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ORCIDICASSP 1989: Glasgow, Scotland
- IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP '89, Glasgow, Scotland, May 23-26, 1989. IEEE 1989
- Kazunaga Yoshida, Takao Watanabe, Shinji Koga:
Large vocabulary word recognition based on demi-syllable hidden Markov model using small amount of training data. 1-4 - Gerhard Rigoll:
Speaker adaptation for large vocabulary speech recognition systems using speaker Markov models. 5-8 - Eleftherios D. Frangoulis:
Vector quantisation of the continuous distributions of an HMM speech recogniser based on mixtures of continuous distributions. 9-12 - Jerome R. Bellegarda, David Nahamoo:
Tied mixture continuous parameter models for large vocabulary isolated speech recognition. 13-16 - Les T. Niles, Harvey F. Silverman, Gary N. Tajchman, Marcia A. Bush:
How limited training data can allow a neural network to outperform an 'optimal' statistical classifier. 17-20 - Andreas Krause, Heidi Hackbarth:
Scaly artificial neural networks for speaker-independent recognition of isolated words. 21-24 - Hidefumi Sawai, Alex Waibel, Masanori Miyatake, Kiyohiro Shikano:
Spotting Japanese CV-syllables and phonemes using time-delay neural networks. 25-28 - Hiroaki Sakoe, Ryosuke Isotani, Kazunaga Yoshida, Ken-ichi Iso, Takao Watanabe:
Speaker-independent word recognition using dynamic programming neural networks. 29-32 - Hervé Bourlard, Christian J. Wellekens:
Speech dynamics and recurrent neural networks. 33-36 - Frédéric Guyot, Frédéric Alexandre, Jean Paul Haton:
Toward a continuous model of the cortical column: Application to speech recognition. 37-40 - E. Ofer, David Malah, Amir Dembo:
A unified framework for LPC excitation representation in residual speech coders. 41-44 - Bishnu S. Atal:
A model of LPC excitation in terms of eigenvectors of the autocorrelation matrix of the impulse response of the LPC filter. 45-48 - Shihua Wang, Allen Gersho:
Phonetically-based vector excitation coding of speech at 3.6 kbps. 49-52 - Anders Bergström, Per Hedelin:
Code-book driven glottal pulse analysis. 53-56 - Mark Ireton, Costas S. Xydeas:
On improving vector excitation coders through the use of spherical lattice codebooks (SLCs). 57-60 - Claude Lamblin, Jean-Pierre Adoul, Dominique Massaloux, Sarto Morissette:
Fast CELP coding based on the Barnes-Wall lattice in 16 dimensions. 61-64 - Nikil S. Jayant, Juin-Hwey Chen:
Speech coding with time-varying bit allocations to excitation and LPC parameters. 65-68 - Bishnu S. Atal, Richard V. Cox, Peter Kroon:
Spectral quantization and interpolation for CELP coders. 69-72 - Luca Cellario, Giuseppe Ferraris, Daniele Sereno:
A 2 ms delay CELP coder. 73-76 - Timothy Thorpe:
The mean squared error criterion: Its effect on the performance of speech coders. 77-80 - Erik McDermott, Shigeru Katagiri:
Shift-invariant, multi-category phoneme recognition using Kohonen's LVQ2. 81-84 - Gert-Jan Vernooij, Gerrit Bloothooft, Yvonne van Holsteijn:
A simulation study on the usefulness of broad phonetic classification in automatic speech recognition. 85-88 - Satoshi Nakamura, Kiyohiro Shikano:
Speaker adaptation applied to HMM and neural networks. 89-92 - Anna Maria Colla:
Automatic extraction of acoustic prototypes for large vocabulary speech recognition by using speaker-independent features. 93-96 - Li Deng, Patrick Kenny, Matthew Lennig, Vishwa Gupta, Paul Mermelstein:
A locus model of coarticulation in an HMM speech recognizer. 97-100 - Abdulmesih Aktas, Harald Höge:
Real-time recognition of subword units on a hybrid multi-DSP/ASIC based acoustic front-end. 101-103 - Kathy L. Brown, V. Ralph Algazi:
Characterization of spectral transitions with applications to acoustic sub-word segmentation and automatic speech recognition. 104-107 - Torbjørn Svendsen, Kuldip K. Paliwal, Erik Harborg, P. O. Husoy:
An improved sub-word based speech recognizer. 108-111 - Alex Waibel, Hidefumi Sawai, Kiyohiro Shikano:
Consonant recognition by modular construction of large phonemic time-delay neural networks. 112-115 - Anne-Marie Derouault, Bernard Mérialdo:
Improving speech recognition accuracy with contextual phonemes and MMI training. 116-119 - Alain Le Guyader, Dominique Massaloux, Jean-Pierre Petit:
Robust and fast code-excited linear predictive coding of speech signals. 120-123 - Redwan Salami, Lajos Hanzo, Derek G. Appleby:
A fully vector quantised self-excited vocoder. 124-127 - Ahmet M. Kondoz, K. Y. Lee, Barry G. Evans:
Improved quality CELP base-band coding of speech at low-bit rates. 128-131 - Jean E. Menez, Claude R. Galand, Michele M. Rosso, F. Bottau:
Adaptive code excited linear predictive coder (ACELPC). 132-135 - Hong Chae Woo, Jerry D. Gibson:
Multipulse-based codebooks for CELP coding at 7 kbps. 136-139 - H. Brehm, Manfred Herbert:
Lattice quantizers in speech coding. 140-143 - Jae H. Chung, Ronald W. Schafer:
A 4.8 Kbps homomorphic vocoder using analysis-by-synthesis excitation analysis. 144-147 - Masumi Akamine, Kimio Miseki:
ARMA model based speech coding at 8 kb/s. 148-151 - Martin Schultheiß, Arild Lacroix:
On the performance of CELP algorithms for low rate speech coding. 152-155 - Tomohiko Taniguchi, Shigeyuki Unagami, Robert M. Gray:
Multimode coding: application to CELP. 156-159 - Yair Shoham:
Cascaded likelihood vector coding of the LPC information. 160-163 - Yoshua Bengio, Régis Cardin, Piero Cosi, Renato De Mori:
Speech coding with multi-layer networks. 164-167 - Nariman Farvardin, Rajiv Laroia:
Efficient encoding of speech LSP parameters using the discrete cosine transformation. 168-171 - Baruch Mazor, C. Hudson, D. Borkowski:
Transform subbands coding with channel error control. 172-175 - Salvatore D. Morgera, Mohammad Reza Soleymani, Yves Normandin:
Combined source-channel coding. 176-179 - Robert E. Bogner, Tzuyin Li:
Pattern search prediction of speech. 180-183 - James L. Dixon, Vijay K. Varma, Nelson Sollenberger, David W. Lin:
Single DSP implementation of a 16 kbps sub-band speech coder for portable communications. 184-187 - Junji Suzuki, Naohisa Ohta:
Variable rate coding scheme for audio signal with subband and embedded coding techniques. 188-191 - Rosario Drogo de Jacovo, Roberto Montagna, Franco Perosino, Daniele Sereno:
Some experiments of 7 kHz audio coding at 16 kbit/s. 192-195 - Takehiro Moriya, Hirohito Suda:
An 8 kbit/s transform coder for noisy channels. 196-199 - David P. Kemp, Retha A. Sueda, Thomas E. Tremain:
An evaluation of 4800 bps voice coders. 200-203 - Y. J. Liu, Joseph Rothweiler:
A high quality speech coder at 400 bps. 204-206 - Thomas F. Quatieri, Robert J. McAulay:
Phase coherence in speech reconstruction for enhancement and coding applications. 207-210 - William M. Kushner, Vladimir Goncharoff, Chung Wu, Vien Nguyen, John N. Damoulakis:
The effects of subtractive-type speech enhancement/noise reduction algorithms on parameter estimation for improved recognition and coding in high noise environments. 211-214 - David M. Howard, Andrew P. Breen:
Methods for dynamic excitation control in parallel formant speech synthesis. 215-218 - Michael S. Scordilis, John N. Gowdy:
Neural network based generation of fundamental frequency contours. 219-222 - Rolf Carlson, Gunnar Fant, Christer Gobl, Björn Granström, Inger Karlsson, Qiguang Lin:
Voice source rules for text-to-speech synthesis. 223-226 - Mazin G. Rahim, Colin C. Goodyear:
Articulatory synthesis with the aid of a neural net. 227-230 - René Carré, Mohamad Mrayati:
New concept in acoustic-articulatory-phonetic relations-perspectives and applications. 231-234 - John Brian Pickering:
Modelling coarticulation. 235-237 - Christian Hamon, Eric Moulines, Francis Charpentier:
A diphone synthesis system based on time-domain prosodic modifications of speech. 238-241 - Hector R. Javkin, Kazue Hata, Lucio Mendes, Steven Pearson, Hisayo Ikuta, Abigail Kaun, Gregory DeHaan, Alan Jackson, Beatrix Zimmermann, Tracy Wise, Caroline Henton, Merrilyn Gow, Kenji Matsui, Noriyo Hara, Masaki Kitano, Der-Hwa Lin, Chun-Hong Lin:
A multi-lingual text-to-speech system. 242-245 - Hirohisa Iijima, Nobuhiro Miki, Nobuo Nagai:
Fundamental consideration of finite element method for the simulation of the vibration of vocal cords. 246-249 - Bert Van Coile:
The DEPES development system for text-to-speech synthesis. 250-253 - Jay G. Wilpon, Chin-Hui Lee, Lawrence R. Rabiner:
Application of hidden Markov models for recognition of a limited set of words in unconstrained speech. 254-257 - Dirk Van Compernolle:
Spectral estimation using a log-distance error criterion applied to speech recognition. 258-261 - Melvyn J. Hunt, Claude Lefèbvre:
A comparison of several acoustic representations for speech recognition with degraded and undegraded speech. 262-265 - John H. L. Hansen, Mark A. Clements:
Stress compensation and noise reduction algorithms for robust speech recognition. 266-269 - Gary E. Kopec, Marcia A. Bush:
An LPC-based spectral similarity measure for speech recognition in the presence of co-channel speech interference. 270-273 - Tomio Takara:
Isolated word recognition using continuous state transition-probability and DP-matching. 274-277 - Paul Cosgrove, J. Patrick Wilson, Roy D. Patterson:
Formant transition detection in isolated vowels with transitions in initial and final position. 278-281 - Y. Guedon, C. Cocozza-Thivent:
Use of the Derin's algorithm in hidden semi-Markov models for automatic speech recognition. 282-285 - Sadaoki Furui:
Unsupervised speaker adaptation method based on hierarchical spectral clustering. 286-289 - D. Hsu, John R. Deller Jr.:
On the use of HMMs to recognize cerebral palsy speech: isolated word case. 290-293 - Stephen J. Cox, John S. Bridle:
Unsupervised speaker adaptation by probabilistic spectrum fitting. 294-297 - Masafumi Nishimura:
HMM-based speech recognition using dynamic spectral feature. 298-301 - Ted H. Applebaum, Brian A. Hanson:
Enhancing the discrimination of speaker independent hidden Markov models with corrective training. 302-305 - X. Zhang, John S. Mason:
Improved training using semi-hidden Markov models in speech recognition. 306-309 - Fred Stentiford, Richard Hemmings:
A piecewise approach to connectionist networks for speech recognition. 310-313 - Piero Demichelis, L. Fissore, Pietro Laface, Giorgio Micca, E. Piccolo:
On the use of neural networks for speaker independent isolated word recognition. 314-317 - Magne Hallstein Johnsen:
A sub-word based speaker independent speech recognizer using a two-pass segmentation scheme. 318-321 - Shigeru Katagiri, Erik McDermott, Manami Yokota:
A new algorithm for representing acoustic feature dynamics. 322-325 - Yoshiharu Abe, Kunio Nakajima:
Speech recognition using dynamic transformation of phoneme templates depending on acoustic/phonetic environments. 326-329 - Ki Chui Kim, Hwang Soo Lee, Jung Wan Cho:
Phonetic recognition using peak weighted binary spectrum. 330-333 - Siyu Zhu, Jiankui Zhao, Cheng Fan:
Feature-based recognition of nonsonorant consonants in Chinese speech. 334-337 - Shinta Kimura, Hitoshi Iwamida, Toru Sanada:
Extraction and evaluation of phonetic-acoustic rules for continuous speech recognition. 338-341 - H. Hasan, J. M. Pardo, S. Alexandres, C. Casado:
Phonetic properties of a large Spanish lexicon and its implications for large vocabulary speech recognition. 342-344 - Lalit R. Bahl, P. S. Gopalakrishnan, Dimitri Kanevsky, David Nahamoo:
Matrix fast match: a fast method for identifying a short list of candidate words for decoding. 345-348 - Boneung Koo, Jerry D. Gibson, Steven D. Gray:
Filtering of colored noise for speech enhancement and coding. 349-352 - Yariv Ephraim, David Malah, Biing-Hwang Juang:
Speech enhancement based upon hidden Markov modeling. 353-356 - Charles H. Rogers, D. Chien, M. Featherstone, Kwang-Shik Min:
Neural network enhancement for a two speaker separation system. 357-360 - Marc A. Zissman, Clifford J. Weinstein, Louis D. Braida, Rosalie M. Uchanski, William M. Rabinowitz:
Speech-state-adaptive simulation of co-channel talker interference suppression. 361-364 - Hidefumi Kobatake, Katsuhisa Tawa, Akira Ishida:
Speech/nonspeech discrimination for speech recognition system under real life noise environments. 365-368 - D. K. Freeman, G. Cosier, C. B. Southcott, I. Boyd:
The voice activity detector for the Pan-European digital cellular mobile telephone service. 369-372 - V. Viswanathan, C. Henry:
Noise-immune multisensor speech input: formal subjective testing in operational conditions. 373-376 - Saeed Vaseghi, Peter J. W. Rayner:
The effects of non-stationary signal characteristics on the performance of adaptive audio restoration systems. 377-380 - Mitsuhiro Yuito, Naoki Matsuo:
A new sample-interpolation method for recovering missing speech samples in packet voice communications. 381-384 - Gérard Faucon, Saïd Tazi Mezalek, Régine Le Bouquin
:
Study and comparison of three structures for enhancement of noisy speech. 385-388 - Victor Zue, James R. Glass, Michael Philips, Stephanie Seneff:
Acoustic segmentation and phonetic classification in the SUMMIT system. 389-392 - Kaichiro Hatazaki, Yasuhiro Komori, Takeshi Kawabata, Kiyohiro Shikano:
Phoneme segmentation using spectrogram reading knowledge. 393-396 - Shigeki Sagayama:
Phoneme environment clustering for speech recognition. 397-400 - K. Frimpong-Ansah, David J. B. Pearce, Wendy J. Holmes, N. G. Dixon:
A stochastic/feature based recogniser and its training algorithm. 401-404 - Lawrence R. Rabiner, Chin-Hui Lee, Biing-Hwang Juang, Jay G. Wilpon:
HMM clustering for connected word recognition. 405-408 - Claude Montacié, Khalid Choukri, Gérard Chollet:
Speech recognition using temporal decomposition and multi-layer feed-forward automata. 409-412 - Steve Renals, Richard Rohwer:
Learning phoneme recognition using neural networks. 413-416 - T. D. Harrison, Frank Fallside:
A connectionist model for phoneme recognition in continuous speech. 417-420 - Joseph Picone:
On modeling duration in context in speech recognition. 421-424 - Michael A. Franzini, Michael J. Witbrock, Kai-Fu Lee:
A connectionist approach to continuous speech recognition. 425-428 - J. Mariani:
Recent advances in speech processing. 429-440 - Stephen E. Levinson, M. Y. Liberman, Andrej Ljolje, Laura G. Miller:
Speaker independent phonetic transcription of fluent speech for large vocabulary speech recognition. 441-444 - Kai-Fu Lee, Hsiao-Wuen Hon, Mei-Yuh Hwang, Sanjoy Mahajan, Raj Reddy:
The SPHINX speech recognition system. 445-448 - Douglas B. Paul:
The Lincoln robust continuous speech recognizer. 449-452 - L. Fissore, Pietro Laface, Giorgio Micca, Roberto Pieraccini:
A word hypothesizer for a large vocabulary continuous speech understanding system. 453-456 - Martin Brenner, Harald Höge, Erwin Marschall, Jorge Romano:
Word recognition in continuous speech using a phonological based two-network matching parser and a synthesis based prediction. 457-460 - Takeshi Kawabata, Kiyohiro Shikano:
Island-driven continuous speech recognizer using phone-based HMM word spotting. 461-464 - Lalit R. Bahl, Raimo Bakis, Jerome R. Bellegarda, Peter F. Brown, David Burshtein, Subrata K. Das, Peter V. de Souza, P. S. Gopalakrishnan, Frederick Jelinek, Dimitri Kanevsky, Robert L. Mercer, Arthur Nádas, David Nahamoo, Michael A. Picheny:
Large vocabulary natural language continuous speech recognition. 465-467 - V. Ralph Algazi, Sang Chung, Michael J. Ready, Kathy L. Brown:
Robust LPC analysis and synthesis using the KL transformation of acoustic subwords spectra. 468-471 - John D. Tardelli:
Intelligibility measurement of interrupted voice communication systems. 472-475 - Jean-Claude Junqua, Hisashi Wakita:
A comparative study of cepstral lifters and distance measures for all pole models of speech in noise. 476-479 - Hynek Hermansky, David J. Broad:
The effective second formant F2' and the vocal tract front-cavity. 480-483 - Hema A. Murthy, K. V. Madhu Murthy, B. Yegnanarayana:
Formant extraction from Fourier transform phase. 484-487 - Vladimir Goncharoff, Suresh Chandran:
Optimization of spectral warping parameters in a speech modification system. 488-491 - A. Lowry, M. C. Hall, P. M. Hughes:
Analysis and encoding of speech for a parallel formant synthesizer. 492-495 - Tatsuya Hirahara, Takashi Komakine:
A computational cochlear nonlinear preprocessing model with adaptive Q circuits. 496-499 - Benjamin Monderer, Aurel A. Lazar:
An ADPCM architecture based on models of the auditory system. 500-503 - Veton Z. Kepuska, John N. Gowdy:
Investigation of phonemic context in speech using self-organizing feature maps. 504-507 - Gérard Bailly, Pierre-François Marteau, Christian Abry:
A new algorithm for temporal decomposition of speech-application to a numerical model of coarticulation. 508-511 - I. Lecomte, Michel Lever, Jérôme Boudy, Alain Tassy:
Car noise processing for speech input. 512-515 - Hideki Noda:
On the use of the information on individual speaker's position in the parameter space for speaker recognition. 516-519 - L. Xu, John Oglesby, John S. Mason:
The optimization of perceptually-based features for speaker identification. 520-523 - Jayant M. Naik, Lorin Netsch, George R. Doddington:
Speaker verification over long distance telephone lines. 524-527 - Fred J. Goodman, Alvin F. Martin, Robert E. Wohlford:
Improved automatic language identification in noisy speech. 528-531 - L. Gillick, Stephen J. Cox:
Some statistical issues in the comparison of speech recognition algorithms. 532-535 - David S. Pallett:
Benchmark tests for DARPA resource management database performance evaluations. 536-539 - Herman J. M. Steeneken, Jeroen G. van Velden:
Objective and diagnostic assessment of (isolated) word recognizers. 540-543 - Trevor J. Thomas, Jeremy Peckham, Eleftherios D. Frangoulis:
A determination of the sensitivity of speech recognisers to speaker variability. 544-547 - Richard M. Schwartz, Owen Kimball, Francis Kubala, Ming-Whei Feng, Yen-Lu Chow, Chris Barry, John Makhoul:
Robust smoothing methods for discrete hidden Markov models. 548-551 - Dina Yashchin, Sara Basson, Niels Lauritzen, Suzi Levas, Annie Loring, Judith Spitz:
Performance of speech recognition devices: evaluating speech produced over the telephone network. 552-555 - George R. Doddington:
Phonetically sensitive discriminants for improved speech recognition. 556-559 - Hisao Kuwabara, Kazuya Takeda, Yoshinori Sagisaka, Shigeru Katagiri, S. Morikawa, T. Watanabe:
Construction of a large-scale Japanese speech database and its management system. 560-563 - Yan Ming Cheng, Douglas D. O'Shaughnessy:
Parameter sensitivity and robust estimation in an ARX model with glottal excitation. 564-567 - J. M. Turnbull, Andrew T. Sapeluk, Robert I. Damper:
A new method of pole-tracking with application to vowel and semivowel recognition. 568-571 - G. Duncan, B. Yegnanarayana, Hema A. Murthy:
A nonparametric method of formant estimation using group delay spectra. 572-575 - C. S. Wu, V. Nguyen, Vladimir Goncharoff, William M. Kushner, John Damoulakis:
Adaptive pitch detection algorithm for noisy signals. 576-579 - Joseph Picone, George R. Doddington:
A phonetic vocoder. 580-583 - Frank K. Soong:
A phonetically labeled acoustic segment (PLAS) approach to speech analysis-synthesis. 584-587 - Juergen Schroeter, M. Mohan Sondhi:
Dynamic programming search of articulatory codebooks. 588-591 - Masanobu Abe, Shin-Ichi Tamura, Hisao Kuwabara:
A new speech modification method by signal reconstruction. 592-595 - H. J. Coetzee, Thomas P. Barnwell III:
An LSP based speech quality measure. 596-599 - Dieter Huber:
A statistical approach to the segmentation and broad classification of continuous speech into phrase-sized information units. 600-603 - Katsuhiko Shirai, Noriyuki Aoki, Naoki Hosaka:
Multi-level clustering of acoustic features for phoneme recognition based on mutual information. 604-607 - James L. Hieronymus:
Explicit modeling of vowel coarticulation in continuous speech recognition. 608-611 - Ming-Whei Feng, Richard M. Schwartz, Francis Kubala, John Makhoul:
Iterative normalization for speaker-adaptive training in continuous speech recognition. 612-615 - Otto Schmidbauer:
Robust statistic modelling of systematic variabilities in continuous speech incorporating acoustic-articulatory relations. 616-619 - E. Chau, Y. K. Chung, Eleftherios D. Frangoulis, A. Lucas:
An enhanced training method for speech recognition in the VODIS project. 620-622 - Roberto Pieraccini, Aaron E. Rosenberg:
Automatic generation of phonetic units for continuous speech recognition. 623-626 - Jan Robin Rohlicek, William Russell, Salim Roukos, Herbert Gish:
Continuous hidden Markov modeling for speaker-independent word spotting. 627-630 - P. S. Gopalakrishnan, Dimitri Kanevsky, Arthur Nádas, David Nahamoo:
A generalization of the Baum algorithm to rational objective functions. 631-634 - John Makhoul, Richard M. Schwartz, Amro El-Jaroudi:
Classification capabilities of two-layer neural nets. 635-638 - X. D. Huang, Mervyn A. Jack:
Unified techniques for vector quantization and hidden Markov modeling using semi-continuous models. 639-642 - Héctor M. Rulot, Natividad Prieto, Enrique Vidal:
Learning accurate finite-state structural models of words through the ECGI algorithm. 643-646 - Peter Brauer, Petter Knagenhjelm:
Infrastructure in Kohonen maps. 647-650 - Michael M. Hochberg, Harvey F. Silverman, David P. Morgan:
A dynamic programming/neural network approach for connected-speech recognition. 651-654 - Mahesan Niranjan, Frank Fallside:
Temporal decomposition: a framework for enhanced speech recognition. 655-658 - Xavier L. Aubert:
Fast look-ahead pruning strategies in continuous speech recognition. 659-662 - Yasuo Ariki, Fergus R. McInnes, Mervyn A. Jack:
Hierarchical phoneme discrimination by hidden Markov modelling using cepstrum and formant information. 663-666 - S. C. Austin, Frank Fallside:
A unified syntax direction mechanism for automatic speech recognition systems using hidden Markov models. 667-670 - M. K. Nasri, Geneviève Caelen-Haumont, Jean Caelen:
Using prosodic rules in speech recognition expert system. 671-674 - Bill J. Stanton, Leah H. Jamieson, George D. Allen:
Robust recognition of loud and Lombard speech in the fighter cockpit environment. 675-678 - Andreas Noll, Hans-Hermann Hamer, H. Piotrowski, Hans-Wilhelm Rühl, Stefan Dobler, J. Weith:
Real-time connected-word recognition in a noisy environment. 679-681 - Chin-Hui Lee, Biing-Hwang Juang, Frank K. Soong, Lawrence R. Rabiner:
Word recognition using whole word and subword models. 683-686 - Mats Blomberg:
Synthetic phoneme prototypes in a connected-word speech recognition system. 687-690 - Miriam De Mattia, Egidio P. Giachin:
Experimental results on large vocabulary continuous speech understanding. 691-694 - Kenneth Ward Church:
A stochastic parts program and noun phrase parser for unrestricted text. 695-698 - Mitch Weintraub, Hy Murveit, Michael Cohen, Patti Price, Jared Bernstein, Gay Baldwin, Don Bell:
Linguistic constraints in hidden Markov model based speech recognition. 699-702 - Kenji Kita, Takeshi Kawabata, Hiroaki Saito:
HMM continuous speech recognition using predictive LR parsing. 703-706 - Marco Ferretti, Giulio Maltese, Stefano Scarci:
Language model and acoustic model information in probabilistic speech recognition. 707-710 - Stephanie Seneff:
TINA. A probabilistic syntactic parser for speech understanding systems. 711-714 - Douglas D. O'Shaughnessy:
Using syntactic information to improve large-vocabulary word recognition. 715-718 - Annedore Paeseler, Hermann Ney:
Continuous-speech recognition using a stochastic language model. 719-722 - Charles T. Hemphill, Joseph Picone:
Speech recognition in a unification grammar framework. 723-726 - Yen-Lu Chow, Salim Roukos:
Speech understanding using a unification grammar. 727-730 - Masami Nakamura, Kiyohiro Shikano:
A study of English word category prediction based on neutral networks. 731-734 - Joseph P. Campbell Jr., Vanoy C. Welch, Thomas E. Tremain:
An expandable error-protected 4800 bps CELP coder (US Federal Standard 4800 bps voice coder). 735-738 - Richard V. Cox, W. Bastiaan Kleijn, Peter Kroon:
Robust CELP coders for noisy backgrounds and noisy channels. 739-742 - Mei Yong, Allen Gersho:
Subband vector excitation coding with adaptive bit-allocation. 743-746 - Richard L. Zinser, Steven R. Koch:
4800 and 7200 bit/sec hybrid codebook multipulse coding. 747-750 - Frank H. Wu, Kalyan Ganesan:
Comparative study of algorithms for VQ design using conventional and neural-net based approaches. 751-754 - Yoshiaki Asakawa, Akira Ichikawa, Shun'ichi Yajima, Katsuya Yamasaki:
Speech coding method using fuzzy vector quantization. 755-758 - Jianjun Li, William A. Pearlman, Constantine N. Manikopoulos:
Fast vector quantization based on lattice prequantization. 759-761 - Luis A. Hernández Gómez, Carmen García-Mateo, Francisco Javier Casajús-Quirós:
Short-time synthesis procedures in vector adaptive transform coding of speech. 762-765 - Tore Fjällbrant, Fisseha Mekuria:
A hierarchical two-level analysis structure for use in speech coding and recognition. 766-769 - Anders Bergström, Per Hedelin:
High temporal resolution in multi-pulse coding. 770-773 - Siddhartha Chatterjee, Prathima Agrawal:
Connected speech recognition on a multiple processor pipeline. 774-777 - David B. Roe, Allen L. Gorin, Padma Ramesh:
Incorporating syntax into the level-building algorithm on a tree-structured parallel computer. 778-781 - Roberto Bisiani, Thomas S. Anantharaman, L. Butcher:
BEAM. An accelerator for speech recognition. 782-784 - Enzo Mumolo, Francesco Pazienti:
Large vocabulary isolated words recognizer on a cellular array processor. 785-788 - Hy Murveit, J. Mankoski, Jan M. Rabaey, Robert W. Brodersen, T. Stoelzle, D. Chen, Shankar Narayanaswamy, R. Yu, P. Schrupp, R. Schwartz, A. Santos:
A large-vocabulary real-time continuous-speech recognition system. 789-792 - Satoshi Miki, Kiyoshi Intoh:
Speaker-independent isolated-word recognition LSI. 793-796 - N. T. Condick, D. T. Chalmers:
A transputer based speech recognition system. 797-800 - Basavaraj I. Pawate, George R. Doddington:
Implementation of a hidden Markov model-based layered grammar recognizer. 801-804 - Clive D. Summerfield, Marwan A. Jabri:
Design and implementation of a formant speech synthesiser ASIC. 805-808 - K. Mervyn Curtis, P. Race, A. Abdul Aziz:
Parallelism and the transputer in the automatic translation of text to speech. 809-811 - Kai Öistämö, Petri Jarske, Jaakko Astola, Yrjö Neuvo:
Vector median filters for complex signal. 813-816 - Maureen P. Quirk:
Efficient computation of aliasing noise in down-sampling filters. 817-820 - Guido M. Cortelazzo, Simone Carmignato, Gian Antonio Mian:
A double-objective differential correction algorithm for IIR transfer function design. 821-824 - Gregory W. Medlin, John W. Adams:
A new design technique for maximally linear differentiators. 825-828 - Roy Chapman, M. A. Rahman:
A generalised design method for orthogonal IIR lattice filters. 829-832 - Max Gerken:
On lossless digital 2×2 transfer functions that can be implemented with allpass transfer functions. 833-836 - Marina Mondin, Ezio Biglieri:
Issues in the design of digital Nyquist filters. 837-840 - Gordon B. Scarth, Gert O. Martens:
Complex wave digital networks using complex port references. 841-844 - Ping Li, J. I. Sewell:
On low sensitivity/noise digital filter structures. 845-848 - Hassan M. Ahmed:
The generalized convergence computation method. 849-852 - Ashok K. Rao, Yih-Fang Huang:
Analysis of finite precision effects on an OBE algorithm. 853-856 - Nizar Bhalwani, Thomas R. Fischer, Michael W. Marcellin:
Hardware implementation of trellis coded quantization. 857-860 - Nevio Benvenuto, Michele Marchesi, Gianni Orlandi, Francesco Piazza, Aurelio Uncini:
Finite wordlength digital filter design using an annealing algorithm. 861-864 - David H. Horrocks, David R. Bull:
Response error estimates for FIR digital filters with floating point coefficients. 865-868 - Chintana Griffin, Preeti Rao, Fred J. Taylor:
Roundoff error analysis of the discrete Wigner distribution using fixed-point arithmetic. 869-871 - Paul H. Moose, Abdul Aziz Al-Bassiouni:
The quantized detection algorithm. 872-875 - Michiel J. Werter, John H. F. Ritzerfeld:
New zero-input overflow stability proofs based on Lyapunov theory. 876-879 - Tamal Bose:
Effects of two's complement truncation quantization in state-space digital filters. 880-883 - Shigeyoshi Kawarai, Toru Murakami:
An optimization procedure to minimize the roundoff noise in cascade floating-point digital filters. 884-887 - Raimund Meyer:
Error analysis and comparison of FFT implementation structures. 888-891 - Akihiko Sugiyama, M. N. S. Swamy, Eugene I. Plotkin:
A fast convergence algorithm for adaptive FIR filters. 892-895 - Neil J. Bershad, Odile Macchi:
Comparison of RLS and LMS algorithms for tracking a chirped signal. 896-899 - John D. DiLullo, S. S. Rao:
A real-time algorithm for separating locally-stationary random processes in the presence of noise. 900-903 - Angelos P. Varvitsiotis, Sergios Theodoridis, George V. Moustakides:
A novel structure for adaptive LS FIR filtering based on QR decomposition. 904-907 - Gregory E. Bottomley, S. Thomas Alexander:
A theoretical basis for the divergence of conventional recursive least squares filters. 908-911 - Carlos E. Davila:
A recursive least-squares algorithm with data-adaptive step size. 912-915 - Irene Y. H. Gu:
RLS lattice and circular lattice with real time variable sliding-window length. 916-919 - Adetokunbo O. Ogunfunmi, Allen M. Peterson:
Fast direct implementation of block adaptive FIR filtering. 920-923 - Jia-Sien Soo, Khee K. Pang:
A multi-step size (MSS) frequency domain adaptive filter for stationary and nonstationary signals. 924-927 - V. S. Somayazulu, Sanjit K. Mitra, John J. Shynk:
Adaptive line enhancement using multirate techniques. 928-931 - Jeffrey C. Strait, W. Kenneth Jenkins:
Filter architectures and adaptive algorithms for 2-D adaptive digital signal processing. 932-935 - Dacheng Yang, Dejung Wang:
Realization of the multichannel and high-order IIR adaptive filters. 936-939 - Piet C. W. Sommen, C. J. van Valburg:
Efficient realisation of adaptive filter using an orthogonal projection method. 940-943 - Steve McLaughlin, Bernard Mulgrew, Colin F. N. Cowan:
A novel adaptive equaliser for nonstationary communication channels. 944-947 - Richard A. Games, Daniel Moulin, Sean D. O'Neil, Joseph J. Rushanan:
Algebraic-integer quantization and residue number system processing. 948-951 - Sangil Park, Garth Hillman:
On acoustic-echo cancellation implementation with multiple cascadable adaptive FIR filter chips. 952-955 - Jin-Yun Zhang, Willem Steenaart:
Realization and implementation of adaptive state-space recursive filters. 956-959 - John R. Treichler, Sally L. Wood, Michael G. Larimore:
Convergence rate limitations in certain frequency-domain adaptive filters. 960-963 - Kenneth W. Martin:
Power normalized update algorithm for adaptive filters-without divisions. 964-967 - Michael J. Rude, Lloyd J. Griffiths:
Incorporation of linear constraints into the constant modulus algorithm. 968-971 - William A. Sethares, Gonzalo A. Rey, C. Richard Johnson Jr.:
Approaches to blind equalization of signals with multiple modulus. 972-975 - Kunio Oh'ishi, Hajime Kubota, Mahoki Onoda:
A new block adaptation algorithm realizing rapid identification system. 976-979 - Hua Ye, Bo-Xiu Wu:
A new family of adaptive gradient lattice algorithms based on LMK criterion. 980-983 - Panos E. Papamichalis, C. Sidney Burrus:
Conversion of digit-reversed to bit-reversed order in FFT algorithms. 984-987 - Christoph Loeffler, Adriaan Ligtenberg, George S. Moschytz:
Practical fast 1-D DCT algorithms with 11 multiplications. 988-991 - Chao Lu, Richard Tolimieri:
Extension of Winograd multiplicative algorithm to transform size N=p2q and its implementation. 992-995 - Anshu J. Gupta, K. R. Rao:
Efficient FFT algorithm based on the DST. 996-998 - Martin Vetterli, Pierre Duhamel, Christine Guillemot:
Trade-off's in the computation of mono- and multi-dimensional DCT's. 999-1002 - Tien T. Wang:
Segmented chirp Z-transform and its applications. 1003-1006 - Yoshiaki Tadokoro, Kenji Nakamura:
A new discrete Fourier transform for unevenly sampled data. 1007-1010 - Gloria Faye Boudreaux-Bartels, Donald W. Tufts, P. Dhir, G. Sadasiv, G. Fischer:
Analysis of errors in the computation of Fourier coefficients using the arithmetic Fourier transform (AFT) and summation by parts (SBP). 1011-1014 - Christopher J. Read, Douglas M. Chabries, Richard W. Christiansen, J. Kelly Flanagan:
A method for computing the DFT of vector quantized data. 1015-1018 - Masahichi Kishi:
The properties and configuration of the short time DFT Hilbert transformers. 1019-1022 - Rong Peng, Bhaskar D. Rao:
A differential equation approach for the analysis of the adaptive lattice filter. 1023-1026 - Serafim Karaboyas, Nicholas Kalouptsidis:
Efficient adaptive transversal algorithms for least squares ARMA identification. 1027-1030 - Ahmed Benallal, André Gilloire:
Improvement of the tracking capability of the numerically stable fast RLS algorithms for adaptive filtering. 1031-1034 - Björn Gudmundson, Svante Signell:
Adaptive algorithms based on exact gradients. 1035-1038 - Dirk T. M. Slock, Luigi Chisci, Hanoch Lev-Ari, Thomas Kailath:
Modular and numerically stable multichannel FTF algorithms. 1039-1042 - Odile Macchi, Nacer K. M'Sirdi, Christine Uhl:
Stability of adaptive IIR predictors with nonstationary inputs. 1043-1046 - Majid Nayeri:
Uniqueness of MSOE estimates in IIR adaptive filtering; a search for necessary conditions. 1047-1050 - Kun Tang, Chongxi Feng:
Using auxiliary signal to remove the SPR condition from recursive adaptive algorithms. 1051-1054 - Mustafa Koparoglu, Yalçin Tanik, Mehmet Ali Tugay:
On the frequency domain LMS adaptive line enhancer. 1055-1058 - Toshihiro Furukawa, Hajime Kubota, Shigeo Tsujii:
The orthogonal projection algorithm for block adaptive signal processing. 1059-1062 - P. A. Ramamoorthy, Brahmaji Potu:
High-speed ADC using residue number system. 1063-1066 - John R. Deller Jr., Souheil F. Odeh:
Implementing the optimal bounding ellipsoid algorithm on a fast processor. 1067-1070 - Y. Z. Zhang, Y. F. Yao:
Fast implementation of recursive DFTs. 1071-1074 - Mike Griffin, Mike Sousa, Fred J. Taylor:
Efficient scaling in the residue number system. 1075-1078 - Thanos Stouraitis:
A hybrid floating-point/logarithmic number system digital signal processor. 1079-1082 - John E. Shore:
An extensible file system for signal processing software. 1083-1086 - Messaoud Benidir, Bernard Picinbono:
Routh's like array for the zero location of continuous- and discrete-time filters. 1087-1090 - Kar-Lik Wong, Wan-Chi Siu:
Fast address generation for the computation of prime factor algorithms. 1091-1094 - G. Robert Redinbo:
Data protection in convolution computations. 1095-1098 - Anne C. Elster:
Fast bit-reversal algorithms. 1099-1102 - Joseph E. Peters, Stanley M. Dunn:
A compiler that easily retargets high level language programs for different signal processing architectures. 1103-1106 - Yu Hen Hu:
Parallel eigenvalue decomposition for Toeplitz and related matrices. 1107-1110 - J. P. Schwartz, D. Degryse, P. A. Comte, Dominique Vicard:
OPAL: a high level language and environment for DSP boards on PC. 1111-1114 - Hui-Min Zhang, Pierre Duhamel:
Doubling Levinson/Schur algorithm and its implementation. 1115-1118 - Kazuo Toraichi, Mamoru Iwaki, Masaru Kamada:
Biorthonormal expansion in signal spaces composed of spline functions. 1119-1122 - Cédric Demeure, Louis L. Scharf:
Fast algorithms to QR factor circulant matrices. 1123-1126 - Robert Michael Owens, Mary Jane Irwin:
Implementing algorithms for convolution on arrays of adders. 1127-1130 - Yanfang Sun:
An order recursive algorithm for synthesizing linear recursive filters. 1131-1133 - Steven L. Gay, John Hartung, Geoffrey L. Smith:
Algorithms for multi-channel DTMF detection for the WE DSP32 family. 1134-1137 - D. Bergmann, D. Boillon, F. Bonifacio, R. Breitschädel:
Experimental speech input/output system. 1138-1141 - Nam Ling, Magdy A. Bayoumi:
The design and implementation of multidimensional systolic arrays for DSP applications. 1142-1145 - Gagan Mirchandani, Peter A. Twombly:
A software development tool for scheduling signal processing algorithms on multiprocessors with arbitrary interconnectivity. 1146-1149 - Matti Karjalainen:
A Lisp-based high-level programming environment for the TMS320C30. 1150-1153 - Min Xu, Yves Grenier:
Time-frequency domain adaptive filters. 1154-1157 - David Starer, Arye Nehorai:
Polynomial factorization algorithms for adaptive root estimation. 1158-1161 - Jar-Fen Yang, Mostafa Kaveh:
Adaptive algorithms for tracking roots of spectral polynomials. 1162-1165 - Patrick Duvaut:
A unifying and general approach to adaptive linear-quadratic discrete time Volterra filtering. 1166-1170 - P. Bragard, Geneviève Jourdain:
Adaptive equalization for underwater data transmission. 1171-1174 - Dimitrios Hatzinakos, Chrysostomos L. Nikias:
Adaptive filtering based on polycepstra. 1175-1178 - Francesco Palmieri:
A backpropagation algorithm for multilayer hybrid order statistic filters. 1179-1182 - Galvin J. Gibson, Sammy Siu, Colin F. N. Cowan:
Multilayer perceptron structures applied to adaptive equalisers for data communications. 1183-1186 - Sharad Singhal, Lance Wu:
Training feed-forward networks with the extended Kalman algorithm. 1187-1190 - Peter J. W. Rayner, Michael R. Lynch:
A new connectionist model based on a non-linear adaptive filter. 1191-1194 - John L. Brown Jr.:
Generalized sampling and the perfect reconstruction problem for maximally decimated filter banks. 1195-1198 - Adewole O. Osinubi, Robert A. King:
Naturalness-preserving transform (NPT) reconstruction of signals degraded by non-stationary noise processes. 1199-1202 - Hua Lee, Douglas P. Sullivan:
Discrete bandlimited signal extrapolation by data segmentation. 1203-1206 - Frederick L. Kitson:
An algorithm for curve and surface fitting using B-splines. 1207-1210 - Neil K. Jablon:
Carrier recovery for blind equalization. 1211-1214 - Yunbiao Wang, Hari Krishna, Bal Krishna:
Split Levinson algorithm is weakly stable. 1215-1218 - Anthony D. Fagan, Paul F. Curran, Brian P. Murray, Michael McLaughlin:
Implementation of a full V32 modem using a single DSP chip. 1219-1222 - Yutaka Miyake, Masafumi Hagiwara, Masao Nakagawa:
A new timing extraction method and data interpolation for block demodulation. 1223-1226 - Rodolfo Mann, Karl-Dirk Kammeyer:
A pole-zero-tracking constant modulus algorithm. 1227-1230 - Fathy F. Yassa, Barbara A. Thompson:
Adaptive phase-insensitive amplitude demodulator for noncoherent DSBSC signals. 1231-1234 - Risto Wichman, Jaakko Astola, Yrjö Neuvo:
The in-place growing median concept for filtering signals with noisy edges. 1235-1238 - Zhongnong Jiang:
FIR filter design and implementation with powers-of-two coefficients. 1239-1242 - Masaru Kamada, Hitoshi Sakai, Kazuo Toraichi:
A quadratic spline function generator based on B-spline functions. 1243-1246 - Ravi Prakash Ramachandran, Peter Kabal:
Minimax design of factorable Nyquist filters. 1247-1250 - B. Khaitan, R. Blasco, C. N. Patel, K. Chan, S. Chen, T. Chiang:
Block digital signal processor balances performance and complexity. 1251-1254 - Uday B. Desai:
A state-space approach to orthogonal digital filters for VLSI implementation. 1255-1258 - Yukio Kadowaki, Shigeki Matsuoka, Shogo Nakamura:
LSI implementation of a programmable FIR digital filter. 1259-1262 - Michael H. Wolf, Dietmar Vogel:
DSP method to receive spread spectrum signals. 1263-1266 - Adam Dabrowski:
Transmission of effective pseudopower in multirate signal processing. 1267-1270 - Paul C. Millar:
The delay paradox when combining the outputs of non-quadrature filter networks and its importance to subband coding schemes. 1271-1274 - Sohail A. Dianat, M. R. Raghuveer:
Bispectrum phase transformation for non-minimum phase signal reconstruction. 1275-1277 - Adam W. Bojanczyk, Allan O. Steinhardt:
Stabilized hyperbolic Householder transformations. 1278-1281 - Anna Z. Baraniecki:
Fast computation of the discrete Hartley transform. 1282-1285 - F. J. Harris, W. H. McKnight, F. M. Tirpak Jr., Harper J. Whitehouse:
Implementation considerations and limitations for dynamic range enhanced analog to digital converters. 1286-1289 - Fuyun Ling:
Efficient least-squares lattice algorithms based on Givens rotation with systolic array implementations. 1290-1293 - Claude Guéguen, François Desbouvries:
Preservation of displacement ranks and the numerical stability of time recursive fast algorithms. 1294-1297 - Jürgen Götze, Uwe Schwiegelshohn:
An orthogonal method for solving systems of linear equations without square roots and with few divisions. 1298-1301 - Kevin L. Kloker, Brett Lindsley, Natan Baron, Guy R. L. Sohie:
Efficient FFT implementation on an IEEE floating-point digital signal processor. 1302-1305 - Andrew E. Yagle:
Fast algorithms for Nevanlinna-Pick interpolation and H∞ optimization. 1306-1309 - Alexander Skavantzos, Zarir B. Sarkari, Thanos Stouraitis:
A complex DSP processor using polynomial encoding. 1310-1313 - Thorsten Selinger, Maati Talmi:
Signal processor SIPRO23 with minimized overhead. 1314-1317 - Alfredo Restrepo Palacios, Alan C. Bovik:
Locally monotonic regression. 1318-1321 - A. Farassopoulos:
Speech enhancement for hearing aids using adaptive beamformers. 1322-1325 - Cheung Auyeung, Russell M. Mersereau:
A dual approach to signal restoration. 1326-1329 - A. Enis Çetin:
An algorithm for signal reconstruction from bispectrum. 1330-1333 - Reinaldo A. Valenzuela, C. N. Animalu:
A new voice-packet reconstruction technique. 1334-1336 - Truong Q. Nguyen, P. P. Vaidyanathan:
Lattice structures for design of three-channel linear-phase perfect-reconstruction FIR QMF banks. 1337-1340 - Tom D. Lookabaugh, Michael G. Perkins, Christie L. Cadwell:
Analysis/synthesis systems in the presence of quantization. 1341-1344 - Chein-I Chang, Lee D. Davisson:
A counterpart of Remez's algorithms in statistical decision theory: Chang-Davisson's algorithms. 1345-1348 - X. Li, Nihat M. Bilgutay, Jafar Saniie:
Frequency diverse statistic filtering for clutter suppression. 1349-1352 - Garth Hillman, J. E. Lane:
Real-time determination of IIR coefficients for cascaded Butterworth filters. 1353-1356 - D. R. Bungard, L. Lau, T. L. Rorabaugh:
Programmable FFT processors for wide-bandwidth HF spread-spectrum communications and radar signal processing. 1357-1359 - Robert C. DiPietro:
An FFT based technique for suppressing narrow-band interference in PN spread spectrum communications systems. 1360-1363 - Ephraim Feig, Fred Mintzer, Arthur Nádas:
Digital implementation of frequency division multiplexing on peak-limited channels. 1364-1367 - Ryuji Kohno, Hideki Imai, Subbarayan Pasupathy:
An automatic equalizer including a Viterbi decoder for trellis coded modulation system. 1368-1371 - Boaz Porat, Benjamin Friedlander:
Blind adaptive equalization of digital communication channels using high-order moments. 1372-1375 - Jiande Chen, Joos Vandewalle:
Study of adaptive nonlinear echo canceller with Volterra expansion. 1376-1379 - Tyseer Aboulnasr, A. Abousaada, Willem J. D. Steenaart:
Efficient implementation of echo cancellers for ISDN applications. 1380-1383 - A. M. Alvarez:
Echo canceller design and implementation of a CCITT V32 modem: software solution. 1384-1387 - G. T. Davis, B. D. Mandalia:
Pseudo-coherent phase shift keyed demodulator. 1388-1391 - Richard P. Gooch, Brian J. Sublett:
Demodulation of cochannel QAM signals. 1392-1395 - Reginald L. Lagendijk, Jan Biemond, Dick E. Boekee:
Blur identification using the expectation-maximization algorithm. 1397-1400 - Piero Zamperoni:
Variations on the rank-order filtering theme for grey-tone and binary image enhancement. 1401-1404 - A. G. Qureshi:
Constrained Kalman filtering for image restoration. 1405-1408 - Simon Clippingdale, Roland G. Wilson:
Least-squares image estimation on a multiresolution pyramid. 1409-1412 - Lang Hong, Dragana Brzakovic:
Bayesian restoration of image sequences using 3-D Markov random fields. 1413-1416 - Tal Simchony, Rama Chellappa, Ze'ev Lichtenstein:
Graduated nonconvexity algorithm for image estimation using compound Gauss Markov field models. 1417-1420 - Yunzin Zhao, Lars S. Andersen, Les E. Atlas:
Parameter estimation and restoration of noisy images using Gibbs distributions in hidden Markov models. 1421-1424 - Guanrong Chen, Rui J. P. de Figueiredo:
Optimal image reconstruction based on general PDE models. 1425-1428 - M. Ibrahim Sezan, H. Joel Trussell:
Use of a priori knowledge in multispectral image restoration. 1429-1432 - C. R. Moloney, M. E. Jernigan:
Nonlinear adaptive restoration of images with multiplicative noise. 1433-1436 - Ioannis Pitas, Costas Kotropoulos:
Texture analysis and segmentation of seismic images. 1437-1440 - Jim Schroeder, John Endsley:
Lp normed spectral estimation residual analysis for sonic well logging. 1441-1444 - Olutayo Ibikunle:
Projection allocation for image reconstruction. 1445-1448 - Michael R. Smith, S. T. Nichols:
The use of modeling as an alternative magnetic resonance image technique. 1449-1452 - Orhan Arikan, David C. Munson Jr.:
Analysis and simulation of a new algorithm for spotlight-mode synthetic aperture radar. 1453-1455 - Stephen G. Azevedo, James M. Brase, Harry E. Martz, Anil K. Jain, K. Wayne Current, Paul J. Hurst:
A Radon transform computer for multidimensional signal processing. 1457-1459 - Nicolas J. Dusaussoy, Ikram E. Abdou:
The extended MENT algorithm: a promising reconstruction algorithm for computerized tomography. 1460-1463 - Lingxiong Shao, Alfred O. Hero III:
Information optimization of projective tomographic imaging systems. 1464-1467 - Jerry L. Prince, Alan S. Willsky:
A hierarchical algorithm for limited-angle reconstruction. 1468-1471 - Yoram Bresler, Carl J. Skrabacz:
Optimal interpolation in helical scan 3D computerized tomography. 1472-1475 - Ken D. Sauer, Bede Liu:
Nonstationary filtering for removal of signal dependent noise in reconstructions from projections. 1476-1478 - Barry J. Sullivan:
Reconstruction of projection images from CCI holograms. 1480-1483 - Peter L. Chu:
A multistage projection structure for multidimensional signal detection. 1484-1487 - Eitan Yudilevich, Henry Stark:
Reconstruction from partial data in multislice magnetic resonance imaging. 1488-1491 - Nak H. Kim, Alan C. Bovik:
Computing 3-D symbolic representation of blood vessels from stereo-microscopic images. 1492-1495 - N. Saeed, Tariq S. Durrani:
A new MRI rotation algorithm for the registration of temporal images. 1496-1499 - Kazuo Toraichi, Satomi Ishiuchi, Mamoru Iwaki:
A method of improving image quality for medical video hardcopy. 1500-1503 - R. de Beer, Dirk van Ormondt, W. W. F. Pijnappel:
Maximum likelihood estimation of poles, amplitudes and phases from 2-D NMR time domain signals. 1504-1507 - Richard Y. Chiao, Hua Lee, Glen Wade:
Image restoration and wave-field error removal in holographic acoustic microscopy. 1508-1511 - Paul S. Lewis:
Adaptive enhancement of magnetoencephalographic signals via multichannel filtering. 1512-1515 - Meir Feder, Jules S. Jaffe:
Limited-angle reconstruction from noisy data using clustering of the solution space. 1516-1519 - Hong Liang, Zhenming Chai, Xiaoying Chen:
The blind pattern recognizing in neural spike train. 1520-1523 - R. M. S. S. Abeysekera, Boualem Boashash:
Time-frequency domain features of ECG signals: their application in P wave detection using the cross Wigner-Ville distribution. 1524-1527 - M. J. Paterson, W. A. Cormack, J. T. Herd, S. M. Beard:
An astronomical imaging application using transputers. 1528-1531 - Mark B. Sandler, L. Hayat, L. Costa, A. A. Naqvi:
A comparative evaluation of DSPs, microprocessors and the transputer for image processing. 1532-1535 - Susan M. Miller, Harvey F. Silverman:
Optimized implementation of the 2-D DFT on loosely-coupled parallel systems. 1536-1539 - Sverre Holm, Arve Måøy:
Synthetic aperture radar processing facility based on a parallel supercomputer. 1540-1543 - Kenneth Adamson, G. Donnan, Norman D. Black:
Biomedical imaging using transputers and the IMSG170 Colour Palette. 1544-1547 - Graham Hall, Trevor J. Terrell, John M. Senior, Lesley M. Murphy:
Transputer implementation of the Radon transform for image enhancement. 1548-1551 - Joseph M. Francos, A. Zvi Meiri:
A 2-D autoregressive, finite support, causal model for texture analysis and synthesis. 1552-1555 - Hani Muammar, Mark Nixon:
Approaches to extending the Hough transform. 1556-1559 - Peter Aschwanden:
Methods for real-time tracking of uncooperative targets. 1560-1563 - Leonard M. Napolitano Jr., David D. Andaleon, K. R. Berry, P. R. Bryson, S. R. Klapp, J. E. Leeper, G. Robert Redinbo:
A special-purpose computer for automatic target recognition. 1564-1567 - Petros Maragos:
Morphological correlation and mean absolute error criteria. 1568-1571 - Xiaoning Nie, Rolf Unbehauen:
2-D IIR filter design using the extended McClellan transformation. 1572-1574 - James V. Krogmeier, Kaxlamangla S. Arun:
Comparison of causal and non-causal linear prediction models for multidimensional spectrum estimation. 1575-1578 - Kari-Pekka Estola:
Variable kernel functions for efficient event detection and classification. 1579-1582 - Samad Moini, Jenö Gazdag:
Use of frequency domain analysis for signal-to-noise ratio enhancement in stacking. 1583-1585 - C. Srinivas, M. D. Srinath:
Compound Gauss Markov random field model for image segmentation and restoration. 1586-1589 - Richard A. Haddad, Bruce G. Nichol:
Efficient filtering of images using binomial sequences. 1590-1593 - Reiner Lenz:
A group theoretical approach to filter design. 1594-1597 - Mahmood R. Azimi-Sadjadi, Kash Khorasani:
Model reduction for two-dimensional systems. 1598-1601 - John P. Oakley, Michael J. Cunningham:
A formula for least-squares projection and its application in image reconstruction. 1602-1605 - Christian Viard-Gaudin, Dominique Barba:
Control of postal parcels separation by artificial vision. 1606-1609 - Harold G. Longbotham, Alan C. Bovik, Alfredo Restrepo Palacios:
Generalized order statistic filters. 1610-1613 - Qinfen Zheng, Rama Chellappa:
Estimation of surface topography from stereo SAR images. 1614-1617 - John W. Betz, Robert W. Pinto, Jerry L. Prince:
A model-based vision system for object recognition with synthetic aperture radar data. 1618-1621 - Langford B. White:
Resolution enhancement in time-frequency signal processing using inverse methods. 1622-1625 - Roberto Cusani, Giovanni Jacovitti:
A double tomographic approach to the estimation and classification of single objects. 1626-1629 - Guo-Qing Wei, Zhenya He, Song De Ma:
Camera calibration by vanishing point and cross ratio. 1630-1633 - G. R. Halsall, D. R. Burton, M. J. Lalor, C. Allan Hobson:
A novel real-time opto-electronic profilometer using FFT processing. 1634-1637 - K. H. Wong, Hudson H. M. Law, Peter Wai-Ming Tsang:
A system for recognising human faces. 1638-1642 - Nanda Gopal, Tomas Emmoth, Alan C. Bovik:
Channel interactions in visible pattern analysis. 1643-1646 - Fabrice Heitz, Henri Maître, Marc Bernard, Charles de Couessin:
Event detection in multisource imaging using contextual estimation. 1647-1650 - Patrick Bouthemy, Patrick Lalande:
Motion detection in an image sequence using Gibbs distributions. 1651-1654 - Tim J. Chown, Paul H. Lewis:
Image analysis by enhanced facet modelling. 1655-1658 - Maha Kadirkamanathan, Peter J. W. Rayner:
A unified approach to on-line cursive script segmentation and feature extraction. 1659-1662 - Wei Qian, Min Shan Lei:
The segmentation of 3-D image and space Markov cubic mesh models. 1663-1666 - Thrasyvoulos N. Pappas, Nikil S. Jayant:
An adaptive clustering algorithm for image segmentation. 1667-1670 - Takahiro Saito, Hideaki Eguchi, Ryuji Abe, Takashi Komatsu, Hiroshi Harashima:
Self-organizing pattern-matching coding for picture signals. 1671-1674 - Fure-Ching Jeng, John W. Woods:
Texture discrimination using doubly stochastic Gaussian random fields. 1675-1678 - Zhigang Fan:
An edge-based hierarchical algorithm for textured image segmentation. 1679-1682 - T. N. Tan, Anthony G. Constantinides:
Texture feature extraction based on primitive analysis. 1683-1686 - Luc de Vos, M. Stegherr, Tobias G. Noll:
VLSI architectures for the full-search blockmatching algorithm. 1687-1690 - Yunzin Zhao, Robert M. Haralick:
Binary shape recognition based on an automatic morphological shape decomposition. 1691-1694 - Ziheng Zhou, Anastasios N. Venetsanopoulos:
Pseudo-Euclidean morphological skeleton transform for machine vision. 1695-1698 - Ian T. Young:
Modern digital image analysis. 1699-1702 - Til Aach, Uwe Franke, Rudolf Mester:
Top-down image segmentation using object detection and contour relaxation. 1703-1706 - Jean-Yves Fouques, Paul Cohen:
Partition filters: a new class of morphological operators for segmenting textured images. 1707-1710 - Hiroshi Kaneko:
A generalized fractal dimension and its application to texture analysis-fractal matrix model. 1711-1714 - Kazuo Toraichi, Tadahiko Kumamoto, Iwao Sekita, Kazuhiko Yamamoto, Hiromitsu Yamada:
Feature analysis of handprinted Chinese characters for a recognition system. 1715-1718 - John A. Vlontzos, Sun-Yuan Kung:
Hidden Markov models for character recognition. 1719-1722 - Vito Cappellini, Alberto Del Bimbo, Alessandro Mecocci:
An object oriented approach for object recognition and classification. 1723-1726 - J. M. Jansen, F. W. Sijstermans:
Template matching with a MIMD computer. 1727-1730 - Robert A. Cohen, John W. Woods:
Sliding block entropy coding of images. 1731-1734 - Eve A. Riskin, Elizabeth M. Daly, Robert M. Gray:
Pruned tree-structured vector quantization in image coding. 1735-1738 - Nasser M. Nasrabadi, Shihkuan E. Lin, Yushu Feng:
Interframe hierarchical vector quantization. 1739-1742 - Dae-Gwon Jeong, Jerry D. Gibson:
Lattice vector quantization for image coding. 1743-1746 - V. John Mathews, Mehrdji Khorchidian:
Multiplication-free vector quantization using L1 distortion measure and its variants. 1745-1750 - Shuitsu Matsumura:
Upper bits separation coding of images using vector quantization. 1751-1754 - Yushu Feng, Nasser M. Nasrabadi:
A dynamic address-vector quantization algorithm based on inter-block and inter-color correlation for color image coding. 1755-1758 - J. Kelly Flanagan, Darryl Morrell, Richard L. Frost, Christopher J. Read, Brent E. Nelson:
Vector quantization codebook generation using simulated annealing. 1759-1762 - Thomas G. Stockham Jr., Zhenhua Xie:
Optimal previsualized image vector quantization. 1763-1766 - A. Madisetti, R. Subramonian, V. Ralph Algazi:
A radius-bucketing approach to fast vector quantization encoding. 1767-1770 - M. Ibrahim Sezan, Henry Stark, Shu-Jen Yeh:
A signal processing perspective to the operational characteristics of perceptron and Hopfield associative memory neural networks. 1771-1774 - Okan K. Ersoy, Daesik Hong:
Neural network learning paradigms involving nonlinear spectral processing. 1775-1778 - Badrinath Roysam, Michael I. Miller:
A unified approach for hierarchical imaging based on joint hypothesis testing and parameter estimation. 1779-1782 - Doug Young Suh, Russell M. Mersereau, Robert L. Eisner, Roderic I. Pettigrew:
Knowledge-based boundary detection applied to cardiac magnetic resonance image sequences. 1783-1786 - Stefanos D. Kollias, Tu-Chih Tsai, Dimitris Anastassiou:
Image halftoning and reconstruction using a neural network. 1787-1790 - Ann Marie Aull, Robert A. Gabel:
Machine intelligence applied to radar image understanding. 1791-1794 - Shigeo Morishima, Kiyoharu Aizawa, Hiroshi Harashima:
An intelligent facial image coding driven by speech and phoneme. 1795-1798 - Daniel J. Tomich, Edmund A. Quincy, Dara Parsavand:
Expert pattern recognition assessment of image quality. 1799-1802 - Vito Roberto, Claudio Chiaruttini:
A knowledge-based system for seismological signal understanding. 1803-1806 - H. B. Zhou, B. Z. Yuan:
Knowledge based parallel recognition of handwritten alphanumerics. 1807-1810 - Peter H. Westerink, Jan Biemond, Dick E. Boekee:
Progressive transmission of images using subband coding. 1811-1814 - Michael G. Perkins, Tom D. Lookabaugh:
A psychophysically justified bit allocation algorithm for subband image coding systems. 1815-1818 - Hamid Gharavi:
Differential sub-band coding of video signals. 1819-1821 - Haibo Li, Zhenya He:
Directional subband coding of images. 1823-1826 - Kofi Mensa-Ababio:
Improved transform coding [imaging coding]. 1827-1830 - Jong Won Kim, Sang Uk Lee:
Discrete cosine transform-classified VQ technique for image coding. 1831-1834 - Wen-Jun Zhang, Song-Yu Yu, Hong-Bin Chen:
A new adaptive classified transform coding method [image coding]. 1835-1837 - Xiancheng Yuan, Vinay K. Ingle:
An image coding algorithm based on a class of doubly stochastic image models. 1838-1841 - Atilla Baskurt, Robert Goutte:
Encoding the location of spectral coefficients using quadtrees in transform image compression. 1842-1845 - V. Ralph Algazi, Glyn Ford, E. Hildum:
Digital representation and storage of high quality color images by anisotropic enhancement and subsampling. 1846-1849 - Jonathan W. Brandt, Anil K. Jain:
A medical axis transform algorithm for compression and vectorization of document images. 1850-1853 - Cheng-Tie Chen:
Adaptive transform coding via quadtree-based variable blocksize DCT. 1854-1857 - K. S. Thyagarajan, Harry Sanchez:
Image sequence coding using interframe VDPCM and motion compensation. 1858-1861 - Ali N. Akansu, Manjunatha S. Kadur:
Discrete Walsh-Hadamard transform vector quantization for motion-compensated frame difference signal coding. 1862-1865 - M. W. Whybray, E. Hanna:
A DSP based videophone for the hearing impaired using valledge processed pictures. 1866-1869 - Sudhir S. Dixit, Yushu Feng:
Adaptive vector quantization of video for packet switched networks. 1870-1873 - Shabbir A. Khakoo:
Signature-based search algorithm. 1874-1877 - George Tziritas, Jean-Christophe Pesquet:
A hybrid image coder: adaptive intra-interframe prediction using motion compensation. 1878-1881 - Mohsen Soryani, Roger J. Clarke:
Image segmentation and motion-adaptive frame interpolation for coding moving sequences. 1882-1885 - Dennis Martinez, Jae S. Lim:
Spatial interpolation of interlaced television pictures. 1886-1889 - Yo-Sung Ho, Allen Gersho:
Classified transform coding of images using vector quantization. 1890-1893 - Constantine N. Manikopoulos, H. Sun:
Finite state vector quantization with an interleaved image source designed for reconstruction of lost packets. 1894-1897 - Bijaoui Albert, Huang Li:
Digital compression and morphological filter image coding. 1898-1900 - Johannes N. Driessen, Jan Biemond, Dick E. Boekee:
A pel-recursive segmentation and estimation algorithm for motion compensated image sequence coding. 1901-1904 - Paul Haskell, Kou-Hu Tzou, T. Russell Hsing:
A lapped-orthogonal-transform based variable bit-rate video coder for packet networks. 1905-1908 - Les Thede, Subhash C. Kwatra:
Image sequence coding by vector quantization. 1909-1912 - Sinan Y. Othman, S. G. Wilson:
Image sequence coding at 64 kbps using vector quantization and block matching. 1913-1916 - Peter J. Cordell, Roger J. Clarke:
An interpolative spatial domain technique for coding image sequences. 1917-1920 - Torbjörn Kronander:
Motion compensated 3-dimensional wave-form image coding. 1921-1924 - Joseph Ronsin:
Original TV coding scheme for 34 Mbit/s. 1925-1928 - Tokumichi Murakami, Naoto Kinjo, Koh Kamizawa, Toshiaki Shimada:
A 24 bit DSP for motion video codec and software development support system. 1929-1932 - Adnan M. Alattar, Sarah A. Rajala:
Primitive-based teleconference image coding technique. 1933-1936 - Cumhur Cengiz Evci:
A videophone codec for ISDN application. 1937-1940 - C. S. Kim, Mark J. T. Smith, Russell M. Mersereau:
An improved SBC/VQ scheme for color image coding. 1941-1944 - Robert J. Safranek, James D. Johnston:
A perceptually tuned sub-band image coder with image dependent quantization and post-quantization data compression. 1945-1948 - Jelena Kovacevic, Didier J. Le Gall, Martin Vetterli:
Image coding with windowed modulated filter banks. 1949-1952 - Mahesh Balakrishnan, William A. Pearlman:
Hexagonal sub-band coding for images. 1953-1956 - Jorge L. Salinas, Richard L. Baker:
Laplacian pyramid encoding: optimum rate and distortion allocations. 1957-1960 - Peter Strobach:
Image coding based on quadtree-structured recursive least-squares approximation. 1961-1964 - Alberto Sanz, Carlos Muñoz, Narciso García:
Analysis of predictive schemes in pyramidal image coding. 1965-1968 - M. Todd, R. Wilson:
An anisotropic multi-resolution image data compression algorithm. 1969-1972 - James W. Modestino, Daniel D. Harrison:
Adaptive entropy-coded 2-D DPCM encoding of images. 1973-1975 - Hiroyuki Yamaguchi, Yasushi Tatehira, Kenji Akiyama, Yukio Kobayashi:
Stereoscopic images disparity for predictive coding. 1976-1979 - B. Bochow, B. Czyrnik:
Multiprocessor implementation of an ATC audio codec. 1981-1984 - Hiroaki Nomura, Hiroyuki Miyata, Tammo Houtgast:
Speech intelligibility and MTF in non-exponential decay fields. 1985-1988 - Donald G. Jamieson, Emmet Raftery:
A general-purpose hearing aid prescription, simulation and testing system. 1989-1992 - James D. Johnston:
Perceptual transform coding of wideband stereo signals. 1993-1996 - Masaru Kamada, Kazuo Toraichi:
Effects of ultrasonic components on perceived tone quality. 1997-2000 - Shinichi Tamura:
An analysis of a noise reduction neural network. 2001-2004 - C. P. Downing, Francis M. Boland, J. B. Foley:
Improved noise canceller performance by means of an adaptive arrangement of IIR and FIR filters. 2005-2008 - Raymond N. J. Veldhuis, Marcel Breeuwer, Robbert G. van der Waal:
Subband coding of digital audio signals without loss of quality. 2009-2012 - Zong Liang Wu, Pierre Escudier, Jean-Luc Schwartz:
Specialized physiology-based channels for the detection of articulatory-acoustic events. A preliminary scheme and its performance. 2013-2016 - Diane K. Bustamante, Thomas L. Worrall, Malcolm J. Williamson:
Measurement and adaptive suppression of acoustic feedback in hearing aids. 2017-2020 - Yannick Mahieux, Jean-Pierre Petit, A. Charbonnier:
Transform coding of audio signals using correlation between successive transform blocks. 2021-2024 - Guy Billoud, Marie-Annick Galland, Michel Sunyach:
The use of time algorithms for the realization of an active sound attenuator. 2025-2028 - Larry J. Eriksson, Mark C. Allie, C. D. Bremigan, J. A. Gilbert:
Weight vector analysis of an RLMS adaptive filter with on-line auxiliary path modelling. 2029-2032 - Joachim Scheuren:
Iterative design of bandlimited FIR filters with gain constraints for active control of wave propagation. 2033-2036 - Mikio Tohyama, Akira Suzuki, Kiyoshi Sugiyama:
Active power minimization of a sound source in a reverberant space. 2037-2040 - Hiroshi Yasukawa, Isao Furukawa, Yasuzou Ishiyama:
Acoustic echo control for high quality audio teleconferencing. 2041-2044 - Dieter Bauer, Dieter Seitzer:
Statistical properties of high quality stereo signals in the time domain. 2045-2048 - J. C. Ventura:
Digital audio gain control for hearing aids. 2049-2052 - Jean Laroche:
A new analysis/synthesis system of musical signals using Prony's method-application to heavily damped percussive sounds. 2053-2056 - Dieter Seitzer, Karl-Heinz Brandenburg, Rolf Kappust, Ernst Eberlein, Heinz Gerhäuser, Stefan Krägeloh, Hartmut Schott:
DSP based real time implementation of an advanced analysis tool for audio channels. 2057-2060 - H. J. Butterweck:
About Doppler nonlinearities in loudspeakers. 2061-2063 - Michèle Basseville, Albert Benveniste:
Multiscale statistical signal processing. 2065-2068 - Ahmed H. Tewfik:
Harmonic retrieval in the presence of colored noise. 2069-2072 - Peter J. Sherman, Kang-Ning Lou:
A new method for point power spectrum estimation. 2073-2076 - Svante Gunnarsson, Lennart Ljung:
Frequency domain description of the tracking capability and disturbance rejection trade-off in recursive identification. 2077-2080 - John B. Kenney, Charles E. Rohrs:
Bias analysis of a combined output error-equation error algorithm. 2081-2084 - Ehud Weinstein, Meir Feder:
Sequential algorithms based on Kullback-Liebler information measure and their application to FIR system identification. 2085-2088 - Hans Wilhelm Schüßler, Y. Dong:
A new method for measuring the performance of weakly nonlinear systems. 2089-2092 - Ralph D. Hippenstiel, Paulo M. Oliveira:
Contributions to time varying spectrum estimation using the instantaneous power spectrum (IPS). 2093-2096 - Constantin Papaodysseus, Elias Koukoutsis, George Carayannis:
Numerical behaviour of Toeplitz solutions. 2097-2100 - Patrick L. Combettes, H. Joel Trussell:
General order moments in set theoretic estimation. 2101-2104 - J. S. Lee, Joe K. Hammond:
Time-varying filter modelling of the sound field due to a moving source and time-delay estimation. 2105-2108 - Jean-François Cardoso:
Source separation using higher order moments. 2109-2112 - Hagit Messer, Yael Adar:
Lower bounds on bearing estimation errors of any number of narrowband far-field sources. 2113-2116 - L. R. Hunt, Darel A. Linebarger, Ronald D. DeGroat:
Nonlinear system identification and characterization. 2117-2120 - Darel A. Linebarger, Ronald D. DeGroat:
A statistical measure of resolution for modern direction finding methods. 2121-2123 - Xiao-Liang Xu, Kevin M. Buckley:
Statistical performance comparison of MUSIC in element-space and beam-space. 2124-2127 - Dov Wulich, Eugene I. Plotkin, M. N. S. Swamy, E. Kashi:
Separation of close sinusoids by cross-coupled phase locked loop. 2128-2131 - Sang-Geun Oh, R. L. Kashyap:
Robust frequency estimation. 2132-2135 - George A. Lampropoulos:
On detection of sinusoids from periodograms. 2136-2139 - Stephen W. Lang, Bruce R. Musicus:
Frequency estimation from phase differences. 2140-2143 - Jonathan S. Abel:
A divide and conquer approach to least-squares estimation with application to range-difference-based localization. 2144-2147 - John Litva:
Superresolution based on the use of deterministic physical modelling. 2148-2151 - S. Lawrence Marple Jr.:
A tutorial overview of modern spectral estimation. 2152-2157 - Will Gersch:
Smoothness priors multichannel autoregressive time series modeling. 2158-2161 - Amro El-Jaroudi, John Makhoul:
Discrete pole-zero modeling and applications. 2162-2165 - Maurice G. Bellanger:
The potential of QR adaptive filter variables for signal analysis. 2166-2169 - Tom W. Parks, Ram G. Shenoy, Nihal I. Wijeyesekera:
Signal class estimation using MLM. 2170-2173 - Miguel Angel Lagunas, Mateo Amengual, Maria E. Forcada:
The periodogram envelope in parametric and non-parametric spectral estimation. 2174-2177 - A. Murat Tekalp, A. Tanju Erdem:
Higher-order spectrum factorization with applications. 2178-2181 - Ananthram Swami, Georgios B. Giannakis, Jerry M. Mendel:
A unified approach to modeling multichannel ARMA processes. 2182-2185 - M. Isabel Ribeiro, Josiane Zerubia, José M. F. Moura, Gérard Alengrin:
Comparison of two ARMA estimators. 2186-2189 - Stephen Konyk Jr., Moeness G. Amin, M. G. Lagunas:
Least squares null space variational characterization for nonminimum norm solutions. 2190-2193 - Donald W. Tufts, R. J. Vacarro, A. C. Kot:
Analysis of estimation of signal parameters by linear-prediction at high SNR using matrix approximation. 2194-2197 - Cédric Demeure, Louis L. Scharf:
Fast least squares solution of Vandermonde systems of equations. 2198-2210 - Salvatore D. Morgera, Bernard Armour:
Structured maximum likelihood autoregressive parameter estimation. 2202-2205 - Yumi Takizawa, Atsushi Fukasawa:
Non-stationary signal spectrum analysis improving maximum entropy estimation error. 2206-2209 - Sathyanarayan S. Rao, K. R. Raghavan:
Estimating the frequencies of sinusoids at low signal to noise ratios using the principal component method in tandem with the Burg algorithm. 2210-2213 - Victor E. DeBrunner, A. A. (Louis) Beex:
Sensitivity of structures for the identification of linear systems from impulse response data. 2214-2217 - Hiroyuki Tsuji, Akira Sano:
Separable estimation of discrete and continuous spectra of signal with mixed spectrum. 2218-2221 - Douglas L. Jones, Thomas W. Parks:
A resolution comparison of several time-frequency representations. 2222-2225 - Shubha Kadambe, Gloria Faye Boudreaux-Bartels, Patrick Duvaut:
Window length selection for smoothing the Wigner distribution by applying an adaptive filter technique. 2226-2229 - Mingui Sun, C. C. Li, Laligam N. Sekhar, Robert J. Sclabassi:
Elimination of cross-components of the discrete pseudo Wigner distribution via image processing. 2231-2233 - Angus Andrews:
Parallel time-frequency analysis. 2234-2237 - Leon Cohen, Chongmoon Lee:
Standard deviation of instantaneous frequency. 2238-2241 - Edgar F. Velez, Richard G. Absher:
Transient analysis of speech signals using the Wigner time-frequency representation. 2242-2245 - Patrick Flandrin, Dominique Garreau, Claude Puyal:
Improving monitoring of PWR electrical power plants 'in core' instrumentation with time-frequency signal analysis. 2246-2249 - A. Krantzik, Dietrich Wolf:
Analysis of a modified Suzuki fading channel model. 2250-2253 - Petar M. Djuric, Steven M. Kay:
A simple frequency rate estimator. 2254-2257 - Liang Zhang, Zhao-Xiong Wu, Hong-Bin Chen:
The decision controlled adaptive LMS Tauberian deconvolution. 2258-2261 - Yongli Sun, Saleem A. Kassam, Fred Haber, Sumit Roy:
A pattern diversity method for multiple coherent source location. 2262-2265 - Björn E. Ottersten, Lennart Ljung:
Asymptotic results for sensor array processing. 2266-2269 - Seth D. Silverstein, S. M. Carroll, Joseph M. Pimbley:
Performance comparisons of the minimum free energy algorithms with the reduced rank modified covariance eigenanalysis algorithm. 2270-2273 - John H. Cozzens, Robert C. DiPietro, Michael J. Sousa:
Enumeration of fully correlated signals by modified rank sequences. 2274-2277 - Stephan V. Schell, Robert A. Calabretta, William A. Gardner, Brian G. Agee:
Cyclic MUSIC algorithms for signal-selective direction estimation. 2278-2281 - Jianguo Huang, Steven Kay:
Frequency estimation using a dynamic programming-type algorithm. 2282-2285 - A. Okhovat, J. R. Cruz:
Statistical analysis of the Tufts-Kumaresan and principal Hankel components methods for estimating damping factors of single complex exponentials. 2286-2289 - Akira Sano, T. Furuya, Hiroyuki Tsuji, Hiromitsu Ohmori:
Simultaneous optimization method of regularization and singular value decomposition in least squares parameter identification. 2290-2293 - Alf J. Isaksson:
Frequency domain accuracy of identified 2-D causal models. 2294-2297 - Bülent Baygün, Yalçin Tanik:
Performance analysis of the MUSIC algorithm in direction finding systems. 2298-2301 - Yves Grenier:
Model expansion applied to speech analysis and source location. 2302-2305 - S. W. Nam, Sung Bae Kim, Edward J. Powers:
Utilization of digital polyspectral analysis to estimate transfer functions of cubically nonlinear systems with nonGaussian inputs. 2306-2309 - Doron Kletter, Hagit Messer:
The role of third order spectrum in maximum likelihood time delay estimation of a random multi-tone signal in noise. 2310-2313 - Benjamin Friedlander, Boaz Porat:
Algorithms for optimal estimation of the parameters of non-Gaussian processes from high-order moments. 2314-2317 - Ananthram Swami, Jerry M. Mendel:
Computation of cumulants of ARMA processes. 2318-2321 - Sohail A. Dianat, M. R. Raghuveer:
Polyspectral factorization: necessary and sufficient condition for finite extent cumulant sequences. 2322-2324 - A. Lee Swindlehurst, Thomas Kailath:
Detection and estimation using the third moment matrix. 2325-2328 - I. Rhodes, Anthony G. Constantinides:
A partitioned approach to spectral estimation. 2329-2332 - Roger F. Dwyer:
Fourth-order spectra of mixture processes. 2333-2336 - Athina P. Petropulu, Chrysostomos L. Nikias:
Analytic performance evaluation of the bicepstrum. 2337-2340 - Der-Shan Luo, Andrew E. Yagle:
Lattice algorithms applied to the blind deconvolution problem. 2341-2344 - Domingo Docampo-Amoedo, Aníbal R. Figueiras-Vidal:
A deconvolution approach to harmonic signal extrapolation. 2345-2348 - Filson H. Glanz, W. Thomas Miller III:
Deconvolution and nonlinear inverse filtering using a neural network. 2349-2352 - Radomir T. Sokolov, James C. Rogers:
Deconvolution in the class of non-stationary repetitive signals. 2353-2356 - Giovanni Jacovitti, Alessandro Neri, Gaetano Scarano:
A deconvolution technique based on nonlinear estimation of hidden Markov chains. 2357-2360 - Zoran B. Banjanin, J. R. Cruz, Dusan S. Zrnic:
Linear prediction approach to Doppler estimation of radar signals in the presence of ground clutter. 2361-2364 - Joseph A. O'Sullivan, Donald L. Snyder, Pierre Moulin:
The role of spectrum estimation in forming high-resolution radar images. 2365-2367 - Ramón García-Gómez, Juan Gómez-Mena, Luís Díez del Río:
Adaptive receivers for removing linear and non-linear intersymbol interference by mean of time delay neural nets (AR-TDNN). 2368-2371 - Hong Wang, Lujing Cai:
Adaptive filtering for moving-target-detection in severely inhomogeneous clutter. 2372-2375 - Claudio Maria Prati:
3-D synthetic aperture radar surveys. 2376-2379 - Vincent Considine:
CORDIC trigonometric function generator for DSP. 2381-2384 - Hassan M. Ahmed, Kin-Ho Fu:
A VLSI array CORDIC architecture. 2385-2388 - Gian Carlo Cardarilli, Roberto Lojacono, Mario Salerno:
RNS approach to fast dividers. 2389-2392 - John Canaris:
A high speed fixed point binary divider. 2393-2396 - Richard R. Shively, Allen L. Gorin:
A reconfigurable fault-tolerant systolic signal processor. 2397-2400 - Rabin Raut, B. B. Bhattacharya, S. M. Faruque:
An application of systolic array design architecture to switched capacitor filter circuits. 2401-2404 - Robert W. Stewart, Roy Chapman, Tariq S. Durrani:
Arithmetic implementation of the Givens QR triarray. 2405-2408 - Eric M. Dowling, Fred J. Taylor:
Matrix methods for the design and analysis of recurrent algorithms for multi-purpose systolic arrays. 2409-2412 - Stewart G. Smith, Ralph W. Morgan:
Generic ASIC architecture and synthesis scheme for DSP. 2413-2416 - Cynthia J. Anfinson, Adam W. Bojanczyk, Franklin T. Luk, Eric K. Torng:
Algorithm-based fault-tolerant techniques for MVDR beamforming. 2417-2420 - Richard C. Jaffe, David S. Johnson, Wen-Tai Lin, Chung-Yin Ho:
Application of VLSI for image processing. 2421-2424 - Kamyar Dezhgosha, Mohsin M. Jamali, Subhash C. Kwatra:
Real-time VLSI architecture for a VQ-based high-quality image coding algorithm. 2425-2428 - Jean-Claude Carlach, Pierre Penard, Jean-Luc Sicre:
TCAD: a 27 MHz 8×8 discrete cosine transform chip. 2429-2432 - Takashi Miyazaki, Takao Nishitani, Shinichi Aikoh, Masaki Ishikawa, Takeshi Yoshimura, Kaoru Mitsuhashi, M. Furuichi:
A single chip VLSI chrominance/luminance separator based on a silicon compiler. 2433-2436 - Kun-Min Yang, Ming-Ting Sun, Lance Wu, I-Fei G. Chuang:
Very high efficiency VLSI chip-pair for full search block matching with fractional precision. 2437-2440 - Chein-Wei Jen, Chi-Min Liu:
Two-level pipeline design for image resampling. 2441-2444 - Ulrich Kleine, Erik De Man:
Two-dimensional FIR filter architectures based on NTT. 2445-2448 - Simon C. Knowles, John G. McWhirter, Roger F. Woods, John V. McCanny:
A bit-level systolic architecture for very high performance IIR filters. 2449-2452 - A. Artieri, Francis Jutand:
A versatile and powerful chip for real-time motion estimation. 2453-2456 - Thomas Komarek, Peter Pirsch:
VLSI architectures for block matching algorithms. 2457-2460 - J. B. G. Roberts:
Recent developments in parallel processing. 2461-2467 - Taiho Koh, Oscar E. Agazzi, Syed S. Haider, Robert W. Walden, Daniel R. Cassiday, Gene A. Wilson, T. Mariano Lalumia, Christine M. Gerveshi, M. R. Dwarakanath, Jitendra Kumar, Ronald E. Crochiere, Robert F. Shaw, Ralph A. Wilson III, William R. McDonald, Noah L. Gottfried, Nallepilli S. Ramesh, M. L. Heisakanen, Rob B. Blake Jr.:
Algorithms and architecture of a VLSI signal processor for ANSI standard ISDN transceiver. 2468-2471 - Nan-Sheng Lin, Bernard Fraenkel, Gordon Jacobs, Leechung Yiu, Daniel Senderowicz, Richard Ulmer:
Sigma-delta A/D and D/A for high speed voiceband modems. 2472-2475 - Robert J. Sluyter, P. J. Snijder, H. Dijkstra, C. M. Huizer, Arthur H. M. van Roermund:
A programmable video signal processor. 2476-2479 - Kevin L. Kloker, Brett Lindsley, Sergio Liberman, Paul Marino, Elchanan Rushinek, Garth D. Hillman:
The Motorola DSP 96002 IEEE floating-point digital signal processor. 2480-2483 - Sayfe Kiaei, Jaisimha K. Durgam:
VLSI design of dynamically reconfigurable array processor-DRAP. 2484-2488 - Joseph B. Evans, Bede Liu:
A CMOS implementation of a variable step size digital adaptive filter. 2489-2492 - A. Picco, J. C. Michalina, B. Laurier, D. Fuin, P. Menut, J. L. Laborie, C. Priol, F. Sforza:
The ST18940/41: an advanced single-chip digital signal processor. 2493-2496 - R. Srinivasan, Malayappan Shridhar, Majid Ahmadi:
A neural implementation for interpolation of stereo data. 2497-2500 - Kurt R. Smith, Michael I. Miller:
Learning regular grammars on connection architectures. 2501-2504 - Sun-Yuan Kung, Jenq-Neng Hwang:
A unifying algorithm/architecture for artificial neural networks. 2505-2508 - Yoshitake Suzuki, Les E. Atlas:
A comparison of processor topologies for a fast trainable neural network for speech recognition. 2509-2512 - Ermanno Di Zitti, Daniele D. Caviglia, Giacomo M. Bisio, Giancarlo Parodi:
Neural networks on a transputer array. 2513-2516 - Andrea Maccato, Rui J. P. de Figueiredo:
Automated design of neural networks for symbolic signal processing. 2517-2520 - Gagan Mirchandani, Wei Cao, Barry Bosworth:
Efficient implementation of neural nets using an optimal relationship between number of patterns, input dimension and hidden nodes. 2521-2523 - D. C. Ashby, R. A. McConnell, C. C. Yu:
An integrated environment for DSP systems design. 2524-2527 - C. C. Yu, Les J. Wu, Y. S. Wu:
Constant capacity - an information theoretic approach to VLSI/DSP architecture. 2528-2531 - Teresa H.-Y. Meng, Robert W. Brodersen, David G. Messerschmitt:
Design of clock-free asynchronous systems for real-time signal processing. 2532-2535 - Konstantinos Konstantinides, Ronald T. Kaneshiro, Jon R. Tani:
Scheduling and task allocation for parallel digital signal processing architectures. 2536-2539 - Edward C. Bronson, Thomas L. Casavant, Leah H. Jamieson:
Experimental analysis of multi-mode fast Fourier transforms on the PASM parallel processing system. 2540-2543 - Aggelos K. Katsaggelos, Srikanta P. R. Kumar, Majid Sarrafzadeh:
Parallel processing architectures for iterative image restoration. 2544-2547 - Mustafa Karaman, Levent Onural, Abdullah Atalar:
Design and implementation of a general purpose VLSI median filter unit and its applications. 2548-2551 - F. Balestro, Gilles Privat, M. S. Tawfik:
A bit-serial approach to VLSI implementation of digital LDI ladder filters. 2552-2555 - Myriam Ba, Dominique Degrugillier, Claude Berrou:
Digital VLSI using parallel architecture for co-occurrence matrix determination. 2556-2559 - Tim A. Williams:
A systolic IIR decimator. 2560-2563 - M. Frenkel, E. Vildenberg, Rafi Retter, I. Shpancer:
A dedicated DSP computation engine based on VLSI vector signal processors. 2564-2568 - Charles D. Thompson:
A VLSI sigma delta A/D converter for audio and signal processing applications. 2569-2572 - Vittorio Rampa, Neviano Dal Degan, Alessandro Balboni:
VLSI implementation of a pel-by-pel motion estimator. 2573-2576 - William Robertson, William J. Phillips:
A systolic MUSIC system for VLSI implementation. 2577-2580 - Michael F. Griffin, Fred J. Taylor:
A residue number system reduced instruction set computer (RISC) concept. 2581-2584 - Ramasamy Krishnan:
An efficient systolic array VLSI cell architecture for the implementation of transversal filter based on the quadratic residue number systems. 2585-2588 - David N. Swingler, Robert S. Walker, Jeffrey L. Krolik:
Implementation considerations for a doubly-steered, coherently-focussed high-resolution broadband beamformer for source location estimation. 2589-2592 - Henry Cox, Robert M. Zeskind, Matthew Myers:
Large aperture matched field processing via subaperture beamforming techniques. 2593-2596 - Victor Barroso, José M. F. Moura:
Adaptive beamforming as an inverse problem. 2597-2600 - Jean-Pierre Le Cadre, Patrice Ravazzola:
Approximated stochastic realization and model reduction methods applied to array processing by means of state space models. 2601-2604 - Petre Stoica, Arye Nehorai:
MUSIC, maximum likelihood and Cramér-Rao bound: further results and comparisons. 2605-2608 - James P. Reilly, Kon Max Wong, P. M. Reilly:
Direction of arrival estimation in the presence of noise with unknown, arbitrary covariance matrices. 2609-2612 - Fu Li, Richard J. Vaccaro, Donald W. Tufts:
Min-norm linear prediction for arbitrary sensor arrays. 2613-2616 - Young-Soo Kim, James A. Cadzow, Han-Kyu Park:
Signal enhancement approach for high resolution of multiple broadband incoherent sources. 2617-2620 - Ehoud Haberman, Lloyd J. Griffiths:
On combining the eigenstructure-based and model-based approaches in direction of arrival estimation problems. 2621-2624 - Maria-João D. Rendas, José M. F. Moura:
Resolving narrowband coherent paths with non-uniform arrays. 2625-2628 - José M. F. Moura, Maria-João D. Rendas:
Sensitivity of range localization in a multipath environment. 2629-2632 - Benoît Champagne, Moshe Eizenman, Subbarayan Pasupathy:
Exact maximum likelihood time delay estimation. 2633-2636 - Jeffrey L. Krolik, Jim Lynch, David N. Swingler:
A robust incoherent matched field processor for source localization in uncertain multipath fields. 2637-2640 - Joongkyu Kim, Alfred O. Hero III:
Error intensity measures for multi-parameter tracking and passive bearing estimation. 2641-2644 - Jean-Michel Passerieux, Denis Pillon, Pierre Blanc-Benon, Claude Jauffret:
Target motion analysis with bearings and frequencies measurements via instrumental variable estimator [passive sonar]. 2645-2648 - Y. Chocheyras:
Near field three dimensional time delay and Doppler target motion analysis. 2649-2652 - G. W. Johnson, E. J. Modugno:
Source localization from multiple independent observables. 2653-2656 - Yoram Bresler, Alexander H. Delaney:
Resolution of overlapping echoes of unknown shape. 2657-2660 - John P. Ianniello, Miriam Hamilton:
Fundamental limitations to broadband time delay resolution for two sensors. 2661-2664 - Mordechai Segal, Ehud Weinstein:
Spatial and spectral parameter estimation of multiple source signals. 2665-2668 - Jerome R. Bellegarda:
Time-frequency properties of extended quadratic congruential frequency hop signals. 2669-2672 - T. T. Kadota:
A nonlinear optimum-detection problem. 2673-2675 - Isabel M. G. Lourtie, G. Clifford Carter:
Signal detection in a multiple time delay environment. 2676-2679 - Herbert Gish, Ronald A. Mucci:
Detection of multiple signals by the significance test. 2680-2683 - Jean-Jacques Fuchs:
Estimation of the number of signals in the presence of unknown correlated sensor noise. 2684-2687 - M. P. Boyer, Bernard Picinbono:
Quantization and distributed detection. 2688-2691 - Giorgio Tacconi, Antonio Tiano:
Identification of an underwater target by acoustic inverse scattering. 2692-2695 - Serge Prosperi:
Stability and convergence of probabilistic data association filters. 2696 - Magnus Moll:
A simple algorithm for track termination. 2700-2703 - Michel Bouvet:
Performance of normalized matched filters. 2704-2707 - V. Nimier, Geneviève Jourdain, B. Faure:
Use of high resolution methods in time delay estimation for tomography. 2708-2711 - Jean-Pierre Hermand, Philippe Nicolas:
Adaptive classification of underwater transients. 2712-2715 - K. C. Sharman, G. D. McClurkin:
Genetic algorithms for maximum likelihood parameter estimation. 2716-2719 - Darryl R. Morrell, Wynn C. Stirling:
Robust tracking of marginally observable targets. 2720-2723 - D. Van Cappel:
Target motion analysis using time delays measured from a nonlinear array. 2724-2727 - Pamela A. Nielsen, John B. Thomas:
Non-parametric detection in underwater environments. 2728-2731 - Amy R. Reibman:
Performance of large distributed detection networks and the sign detector. 2732-2735 - Yiu Tong Chan, Pak-Chung Ching:
Non-stationary time delay estimation with a multipath. 2736-2739 - Ju-Hong Lee, Jen-Fu Wu:
A novel approach for robust adaptive beamforming. 2740-2743 - Isabelle Guelle, Denis Pillon:
Inter array multi-tracks association. 2744-2747 - Jean-Louis Berrou, Christine Debaillon-Vesque:
Acoustical imaging and far-field estimation of the noise radiated by a moving source using a linear array. 2748-2751 - Alireza Moghaddamjoo:
Spatial filtering approach to the direction of arrival estimation in a multipath environment. 2752-2755 - Sylvie Marcos, Jacques Munier:
Source localization using a distorted antenna. 2756-2759 - Bhaskar D. Rao, K. V. S. Hari:
Statistical performance analysis of the minimum-norm method. 2760-2763 - Henri Clergeot, Olivier Michel:
New simple implementation of the coherent signal subspace method for wide band direction of arrival estimation. 2764-2767 - Leopold Bömer, Markus Antweiler:
Two-dimensional binary arrays with constant sidelobe in their PACF. 2768-2771 - S. Sivanand, Jar-Fen Yang, Mostafa Kaveh:
Time-domain coherent signal-subspace wideband direction-of-arrival estimation. 2772-2775 - Byong Kun Chang, Nasir Ahmed:
Adaptive broadband beamforming: an integral nulling approach. 2776-2778 - Dong-Chang Shiue, James A. Cadzow, George R. Davis:
Estimation of direction-of-arrival using spatially large sparse array. 2779-2782 - Philippe Forster, Georges Vezzosi:
Optimal Toeplitzification with a given rank constraint. 2783-2786 - Dennis R. Morgan, Thomas M. Smith:
Coherence effects on the detection performance of quadratic array processors, with applications to large-array matched-field beamforming. 2787-2790 - P. Germain, Alain Maguer, Laurent Kopp:
Comparison of resolving power of array processing methods by using an analytical criterion. 2791-2794 - David R. Farrier, R. Mardani:
A suboptimal, low cost maximum likelihood algorithm. 2795-2798 - Georges Bienvenu, P. Fuerxer, Georges Vezzosi, Laurent Kopp, F. Florin:
Coherent wide band high resolution processing for linear array. 2799-2802 - Qin-Ye Yin, R. W. Newcomb, Li-He Zou:
Estimating 2-D angles of arrival via two parallel linear arrays. 2803-2806 - Björn E. Ottersten, Mats Viberg:
Analysis of subspace fitting based methods for sensor array processing. 2807-2810 - Benjamin Friedlander:
A sensitivity analysis of the MUSIC algorithm. 2811-2814 - Alain Maguer:
Detection of targets in presence of strong jammers by adaptive beamforming. 2815-2818 - Bin Yang, Johann F. Böhme:
On a systolic implementation and the numerical properties of a multiple constrained adaptive beamformer. 2819-2822 - John W. Fau, Joseph J. Wolcin:
Bearing estimation accuracy with a synthetic aperture. 2823-2825 - Pascal Chevalier, Bernard Picinbono:
Optimal linear-quadratic array for detection. 2826-2829 - Douglas B. Williams, Don H. Johnson:
Narrowband array processing algorithms for arbitrary noise distributions. 2830-2833
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