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ORCIDICASSP 1986: Tokyo, Japan
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 1986, Tokyo, Japan, April 7-11, 1986. IEEE 1986
- K. Odaka, T. T. Doi, H. Nakajima:
Digital audio magnetic recording: Progress in packing density and key technology. 1-4 - Toshiyuki Suzuki, Takeharu Niioka:
Perpendicular magnetic recording and its application to digital recording. 5-8 - Kenji Hayashi, Takao Arai, Takaharu Noguchi, Hiroo Okamoto, Masaharu Kobayashi:
Error correction method for R-DAT and its evaluation. 9-12 - John P. Stautner, David M. Horowitz:
Efficient data reduction for digital audio using a digital filter array. 13-16 - Roger Lagadec:
Digital data equalizing in multitrack digital audio recording. 17-20 - Joël Soumagne, Philippe Mabilleau, Sarto Morissette, Gerald Chouinard, David Bennett:
A comparative study of the proposed high quality coding schemes for digital music. 21-24 - Yoshiharu Hoshino, Toshiyuki Takegahara:
Permissible value of group delay distortion on tone quality due to low-pass filters. 25-28 - Takao Arai, Takaharu Noguchi, Masaharu Kobayashi, Nobutaka Amada, Yasufumi Yumde, Kuniaki Miura:
A study on the digitization of audio signals for video tape recorder. 29-32 - Yasushi Katsumata, Osamu Hamada:
An audio sampling frequency conversion using digital signal processors. 33-36 - Masao Kasuga:
An approach to high resolution D/A converter utilizing a linear predictive coding. 37-40 - Biing-Hwang Juang, Lawrence R. Rabiner:
Mixture autoregressive hidden Markov models for speaker independent isolated word recognition. 41-44 - Serge Soudoplatoff:
Markov modeling of continuous parameters in speech recognition. 45-48 - Lalit R. Bahl, Peter F. Brown, Peter V. de Souza, Robert L. Mercer:
Maximum mutual information estimation of hidden Markov model parameters for speech recognition. 49-52 - Amir Averbuch, Lalit R. Bahl, Raimo Bakis, Peter F. Brown, A. G. Cole, Gregg 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:
An IBM PC based large-vocabulary isolated-utterance speech recognizer. 53-56 - Jean-Luc Gauvain:
A syllable-based isolated word recognition experiment. 57-60 - Jean-Pierre Tubach, Louis-Jean Boë:
Quantitative knowledge on word structure, from a phonetic corpus, with application to large vocabularies recognition systems. 61-64 - Roberto Billi, G. Massia, F. Nesti:
Word preselection for large vocabulary speech recognition. 65-68 - Shoji Hiraoka, Shuji Morii, Masakatsu Hoshimi, Katsuyuki Niyada:
Compact isolated word recognition system for large vocabulary. 69-72 - Lynn Wilcox, Bruce T. Lowerre:
Coarse classification using a hierarchical decision tree and top down parsing. 73-76 - Kai-Fu Lee:
Incremental network generation in word recognition. 77-80 - Mitchel Weintraub:
A computational model for separating two simultaneous talkers. 81-84 - Vishu Viswanathan, Claudia M. Henry, Richard M. Schwartz, Salim E. Roucos:
Evaluation of multisensor speech input for speech recognition in high ambient noise. 85-88 - J. W. Kim, C. K. Un:
Enhancement of noisy speech by forward/Backward adaptive digital filtering. 89-92 - Gagan Mirchandani, Richard C. Gaus Jr., L. Kathy Bechtel:
Performance characteristics of a hardware implementation of the cross-talk resistant adaptive noise canceller. 93-96 - Yasuo Ariki, K. Kajimoto, Toshiyuki Sakai:
Acoustic noise reduction by two dimensional spectral smoothing and spectral amplitude transformation. 97-100 - Kuldip K. Paliwal:
Speech enhancement using multi-pulse excited linear prediction system. 101-104 - David J. Goodman, O. G. Jaffe, Gordon B. Lockhart, W. C. Wong:
Waveform substitution techniques for recovering missing speech segments in packet voice communications. 105-108 - Dimitrios P. Prezas, Joe Picone, David L. Thomson:
Fast and accurate pitch detection using pattern recognition and adaptive time-domain analysis. 109-112 - Francis Charpentier:
Pitch detection using the short-term phase spectrum. 113-116 - Masakazu Imai, Seiji Inokuchi:
Frequency identification by complex spectrum. 117-120 - Werner Verhelst, Fernand De Decker, Bart Francq, Oscar Steenhaut:
An adaptive non-uniform sign clipping preprocessor (ANUSC) for real-time autocorrelative pitch detection. 121-124 - Daniel W. Griffin, Jae S. Lim:
A high quality 9.6 kbps speech coding system. 125-128 - Takahiro Saito, Hideya Takeo, Kiyoharu Aizawa, Hiroshi Harashima, Hiroshi Miyakawa:
Adaptive discrete cosine transform image coding using gain/Shape vector quantizers. 129-132 - Tokumichi Murakami, Kohtaro Asai, Atsushi Itoh:
Vector quantization of color images. 133-136 - R. Aravind, Allen Gersho:
Low-rate image coding with finite-state vector quantization. 137-140 - Ho John Lee, Daniel T. L. Lee:
A gain-shape vector quantizer for image coding. 141-144 - J. P. Marescq, Claude Labit:
Vector quantization in transformed image coding. 145-148 - F. Oliveri, G. Conte, Mario Guglielmo:
A technique using a one-dimensional mapping for vector quantisation of images. 149-152 - P. A. Ramamoorthy, T. Tran:
A hybrid coding involving ADM and vector quantization for digital video image compression. 153-156 - R. H. J. M. Plompen, J. G. P. Groenveld, Dick E. Boekee, F. Booman:
The performance of a hybrid videoconferencing coder using displacement estimation in the transform domain. 157-160 - Stefan Carlsson, Christian Reillo:
Contour based representation of the displacement field for motion compensated image coding. 161-164 - Sharaf E. Elnahas, Kou-Hu Tzou:
Hybrid interframe coding of video signals with backward-acting motion detection. 165-168 - Lloyd J. Griffiths:
High-resolution spectral estimation: Rethinking the Fourier transform. 169-172 - Genshen Xu, Yoh-Han Pao:
Single vector approaches to eigenstructure analysis for harmonic retrieval. 173-176 - Hideaki Sakai:
Estimation of frequencies of sinusoids in colored noise. 177-180 - Peter J. Sherman, Arthur E. Frazho:
High resolution spectral estimation of sinusoids in colored noise using a modified Pisarenko decomposition. 181-184 - Eugene Church, P. Z. Takacs:
Spectral and parameter estimation problems arising in the metrology of high performance mirror surfaces. 185-188 - Cory Myers:
On the use of linear programming for spectral estimation. 189-192 - Sun-Yuan Kung, C. K. Lo, R. Foka:
A Toeplitz approximation approach to coherent source direction finding. 193-196 - S. Lawrence Marple Jr.:
Performance of multichannel autoregressive spectral estimators. 197-200 - Jim E. Schroeder, Rao K. Yarlagadda:
Linear predictive spectral analysis via the Lp norm. 201-204 - Yoshikazu Miyanaga, Nobuo Nagai, Nobuhiro Miki:
Refined ARMA digital lattice filter. 205-208 - Yonina Rosen, Boaz Porat:
ARMA parameter estimation based on sample covariances, for missing data. 209-212 - John W. Adams:
A new FFT approach to the interpolation of discrete-time signals. 213-216 - Naoki Suehiro, Mitsutoshi Hatori:
Fast algorithms for the discrete Fourier transform and for other transforms. 217-220 - Peter Kabal, B. Sayar:
Performance of fixed-point FFT's: Rounding and scaling considerations. 221-224 - C. X. Fan, S. H. Wang:
A fast Fourier transform algorithm using Hadamard transform. 225-228 - Pierre Duhamel, Martin Vetterli:
Cyclic convolution of real sequences: Hartley versus Fourier and new schemes. 229-232 - Ramasamy Krishnan, Graham A. Jullien, William C. Miller:
Computation of complex number theoretic transforms using quadratic residue number systems. 233-236 - Wan-Chi Siu, Anthony G. Constantinides:
A hardware efficient realisation of number theoretic convolvers. 237-240 - Vijay K. Jain, T. E. McClellan, Tapan K. Sarkar:
Half-Fourier transform and application to radar signals. 241-244 - Wirendre A. Perera, Peter J. W. Rayner:
Optimal design of multiplierless DFTS and FFTS. 245-248 - Ramesh C. Agarwal, James W. Cooley:
An efficient vector implementation of the FFT algorithm on IBM 3090VF. 249-252 - Yuval Bistritz, Hanoch Lev-Ari, Thomas Kailath:
Immitance-domain Levinson algorithms. 253-256 - Wonyong Sung, Sanjit K. Mitra:
Efficient multi-processor implementation of recursive digital filters. 257-260 - Jin-Der Wang, H. Joel Trussell:
A unified derivation of the fast RLS algorithms. 261-264 - De-Yuan Cheng, Allen Gersho:
A fast codebook search algorithm for nearest-neighbor pattern matching. 265-268 - Hiroshi Nagaoka, Yoshimi Monden, Suguru Arimoto:
The continuous-time limit of the discrete-time stability theory. 269-272 - Cheung Auyeung, Russell M. Mersereau, Ronald W. Schafer:
Maximum entropy deconvolution. 273-276 - Jesús M. Alcázar-Fernández, José R. Casar Corredera, Ramón García Gómez:
L1-norm noisy Tauberian deconvolution. 277-280 - Zhao-Xiong Wu:
A new homomorphic deconvolution system. 281-284 - Atsushi Imiya, Toshio Sasaki, Hidemitu Ogawa:
Generalized Lanczos method for signal smoothing. 285-288 - Zhenyu Li, Henrik V. Sorensen, C. Sidney Burrus:
FFT and convolution algorithms on DSP microprocessors. 289-292 - Paul M. Farrelle, S. Srinivasan, Anil K. Jain:
A unified transform architecture. 293-296 - William J. Vetter:
The array matrix generalization for signal processing. 297-300 - Kazuo Toraichi, Kazuki Katagishi, Ryoichi Mori:
A fast B-spline transform and its applications. 301-304 - Yasuo Sugiyama:
A generalization of Levinson algorithm for solving Toeplitz systems. 305-308 - Kazuyo Tanaka, Satoru Hayamizu, Kozo Ohta:
A demiphoneme network representation of speech and automatic labeling techniques for speech data base construction. 309-312 - Michael F. Guyote, Keith A. Lewis, Donald Lijana:
A speech data base at the united states air force academy. 313-316 - David S. Pallett:
A PCM/VCR speech database exchange format. 317-320 - Shuichi Itahashi:
A Japanese language speech database. 321-324 - Guy Perennou:
B.D.L.E.X. : A data and cognition base of spoken French. 325-328 - Victor W. Zue, D. Scott Cyphers, Robert H. Kassel, David H. Kaufman, Hong C. Leung, Mark Randolph, Stephanie Seneff, John E. Unverferth III, Timothy Wilson:
The development of the MIT Lisp-machine based speech research workstation. 329-332 - Kari Torkkola, Heikki Riittinen:
A microprocessor-based word recognition system for large vocabularies. 333-336 - A. Fukui, Y. Fujihashi, F. Nakagawa:
Signal processor application to voice dialing equipment. 337-340 - Richard V. Cox, D. Bock, K. B. Bauer, James D. Johnston, Jeffrey Snyder:
The analog voice privacy system. 341-344 - N. Morgan, Baruch Mazor:
Development of a 16-kb/s speech codec for telephone applications. 345-348 - Sami Aly:
24-channel 32 kb/s ADPCM transcoder using the CCITT recommendation G.721. 349-352 - Ph. Missakian, M. Milgram, B. Zavidovique:
A special architecture for dynamic programming. 353-356 - William G. Bliss, J. Girard, J. Avery, M. Lightner, Louis Scharf:
A modular architecture for dynamic programming and maximum likelihood sequence estimation. 357-360 - Allen L. Gorin, J. E. Shoenfelt, R. N. Lewine:
Speech recognition on the DADO/DSP multiprocessor. 361-364 - Evert Dijkstra, Christian Piguet:
A new systolic decomposition for the dynamic time warping algorithm. 365-368 - Francis Jutand, Nicolas Demassieux, Dominique Vicard:
VDP : A versatile high performance vector distance processor. 369-372 - Yoshitake Suzuki:
Design of an efficient dynamic time warping LSI. 373-376 - Makoto Morito, Kozo Yamada, Akihiko Fujisawa, Masao Takeuchi:
A single-chip speaker independent voice recognition system. 377-380 - Henri Barral, Nicolas Moreau:
VLSI Architecture for a real-time LPC-based feature extractor. 381-384 - Frans 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:
On the IC architecture and design of a 2 µm CMOS 8 MIPS digital signal processor with parallel processing capability: The PCB5010/5011. 385-388 - J. L. Laborie, D. F. Martin, J. C. Michalina, A. Picco:
VLSI Digital signal processor (PSI). 389-392 - S. Abiko, M. Hashizume, Y. Matsushita, K. Shinozaki, T. Takamizawa, Cole Erskine, Surendar Magar:
Architecture and applications of a 100-ns CMOS VLSI digital signal processor. 393-396 - John P. Roesgen:
A high performance microprocessor for DSP applications. 397-400 - Takao Kaneko, Hironori Yamauchi, Atsushi Iwata:
A 50ns floating-point signal processor VLSI. 401-404 - Yoshikazu Mori, Toshio Jufuku, Masao Iida, Akira Nomura, Noboru Ichiura, Takao Nakamura:
Architecture of high-speed 22-bit floating-point digital signal processor. 405-408 - Takao Nishitani, Ichiro Kuroda, Yuichi Kawakami, H. Tanaka, Tom Nukiyama:
Advanced single-chip signal processor. 409-412 - Shingo Tsujimichi, TakaHide Ohkami, Yukihiko Shimazu:
A next-generation 32-bit VLSI signal processor. 413-416 - Robert E. Owen, Bruce E. Miller:
Architectural considerations for a sub 10 nanosecond DSP building block family. 417-420 - James R. Boddie, W. Patrick Hays, James Tow:
The architecture, instruction set and development support for the WE®DSP32 digital signal processor. 421-424 - Yoshihiro Tomita, Shigeyuki Unagami, Tomohiko Taniguchi, Yasuhiko Tada, Masahiro Taka:
Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3). 425-428 - Christopher R. Cole, Amine Haoui, Peter L. Winship:
A high performance digital voice echo canceller on a single TMS32020. 429-432 - Eric P. Farges, Mark A. Clements:
Hidden Markov models applied to very low bit rate speech coding. 433-436 - Biing-Hwang Juang:
Design and performance of trellis vector quantizers for speech signals. 437-440 - Yasuo Matsuyama:
Joint time-spectral vector quantization and inverse filter set. 441-444 - Hitoshi Koyama, Allen Gersho:
Fully vector-quantized multipulse LPC at 4800 bps. 445-448 - Hüseyin Abut, Siegfried Ergezinger:
Low-rate speech encoding using vector quantization and subband coding. 449-452 - Richard C. Rose, Thomas P. Barnwell III:
The self excited vocoder - an alternate approach to toll quality at 4800 bps. 453-456 - Kazunori Ozawa, Takashi Araseki:
Low bit rate multi-pulse speech coder with natural speech quality. 457-460 - Daniel Lin:
A novel LPC synthesis model using a binary pulse source excitation. 461-464 - Per Hedelin:
High quality glottal LPC-vocoding. 465-468 - Luis A. Hernández Gómez, Francisco Javier Casajús-Quirós, Aníbal R. Figueiras-Vidal, Ramón García-Gómez:
On the behaviour of reduced complexity code-excited linear prediction (CELP). 469-472 - Joseph P. Campbell Jr., Thomas E. Tremain:
Voiced/Unvoiced classification of speech with applications to the U.S. government LPC-10E algorithm. 473-476 - Karl-Heinz Brenner, Adolf W. Lohmann:
Optical circuitry and architectures for digital optical computing. 477-480 - Yoshiki Ichioka, Jun Tanida:
Optical parallel array logic system. 481-484 - Keikichi Hirose, Hiroya Fujisaki, Yasuhiro Kosugi:
Use of optical signal processing techniques to spectrum analysis of speech. 485-488 - Nobuo Goto, Yasumitsu Miyazaki:
Optical signal processors consisting of collinear acousto-optic channel waveguides. 489-492 - T. Asakura, T. Iwai, Tohru Ifukube, Toshio Kawashima:
Reproduction of the sounds from old wax phonographic cylinders using the laser-beam reflection method. 493-496 - P. P. Vaidyanathan:
New cascaded lattice structures for FIR filters having extremely low coefficent sensitivity. 497-500 - Masami Iwatsuki, Masayuki Kawamata, Tatsuo Higuchi:
Synthesis of minimum sensitivity structures in linear systems using controllability and observability measures. 501-504 - Kou-Hu Tzou:
Embedded max quantization. 505-508 - Kaveh Pahlavan:
Nonlinear quantization and data communication. 509-512 - Sasan H. Ardalan:
Floating point error analysis of recursive least squares and least means squares adaptive filters. 513-516 - Akinori Nishihara:
Design of limit-cycle-free digital biquad filters. 517-520 - Michael E. Lukacs:
Predictive coding of multi-viewpoint image sets. 521-524 - Leonardo Chiariglione, Luigi Corgnier, Mario Guglielmo:
An architecture for the universal video codec. 525-528 - Eugene Walach, Ehud D. Karnin:
A fractal based approach to image compression. 529-532 - Kou-Hu Tzou, Sharaf E. Elnahas:
Bit-sliced progressive transmission and reconstruction of transformed images. 533-536 - Yi-Hsin Chen, Frederick C. Mintzer, Keith S. Pennington:
A binary representation of mixed documents (text/Graphic/Image) that compresses. 537-540 - K. Yamada, Michihiko Minoh, Toshiyuki Sakai:
A representation method of color pictures by arranging fixed size dots of the primary colors. 541-544 - Michael E. Zervakis, Anastasios N. Venetsanopoulos:
Design of 3-D IIR filters via transformations of 2-D circularly symmetric rotated filters. 545-548 - Christakis Charalambous, Hana Khreishi:
A computer program for designing optimum 2-D FIR linear phase digital filters. 549-552 - Acyl Benslimane, Pierre Siohan:
An algorithm for the design of optimal finite wordlength 2-D FIR digital filters. 553-556 - Takao Hinamoto, Sadao Maekawa:
Reduction of the number of parameters in the design of 2-D digital filters. 557-560 - Myung-Ho Pee, B. A. Shenoi, E. Bruce Lee:
On the stability test for 2-D digital recursive filters. 561-564 - Nobumoto Yamane, Yoshitaka Morikawa, Hiroshi Hamada:
A fast image filtering processor-FIFP. 565-568 - Anthony J. Weiss, Ehud Weinstein:
Lower bounds in parameter estimation - summary of results. 569-572 - Peter Strobach:
New forms of least squares lattice algorithms and a comparison of their round-off error characteristics. 573-576 - Yu Hen Hu, Pin-Kuan Chou:
Effective adaptive Pisarenko spectrum estimate. 577-580 - Claude Guéguen, Nicolas Moreau:
Determining MA models as salvos of pulses. 581-584 - Boaz Porat, Benjamin Friedlander:
On the limiting behavior of estimates based on sample covariances. 585-588 - Hong Wang, Mostafa Kaveh:
Performance of narrowband signal-subspace processing. 589-592 - Scott A. Merritt:
Phase-strip sequence estimation: A phase unwrapping/Tracking algorithm. 593-596 - Shin'Ya Kuwahara, Mitsuo Ohta:
A new functional method of parameter estimation for the energy acoustic system based on an Lx evaluation criterion. 597-600 - Masuzo Yanagida, Osamu Kakusho:
Estimation of anti-resonance frequencies by using an over-determined high-order Yule-Walker equation. 601-604 - E. Uchino, M. Ohta:
A new methodological trial on state estimation of linear structure vibration model with noisy power observation mechanism of non-Gaussian type. 605-608 - Peter Stoica, Benjamin Friedlander, Torsten Söderström:
Asymptotic properties of high-order Yule-Walker estimates of frequencies of multiple sinusoids. 609-612 - Hiroshi Hosoi, H. Abe, Fumihiko Ohta, Satoshi Imaizumi:
Assessment of speech discrimination ability of hearing impaired subjects using FIR digital filter. 613-616 - Tohru Ifukube, Robert L. White:
A speech processor with lateral inhibition for an eight channel cochlea implant and its evaluation by subject testing. 617-620 - J. Genin, R. Charachon:
Sound processing for cochlear implant. 621-624 - A. Maynard Engebretson, Robert E. Morley Jr., Michael P. O'Connell:
A wearable, pocket-sized processor for digital hearing aid and other hearing prostheses applications. 625-628 - Yuichi Ueda, Akira Watanabe:
Assessment of electronic aids for the hearing impaired which transmit visible or tactile speech. 629-632 - Lynne E. Bernstein, James B. Ferguson III, Moise H. Goldstein Jr.:
Speech training devices for profoundly deaf children. 633-636 - Minoru Shigenaga, Hirotaka Kubo:
Speech training systems for handicapped children using vocal tract lateral shapes. 637-640 - Santiago Aguilera, A. Borrajo, José M. Pardo, E. Muñoz:
Speech-analysis-based devices for diagnosis and education of speech and hearing impaired people. 641-644 - Marc El-Bèze:
Refutation based recognition to help vowel articulation. 645-648 - Francis Destombes:
Memory-intensive recognition for word articulation training. 649-652 - Robert I. Damper:
Speech control of assistive devices for the physically disabled. 653-656 - Guido Castellini, Pier Luigi Emiliani, Paolo Graziani, A. Tronconi:
Integrated communication aids for the blind. 657-660 - Yukio Mitome, Katsunobu Fushikida:
Japanese speech synthesis system in a book reader for the blind. 661-664 - Sheri Hunnicutt, Lennart Neovius:
A lexical prediction system. 665-668 - Hideki Kasuya, Shigeki Ogawa, Yoshinobu Kikuchi:
An adaptive comb filtering method as applied to acoustic analyses of pathological voice. 669-672 - Sergio Feijóo, C. Hernández, R. Carmelo:
Multidimensional analysis of phonological degeneration in pathological voices. 673-676 - Satoshi Imaizumi:
Acoustic measurement of pathological voice qualities for medical purposes. 677-680 - Jan Gauffin, Britta Hammarberg, Satoshi Imaizumi:
A microcomputer based system for acoustic analysis of voice characteristics. 681-684 - Yuki Kakita:
Wave-flow index: A measure of vocal efficiency. 685-688 - D. G. Childers, D. M. Hicks, G. P. Moore, Yacoub A. Alsaka:
Electroglottography: Assessing articulatory function to aid the handicapped. 689-692 - A. Cohen, A. Lieberman:
Analysis and classification of snoring signals. 693-696 - M. Elghonemy, M. Fikri, M. Hashish, El-Sayed A. Talkhan:
Speaker independent isolated Arabic word recognition system. 697-700 - Michael Wagner, Wei Wang, Helen Ho, Mary O'Kane:
Isolated-word recognition of the complete vocabulary of spoken Chinese. 701-704 - Alan B. Poritz, Alan G. Richter:
On hidden Markov models in isolated word recognition. 705-708 - Abdulmesih Aktas, Bernhard R. Kämmerer, Wolfgang A. Küpper, Helmut Lagger:
Large-vocabulary isolated word recognition with fast coarse time alignment. 709-712 - Shigeo Morishima, Hiroshi Harashima, Hiroshi Miyakawa:
A proposal of a knowledge based isolated word recognition. 713-716 - Won Sik Youn, C. K. Un:
Isolated word recognition based on finite-state vector quantization. 717-720 - J. F. Trehern, Mervyn A. Jack, John Laver:
Speech processing with a Boltzmann machine. 721-724 - Susan L. Banner:
Simulating an acoustic recognizer. 725-728 - Daniel Kahn, Anand Gnanadesikan:
Experiments in speech recognition over the telephone network. 729-732 - Periagaram K. Rajasekaran, George R. Doddington, Joseph Picone:
Recognition of speech under stress and in noise. 733-736 - Salim E. Roucos, Vishu Viswanathan, Claudia M. Henry, Richard M. Schwartz:
Word recognition using multisensor speech input in high ambient noise. 737-740 - John N. Holmes, Nigel C. Sedgwick:
Noise compensation for speech recognition using probabilistic models. 741-744 - Janet M. Baker, David F. Pinto:
Optimal and suboptimal training strategies for automatic speech recognition in noise, and the effects of adaptation on performance. 745-748 - B. Patrick Landell, Robert E. Wohlford, Lawrence G. Bahler:
Improved speech recognition in noise. 749-752 - David Nahamoo:
Clustering acoustic prototypes with self organizing distortion measures. 753-756 - Brian A. Hanson, Hisashi Wakita:
Spectral slope based distortion measures for all-pole models of speech. 757-760 - Yoh'ichi Tohkura:
A weighted cepstral distance measure for speech recognition. 761-764 - Biing-Hwang Juang, Lawrence R. Rabiner, Jay G. Wilpon:
On the use of bandpass liftering in speech recognition. 765-768 - Hiroshi Matsumoto, Haruyuki Imai:
Comparative study of various spectrum matching measures on noise robustness. 769-772 - Itsuo Kumazawa, Taizo Iijima:
An adaptive filtering approach to speaker independent vowel recognition. 773-776 - James H. Hesson, K. Andrew Harrington:
Systolic image processor for automatic detection and recognition of traffic violations. 777-780 - Roy Chapman, T. Willey, J. G. Bartkowiak, Tariq S. Durrani:
Image processing on linear transputer arrays. 781-784 - Masao Iwashita, Tsutomu Temma:
Data flow chip ImPP and its system for image processing. 785-788 - Tadashi Fukushima, Yoshiki Kobayashi, Seiji Kashioka, Kazuyoshi Asada:
Construction methods and performance evaluation for image processing systems with ISP LSIs. 789-792 - Kazumasa Enami, Nobuyuki Yagi, Keinosuke Murakami:
Architecture of a real-time video signal processor. 793-796 - Takao Nishitani, Ichiro Tamitani, Hidenobu Harasaki, Masakatsu Yamashina, Tadayoshi Enomoto:
Video signal processor configuration by multiprocessor approach. 797-800 - Peter A. Ruetz, Robert W. Brodersen:
A custom chip set for real-time image processing. 801-804 - Francis Jutand, Nicolas Demassieux, Gilles Concordel, Jacques Guichard, Eric Cassimatis:
A single chip video rate 16×16 discrete cosine transform. 805-808 - Shin-ichi Maki, Kiichi Matsuda, Toshitaka Tsuda, Hirokazu Fukui, Hirohisa Gambe:
Video signal processing LSI and its application to TV CODEC. 809-812 - Naoki Mukawa, Yutaka Suzuki, Hideo Kuroda, Hiroshi Yoshimura:
A study of VLSI logic design for DPCM coding. 813-816 - Masahiro Taka, Xavier Maitre:
CCITT Standardizing activities on speech coding. 817-820 - Atsushi Fukasawa, Kenichiro Hosoda, Takashi Kanda, Yohtaro Yatsuzuka, Makoto Takahashi:
An advanced 32 kbit/s ADPCM coding to transmit speech and high-speed voiceband data. 821-824 - Frederick L. Kitson, Kenneth A. Zeger:
A real-time ADPCM encoder using variable order prediction. 825-828 - Nikil S. Jayant, Venkatasubbarao Ramamoorthy:
Adaptive postfiltering of 16 kb/s-ADPCM speech. 829-832 - Jaswant R. Jain:
Amplitude normalization and its application to speech coding. 833-836 - Masafumi Hagiwara, Masao Nakagawa:
Range extended speech decoder for Modulo-PCM. 837-840 - Tor A. Ramstad:
Considerations on quantization and dynamic bit-allocation in subband coders. 841-844 - Maurizio Copperi, Daniele Sereno, Luciano Bertorello:
16 kbit/s Split-band APC coder using vector quantization and dynamic bit allocation. 845-848 - Yasuhiko Tada, Masahiro Taka:
Adaptive predictive coding with dynamic quantization adjustment (APC-DQA) at 16 kbits/s. 849-852 - Shigeru Iizuka, Yohtaro Yatsuzuka, Tomohiro Yamazaki:
Performance of 16 kbit/s APC codec with maximum likelihood quantization for low-speed voiceband data. 853-856 - Alain Le Guyader, Pierre Combescure, Claude Lamblin, M. Mouly, Jean-Frédéric Zurcher:
A robust 16 kbits/s vector adaptive predictive coder for mobile communications. 857-860 - Jerry T. Foil:
Language identification using noisy speech. 861-864 - Herbert Gish, Michael A. Krasner, William Russell, Jared J. Wolf:
Methods and experiments for text-independent speaker recognition over telephone channels. 865-868 - Alan L. Higgins, Robert E. Wohlford:
A new method of text-independent speaker recognition. 869-872 - Aaron E. Rosenberg, Frank K. Soong:
Evaluation of a vector quantization talker recognition system in text independent and text dependent modes. 873-876 - Frank K. Soong, Aaron E. Rosenberg:
On the use of instantaneous and transitional spectral information in speaker recognition. 877-880 - Jayant M. Naik, George R. Doddington:
High performance speaker verification using principal spectral components. 881-884 - Lü Shinan, A. Almeida:
The effects of voice disguise upon formant transition. 885-888 - Tohru Takagi, Hisao Kuwabara:
Contributions of pitch, formant frequency and bandwidth to the perception of voice-personality. 889-892 - C. B. Huang:
The effect of formant trajectory and spectral shape on the tense/Lax distinction in American vowels. 893-896 - Alain de Cheveigné:
A pitch perception model. 897-900 - Louis C. W. Pols, Gerard Boxelaar:
Comparative evaluation of the speech quality of speech coders and text-to-speech synthesizers. 901-904 - Ole Roth:
Acoustical transformation technique as a practical tool. 905-907 - Kenji Endoh, Yoshio Yamasaki, Takeshi Itow:
Grasp and development of spatial informations in a room by closely located four-point microphone method. 909-912 - Gerhardus S. Müller, Christoff K. Pauw:
Acoustic noise cancellation. 913-916 - Masato Miyoshi, Yutaka Kaneda:
Inverse control of room acoustics using multiple loudspeakers and/Or microphones. 917-920 - Olivier Muron, Jacques Sikorav:
Modelling of reverberators and audioconference rooms. 921-924 - Takashi Nishi:
Extraction of fundamental physical parameters in the sound field by multidimensional analysis. 925-928 - Scott Foster:
Impulse response measurement using Golay codes. 929-932 - Masato Abe, Yoshifumi Nagata, Ken'iti Kido:
A new method to locate sound sources by searching the minimum value of error function. 933-936 - Kaoru Okabe, Hareo Hamada, Tanetoshi Miura:
Headphone responses on real ears and a head and torso simulator. 937-940 - Satoru Ibaraki, Hiroki Furukawa, Hiroyuki Naono:
Pre-howling howlback detection method. 941-944 - Steven B. Davis:
Octave and fractional octave band digital filtering based on the proposed ANSI standard. 945-948 - Michael G. Larimore, John R. Treichler:
Data equalization based on the constant modulus adaptive filter. 949-952 - Brian G. Agee:
The least-squares CMA: A new technique for rapid correction of constant modulus signals. 953-956 - Akira Kanemasa, Akihiko Sugiyama:
An adaptive filter convergence method for echo cancellation and decision feedback equalization. 957-960 - Alex C. Orgren, N. R. Malik, Dae H. Youn:
Convergence of an adaptive echo cancellation system with an augmented predictor. 961-964 - Takashi Yahagi:
An adaptive echo canceller using parallel Kalman filters. 965-968 - Tomoki Ohsawa, Junji Namiki:
Adaptive rate conversion filter. 969-972 - Ronald A. Iltis:
Interference cancellation for enhanced detection of frequency-hopped signals. 973-976 - Dan Chazan, Yoav Medan, Uzi Shvadron:
Noise cancellation for hearing aids. 977-980 - Man Mohan Sondhi, Gary W. Elko:
Adaptive optimization of microphone arrays under a nonlinear constraint. 981-984 - Kiyoharu Aizawa, Hiroshi Harashima, Hiroshi Miyakawa:
Adaptive discrete cosine transform coding with vector quantization for color images. 985-988 - Rey R. Furner, Richard W. Christiansen, Douglas M. Chabries:
Motion compensated vector quantization. 989-992 - K. Veerabhadra Rao, V. Umapathi Reddy:
A novel method for constructing a set of Hadamard matrices. 993-996 - Zhongde Wang:
A nearly optimal transformation. 997-1000 - King N. Ngan:
Two-dimensional transform domain decimation techniques. 1001-1004 - John W. Woods, Sean D. O'Neil:
Sub-band coding of images. 1005-1008 - Stephen Marshall, Roy Chapman, Tariq S. Durrani, Louis L. Scharf:
Contour coding of images. 1009-1012 - Shan Rong Dai:
An adaptive state-prediction digital facsimile coding technique. 1013-1016 - Eiichirou Murakami, Tatsumi Mashimo, Norikazu Onda, Shinji Ozawa:
Proposal of line element chain coding. 1017-1020 - Chow-Ming Lin, Subhash C. Kwatra, Wayne A. Whyte:
Two alternatives for motion compensated coding of composite NTSC video signal. 1021-1024 - Ari Nieminen, Pekka Heinonen, Yrjö Neuvo:
2-D Multilevel FIR-median hybrid filters. 1025-1028 - Hon Keung Kwan, Kotaro Hirano:
High speed delayed multipath 2-D digital filter structures. 1029-1032 - David J. Rossi, Stephen W. Lang:
Optimized shift-varying filters for image interpolation. 1033-1036 - Takuro Kida:
New interpolatory approximation method with application to the design of multi-dimensional FIR filters. 1037-1040 - Joël Le Roux:
2D Spectral factorization and stability test for 2D matrix polynomials based on the radon projection. 1041-1044 - Tao Lin, Masayuki Kawamata, Tatsuo Higuchi:
A unified study on the roundoff noise in 2-D state space digital filters. 1045-1048 - Yoichi Uetake, Shigenori Okubo:
A generalized state-space model for noncausal 2-D systems. 1049-1052 - Sudhir S. Dixit, Robert D. Klein:
On compression of multi-level document images. 1053-1056 - Hon Keung Kwan, Ying Chun Lui:
3-D Lattice predictive modeling of random fields. 1057-1060 - Giovanni L. Sicuranza, G. Ramooni:
Theory and realization of M-D nonlinear digital filters. 1061-1064 - Lawrence R. Rabiner, Jay G. Wilpon, Biing-Hwang Juang:
A continuous training procedure for connected digit recognition. 1065-1068 - Kalyan Ganesan, M. Marlot, P. Mehta:
An efficient algorithm for combining vector quantization and stochastic modeling for speaker-independent speech recognition. 1069-1071 - Hiromi Fujii, Masao Watari, Hiroaki Sakoe, Seibi Chiba:
Connected digit speech recognition by using demi-word pair reference pattern (DWPR). 1073-1076 - Teruhiko Ukita, Tsuneo Nitta, Sadakazu Watanabe:
A speaker independent recognition algorithm for connected word using word boundary hypothesizer. 1077-1080 - Christian Wellekens:
Global connected digit recognition using Baum-Welch algorithm. 1081-1084 - David M. Lubensky, Wolfgang Feix:
Fast feature-based preclassification of segments in continuous digit recognition. 1085-1088 - Francine R. Chen:
Lexical access and verification in a broad phonetic approach to continuous digit recognition. 1089-1092 - Gunnar A. Hult:
Speaker-independent recognition of connected Swedish digits. 1093-1096 - Marcia A. Bush, Gary E. Kopec:
Network-based connected digit recognition using explicit acoustic-phonetic modeling. 1097-1100 - Koji Tajima, Mitsuo Komura, Yasuo Sato:
Connected word recognition by overlap and split of reference patterns and its performance evaluation tests. 1101-1104 - Carlo Scagliola, Donatella Sciarra:
Two novel algorithms for variable frame analysis and word matching for connected word recognition. 1105-1108 - Denis Jouvet, Jean Monné, Dominique Dubois:
A new network-based speaker-independent connected-word recognition system. 1109-1112 - Masao Watari:
New DP matching algorithms for connected word recognition. 1113-1116 - Sei-ichi Nakagawa, Mohammed M. Jilan:
Syllable-based connected spoken word recognition by two pass O(n) DP matching and hidden Markov models. 1117-1120 - Fumio Togawa, Mitsuhiro Hakaridani, Hiroyuki Iwahashi, Toru Ueda:
Voice-activated word processor with automatic learning for dynamic optimization of syllable-templates. 1121-1124 - Noboru Sugamura, Toshiaki Tsuboi, Ryohei Nakatsu:
Japanese text input system based on continuous speech recognition. 1125-1128 - Akio Ando, Kazuhiko Ozeki:
A "Bunsetsu" speech recognition system using a top-down processor controlled by bottom-up information. 1129-1132 - Fumihiro Yato, Tohru Asami, Norio Higuchi:
Speech understanding system using knowledge engineering techniques. 1133-1136 - Kaichiro Hatazaki, Takao Watanabe:
A linguistic processor for Japanese continuous speech recognition. 1137-1140 - Thomas M. English, Lois C. Boggess:
A grammatical approach to reducing the statistical sparsity of language models in natural domains. 1141-1144 - Olli Ventä:
N-gram driven search for sentences in a syntactic network. 1145-1148 - Juan-Manuel Jover, Thomas Kailath:
On the analysis of systolic arrays. 1149-1152 - Teresa H.-Y. Meng, David G. Messerschmitt:
Implementations of arbitrarily fast adaptive lattice filters with multiple slow processing elements. 1153-1156 - John J. Vaccaro, Bruce L. Johnson, Carol L. Nowacki:
A systolic discrete Fourier transform using residue number systems over the ring of Gaussian integers. 1157-1160 - J. C. White, John V. McCanny, A. McCabe, John G. McWhirter, R. Evans:
A high speed CMOS/SOS implementation of a bit level systolic correlator. 1161-1164 - Juan José Navarro-Guerrero, Vicente Casares Giner:
Systolic implementation for deconvolution iterative algorithm. 1165-1168 - Tze-Yun Sung, Yu-Hen Hu, H. J. Yu:
Doubly pipelined Cordic array for digital signal processing algorithms. 1169-1172 - Mehdi Hatamian, Glenn L. Cash:
High speed signal processing, pipelining, and VLSI. 1173-1176 - Leah H. Jamieson, Edward A. Ashcroft:
A dataflow algorithm for digital filtering. 1177-1180 - Masaru Kamada, Kazuo Toraichi, Kazuhiko Yamamoto, Hiromitsu Yamada, Ryoichi Mori:
A round-robin architecture for relaxation operation. 1181-1184 - K. Hashimoto, H. Kimura:
A parallel architecture for recursive least square identification. 1185-1188 - G. Robert Redinbo:
Fault-tolerant digital filtering structures for wafer scale VLSI. 1189-1192 - Paul-Eric Stern, Maxine Eskénazi, Daniel Memmi:
An expert system for speech spectrogram reading. 1193-1196 - Victor W. Zue, Lori Lamel:
An expert spectrogram reader: A knowledge-based approach to speech recognition. 1197-1200 - Noëlle Carbonell, Jean-Paul Damestoy, Dominique Fohr, Jean-Paul Haton, François Lonchamp:
APHODEX, design and implementation of an acoustic-phonetic decoding expert system. 1201-1204 - Phil D. Green, A. R. Wood:
A representational approach to knowledge-based acoustic-phonetic processing in speech recognition. 1205-1208 - Jean Caelen, Nadine Vigouroux:
Producing and organizing phonetic knowledge from acoustics facts in multi-level data-information. 1209-1212 - Michael Allerhand, Frank Fallside:
A hybrid recogniser for speech patterns. 1213-1216 - Renato De Mori, Lily Lam:
Plan refinement in a knowledge-based system for automatic speech recognition. 1217-1220 - Riichiro Mizoguchi, Katsuhiko Tsujino, Osamu Kakusho:
A continuous speech recognition system based on knowledge engineering techniques. 1221-1224 - Gary E. Kopec:
A family of formant trackers based on hidden Markov models. 1225-1228 - Gerhard Rigoll:
A new algorithm for estimation of formant trajectories directly from the speech signal based on an extended Kalman-filter. 1229-1232 - Jorge S. Marques, Luís B. Almeida:
A background for sinusoid based representation of voiced speech. 1233-1236 - John D. Tardelli, Charlton M. Walter:
A speech waveform analysis and reconstruction process based on non-euclidean error minimization and matrix array processing techniques. 1237-1240 - Stephen E. Levinson:
Continuously variable duration hidden Markov models for speech analysis. 1241-1244 - Yoshiaki Miyoshi, Kazuharu Yamato, Masuzo Yanagida, Osamu Kakusho:
Analysis of speech signals of short pitch period by the sample-selective linear prediction. 1245-1248 - Dimitris Manolakis, George Carayannis:
Efficient implementation of the multipulse LPC method for speech. 1249-1252 - Hiroyoshi Morikawa, Hiroya Fujisaki:
A speech analysis-synthesis system based on the ARMA model and its evaluation. 1253-1256 - Taizo Umezaki, Fumitada Itakura:
Analysis of time fluctuating characteristics of linear predictive coefficients. 1257-1260 - Shigeki Sagayama, Fumitada Itakura:
Duality theory of composite sinusoidal modeling and linear prediction. 1261-1264 - A. K. Mahalanabis, Takeo Kanai:
Optimal speech data reduction using a stochastic realization algorithm. 1265-1268 - Jean-Claude Risset:
Musical sound models for digital synthesis. 1269-1271 - Rokuya Ishii, Hiroshi Katsumoto, Yoshinori Kihara:
A signal source in a digital musical synthesizer. 1273-1276 - Takami Niihara, Seiji Inokuchi:
Transcription of sung song. 1277-1280 - Jun Miura, Yasuhiko Yahata, Kiminori Yamaguchi:
A study on annoyance of musical signal using LAeq measurement and digital signal processing. 1281-1284 - Hideo Suzuki:
Model analysis of a hammer-string interaction. 1285-1288 - Chris Chafe, David A. Jaffe:
Source separation and note identification in polyphonic music. 1289-1292 - Isao Nakamura, Soichiro Iwaoka:
Piano tone synthesis using digital filters by computer simulation. 1293-1296 - Gabriel Weinreich, René Caussé:
Digital and analog bows: Hybrid mechanical-electrical systems. 1297-1299 - Makoto Itami, Takashi Mochizuki, Mitsutoshi Hatori:
Hardware implementation of FM multipath distortion canceller. 1301-1304 - Karl-Dirk Kammeyer:
Multipath echo cancellation in FM transmission using adaptive FIR filters. 1305-1308 - Hiroshi Yasukawa, Masakazu Nishino, Kaoru Ishimaru, Hideyo Murakami:
Echo return loss required for acoustic echo canceller based on subjective assessment. 1309-1312 - Vijay K. Madisetti, David G. Messerschmitt, Niklas Nordström:
Dynamically-reduced complexity implementation of echo cancelers. 1313-1316 - José R. Casar Corredera, Jesús M. Alcázar-Fernández:
An acoustic echo canceller for teleconference systems. 1317-1320 - Radamis Botros, Onsy A. Abdel-Alim, Peter Damaske:
Stereophonic speech teleconferencing. 1321-1324 - To R. Hsing:
A new adaptive summing technique for audio teleconferencing. 1325-1328 - Shoji Shimada, Tetsurou Fujii:
Stability and insertion loss required for audio conference bridges. 1329-1332 - P. J. Wilson, J. M. Puetz, Alan McCree, D. T. Wang:
An integrated voice codec and echo canceller implemented in a single DSP processor. 1333-1336 - Thomas F. Quatieri, Gerald C. O'Leary:
Decision-directed echo cancellation for full-duplex data transmission at 4800bps. 1337-1340 - T. D. Hopmann, R. J. Canniff, M. A. Derrenberger, P. A. Stiling:
A system design for real time signal processing. 1341-1344 - J. R. Cruz:
Spectral analysis using the QD algorithm. 1345-1348 - Kazuo Toraichi, Masaru Kamada, Ryoichi Mori:
A new series of digital filter windows B-spline windows. 1349-1352 - Alfredo Restrepo, Alan C. Bovik:
Spectral analysis of order statistic filters. 1353-1356 - Y. F. Huang, Mawlin Yeh:
Spectral analysis for speech signals with bounded disturbance. 1357-1360 - Tsuyoshi Usagawa, Seiji Nishimura, Masanao Ebata, Josuke Okda:
Analysis of a moving sound source - compensation of the Doppler effect. 1361-1364 - Alastair D. McAulay:
Parallel AR computation with a reconfigurable signal processor. 1365-1368 - Kuldip K. Paliwal:
A noise-compensated long correlation matching method for AR spectral estimation of noisy signals. 1369-1372 - Piet M. T. Broersen:
Subsets of reflection coefficients. 1373-1376 - Paul F. Fougere:
Maximum-entropy spectrum from a non-extendable autocorrelation function. 1377-1379 - Hiroshi Kanai, Masato Abe, Ken'iti Kido:
Accurate estimation of transfer function of all-pole model system from the signal buried in noise. 1381-1384 - Raziel Haimi-Cohen, Arnon Cohen:
New relationships between ARMA and AR processes. 1385-1388 - Bruce R. Musicus, Allan M. Kabel:
Maximum entropy pole-zero estimation. 1389-1392 - Keiichiroh Minami, Satoshi Kawata:
Prony's method based on eigenanalysis and overdetermined system approach. 1393-1396 - Md. Anisur Rahman, Kai-Bor Yu:
Improved frequency estimation using total least square approach. 1397-1400 - Ken C. Sharman:
Adaptive algorithms for estimating the complete covariance eigenstructure. 1401-1404 - Yves G. Kamp:
Some results on constrained maximum likelihood estimation. 1405-1407 - Theagenis J. Abatzoglou:
"Fast maximum likelihood joint estimation of frequency and frequency rate". 1409-1412 - Ken Tomiyama:
Estimation of partitioned set of parameters. 1413-1416 - R. L. Johnson, G. E. Miner:
Multidimensional spectral analysis and the decomposition of ionospheric mode structures. 1417-1420 - Kanenori Imai, Kazuhiro Kuno, Kazuo Ikegaya:
Complexity of waveform formulated from Fourier spectrum - Theory and experiment. 1421-1424 - Ronald R. Mohler, Z. Tang:
On estimation with bilinear time series. 1425-1428 - Miquel Bertran-Salvans, Climent Nadeu:
Adaptive spectrum estimation by including prior information into the window method. 1429-1431 - Marc Richetin, Philippe Saint-Marc, Jean-Thierry Lapresté:
Describing greylevel textures through curvature primal sketching. 1433-1436 - Hidefumi Kobatake, Jun Moroo:
Partitioning of texture image using two-dimensional linear prediction model. 1437-1440 - P. Simard, P. Gaillard:
Automatic classification and recognition of defects in an eddy current non destructive testing. 1441-1444 - Toshiaki Ejima, Masayuki Kimura:
Feature extraction by uniform structure threshold logic networks. 1445-1448 - J.-P. Cassou, Jean-Pierre Gambotto:
Non-stationary 2D estimation: An application to turbulence field analysis. 1449-1452 - Tetsuo Hattori, Yoshiaki Hidaka:
An automatic visual inspection method of pattern using a circular feature extraction filter. 1453-1456 - Jacques G. Verly, Patrick L. Van Hove, Robert L. Walton, Dan E. Dudgeon:
Silhouette understanding system. 1457-1460 - Robert J. Marks II, Les E. Atlas:
Image recognition with inexact processing. 1461-1464 - Yitong Zhou, Rama Chellappa, V. Venkateswar:
Edge detection using zero crossings of directional derivatives of a random field model. 1465-1468 - Sally L. Wood:
Fourier analysis of object boundaries from two dimensional digitized images. 1469-1472 - D. B. Sharman, K. A. Stewart, Tariq S. Durrani:
The use of context in image restoration. 1473-1476 - K. Chehdi, Jean-Marc Boucher, Alain Hillion:
Automatic classification of zooplancton by image analysis. 1477-1480 - Salvatore D. Morgera, Hari Krishna:
An iterative method for restoring noisy images. 1481-1484 - Jan Biemond, Reginald L. Lagendijk:
Regularized iterative image restoration in a weighted Hilbert space. 1485-1488 - Jan Biemond, F. G. van der Putten, John W. Woods:
A parallel identification procedure for images with noncausal symmetric blurs. 1489-1492 - Hari Krishna, Salvatore D. Morgera, Bal Krishna:
Karhunen-Loeve transform-based fast algorithms for image restoration. 1493-1496 - Erkki Oja, Jouko Lampinen:
A fast local PPF restoration filter. 1497-1500 - John C. McKeeman, Matthew Kabrisky:
Two-dimensional target recognition using normalization and recorrelation to eliminate noise. 1501-1504 - Shozo Kondo, Kiyoaki Atsuta:
Restoration theory for digital images under additive noises by a constrained generalized inverse matrix. 1505-1508 - P. G. Rogers, S. A. Edwards, John D. O'Sullivan:
A projection operator for reconstruction images from sparse, ungridded data. 1509-1512 - Y. Asayama, Sadaaki Miyamoto, Ko Oi, Y. Ikebe:
Least square method for enhancement of laser radar images based on piecewise linear transformations of gray scales. 1513-1516 - Shiv K. Balakrishnan:
A real-time digital mixer IC. 1517-1520 - Kenji Nakayama, Yutaka Takahashi, Yayoi Satoh, Masahiro Naka, Yasuaki Nukada:
A single-chip SC line equalizer system for full duplex multi-bit rate digital transmission. 1521-1524 - Masayuki Ishikawa, Tsuneo Tsukahara, Tadakatsu Kimura:
VLSI Architecture for an adaptive equalizer in ISDN line termination. 1525-1528 - Noboru Kobayashi, Hirohisa Gambe, Koji Aoki, Masami Koshikawa, Shigeyuki Unagami, Toshi Ikezawa:
An adaptive transversal filter VLSI. 1529-1532 - A. Huber, Erik De Man, E. Schiller, Walter Ulbrich:
FIR Lowpass filter for signal decimation with 15 MHz clock frequency. 1533-1536 - J. Van Ginderdeuren, H. De Man, B. De Loore, G. Van Den Audenaerde:
Application specific integrated filters for HIFI digital audio signal processing. 1537-1540 - Jan M. Rabaey, Robert W. Brodersen:
Experiences with automatic generation of audio band digital signal processing circuits. 1541-1544 - Kuniharu Uchimura, Toshio Hayashi, Tadakatsu Kimura, Atsushi Iwata:
VLSI- A to D and D to A converters with multi-stage noise shaping modulators. 1545-1548 - Georges Quénot, Jean-Luc Gauvain, Jean-Jacques Gangolf, Joseph Mariani:
A dynamic time warp VLSI processor for continuous speech recognition. 1549-1552 - Yoshiaki Kitazume, Hideo Hara, Toshikazu Yasue, Takeyuki Endo:
System design for large-vocabulary speech recognition LSI. 1553-1556 - James R. Mann, F. Matthew Rhodes:
A wafer scale DTW multiprocessor. 1557-1560 - Harald Höge, B. Littel, Erwin Marschall, Otto Schmidbauer, R. Sommer:
Syllable-based acoustic-phonetic decoding and wordhypotheses generation in fluently spoken speech. 1561-1564 - Ernst Günter Schukat-Talamazzini, Heinrich Niemann:
Generating word hypotheses in continuous speech. 1565-1568 - Masaru Tomita:
An efficient word lattice parsing algorithm for continuous speech recognition. 1569-1572 - Fil Alleva, Roberto Bisiani, S. Forin, R. Lerner:
A distributed system architecture for speech recognition. 1573-1576 - Minoru Shigenaga, Yoshihiro Sekiguchi, Tsuyosi Yagisawa, Kinji Kato:
A speech recognition system of continuously spoken Japanese sentences and an application to a speech input device. 1577-1580 - Heinrich Niemann, Astrid Brietzmann, Ute Ehrlich, Gerhard Sagerer:
Representation of a continuous speech understanding and dialog system in a homogeneous semantic net achitecture. 1581-1584 - Yutaka Kobayashi, Yasuhisa Niimi:
A new architecture of a speech understanding system - A hybrid of a hierarchical and a network models. 1585-1588 - Tetsunori Kobayashi, Katsuhiko Shirai:
A network model dealing with focus of conversation for speech understanding system. 1589-1592 - Yen-Lu Chow, Richard M. Schwartz, Salim E. Roucos, Owen Kimball, Patti Price, Francis Kubala, Mari O. Dunham, Michael A. Krasner, John Makhoul:
The role of word-dependent coarticulatory effects in a phoneme-based speech recognition system. 1593-1596 - Ettore Merlo, Renato De Mori, Mathew Palakal, Guy Mercier:
A continuous parameter and frequency domain based Markov model. 1597-1600 - Eric Bognar, Hiroya Fujisaki:
Analysis, synthesis and perception of the French nasal vowels. 1601-1604 - Hiroya Fujisaki, Mats Ljungqvist:
Proposal and evaluation of models for the glottal source waveform. 1605-1608 - Edward P. Neuburg, William R. Bauer:
On the source-filter model of the vocal tract. 1609-1612 - Nobuhiro Miki, Sato Saga, Kunitoshi Motoki, Yoshikazu Miyanaga, Nobuo Nagai:
Area estimation from ARMA analysis based on a vocal-tract model. 1613-1615 - Yi-Teh Lee, Harvey F. Silverman:
A model for nonstationary analysis of speech. 1617-1620 - Unto K. Laine, Matti Karjalainen:
Measurements on the effects of glottal opening and flow on the glottal impedance. 1621-1624 - Hisayoshi Suzuki, Takayoshi Nakai, Keiji Shimizu:
Measurement and analysis of speech sound radiated from vocal tract wall. 1625-1628 - Shinobu Masaki, Katsuhiko Shirai, Shigeru Kiritani:
Effects of tempo and context on jaw openings for vowels in vowel sequence words. 1629-1632 - Shigeru Kiritani, Kiyoshi Honda, Hiroshi Imagawa, Hajime Hirose:
Simultaneous high-speed digital recording of vocal fold vibration and speech signal. 1633-1636 - Tomohiko Uyematsu, Kohichi Sakaniwa:
A hierarchical classification of signals and corresponding approximation method based on minimum norm criterion. 1637-1640 - C. R. Fuller, K. B. Elliott, S. Tavakkoli, W. F. O'Brien, C. J. Hurst:
Effects of representative distortion and noise on wavelet recovery via the complex cepstrum. 1641-1644 - Hua Lee, Zse-Cherng Lin, Thomas S. Huang:
Performance and limitation of discrete band-limited signal extrapolation. 1645-1648 - Farokh A. Marvasti:
Signal recovery from nonuniform samples and spectral analysis on random nonuniform samples. 1649-1652 - Ravi Kumar:
Spline fitting in presense of uniform noise. 1653-1656 - Hidemitsu Ogawa:
A unified approach to generalized sampling theorems. 1657-1660 - Katsuji Uosaki:
Integer programming approach to optimal smoothing of two-state Markov sequences. 1661-1664 - Alfred M. Bruckstein, Tie-Jun Shan, Thomas Kailath:
A time-domain signal resolution problem. 1665-1668 - Zhongze Wu, Yenta Li, Tong Chang:
Discrete signal reconstruction from its spectral magnitude and some samples. 1669-1672 - Nelson Morgan, Alan S. Gevins:
Ignorance-based signal estimation given multiple noisy realizations. 1673-1676 - B. Yegnanarayana, S. Tanveer Fathima:
An algorithm for bandlimited signal interpolation. 1677-1680 - Bishnu S. Atal:
High-quality speech at low bit rates: Multi-pulse and stochastically excited linear predictive coders. 1681-1684 - Maurizio Copperi, Daniele Sereno:
CELP Coding for high-quality speech at 8 kbit/s. 1685-1688 - Kazunori Ozawa, Takashi Araseki:
High quality multi-pulse speech coder with pitch prediction. 1689-1692 - Juin-Hwey Chen, Allen Gersho:
Vector adaptive predictive coding of speech at 9.6 kb/s. 1693-1696 - Satoru Iai, Kazunari Irie:
8 kbits/s Speech coder with pitch adaptive vector quantizer. 1697-1700 - Takehiro Moriya, Masaaki Honda:
Speech coder using phase equalization and vector quantization. 1701-1704 - John Makhoul, Amro El-Jaroudi:
Time-scale modification in medium to low rate speech coding. 1705-1708 - Isabel Trancoso, Luís B. Almeida, José M. Tribolet:
A study on the relationships between stochastic and harmonic coding. 1709-1712 - Robert J. McAulay, Thomas F. Quatieri:
Phase modelling and its application to sinusoidal transform coding. 1713-1716 - Takanori Miyamoto, Kazuhiro Kondo, Toshiro Suzuki, Yoshiaki Asakawa, Akira Ichikawa:
Single DSP 8kbps speech CODEC. 1717-1720 - Tomohiko Taniguchi, Kohei Iseda, Shigeyuki Unagami, Syozi Tominaga:
A high-efficiency speech coding algorithm based on ADPCM with multi-quantizer. 1721-1724 - Itzhak Shenberg, Albert Macovski:
Rapid magnetic resonance imaging using the Hankel tranform approach. 1725-1728 - M. Reha Civanlar, H. Joel Trussell:
Consistency in tomographic reconstruction by iterative methods. 1729-1732 - Nagaraja Srinivasa, V. Krishnan, K. R. Ramakrishnan, Kasi Rajgopal:
Image reconstruction from truncated projections: A linear prediction approach. 1733-1736 - Noboru Niki, Yoshizo Takahashi, Tamon Inouye:
High speed image reconstruction using direct two-dimensional Fourier transform method. 1737-1740 - Akira Iwata, Isao Horiba, Nobuo Suzumura, Norio Takagi:
3-Dimensional reconstructing algorithm for digital tomo-synthesis. 1741-1744 - Jacques Tanguy, Christian Roux:
Detector width artifact correction in computer assisted tomography. 1745-1748 - Ali Mohammad-Djafari, Guy Demoment:
Maximum entropy diffraction tomography. 1749-1752 - Satoshi Kawata, Osamu Nakamura, Shigeo Minami:
Constrained resolution enhancement in optical microscopic tomography. 1753-1756 - Isao Horiba, Akira Iwata, Nobuo Suzumura, Yoshiyuki Yamagishi, Yutaka Miyagi, Tadashi Uwatoko, Yasushi Mizuno:
Blood flow phase detection on digital subtraction angiography. 1757-1760 - D. Lecomte, J. Beullier:
Computation of synthetical radiographs application to the analysis of high velocity flows. 1761-1764 - Kazuki Katagishi, Kazuo Toraichi, Ryoichi Mori, Sai Yang:
An automatic analyzing system of left ventricular cineangiograms. 1765-1768 - Motoyasu Nagata, Kenji Takada:
Signal restoration of distorted kinesiographic measurement. 1769-1772 - Kawan Soetanto, Shigeo Ohtsuki, Motoyoshi Okujima, Motonao Tanaka:
Ultrasonic beamwidth effect on evaluation of attenuation by sampling medical B-mode imaging data. 1773-1776 - Jong-Ho Choi, Si-Whan Kim, Jong-Soo Choi:
Measurement methods of ultrasonic velocity by dip point analysis from echo spectrum and ultrasonic attenuation considering a dispersion. 1777-1780 - Masuyoshi Yachida, Nagaaki Ohyama, Eric Badiqué, Toshio Honda, Jumpei Tsujiuchi, R. Ono, S. Ikeda:
One approach to color image processing. 1781-1784 - Eric Badiqué, Nagaaki Ohyama, Masuyoshi Yachida, Toshio Honda, Jumpei Tsujiuchi:
Compensation of motion related blur in CCD color endoscope images. 1785-1788 - A. Haouari, Mohamed Najim:
A semi-automatic system for schistosome eggs detection. 1789-1792 - Saman S. Abeysekera, R. J. Bolton, L. C. Westphal, Boualem Boashash:
Patterns in Hilbert transforms and Wigner-Ville distributions of electrocardiogram data. 1793-1796 - M. N. Neelakantan, P. Janardhanan:
Novel method for real-time detection of QRS complex in the presence of strong noise. 1797-1800 - S. C. Chen, H. J. Wei, Y. Tan:
New signal processing methods applied in ECG microcomputer automatic diagnosis system recently designed. 1801-1804 - Yitong Zhou, Rama Chellappa, George A. Bekey:
Estimation of intramuscular EMG signals from surface EMG signal analysis. 1805-1808 - Kaoru Arakawa, Derek H. Fender, Hiroshi Harashima, Hiroshi Miyakawa, Yoichi Saitoh:
A nonlinear digital filter for separating nonstationary waves from stationary waves of the EEG. 1809-1812 - John C. McKeeman, Joseph G. Tront, Matthew Kabrisky:
Design and construction of a multiplexed electrode for monitoring evoked responses in cerebral cortex. 1813-1816 - Qihu Li:
A precise bearing estimation method of digital split-beam array system. 1817-1820 - Chi Chong Ko, Yong Ching Lim, King N. Ngan:
A simple fast adaptive array based on a null steering beamformer. 1821-1824 - M. E. Bouhier, Manell Zakharia:
Noise influence on the accuracy of source localising, comparison of various receivers with an imposed achitecture. 1825-1828 - Dae Hee Youn, B. Chang:
Multichannel lattice filter for an adaptive array processor with linear constraints. 1829-1832 - Daniel Spielman, Arogyaswami Paulraj, Thomas Kailath:
Eigenstructure approach to direction-of-arrival estimation in IR detector arrays. 1833-1836 - Binan Yang:
Processing technique for a linear FM pulse array. 1837-1840 - D. Billon:
Bearing estimation with a linear array. 1841-1844 - Siew Kok Hui, Yong Ching Lim:
An adaptive recursive algorithm for array processing of coherent signal. 1845-1848 - Bindu Chandna, Rajiv Kumar Jain, Rajendar Bahl:
A novel compact beamformer for a high resolution sonar. 1849-1852 - Gordon L. DeMuth:
A partitionable distributed beamformer and signal conditioner architecture. 1853-1856 - Yasutaka Tamura, Hiro Yamasaki:
A digital signal processing approach to data handling for ultrasound beam-forming. 1857-1860 - Ramutis A. Zakarevicius, E. H. Fooks, C. J. E. Phillips:
The integrator as a time delay estimator. 1861-1864 - Robert A. Muir, Douglas M. Chabries, Richard W. Christiansen:
Frequency domain compensation for failed elements in linear sonar arrays. 1865-1868 - Kevin M. Buckley, Lloyd J. Griffiths:
Eigenstructure based broadband source location estimation. 1869-1872 - Kazuaki Takao, Nobuyoshi Kikuma, Tatsuro Yano:
Toeplitzization of correlation matrix in multipath environment. 1873-1876 - T. Ishihara, Leopold B. Felsen:
High frequency acoustic propagation in a range dependent guiding-to-antiguiding ocean channel transition. 1877-1880 - John G. McWhirter, T. J. Shepherd, C. R. Ward, R. A. Jones, P. J. Hargrave:
The application of linear constraints to an adaptive beamformer. 1881-1884 - Yosef Rockah, Peter M. Schultheiss:
Source localization with two dimensional array subject to uncertainty in sensor location. 1885-1888 - James S. Byrnes, Edmund J. Sullivan:
The effect of the character of the noise field upon the error in the optimum gain of an antenna array. 1889-1892 - Jeffrey L. Krolik, Moshe Eizenman, Subbarayan Pasupathy:
Time delay estimation via generalized correlation with adaptive spatial prefiltering. 1893-1896 - Seiji Nishimura, Tsuyoshi Usagawa, Masanao Ebata, Josuke Okda:
Analysis of a moving sound source-Orbit estimation using linear sensor array. 1897-1900 - B. Dellery, P. Mennecier, J. P. Rousselot:
Characterization and identification of trajectories using a leg by leg approach of bearings only target motion analysis. 1901-1904 - Tie-Jun Shan, Arogyaswami Paulraj, Thomas Kailath:
On smoothed rank profile tests in eigenstructure approach to directions-of-arrival estimation. 1905-1908 - Daniel Spielman, Arogyaswami Paulraj, Thomas Kailath:
Performance analysis of the MUSIC algorithm. 1909-1912 - Akira Maeda, Akira Tsuboi, Fuminobu Komura:
A fast and accurate algorithm for SAR Doppler center frequency estimation. 1913-1916 - Frank McCarthy, Arogyaswami Paulraj, Thomas Kailath:
Eigenstructure approach to Doppler estimation for wideband signals. 1917-1920 - Kai-Bor Yu, M. K. Sistanizadeh:
Complex deconvolution in non-coherent radar system. 1921-1924 - M. K. Sistanizadeh, Kai-Bor Yu:
Constraint optimization filter design for radar target enhancement. 1925-1928 - Mark A. Richards, Craig E. Morris, Monson H. Hayes:
Iterative enhancement of noncoherent radar data. 1929-1932 - K. B. Raju, G. R. Reddy, V. V. Rao:
Group delay processing of radar signals. 1933-1936 - Stanislav B. Kesler, Jelisaveta Kesler, Gideon Levita:
Resolution of coherent radar targets in the near field of an array antenna. 1937-1940 - Hong Li, Zhong Wang:
A fast algorithm for evaluating the false alarm probability of a CFAR detector. 1941-1942 - Neil J. Malloy:
Fundamental limits on multiple PRF radar performance. 1943-1946 - Mao-Yung Iin, Long-Quan Ruen, Ru-Mei Xiong:
A microprocessor controlled CCD programmable transversal filter for radar applications. 1947-1950 - Yoshimasa Oh-hashi, Takahiko Fujisaka, Michimasa Kondo, Natsuki Kondo:
An improvement of phase correction for spotlight mapping radar. 1951-1954 - Richard S. Goldhor:
A model of peripheral auditory transduction using a phase vocoder with modified channel signals. 1955-1958 - Hiroya Fujisaki, Keikichi Hirose, Hirofumi Udagawa, Noboru Kanedera:
A new approach to continuous speech recognition based on considerations on human processes of speech perception. 1959-1962 - Louis C. W. Pols, Reinier Plomp:
How to make more efficient use of the fact that the speech signal is dynamic and redundant. 1963-1966 - Maurice K. Wong, Hon Wai Chun:
Toward a massively parallel system for word recognition. 1967-1970 - Hynek Hermansky, Kazuhiro Tsuga, Shozo Makino, Hisashi Wakita:
Perceptually based processing in automatic speech recognition. 1971-1974 - Richard F. Lyon, Lounette M. Dyer:
Experiments with a computational model of the cochlea. 1975-1978 - Melvyn J. Hunt, Claude Lefèbvre:
Speech recognition using a cochlear model. 1979-1982 - Stephanie Seneff:
A computational model for the peripheral auditory system: Application to speech recognition research. 1983-1986 - Hisao Kuwabara:
Representation of vowels in connected speech based on speech perception experiments. 1987-1990 - Sadaoki Furui:
Speaker-independent isolated word recognition based on emphasized spectral dynamics. 1991-1994 - Oded Ghitza:
Speech analysis/Synthesis based on matching the synthesized and the original representations in the auditory nerve level. 1995-1998 - Man Mohan Sondhi, Juergen Schroeter:
A nonlinear articulatory speech synthesizer using both time- and frequency-domain elements. 1999-2002 - Yan Ming Cheng, Bernard Guérin:
Dynamically controlled excitation source for a time-varying formant synthesizer. 2003-2006 - Werner Verhelst, Patrick Nilens:
A modified-superposition speech synthesizer and its applications. 2007-2010 - Werner Frank, Arild Lacroix:
Improved vocal tract models for speech synthesis. 2011-2014 - Francis Charpentier, M. G. Stella:
Diphone synthesis using an overlap-add technique for speech waveforms concatenation. 2015-2018 - Takashi Yazu, Kozo Yamada:
The speech synthesis system for an unlimited Japanese vocabulary. 2019-2022 - Jialu Zhang:
Acoustic parameters and phonological rules of a text-to-speech system for Chinese. 2023-2026 - Douglas D. O'Shaughnessy:
The effects of speaking rate on formant transitions in French synthesis-by-rule. 2027-2030 - Toshimitsu Minowa, Yasuhiko Arai:
The Japanese CV-Syllable positioning rule for speech synthesis. 2031-2034 - Pier Luigi Salza, Stefano Sandri:
Microprosodic timing rules for Italian consonant clusters. 2035-2038 - Hiroya Fujisaki, Keikichi Hirose, Noboru Takahashi, Hiroyoshi Morikawa:
Acoustic characteristics and the underlying rules of intonation of the common Japanese used by radio and television announcers. 2039-2042 - Katsuhiko Shirai, Kazuhiko Iwata, Takeshi Ohno:
Pitch contour control in Japanese conversational speech. 2043-2046 - Xinhua Zhuang, Thomas S. Huang:
Multi-scale edge detection with Gaussian filters. 2047-2050 - O. J. Morris, M. de J. Lee, Anthony G. Constantinides:
A unified method for segmentation and edge detection using graph theory. 2051-2054 - Alan C. Bovik:
Detection of object boundaries in synthetic aperture radar imagery using a human visual model. 2055-2058 - John R. M. Mason:
Contour recognition: A syntax-directed approach. 2059-2062 - Hungwen Li:
Fast Hough transform for multidimensional signal processing. 2063-2066 - Petros Maragos, Ronald W. Schafer:
Applications of morphological filtering to image analysis and processing. 2067-2070 - R. Nag, K. H. Wong, Frank Fallside:
Script recognition using hidden Markov models. 2071-2074 - P. Adam, B. Burg, Bertrand Y. Zavidovique:
Dynamic programming for region based pattern recognition. 2075-2078 - Darwin T. Kuan:
Constraint and consistency in stereo matching. 2079-2082 - Indranil Chakravarty, Bruce G. Nichol:
A multi-resolution representation of 3D imagery for efficient image enhancement and processing. 2083-2086 - Hidetomo Suzuki, Jun-ichiro Toriwaki:
Iterative local parallel operations for structural analysis of line patterns using distance transformation. 2087-2090 - Kaoru Kurosawa, Shigeo Tsujii:
A new IIR type adaptive algorithm of parallel type structure. 2091-2094 - Sasan H. Ardalan:
Derivation of the fast pole-zero (ARMA) recursive least squares algorithm using geometric projections. 2095-2098 - Bernard Mulgrew, Colin F. N. Cowan:
An adaptive IIR channel equaliser: A Kalman filter approach. 2099-2102 - W. Kenneth Jenkins, Majid Nayeri:
Adaptive filters realized with second order sections. 2103-2106 - John J. Shynk, Bernard Widrow:
Bandpass adaptive pole-zero filtering. 2107-2110 - Mohamed El-Sharkawy, Behrouz Peikari:
An adaptive stochastic filter with no positive real condition. 2111-2114 - John M. Cioffi:
A covariance RLS lattice for adaptive filtering. 2115-2118 - Youji Iiguni, Hideaki Sakai, Hidekatsu Tokumaru:
Rounding error analysis of the triangular lattice and escalator algorithms. 2119-2122 - Delores M. Etter, C. T. Huang:
An adaptive delay filter with varibale gains and variable taps. 2123-2126 - Fuyun Ling, Dimitris Manolakis, John G. Proakis:
A flexible, numerically robust array processing algorithm and its relationship to the Givens transformation. 2127-2130 - Mark Maulhardt, Artice M. Davis, Jack May:
Numerical design of nonlinear adaptive filters. 2131-2134 - P. A. Ramamoorthy, Tau Chen:
A high-speed programmable architecture using barrel shifters for FFT and digital filtering. 2135-2138 - Hen-Geul Yeh:
Kalman filtering and systolic processors. 2139-2142 - Peter R. Cappello, Grant A. Davidson, Allen Gersho, Çetin Kaya Koç, V. Srinivasa Somayazulu:
A systolic vector quantization processor for real-time speech coding. 2143-2146 - Magdy A. Bayoumi, Graham A. Jullien, William C. Miller:
A VLSI array for computing the DFT based on RNS. 2147-2150 - Howard M. Shao, Trieu-Kien Truong, In-Shek Hsu, Leslie J. Deutsch, Irving S. Reed:
A single chip VLSI Reed-Solomon decoder. 2151-2154 - Hideki Mori, Hiroto Ouchi, Shinichiro Mori:
A WSI oriented two dimensional systolic array for FFT. 2155-2158 - John V. McCanny:
Mapping system level functions on to bit level systolic arrays. 2159-2162 - Luc de Vos, Rajeev Jain, Hugo De Man, Walter Ulbrich:
A fast adder-based multiplication unit for customised digital signal processors. 2163-2166 - Osamu Hamada, Naohisa Kitazato, Taro Nakagami:
Two chip DSP implementation on gate array. 2167-2170 - B. Barazesh, L. Mary:
A cascadable multimode VLSI signal processor. 2171-2174 - Katsuhiko Ueda, Takashi Sakao, Takao Suzuki, Osamu Nishijima:
A high performance digital signal processor VLSI: MN 1900 series. 2175-2178 - Toshitaka Tsuda, Yukou Mochida:
A practical study on the architecture of VLSI DSP. 2179-2182 - Juan Schiappacasse, Michael Schwartz, Graham Baskerville:
An open architecture CMOS digital signal processor. 2183-2186 - Tomoyoshi Takebayashi, Kazuo Murano, Kaoru Yamamoto, Hiromi Mori, Yukihiro Andoh, Tsuyoshi Miyazaki:
A 32kbps ADPCM with improvement in coding characteristics for 9600bps modem signal. 2187-2190 - Yukiharu Kanayama, Sadayasu Ono:
Wideband speech codec at 64kb/s using an advanced digital signal processor "DSSP-1". 2191-2194 - Shinji Ono, T. Aoyama, Masafumi Hagiwara, Masao Nakagawa:
Implementation of a new type DSP PLL using high performance DSP DSSP-1. 2195-2198 - David P. Morgan, Kevin W. Leary:
A dynamic programming-based DUR recognizer implemented on analog devices' DSP. 2199-2202 - Steven G. Smith, Peter B. Denyer:
Efficient bit-serial complex multiplication and sum-of-products computation using distributed arithmetic. 2203-2206 - Steven G. Smith, A. Fitzgerald, Peter B. Denyer, David Renshaw, N. P. Wootton, R. Creasey:
A comparison of micro-DSP and Silicon compiler implementations of a polyphase network filter bank. 2207-2210 - Brent E. Nelson, Christopher J. Read:
A bit-serial VLSI vector quantizer. 2211-2214 - Shishpal Rawat, Poras T. Balsara, Mary Jane Irwin, Tom Mackowiak:
Design and implementation of real time video processor. 2215-2218 - Mohammad Reza Asharif, Tomoyoshi Takebayashi, Takafumi Chujo, Kazuo Murano:
Frequency domain noise canceller: Frequency bin adaptive filtering (FBAF). 2219-2222 - Tze-Yun Sung, Yu-Hen Hu:
VLSI Implementation of real-time Kalman filter. 2223-2226 - Mary O'Kane, Judy Gillis, Philip Rose, Michael Wagner:
Deciphering speech waveforms. 2227-2230 - Atsuo Tanaka, Shin Kamiya:
A speech processing based on syllable identification by using phonological patterns. 2231-2234 - M. Cravero, Roberto Pieraccini, F. Raineri:
Definition and evaluation of phonetic units for speech recognition by hidden Markov models. 2235-2238 - R. Nag, S. C. Austin, Frank Fallside:
Using hidden Markov models to define linguistic units. 2239-2242 - Kazunori Mano, Shunichi Ishige, Katsuhiko Shirai:
Phoneme recognition in connected speech using both static and dynamic properties of spectrum described by vector quantization. 2243-2246 - Katsuhiko Shirai, Tetsunori Kobayashi, J. Yazawa:
Estimation of articulatory parameters by table look-up method and its application for speaker independent phoneme recognition. 2247-2250 - Satoshi Moriai, Shozo Makino, Ken'iti Kido:
Phoneme recognition in continuous speech using phoneme discriminant filters. 2251-2254 - Ron Cole, Mike Phillips, Bob Brennan, Ben Chigier:
The C-MU phonetic classification system. 2255-2258 - Daniel P. Huttenlocher:
A broad phonetic classifier. 2259-2262 - Dominique Vicard, Laurent Miclet:
Steady part recognition of continuous speech for acoustic-phonetic decoding. 2263-2266 - José Manuel Pardo:
On the determination of speech boundaries: A tool for providing anchor time points in speech recognition. 2267-2270 - Chorkin Chan:
Voiced/Unvoiced segmentation. 2271-2274 - Régine André-Obrecht:
Automatic segmentation of continuous speech signals. 2275-2278 - Shinta Kimura, Yasuhiro Nara:
Interactive extraction of phonemic variation rules in continuous speech. 2279-2282 - Akio Komatsu, Eiji Oohira, Akira Ichikawa, Hirohide Endoh:
Prosodic aids to structural analysis of conversational speech. 2283-2286 - Alex Waibel:
Recognition of lexical stress in a continuous speech understanding system - A pattern recognition approach. 2287-2290 - Ryu-ichi Oka:
Phonemic recognition of each frame with vector field feature using continuous dynamic programming. 2291-2294 - Takao Watanabe:
Syllable recognition for continuous Japanese speech recognition. 2295-2298 - Nobuo Hataoka, Akio Amano, Shun'ichi Yajima, Hirohide En'doh:
VCV Segmentation and phoneme recognition in continuous speech. 2299-2302 - Satoshi Kabasawa, Noriyo Hara, Kiichi Hasegawa, Yutaka Uekawa, Eiichi Tsuboka:
Top-down controlled and optimal syllable-template chaining algorithm for large vocabulary speech recognition. 2303-2306 - Takeshi Kawabata, Masaki Kohda:
Word spotting method taking account of duration change characteristics for stable and transient parts of speech. 2307-2310 - Sei-ichi Nakagawa, Alexander G. Hauptmann, Masaru Tomita:
On quick word spotting techniques. 2311-2314 - Shiro Ehara:
A stochastic process model of sound program signals. 2315-2318 - P. Davies, Joe K. Hammond:
Envelope and instantaneous phase methods for the propagation of transients through oscillatory nonlinear systems. 2319-2322 - Boualem Boashash, Langford B. White, Jörg Imberger:
Wigner-Ville analysis of non-stationary random signals. (with application to turbulent microstructure signals). 2323-2326 - Joe K. Hammond, J. S. Lee, R. P. Harrison:
The relationship between Wigner-Ville and evolutionary spectra for frequency modulated random processes. 2327-2330 - Patrick Flandrin:
On detection-estimation procedures in the time-frequency plane. 2331-2334 - Erlendur Karlsson, Monson H. Hayes:
ARMA Modeling of time varying systems with lattice filters. 2335-2338 - Yves Grenier:
Nonstationary ARMA models via simultaneous AR and MA estimation. 2339-2342 - James A. Cadzow, Otis M. Solomon Jr., Samuel D. Stearns:
Estimation of time-varying coherence functions. 2343-2346 - D. Mitchell Wilkes, Monson H. Hayes III:
Spectral line tracking for nonstationary random processes. 2347-2350 - R. J. Rothacker, Richard J. Mammone, S. Davidovici:
Spectrum enhancement using linear programming. 2351-2354 - Yasuo Mitani, Mitsuo Ohta:
Time series modeling for acoustic signals by a generalized adaptive function and its application to actual random noise data. 2355-2358 - Luzheng Lu, Gérard D. Cohen, Philippe Godlewski:
Composite permutation coding of speech waveforms. 2359-2362 - Sharad Singhal:
Reducing computation in optimal amplitude multipulse coders. 2363-2366 - Herbert Alrutz:
Implementation of a multi-pulse coder on a single chip floating-point signal processor. 2367-2370 - Jim K. Omura, S. J. Moon:
Low cost voice compression for mobile digital radios. 2371-2374 - Isabel Trancoso, Bishnu S. Atal:
Efficient procedures for finding the optimum innovation in stochastic coders. 2375-2378 - Claude R. Galand, Henri J. Nussbaumer, J. B. Perini:
Magnitude-phase coding of base-band speech signals. 2379-2382 - Gianpiero Mensa, Roberto Montagna, Fulvio Rusina:
Comparison between vector and scalar quantization in variable rate subband coders. 2383-2386 - Frank K. Soong, Richard V. Cox, Nikil S. Jayant:
A high quality subband speech coder with backward adaptive predictor and optimal time-frequency bit assignment. 2387-2390 - Shigeru Ono, Takashi Araseki:
Linear transformation for low bit rate speech coding. 2391-2394 - V. Ramamoorthy:
An improved modulo-PCM speech encoder with an imbedded vector quantizer. 2395-2398 - Francisco Javier Casajús-Quirós, Ramón García-Gómez:
Dynamic time-scale compression for waveform speech coding. 2399-2401 - Rolf Carlson, B. Granstorm:
Linguistic processing in the KTH multi-lingual text-to-speech system. 2403-2406 - Rodolfo Delmonte, Gian Antonio Mian, Graziano Tisato:
A grammatical component for a text-to-speech system. 2407-2410 - Yoshinori Sagisaka, Hirokazu Sato:
Word identification method for Japanese text-to-speech conversion system. 2411-2414 - Keikichi Hirose, Hiroya Fujisaki, Hisashi Kawai:
Generation of prosodic symbols for rule-synthesis of connected speech of Japanese. 2415-2418 - Gérard Bailly:
Multiparametric generation of French prosody from unrestricted text. 2419-2422 - Kenneth Church:
Stress assignment in letter to sound rules for speech synthesis. 2423-2426 - Susan R. Hertz:
English text to speech conversion with delta. 2427-2430 - Kazuo Hakoda, Kiyoshi Kabeya, Tatsuya Hirahara, Keiichi Nagakura:
Japanese text-to-speech synthesizer based on residual excited speech synthesis. 2431-2434 - Ichiro Mikuni, Kozo Ohta:
Phoneme based text-to-speech synthesis system. 2435-2438 - Ming Ouhyoung, Chin-jiang Shie, Chiu-yu Tseng, Lin-Shan Lee:
A Chinese text-to-speech system based upon a syllable concatenation model. 2439-2442 - Jean-Paul Lefèvre:
A diphone speech synthesis approach applicable to different languages. 2443-2446 - Li Han, Wang Yenping:
A new model of motion blurred images and estimation of its parameter. 2447-2450 - S. S. H. Naqvi, Neal C. Gallagher, Edward J. Coyle:
An application of median filters to digital television. 2451-2454 - Didier Saint-Félix, Du-Xue-Cheng, Guy Demoment:
Image deconvolution using 2-D non-causal fast Kalman filters. 2455-2458 - Friedrich M. Wahl:
A binary image preprocessor for document quality improvement and data reduction. 2459-2462 - Sanjit K. Mitra, Tian-Hu Yu:
Image enhancement via transform amplitude sharpening. 2463-2466 - Jianzhong Qian, Kai-Bor Yu, Robert M. Haralick:
A multi-threshold adaptive filtering for image enhancement. 2467-2470 - V. Ralph Algazi:
FIR Anisotropic filters for image enhancement. 2471-2474 - Mutsumi Watanabe, Eiji Osaki, Shigekatsu Horii, Satoshi Kageyama:
A new recursive estimation method for digital image restoration filter and its applications. 2475-2478 - Robert T. Frankot, Rama Chellappa:
Lognormal random field models and their applications to radar image synthesis. 2479-2482 - David C. Munson Jr., Eric A. Ullman:
Support-limited extrapolation of offset Fourier data. 2483-2486 - Ken D. Sauer, Jan P. Allebach:
Iterative reconstruction of multidimensional signals from nonuniformly spaced samples. 2487-2490 - B. W. Dahanayake, Kon Max Wong:
Proper orthogonal projection - multiple signal classification (POP-MUSIC). 2491-2494 - Richard H. Roy III, Arogyaswami Paulraj, Thomas Kailath:
Direction-of-arrival estimation by subspace rotation methods - ESPRIT. 2495-2498 - Otis L. Frost:
Adaptive interference and multipath rejection in linear direction-finding arrays. 2499-2502 - Meir Feder, Ehud Weinstein:
Multipath and multiple source array processing via the EM algorithm. 2503-2506 - Ken'iti Kido, Hiroyuki Noto, Akihiro Shima, Masato Abe:
Detection of acoustic signal source by use of cross spectra between every pair of sensors distributed in space. 2507-2510 - Charles M. Rader, Allan O. Steinhardt:
Hyperbolic householder transformations, definition and application. 2511-2514 - Anas Abo-zena, Eugene Church:
Multiple target detection - optimum resolution. 2515-2518 - David N. Swingler, Robert S. Walker, G. B. Lewis:
Linear prediction for aperture extrapolation in line array beamforming. 2519-2522 - Richard Gooch, John Lundell:
The CM array: An adaptive beamformer for constant modulus signals. 2523-2526 - Meng Hwa Er, Antonio Cantoni:
A new approach to the design of robust narrow-band array processors. 2527-2530 - William J. Graham:
Fresnel region imaging with a linear array. 2531-2534 - Jaakko Astola, Pekka Heinonen, Yrjö Neuvo:
On the oscillatory roots of one and two dimensional median type filters. 2535-2538 - Edward J. Coyle:
On the optimality of rank order operations. 2539-2542 - Kumar Swaminathan, P. P. Vaidyanathan:
Theory of uniform DFT, parallel, quadrature mirror filter banks. 2543-2546 - P. P. Vaidyanathan, Phillip A. Regalia, Sanjit K. Mitra:
Design of doubly-complementary IIR digital filters, using a single complex allpass filter. 2547-2550 - R. E. Leonard Jr., Charles M. Loeffler:
Bandwidth compression with periodically time-varying FIR filters. 2551-2554 - Markku Renfors, Yrjö Neuvo, Sanjit K. Mitra:
A new class of uniform filter banks based on recursive Nth-band filters. 2555-2558 - Thomas L. Marzetta, Thomas W. Parks, Xiangkun Chen:
FIR Filters with minimum coefficient sensitivity. 2559-2562 - Haruhiko Takeuchi, Kazuo Toraichi, Masaru Kamada, Ryoichi Mori:
Spline filter-A time-varying digital filter and its application. 2563-2566 - Martin Vetterli:
Perfect transmultiplexers. 2567-2570 - Kohei Nomoto, Tetsuo Kirimoto, Michimasa Kondo:
An adaptive tracking filter controlling sampling frequency. 2571-2574 - Kari-Pekka Estola:
Design of computationally efficient FIR filters for sampling rate alteration and multiband filtering with arbitrary passbands and time response. 2575-2578 - Qiangfu Zhao, Masayuki Kawamata, Tatsuo Higuchi:
A unified design method of state-space digital filters using system balancing concept. 2579-2582 - Ramasamy Krishnan, Graham A. Jullien, William C. Miller:
The implementation of the generalized Lagrange FIR filter structure defined over finite fields or rings. 2583-2586 - Masayoshi Sakai, Nozomu Hamada:
Design of IIR group delay digital filters using the iterative near least-square criteria. 2587-2590 - Markku Renfors, Tapio Saramäki, Kari-Pekka Estola:
Efficient FIR, IIR, and hybrid nyquist filters with zero intersymbol interference. 2591-2594 - Leonid M. Blumberg:
Antialiasing filters for continuously varying sampling rate conversion. 2595-2598 - Pierre Comon, Jean-Louis Lacoume:
A robust adaptive filter for noise reduction problems. 2599-2602 - Nobuo Nagai, Masakiyo Suzuki:
Complex wave digital filter for the analysis of complex signals. 2603-2606