 [doi]
• Subsets of reflection coefficientsPiet M. T. Broersen. 1373-1376 [doi]
• Maximum-entropy spectrum from a non-extendable autocorrelation functionPaul F. Fougere. 1377-1379 [doi]
• Accurate estimation of transfer function of all-pole model system from the signal buried in noiseHiroshi Kanai, Masato Abe, Ken'iti Kido. 1381-1384 [doi]
• New relationships between ARMA and AR processesRaziel Haimi-Cohen, Arnon Cohen. 1385-1388 [doi]
• Maximum entropy pole-zero estimationBruce R. Musicus, Allan M. Kabel. 1389-1392 [doi]
• Prony's method based on eigenanalysis and overdetermined system approachKeiichiroh Minami, Satoshi Kawata. 1393-1396 [doi]
• Improved frequency estimation using total least square approachMd. Anisur Rahman, Kai-Bor Yu. 1397-1400 [doi]
• Adaptive algorithms for estimating the complete covariance eigenstructureKen C. Sharman. 1401-1404 [doi]
• Some results on constrained maximum likelihood estimationYves G. Kamp. 1405-1407 [doi]
• "Fast maximum likelihood joint estimation of frequency and frequency rate"Theagenis J. Abatzoglou. 1409-1412 [doi]
• Estimation of partitioned set of parametersKen Tomiyama. 1413-1416 [doi]
• Multidimensional spectral analysis and the decomposition of ionospheric mode structuresR. L. Johnson, G. E. Miner. 1417-1420 [doi]
• Complexity of waveform formulated from Fourier spectrum - Theory and experimentKanenori Imai, Kazuhiro Kuno, Kazuo Ikegaya. 1421-1424 [doi]
• On estimation with bilinear time seriesRonald R. Mohler, Z. Tang. 1425-1428 [doi]
• Adaptive spectrum estimation by including prior information into the window methodMiquel Bertran-Salvans, Climent Nadeu. 1429-1431 [doi]
• Describing greylevel textures through curvature primal sketchingMarc Richetin, Philippe Saint-Marc, Jean-Thierry Lapresté. 1433-1436 [doi]
• Partitioning of texture image using two-dimensional linear prediction modelHidefumi Kobatake, Jun Moroo. 1437-1440 [doi]
• Automatic classification and recognition of defects in an eddy current non destructive testingP. Simard, P. Gaillard. 1441-1444 [doi]
• Feature extraction by uniform structure threshold logic networksToshiaki Ejima, Masayuki Kimura. 1445-1448 [doi]
• Non-stationary 2D estimation: An application to turbulence field analysisJ.-P. Cassou, Jean-Pierre Gambotto. 1449-1452 [doi]
• An automatic visual inspection method of pattern using a circular feature extraction filterTetsuo Hattori, Yoshiaki Hidaka. 1453-1456 [doi]
• Silhouette understanding systemJacques G. Verly, Patrick L. Van Hove, Robert L. Walton, Dan E. Dudgeon. 1457-1460 [doi]
• Image recognition with inexact processingRobert J. Marks II, Les E. Atlas. 1461-1464 [doi]
• Edge detection using zero crossings of directional derivatives of a random field modelYitong Zhou, Rama Chellappa, V. Venkateswar. 1465-1468 [doi]
• Fourier analysis of object boundaries from two dimensional digitized imagesSally L. Wood. 1469-1472 [doi]
• The use of context in image restorationD. B. Sharman, K. A. Stewart, Tariq S. Durrani. 1473-1476 [doi]
• Automatic classification of zooplancton by image analysisK. Chehdi, Jean-Marc Boucher, Alain Hillion. 1477-1480 [doi]
• An iterative method for restoring noisy imagesSalvatore D. Morgera, Hari Krishna. 1481-1484 [doi]
• Regularized iterative image restoration in a weighted Hilbert spaceJan Biemond, Reginald L. Lagendijk. 1485-1488 [doi]
• A parallel identification procedure for images with noncausal symmetric blursJan Biemond, F. G. van der Putten, John W. Woods. 1489-1492 [doi]
• Karhunen-Loeve transform-based fast algorithms for image restorationHari Krishna, Salvatore D. Morgera, Bal Krishna. 1493-1496 [doi]
• A fast local PPF restoration filterErkki Oja, Jouko Lampinen. 1497-1500 [doi]
• Two-dimensional target recognition using normalization and recorrelation to eliminate noiseJohn C. McKeeman, Matthew Kabrisky. 1501-1504 [doi]
• Restoration theory for digital images under additive noises by a constrained generalized inverse matrixShozo Kondo, Kiyoaki Atsuta. 1505-1508 [doi]
• A projection operator for reconstruction images from sparse, ungridded dataP. G. Rogers, S. A. Edwards, J. D. O'Sullivan. 1509-1512 [doi]
• Least square method for enhancement of laser radar images based on piecewise linear transformations of gray scalesY. Asayama, Sadaaki Miyamoto, Ko Oi, Y. Ikebe. 1513-1516 [doi]
• A real-time digital mixer ICShiv K. Balakrishnan. 1517-1520 [doi]
• A single-chip SC line equalizer system for full duplex multi-bit rate digital transmissionKenji Nakayama, Yutaka Takahashi, Yayoi Satoh, Masahiro Naka, Yasuaki Nukada. 1521-1524 [doi]
• VLSI Architecture for an adaptive equalizer in ISDN line terminationMasayuki Ishikawa, Tsuneo Tsukahara, Tadakatsu Kimura. 1525-1528 [doi]
• An adaptive transversal filter VLSINoboru Kobayashi, Hirohisa Gambe, Koji Aoki, Masami Koshikawa, Shigeyuki Unagami, Toshi Ikezawa. 1529-1532 [doi]
• FIR Lowpass filter for signal decimation with 15 MHz clock frequencyA. Huber, E. De Man, E. Schiller, Walter Ulbrich. 1533-1536 [doi]
• Application specific integrated filters for HIFI digital audio signal processingJ. Van Ginderdeuren, H. De Man, B. De Loore, G. Van Den Audenaerde. 1537-1540 [doi]
• Experiences with automatic generation of audio band digital signal processing circuitsJan M. Rabaey, Robert W. Brodersen. 1541-1544 [doi]
• VLSI- A to D and D to A converters with multi-stage noise shaping modulatorsKuniharu Uchimura, Toshio Hayashi, Tadakatsu Kimura, Atsushi Iwata. 1545-1548 [doi]
• A dynamic time warp VLSI processor for continuous speech recognitionGeorges Quénot, Jean-Luc Gauvain, Jean-Jacques Gangolf, Joseph Mariani. 1549-1552 [doi]
• System design for large-vocabulary speech recognition LSIYoshiaki Kitazume, Hideo Hara, Toshikazu Yasue, Takeyuki Endo. 1553-1556 [doi]
• A wafer scale DTW multiprocessorJames R. Mann, F. Matthew Rhodes. 1557-1560 [doi]
• Syllable-based acoustic-phonetic decoding and wordhypotheses generation in fluently spoken speechHarald Höge, B. Littel, Erwin Marschall, Otto Schmidbauer, R. Sommer. 1561-1564 [doi]
• Generating word hypotheses in continuous speechErnst Günter Schukat-Talamazzini, Heinrich Niemann. 1565-1568 [doi]
• An efficient word lattice parsing algorithm for continuous speech recognitionMasaru Tomita. 1569-1572 [doi]
• A distributed system architecture for speech recognitionFil Alleva, Roberto Bisiani, S. Forin, R. Lerner. 1573-1576 [doi]
• A speech recognition system of continuously spoken Japanese sentences and an application to a speech input deviceMinoru Shigenaga, Yoshihiro Sekiguchi, Tsuyosi Yagisawa, Kinji Kato. 1577-1580 [doi]
• Representation of a continuous speech understanding and dialog system in a homogeneous semantic net achitectureHeinrich Niemann, Astrid Brietzmann, Ute Ehrlich, Gerhard Sagerer. 1581-1584 [doi]
• A new architecture of a speech understanding system - A hybrid of a hierarchical and a network modelsYutaka Kobayashi, Yasuhisa Niimi. 1585-1588 [doi]
• A network model dealing with focus of conversation for speech understanding systemTetsunori Kobayashi, Katsuhiko Shirai. 1589-1592 [doi]
• The role of word-dependent coarticulatory effects in a phoneme-based speech recognition systemYen-Lu Chow, Richard M. Schwartz, Salim E. Roucos, Owen Kimball, Patti Price, Francis Kubala, Mari O. Dunham, Michael A. Krasner, John Makhoul. 1593-1596 [doi]
• A continuous parameter and frequency domain based Markov modelEttore Merlo, Renato de Mori, Mathew Palakal, Guy Mercier. 1597-1600 [doi]
• Analysis, synthesis and perception of the French nasal vowelsEric Bognar, Hiroya Fujisaki. 1601-1604 [doi]
• Proposal and evaluation of models for the glottal source waveformHiroya Fujisaki, Mats Ljungqvist. 1605-1608 [doi]
• On the source-filter model of the vocal tractEdward P. Neuburg, William R. Bauer. 1609-1612 [doi]
• Area estimation from ARMA analysis based on a vocal-tract modelNobuhiro Miki, Sato Saga, Kunitoshi Motoki, Yoshikazu Miyanaga, Nobuo Nagai. 1613-1615 [doi]
• A model for nonstationary analysis of speechYi-Teh Lee, Harvey F. Silverman. 1617-1620 [doi]
• Measurements on the effects of glottal opening and flow on the glottal impedanceUnto K. Laine, Matti Karjalainen. 1621-1624 [doi]
• Measurement and analysis of speech sound radiated from vocal tract wallHisayoshi Suzuki, Takayoshi Nakai, Keiji Shimizu. 1625-1628 [doi]
• Effects of tempo and context on jaw openings for vowels in vowel sequence wordsShinobu Masaki, Katsuhiko Shirai, Shigeru Kiritani. 1629-1632 [doi]
• Simultaneous high-speed digital recording of vocal fold vibration and speech signalShigeru Kiritani, Kiyoshi Honda, Hiroshi Imagawa, Hajime Hirose. 1633-1636 [doi]
• A hierarchical classification of signals and corresponding approximation method based on minimum norm criterionTomohiko Uyematsu, Kohichi Sakaniwa. 1637-1640 [doi]
• Effects of representative distortion and noise on wavelet recovery via the complex cepstrumC. R. Fuller, K. B. Elliott, S. Tavakkoli, W. F. O'Brien, C. J. Hurst. 1641-1644 [doi]
• Performance and limitation of discrete band-limited signal extrapolationHua Lee, Zse-Cherng Lin, Thomas S. Huang. 1645-1648 [doi]
• Signal recovery from nonuniform samples and spectral analysis on random nonuniform samplesFarokh A. Marvasti. 1649-1652 [doi]
• Spline fitting in presense of uniform noiseRavi Kumar. 1653-1656 [doi]
• A unified approach to generalized sampling theoremsHidemitsu Ogawa. 1657-1660 [doi]
• Integer programming approach to optimal smoothing of two-state Markov sequencesKatsuji Uosaki. 1661-1664 [doi]
• A time-domain signal resolution problemAlfred M. Bruckstein, Tie-Jun Shan, Thomas Kailath. 1665-1668 [doi]
• Discrete signal reconstruction from its spectral magnitude and some samplesZhongze Wu, Yenta Li, Tong Chang. 1669-1672 [doi]
• Ignorance-based signal estimation given multiple noisy realizationsNelson Morgan, Alan S. Gevins. 1673-1676 [doi]
• An algorithm for bandlimited signal interpolationB. Yegnanarayana, S. Tanveer Fathima. 1677-1680 [doi]
• High-quality speech at low bit rates: Multi-pulse and stochastically excited linear predictive codersBishnu S. Atal. 1681-1684 [doi]
• CELP Coding for high-quality speech at 8 kbit/sMaurizio Copperi, Daniele Sereno. 1685-1688 [doi]
• High quality multi-pulse speech coder with pitch predictionKazunori Ozawa, Takashi Araseki. 1689-1692 [doi]
• Vector adaptive predictive coding of speech at 9.6 kb/sJuin-Hwey Chen, Allen Gersho. 1693-1696 [doi]
• 8 kbits/s Speech coder with pitch adaptive vector quantizerSatoru Iai, Kazunari Irie. 1697-1700 [doi]
• Speech coder using phase equalization and vector quantizationTakehiro Moriya, Masaaki Honda. 1701-1704 [doi]
• Time-scale modification in medium to low rate speech codingJohn Makhoul, Amro El-Jaroudi. 1705-1708 [doi]
• A study on the relationships between stochastic and harmonic codingIsabel Trancoso, Luís B. Almeida, José M. Tribolet. 1709-1712 [doi]
• Phase modelling and its application to sinusoidal transform codingRobert J. McAulay, Thomas F. Quatieri. 1713-1716 [doi]
• Single DSP 8kbps speech CODECTakanori Miyamoto, Kazuhiro Kondo, Toshiro Suzuki, Yoshiaki Asakawa, Akira Ichikawa. 1717-1720 [doi]
• A high-efficiency speech coding algorithm based on ADPCM with multi-quantizerTomohiko Taniguchi, Kohei Iseda, Shigeyuki Unagami, Syozi Tominaga. 1721-1724 [doi]
• Rapid magnetic resonance imaging using the Hankel tranform approachItzhak Shenberg, Albert Macovski. 1725-1728 [doi]
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• The relationship between Wigner-Ville and evolutionary spectra for frequency modulated random processesJoe K. Hammond, J. S. Lee, R. P. Harrison. 2327-2330 [doi]
• On detection-estimation procedures in the time-frequency planePatrick Flandrin. 2331-2334 [doi]
• ARMA Modeling of time varying systems with lattice filtersErlendur Karlsson, Monson H. Hayes. 2335-2338 [doi]
• Nonstationary ARMA models via simultaneous AR and MA estimationYves Grenier. 2339-2342 [doi]
• Estimation of time-varying coherence functionsJames A. Cadzow, Otis M. Solomon Jr., Samuel D. Stearns. 2343-2346 [doi]
• Spectral line tracking for nonstationary random processesD. Mitchell Wilkes, Monson H. Hayes III. 2347-2350 [doi]
• Spectrum enhancement using linear programmingR. J. Rothacker, Richard J. Mammone, S. Davidovici. 2351-2354 [doi]
• Time series modeling for acoustic signals by a generalized adaptive function and its application to actual random noise dataYasuo Mitani, Mitsuo Ohta. 2355-2358 [doi]
• Composite permutation coding of speech waveformsLuzheng Lu, Gérard D. Cohen, Philippe Godlewski. 2359-2362 [doi]
• Reducing computation in optimal amplitude multipulse codersSharad Singhal. 2363-2366 [doi]
• Implementation of a multi-pulse coder on a single chip floating-point signal processorHerbert Alrutz. 2367-2370 [doi]
• Low cost voice compression for mobile digital radiosJim K. Omura, S.-J. Moon. 2371-2374 [doi]
• Efficient procedures for finding the optimum innovation in stochastic codersIsabel Trancoso, Bishnu S. Atal. 2375-2378 [doi]
• Magnitude-phase coding of base-band speech signalsClaude R. Galand, Henri J. Nussbaumer, J. B. Perini. 2379-2382 [doi]
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• A high quality subband speech coder with backward adaptive predictor and optimal time-frequency bit assignmentFrank K. Soong, Richard V. Cox, Nikil S. Jayant. 2387-2390 [doi]
• Linear transformation for low bit rate speech codingShigeru Ono, Takashi Araseki. 2391-2394 [doi]
• An improved modulo-PCM speech encoder with an imbedded vector quantizerV. Ramamoorthy. 2395-2398 [doi]
• Dynamic time-scale compression for waveform speech codingFrancisco Javier Casajús-Quirós, Ramón García Gómez. 2399-2401 [doi]
• Linguistic processing in the KTH multi-lingual text-to-speech systemRolf Carlson, B. Granstorm. 2403-2406 [doi]
• A grammatical component for a text-to-speech systemRodolfo Delmonte, Gian Antonio Mian, Graziano Tisato. 2407-2410 [doi]
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• Generation of prosodic symbols for rule-synthesis of connected speech of JapaneseKeikichi Hirose, Hiroya Fujisaki, Hisashi Kawai. 2415-2418 [doi]
• Multiparametric generation of French prosody from unrestricted textGérard Bailly. 2419-2422 [doi]
• Stress assignment in letter to sound rules for speech synthesisKenneth Church 0001. 2423-2426 [doi]
• English text to speech conversion with deltaSusan R. Hertz. 2427-2430 [doi]
• Japanese text-to-speech synthesizer based on residual excited speech synthesisKazuo Hakoda, Kiyoshi Kabeya, Tatsuya Hirahara, Keiichi Nagakura. 2431-2434 [doi]
• Phoneme based text-to-speech synthesis systemIchiro Mikuni, Kozo Ohta. 2435-2438 [doi]
• A Chinese text-to-speech system based upon a syllable concatenation modelMing Ouhyoung, Chin-jiang Shie, Chiu-yu Tseng, Lin-Shan Lee. 2439-2442 [doi]
• A diphone speech synthesis approach applicable to different languagesJean-Paul Lefèvre. 2443-2446 [doi]
• A new model of motion blurred images and estimation of its parameterLi Han, Wang Yenping. 2447-2450 [doi]
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• A binary image preprocessor for document quality improvement and data reductionFriedrich M. Wahl. 2459-2462 [doi]
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• A multi-threshold adaptive filtering for image enhancementJianzhong Qian, Kai-Bor Yu, Robert M. Haralick. 2467-2470 [doi]
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• Lognormal random field models and their applications to radar image synthesisRobert T. Frankot, Rama Chellappa. 2479-2482 [doi]
• Support-limited extrapolation of offset Fourier dataDavid C. Munson Jr., Eric A. Ullman. 2483-2486 [doi]
• Iterative reconstruction of multidimensional signals from nonuniformly spaced samplesKen D. Sauer, Jan P. Allebach. 2487-2490 [doi]
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• Direction-of-arrival estimation by subspace rotation methods - ESPRITRichard H. Roy III, Arogyaswami Paulraj, Thomas Kailath. 2495-2498 [doi]
• Adaptive interference and multipath rejection in linear direction-finding arraysOtis L. Frost. 2499-2502 [doi]
• Multipath and multiple source array processing via the EM algorithmMeir Feder, Ehud Weinstein. 2503-2506 [doi]
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• Perfect transmultiplexersMartin Vetterli. 2567-2570 [doi]
• An adaptive tracking filter controlling sampling frequencyKohei Nomoto, Tetsuo Kirimoto, Michimasa Kondo. 2571-2574 [doi]
• Design of computationally efficient FIR filters for sampling rate alteration and multiband filtering with arbitrary passbands and time responseKari-Pekka Estola. 2575-2578 [doi]
• A unified design method of state-space digital filters using system balancing conceptQiangfu Zhao, Masayuki Kawamata, Tatsuo Higuchi 0001. 2579-2582 [doi]
• The implementation of the generalized Lagrange FIR filter structure defined over finite fields or ringsRamasamy Krishnan, Graham A. Jullien, William C. Miller. 2583-2586 [doi]
• Design of IIR group delay digital filters using the iterative near least-square criteriaMasayoshi Sakai, Nozomu Hamada. 2587-2590 [doi]
• Efficient FIR, IIR, and hybrid nyquist filters with zero intersymbol interferenceMarkku Renfors, Tapio Saramäki, Kari-Pekka Estola. 2591-2594 [doi]
• Antialiasing filters for continuously varying sampling rate conversionLeonid M. Blumberg. 2595-2598 [doi]
• A robust adaptive filter for noise reduction problemsPierre Comon, Jean-Louis Lacoume. 2599-2602 [doi]
• Complex wave digital filter for the analysis of complex signalsNobuo Nagai, Masakiyo Suzuki. 2603-2606 [doi]
• Analysis of a periodically time-varying digital filterRokuya Ishii, Masahiro Kakishita. 2607-2610 [doi]
• Noncausal digital filters with antisymmetrical impulse responseChok-Ki Chan. 2611-2614 [doi]
• Design of quadrature mirror filters by linear programmingFrancis Grenez. 2615-2618 [doi]
• A simple phase-reversal tone disablerJames L. Melsa, Roderick J. Ragland. 2619-2622 [doi]
• Techniques for realization of high-speed recursive digital filters using residue number system arithmeticThomas G. Johnson, Michael A. Soderstrand, Gregory A. Clark. 2623-2626 [doi]
• On the dimensionality of steady-state vowel normalizationDavid H. Friedman. 2627-2630 [doi]
• Nonlinear frequency warp for speech recognitionMats Blomberg, Kjell Elenius. 2631-2634 [doi]
• Speech recognition using dual-stage hierarchical structure: Gross and fine phonetic featuresChiu-Kuang Chuang. 2635-2638 [doi]
• Unsupervised speaker adaptation methods for vowel templatesMasahide Sugiyama. 2639-2642 [doi]
• Speaker adaptation through vector quantizationKiyohiro Shikano, Kai-Fu Lee, Raj Reddy. 2643-2646 [doi]
• Speaker-adpatation methods using selective linear predictionNorio Higuchi, Fumihiro Yato. 2647-2650 [doi]
• Synthesis of speaker-adaptive word templates by concatenation of the monosyllabic soundsYasuhisa Niimi, Yutaka Kobayashi. 2651-2654 [doi]
• Speaker-adaptive connected syllable recognition based on the multiple similarity methodHiroyuki Tsuboi, Yoichi Takebayashi, Hiroshi Matsu'ura, Tsuneo Nitta, Shouichi Hirai. 2655-2658 [doi]
• Spectral transformations through canonical correlation analysis for speaker adptation in ASRK. Choukri, Gérard Chollet, Yves Grenier. 2659-2662 [doi]
• New classification method of place of articulation of consonants in connected speech using formantsShigeru Chiba. 2663-2666 [doi]
• Speaker adaptation for a hidden Markov modelKazuhide Sugawara, Masafumi Nishimura, Akihiro Kuroda. 2667-2670 [doi]
• Multi-speaker validation of coarticulation models of syllabic nucleiFrantz Clermont, Bruce Millar. 2671-2674 [doi]
• A relaxation technique for seeking optimal vowel candidate sequenceRong Yu, Masayuki Kimura. 2675-2678 [doi]
• Speaker-independent isolated word recognition using label histogramsOsaaki Watanuki, Toyohisa Kaneko. 2679-2682 [doi]
• Speaker-independent French digit recognition using word-based vector quantization and hidden Markov modelsAlain Tassy, Laurent Miclet. 2683-2686 [doi]
• Speaker-independent isolated word recognition for telephone voice using phoneme-like templatesTetsuya Nomura, Ryohei Nakatsu. 2687-2690 [doi]
• New clustering algorithms applied to speaker independent isolated word recognitionAbdelatif Mokeddem, Heinz Hügli, Fausto Pellandini. 2691-2694 [doi]
• Speaker-independent word recognition for large vocabulary using pre-selection and non-linear spectral matchingJouji Miwa, Ken'iti Kido. 2695-2698 [doi]
• Speaker independent digit recognition with reference frame-specific distance measuresEnrico Bocchieri, George R. Doddington. 2699-2702 [doi]
• Discrimination of isolated vowels and stop consonants using features extracted from onset spectrumShigeyoshi Kitazawa, Shuji Doshita. 2703-2706 [doi]
• A unified theory of deterministic system parameter identificationNan K. Loh, Manohar Das. 2707-2710 [doi]
• Structural identification algorithms of nonlinear systems using volterra functional seriesKazuyuki Murase, Shiro Usui, Naohiro Toda. 2711-2714 [doi]
• Results of least-squares identification algorithm for unstable systemsJ. Jiang, R. Doraiswami. 2715-2718 [doi]
• A dynamical identification method based on conditioned observations for a sound insulation system with the order determined by a new criterion indexAkira Ikuta, Mituo Ohta, Noboru Nakasako. 2719-2722 [doi]
• On-line identification of time variant parameters using covariance informationSeiichi Nakamori. 2723-2726 [doi]
• Digital estimation of linear/Quadratic transfer functions with a general random inputKyoung Il Kim, Edward J. Powers. 2727-2730 [doi]
• Some properties of the lattice algorithm for the direct computation of the matricial spectral factorJoël Le Roux. 2731-2734 [doi]
• The locally quasi-stationary processing by the sigular value decompositionTohru Kiryu, Taizo Iijima. 2735-2738 [doi]
• Modeling of a nonstationary stochastic process produced by a linear time-varying systemTosiro Koga, Akio Miyazaki. 2739-2742 [doi]
• A semi-Markovian parametric model for slowly time-varying binary switchingsDavid L. Wang. 2743-2746 [doi]
• ARMA Covariance realization from noisy dataA. A. (Louis) Beex. 2747-2750 [doi]
• Visual characterization of speech spectrogramsHong C. Leung, Victor W. Zue. 2751-2754 [doi]
• Automatic generation of consonant discrimination rulesKiyoaki Aikawa. 2755-2758 [doi]
• Phoneme classification using Markov modelsBernard Mérialdo, Anne-Marie Derouault, Serge Soudoplatoff. 2759-2762 [doi]
• Phonetic properties of the basic vocabulary of five European languages: Implications for speech recognitionRolf Carlson, Kjell Elenius, Björn Granström, Sheri Hunnicutt. 2763-2766 [doi]
• Detection and recognition of nasal consonants in American EnglishJames R. Glass, Victor W. Zue. 2767-2770 [doi]
• Recognition of word-final unstressed syllablesJohn F. Pitrelli. 2771-2774 [doi]
• A phonetically based semivowel recognition systemCarol Y. Espy-Wilson. 2775-2778 [doi]
• Fine phonetic labeling methodology for speech recognition researchPatricia Collins, Susan Barber. 2779-2782 [doi]
• Automatic labeling system using speaker-dependent phonetic unit referencesShozo Makino, Hisashi Wakita. 2783-2786 [doi]
• Compensating for vowel coarticulation in continuous speech recognitionJames L. Hieronymus, William J. Majurski. 2787-2790 [doi]
• Characterization and modeling of speech-segment durationsThomas H. Crystal, Arthur S. House. 2791-2794 [doi]
• Detection in underwater noises modeled as a Gaussian-Gaussian mixtureMichel Bouvet, Stuart C. Schwartz. 2795-2798 [doi]
• Time delay estimation with nonstationary signalsGeorge A. Lampropoulos, Y. T. Chan. 2799-2802 [doi]
• Determining time delay via frequency estimationY. T. Chan, J. G. Bryan, George A. Lampropoulos. 2803-2806 [doi]
• Detection of unknown-frequency sinusoids in noise via autoregressive modelingAndreas Polydoros, Chrysostomos L. Nikias. 2807-2810 [doi]
• A new method for signal detection and estimation using the eigenstructure of the covariance differenceFranz B. Tuteur, Yosef Rockah. 2811-2814 [doi]
• Application of the block Kalman filter to multisensor estimation with uncertain measurementsSumit Roy, Ronald A. Iltis. 2815-2818 [doi]
• Separated estimation of wave parameters and spectral parameters by maximum likelihoodJohann F. Böhme. 2819-2822 [doi]
• Range and speed estimation from frequency measurements aloneD. E. Ohlms, G. W. Johnson. 2823-2826 [doi]
• Direct estimation of multipath signalsDavid R. Farrier, Fraser N. McLeod. 2827-2830 [doi]
• The estimation of wide-angle bottom loss by a normal incident acoustic pulseAkio Kaya, Shunji Ozaki, Masao Igarashi. 2831-2834 [doi]
• Cepstral methods for the determination of reflection coefficients for dispersive systemsJoe K. Hammond, N. Khalili, Peter M. Clarkson. 2835-2838 [doi]
• A new approach to three-dimensional underwater tracking that considers acoustic medium effectsP. T. Liu, R. E. Silva. 2839-2842 [doi]
• Three-dimensional imaging with a multifrequency holographic methodToyokatsu Miyashita, Toshifumi Hamaguchi, Hisanao Ogura. 2843-2846 [doi]
• Fourier domain reconstruction of ultrasonic cross-sectional imagesKeinosuke Nagai. 2847-2850 [doi]
• Three dimensional reconstruction system from TV camera acquired projections for the dynamic evaluation of scoliosisE. Balbi, S. Becchetti, F. Beltrame, Gianni Vernazza. 2851-2854 [doi]
• FFT Processor as a digital lens for radio patrol camera in astrophysicsTsuneaki Daishido, Kuniyuki Asuma, Hiroyoshi Ohara, Shinichi Komatsu, Kiyoshi Nagane. 2855-2857 [doi]
• Accelerated skew coordinate transformation for high resolution SAR image formationAkira Tsuboi, Akira Maeda, Fuminobu Komura. 2859-2862 [doi]
• High speed SAR data processing on data-flow computer NEDIPSH. Kashihara, K. Nakada, M. Murata, T. Kikuchi, S. Hanaki, T. Temma. 2863-2866 [doi]
• Fast SAR signal processing on the DAPDerek G. Appleby, John J. Soraghan. 2867-2870 [doi]
• Three-dimensional underwater imaging method - synthetic aperture image holography using an acoustic lensTomomasa Sato, Shigeru Igarashi. 2871-2874 [doi]
• Reconstruction of propagating complex wave fields using the Hilbert-Hankel transformMichael S. Wengrovitz, Alan V. Oppenheim, George V. Frisk. 2875-2878 [doi]
• An underground object imaging system with computerized reconstructionKeiichi Ueno, Noriyoshi Osumi, Takunori Mashiko. 2879-2882 [doi]
• Signal processing of holographic under-snow radar with a displaying systems of three-dimensional informationYoshinao Aoki, Yuji Sakamoto, Katsuhiro Tajiri, Takaya Sawai. 2883-2886 [doi]
• A parallel signal processor systemWanda Gass, Richard Tarrant, George R. Doddington. 2887-2890 [doi]
• The optimal synchronous cyclo-static array: A multiprocessor supercomputer for digital signal processingD. A. Schwartz, Thomas P. Barnwell III, C. J. M. Hodges. 2891-2894 [doi]
• Using warp as a supercomputer in signal processingMarco Annaratone, Emmanuel A. Arnould, H. T. Kung 0001, Onat Menzilcioglu. 2895-2898 [doi]
• Arithmetic performance of floating point formats available in VLSIFrederick A. Williams. 2899-2902 [doi]
• On the difficulties of utilizing current technologies to perform 100MHz DSPLouis Schirm IV. 2903-2906 [doi]
• A very fast FFT spectrum analyzer for radio astronomyYoshihiro Chikada, Masato Ishiguro, Hisashi Hirabayashi, Masaki Morimoto, Koh-Ichiro Morita, Tomio Kanzawa, Hiroyuki Iwashita, Kiyoshi Nakazima, Shin-ichi Ishikawa, Toshikazu Takahashi, Kazuyuki Handa, Takashi Kasuga, Sachiko Okumura, Tatsushi Miyazawa, Toshiro Nakazuru, Kenichi Miura, Shigeru Nagasawa. 2907-2910 [doi]
• Conceptual diagnosis of signal processing systemsHamid Nawab, Victor R. Lesser, Evangelos E. Milios. 2911-2914 [doi]
• Evaluation of a high level language oriented program development system for high performance DSP DSSP1S. Ono, Y. Kanayama. 2915-2918 [doi]
• The DSP workbench: Modeling parallel architectures as concurrent processesHenning Schulz-Rinne. 2919-2922 [doi]
• Iteration independent subroutine form of Durbin's recursion for programmable signal processorsJohn G. Ackenhusen. 2923-2926 [doi]
• A topological sorting and loop cleansing algorithm for a constrained MIMD compiler of shift-invariant flow graphsS.-H. Lee, Thomas P. Barnwell III. 2927-2930 [doi]
• A highly parallel architecture for adaptive multichannel algorithmsDavid Mansour. 2931-2934 [doi]
• A variable length lattice filter for adaptive noise cancellationMohammad Hasan Savoji. 2935-2938 [doi]
• Adaptive lattice noise canceller and optimal step sizeHeping Ding, Chong-Zhi Yu. 2939-2942 [doi]
• A family of pseudo-least squares estimation algorithms without divisionFuyun Ling, Dimitris Manolakis 0001, John G. Proakis. 2943-2946 [doi]
• Continuous-time discrete-order lattice filtersHanoch Lev-Ari. 2947-2950 [doi]
• Extended Kalman frequency domain adaptive filtering with data-aided state initializationR. A. Katz, S. J. MacMullan. 2951-2954 [doi]
• On the rectangular transform approach for BLMS adaptive filteringGanapati Panda, Bernard Mulgrew, Colin F. N. Cowan, Peter N. Grant. 2955-2958 [doi]
• Adaptive noise cancelling in the spectrum domainMoeness G. Amin. 2959-2962 [doi]
• Suboptimal frequency domain adaptive array processing in a broadband environmentDavid Nunn. 2963-2966 [doi]
• Chandrasekhar adaptive regularizer for adaptive filteringAmrane Houacine, Guy Demoment. 2967-2970 [doi]
• Adaptive filter for concurrent echo cancellation and equalization using fast Kalman estimationJ. P. Agrawal, N. E. Heckman. 2971-2974 [doi]
• Adaptive signal processing using a modified gradient estimation techniqueM. Yaminysharif, Tariq S. Durrani. 2975-2978 [doi]
• A TAP selection algorithm for adaptive filtersShin-ichi Kawamura, Mitsutoshi Hatori. 2979-2982 [doi]
• Performance analysis of the minimal parameter adaptive notch filter with constrained poles and zerosPetre Stoica, Arye Nehorai. 2983-2986 [doi]
• Recursive equalizers using a multistage adaptive algorithmAnthony W. Seeto, Bevan B. Jones. 2987-2990 [doi]
• A microprocessor-based memory-tap echo cancelerXixian Chen, Guangxin Yue. 2991-2994 [doi]
• Estimation of random spectral misadjustment of an adaptive filterC. R. South. 2995-2998 [doi]
• A stability problem in sign-sign adaptive algorithmsCharles E. Rohrs, C. Richard Johnson Jr., James D. Mills. 2999-3001 [doi]
• Stability margins of linear prediction polynomialsPhilippe Delsarte, Yves V. Genin, Yves G. Kamp. 3003-3005 [doi]
• Long term stability in fractionally-space adaptive equalizersL. Vergara-Domínguez, Ramón García Gómez, Francisco Javier Casajús-Quirós, R. Martín-Arcos. 3007-3010 [doi]
• Coefficient wordlength limitation in FLS adaptive filtersMaurice G. Bellanger, Cumhur Cengiz Evci. 3011-3014 [doi]
• Trade-off studies on adaptive filtering algorithms used for adaptive moving target indicatorsHideaki Watanabe, Tetsuo Kirimoto, Michimasa Kondo. 3015-3018 [doi]
• Adaptive phase locked loop employing frequency and phase processing techniqueSaleh R. Al-Araji, Janan E. Allos, Saleh H. Khadhouri. 3019-3022 [doi]
• A new structure for adaptive signal processing with combined FIR & IIR filtering algorithmsJ. Jiang, R. Doraiswami. 3023-3026 [doi]
• A continuously adaptive vector predictive coder (AVPC) for speech encodingEnrique Masgrau, José B. Mariño, Francesc Vallverdú. 3027-3030 [doi]
• A frequency domain waveform speech compression system based on product vector quantizersNing He, Andres Buzo, Federico Kuhlmann. 3031-3034 [doi]
• A reduced search vector quantizer for speechV. Ramamoorthy, Shakir Abdul-Jabbar. 3035-3038 [doi]
• Performance of a real time low rate voice codecJoseph Rothweiler. 3039-3042 [doi]
• Reducing signal delay in multipulse coding at 16kb/sMichael G. Berouti, J. Jachner, D. Sloan, Paul Mermelstein. 3043-3046 [doi]
• Multi-pulse LPC using linear programming in the frequency domainRichard J. Mammone, G. T. Sentman. 3047-3050 [doi]
• Envelope constrained multipulse speech codingRamón García Gómez, Jesús M. Alcázar-Fernández, Aníbal R. Figueiras-Vidal. 3051-3054 [doi]
• Complexity reduction methods for vector excitation codingGrant A. Davidson, Allen Gersho. 3055-3058 [doi]
• An improved CELP by the separate coding of pulsive and random residualsKazuo Nakata, Satoshi Komine. 3059-3062 [doi]
• Multi-dimensional quantization applied to predictive coding of speechTorbjørn Svendsen. 3063-3066 [doi]
• A novel approach to estimating excitation code in code-excited linear prediction codingGhen Ohyama. 3067-3070 [doi]
• A variable rate coding by APC with maximum likelihood quantization from 4.8 kbits/s to 16 kbits/sYohtaro Yatsuzuka, Shigeru Iizuka, Tomohiro Yamazaki. 3071-3074 [doi]
• Adaptive subbands excited transform (ASET) codingBaruch Mazor, Dale E. Veeneman, E. Mintz. 3075-3078 [doi]
• Variable rate coding for speech storageLars M. Lundheim, Tor A. Ramstad. 3079-3082 [doi]
• A low complexity regular pulse coding scheme with a reduced transmission delayPeter Kroon, Robert J. Sluyter, Ed F. Deprettere. 3083-3086 [doi]
• Selective modeling of the LPC residual during unvoiced frames: White noise or pulse excitationDavid L. Thomson, Dimitrios P. Prezas. 3087-3090 [doi]
• A robust ADPCM system using an error-correcting codeRyuji Kohno, Subbarayan Pasupathy, Hideki Imai, Mitsutoshi Hatori. 3091-3094 [doi]
• Hardware realization of a multistage speech waveform vector quantizaterLiu Tian-ming, Hu Zheng. 3095-3097 [doi]
• Subband coding with vector quantizationI. Versvik, Hans C. Guren. 3099-3102 [doi]
• Two-dimensional spectrum analysis in sonic loggingJames H. McClellan. 3105-3112 [doi]
• On VLSI array architectures for digital image processingS. Y. Kung. 3113-3126 [doi]
• New applications of digital signal processing in communicationsMaurice G. Bellanger. 3127-3134 [doi]
• Speech enhancementJae S. Lim. 3135-3142 [doi]
• Report of the 1985 IEEE ASSPS workshop on speech recognitionStephen E. Levinson. 3143-3144 [doi]
• Use of statistical models for time series analysisHirotugu Akaike. 3147-3155 [doi]
• Predictive coding of speech: Historical review and directions for future researchManfred R. Schroeder. 3157-3164 [doi]
• Signal processing for radio astronomy at the Nobeyama radio observatoryMasato Ishiguro. 3165-3173 [doi]