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ORCIDICASSP 1991: Toronto, Ontario, Canada
- 1991 International Conference on Acoustics, Speech, and Signal Processing, ICASSP '91, Toronto, Ontario, Canada, May 14-17, 1991. IEEE Computer Society 1991, ISBN 0-7803-0003-3
- Schuyler Quackenbush:
A 7 kHz bandwidth, 32 kbps speech coder for ISDN. 1-4 - Yair Shoham:
On the use of direct vector quantization in LPC-based analysis-by-synthesis coding systems. 5-8 - Erik Ordentlich, Yair Shoham:
Low-delay code-excited linear-predictive coding of wideband speech at 32 kbps. 9-12 - Claude Laflamme, Jean-Pierre Adoul, Redwan Salami, Sarto Morissette, Philippe Mabilleau:
16 kbps wideband speech coding technique based on algebraic CELP. 13-16 - Guylain Roy, Peter Kabal:
Wideband CELP speech coding at 16 kbits/sec. 17-20 - Juin-Hwey Chen, Yen-Chun Lin, Richard V. Cox:
A fixed-point 16 kb/s LD-CELP algorithm. 21-24 - Majid Foodeei, Peter Kabal:
Low-delay CELP and tree coders: comparison and performance improvements. 25-28 - Rong Peng, Vladimir Cuperman:
Variable-rate low-delay analysis-by-synthesis speech coding at 8-16 kb/s. 29-32 - Alv I. Aarskog, Arild Nilsen, Ole Berg, Hans C. Guren:
A long-term predictive ADPCM coder with short-term prediction and vector quantization. 37-40 - Carl Rudolph Nassar, Mohammad Reza Soleymani:
Globally optimal trellis quantizers. 41-44 - Hidefumi Sawai:
Frequency-time-shift-invariant time-delay neural networks for robust continuous speech recognition. 45-48 - Nelson Morgan, Hynek Hermansky
, Hervé Bourlard, Phil Kohn, Chuck Wooters:
Continuous speech recognition using PLP analysis with multilayer perceptrons. 49-52 - Hidefumi Sawai:
TDNN-LR continuous speech recognition system using adaptive incremental TDNN training. 53-56 - Ken-ichi Iso, Takao Watanabe:
Large vocabulary speech recognition using neural prediction model. 57-60 - Joe Tebelskis, Alex Waibel, Bojan Petek, Otto Schmidbauer:
Continuous speech recognition using linked predictive neural networks. 61-64 - Gerhard Rigoll:
Information theory-based supervised learning methods for self-organizing maps in combination with hidden Markov modeling. 65-68 - Jhing-Fa Wang, Chung-Hsien Wu, Chaug-Ching Haung, Jau-Yien Lee:
Integrating neural nets and one-stage dynamic programming for speaker independent continuous Mandarin digit recognition. 69-72 - Frédéric Bimbot, Gérard Chollet, Jean-Pierre Tubach:
TDNNs for phonetic features extraction: a visual exploration. 73-76 - Sung Jun Lee, Ki Chul Kim, Hyunsoo Yoon, Jung Wan Cho:
Application of fully recurrent neural networks for speech recognition. 77-80 - Fabio Greco, Andrea Paoloni, Giacomo Ravaioli:
A recurrent time-delay neural network for improved phoneme recognition. 81-84 - Masami Nakamura, Shinichi Tamura, Shigeki Sagayama:
Phoneme recognition by phoneme filter neural networks. 85-88 - Jun-ichi Takami, Shigeki Sagayama:
A pairwise discriminant approach to robust phoneme recognition by time-delay neural networks. 89-92 - Luc Mathan, Laurent Miclet:
Rejection of extraneous input in speech recognition applications, using multi-layer perceptrons and the trace of HMMs. 93-96 - Younès Bennani, Nasser Chaourar, Patrick Gallinari, Abdelhamid Mellouk:
Validation of neural net architectures on speech recognition tasks. 97-100 - Jari Kangas:
Phoneme recognition using time-dependent versions of self-organizing maps. 101-104 - Patrick Haffner, Michael A. Franzini, Alex Waibel:
Integrating time alignment and neural networks for high performance continuous speech recognition. 105-108 - Michael M. Hochberg, Les T. Niles, J. T. Foote, Harvey F. Silverman:
Hidden Markov model/neural network training techniques for connected alphadigit speech recognition. 109-112 - Padma Ramesh, Shigeru Katagiri, Chin-Hui Lee:
A new connected word recognition algorithm based on HMM/LVQ segmentation and LVQ classification. 113-116 - Ulrich Bodenhausen, Alex Waibel:
Learning the architecture of neural networks for speech recognition. 117-120 - Yifan Gong, Jean-Paul Haton:
Non-linear vector interpolation by neural network for phoneme identification in continuous speech. 121-124 - Yasuhiro Komori:
Time-state neural networks (TSNN) for phoneme identification by considering temporal structure of phonemic features. 125-128 - Raymond L. Watrous:
Source decomposition of acoustic variability in a modular connectionist network. 129-131 - Peter Brauer, Per Hedelin, Dieter Huber, Petter Knagenhjelm:
Probability based optimization for network classifiers. 133-136 - Jan-Erik Strömberg, Jalel Zrida, Alf J. Isaksson:
Neural trees-using neural nets in a tree classifier structure. 137-140 - David M. Lubensky:
Word recognition using neural nets, multi-state Gaussian and k -nearest neighbor classifiers. 141-144 - Laurent Barbier, Gérard Chollet:
Robust speech parameters extraction for word recognition in noise using neural networks. 145-148 - Timothy R. Anderson:
Speaker independent phoneme recognition with an auditory model and a neural network: a comparison with traditional techniques. 149-152 - Yifan Gong, Ying Cheng, Jean-Paul Haton:
Neural network coupled with IIR sequential adapter for phoneme recognition in continuous speech. 153-156 - Eric R. Buhrke, Joseph L. LoCicero:
Speech recognition with neural networks and network fusion. 157-160 - Chin-Hui Lee, Egidio P. Giachin, Lawrence R. Rabiner, Roberto Pieraccini, Aaron E. Rosenberg:
Improved acoustic modeling for speaker independent large vocabulary continuous speech recognition. 161-164 - Yves Laprie:
Phonetic triplets in acoustic-phonetic decoding of continuous speech. 165-168 - Tomokazu Yamada, Toshiyuki Hanazawa, Takeshi Kawabata, Shoichi Matsunaga, Kiyohiro Shikano:
Phonetic typewriter based on phoneme source modeling. 169-172 - Lalit R. Bahl, Subrata K. Das, Peter DeSouza, M. Epstein, Robert L. Mercer, Bernard Mérialdo, David Nahamoo, Michael A. Picheny, J. Powell:
Automatic phonetic baseform determination. 173-176 - Lalit R. Bahl, Jerome R. Bellegarda, Peter V. de Souza, P. S. Gopalakrishnan, David Nahamoo, Michael A. Picheny:
A new class of fenonic Markov word models for large vocabulary continuous speech recognition. 177-180 - Lynn C. Wood, David J. B. Pearce, Frederic Novello:
Improved vocabulary-independent sub-word HMM modelling. 181-184 - Lalit R. Bahl, Peter V. de Souza, P. S. Gopalakrishnan, David Nahamoo, Michael A. Picheny:
Decision trees for phonological rules in continuous speech. 185-188 - Emilio Sanchis Arnal, Francisco Casacuberta, Isabel Galiano, Encarna Segarra:
Learning structural models of subword units through grammatical inference techniques. 189-192 - Li Deng, Kevin Erler:
Microstructural speech units and their HMM representation for discrete utterance speech recognition. 193-196 - Paul Dalsgaard, Ove Andersen, William J. Barry:
Multi-lingual label alignment using acoustic-phonetic features derived by neural-network technique. 197-200 - W. Bastiaan Kleijn:
Continuous representations in linear predictive coding. 201-204 - Ira A. Gerson, Mark A. Jasiuk:
Techniques for improving the performance of CELP type speech coders. 205-208 - Alain Le Guyader, Renaud J. Di Francesco, Claude Lamblin:
Derivation of efficient CELP coding algorithms using the Z-transform approach. 209-212 - Kazunori Ozawa, Toshiki Miyano:
4 kb/s improved CELP coder with efficient vector quantization. 213-216 - W. Granzow, Bishnu S. Atal, Kuldip K. Paliwal, Juergen Schroeter:
Speech coding at 4 kb/s and lower using single-pulse and stochastic models of LPC excitation. 217-220 - Mark Johnson, Tomohiko Taniguchi:
Low-complexity multi-mode VXC using multi-stage optimization and mode selection [speech coding]. 221-224 - Per Hedelin, Anders Bergström:
Amplitude quantization for CELP excitation signals. 225-228 - Kimio Miseki, Masami Akamine:
Adaptive bit-allocation between the pole-zero synthesis filter and excitation in CELP. 229-232 - Maurizio Copperi:
Efficient excitation modeling in a low bit-rate CELP coder. 233-236 - U. Kipper, Herbert Reininger, Dietrich Wolf:
Improved CELP coding using adaptive excitation codebooks. 237-240 - Tomohiko Taniguchi, Mark Johnson, Yasuji Ohta:
Pitch sharpening for perceptually improved CELP, and the sparse-delta codebook for reduced computation. 241-244 - Philipos C. Loizou, Andreas S. Spanias:
Vector quantization of transform components for speech coding at 1200 bps. 245-248 - John C. Hardwick, Jae S. Lim:
The application of the IMBE speech coder to mobile communications. 249-252 - Lorenzo Fissore, Pietro Laface, Giorgio Micca:
Comparison of discrete and continuous HMMs in a CSR task over the telephone. 253-256 - Yuqing Gao, Taiyi Huang, Zhiwei Lin, Bo Xu, Dongxin Xu:
A real-time Chinese speech recognition system with unlimited vocabulary. 257-260 - Kari Torkkola, Mikko Kokkonen:
Using the topology-preserving properties of SOFMs in speech recognition. 261-264 - Tatsuo Matsuoka, Kiyohiro Shikano:
Robust HMM phoneme modeling for different speaking styles. 265-268 - Kenji Kita, Wayne H. Ward:
Incorporating LR parsing into SPHINX. 269-272 - Shozo Makino, Akinori Ito, Mitsuru Endo, Ken'iti Kido:
A Japanese text dictation system based on phoneme recognition and a dependency grammar. 273-276 - John S. Bridle, L. Dodd:
An Alphanet approach to optimising input transformations for continuous speech recognition. 277-280 - Jeffrey N. Marcus, Victor W. Zue:
A variable duration acoustic segment HMM for hard-to-recognize words and phrases. 281-284 - Helen M. Meng, Victor W. Zue:
Signal representation comparison for phonetic classification. 285-288 - Vassilios Digalakis, Jan Robin Rohlicek, Mari Ostendorf:
A dynamical system approach to continuous speech recognition. 289-292 - Kathy L. Brown, V. Ralph Algazi:
Speech recognition using dynamic features of acoustic subword spectra. 293-296 - Helene Cerf-Danon, Marc El-Bèze:
Three different probabilistic language models: comparison and combination. 297-300 - Stephan Euler, Joachim Zinke:
Extending the vocabulary of a speaker independent recognition system. 301-304 - Ayman Asadi, Richard M. Schwartz, John Makhoul:
Automatic modeling for adding new words to a large-vocabulary continuous speech recognition system. 305-308 - Jay G. Wilpon, Laura G. Miller, P. Modi:
Improvements and applications for key word recognition using hidden Markov modeling techniques. 309-312 - David P. Morgan, Christopher L. Scofield, John E. Adcock:
Multiple neural network topologies applied to keyword spotting. 313-316 - Richard C. Rose, Eric I. Chang, Richard Lippmann:
Techniques for information retrieval from voice messages. 317-320 - Colin W. Wightman, Mari Ostendorf:
Automatic recognition of prosodic phrases. 321-324 - Ronald A. Cole, Mark A. Fanty, M. Gopalakrishnan, Rik D. T. Janssen:
Speaker-independent name retrieval from spellings using a database of 50000 names. 325-328 - Douglas B. Paul:
The Lincoln tied-mixture HMM continuous speech recognizer. 329-332 - Yunxin Zhao, Hisashi Wakita, Xinhua Zhuang:
An HMM based speaker-independent continuous speech recognition system with experiments on the TIMIT database. 333-336 - William S. Meisel, Mark T. Anikst, S. S. Pirzadeh, J. E. Schumacher, Matthew C. Soares, David J. Trawick:
The SSI large-vocabulary speaker-independent continuous speech recognition system. 337-340 - Vishwa Gupta, Matthew Lennig, Paul Mermelstein, Patrick Kenny, Franz Seitz, Douglas D. O'Shaughnessy:
Using phoneme duration and energy contour information to improve large vocabulary isolated-word recognition. 341-344 - Xuedong Huang, Kai-Fu Lee, Hsiao-Wuen Hon, Mei-Yuh Hwang:
Improved acoustic modeling with the SPHINX speech recognition system. 345-348 - Jay G. Wilpon, Chin-Hui Lee, Lawrence R. Rabiner:
Improvements in connected digit recognition using higher order spectral and energy features. 349-352 - Yeshwant K. Muthusamy, Ronald A. Cole, M. Gopalakrishnan:
A segment-based approach to automatic language identification. 353-356 - Horacio Franco, António Joaquim Serralheiro:
Training HMMs using a minimum recognition error approach. 357-360 - Jean-Claude Junqua:
The influence of psychoacoustic and psycholinguistic factors on listener judgments of intelligibility of normal and Lombard speech. 361-364 - Wayne H. Ward:
Understanding spontaneous speech: the Phoenix system. 365-367 - Steve Renals, David McKelvie, Fergus McInnes:
A comparative study of continuous speech recognition using neural networks and hidden Markov models. 369-372 - Dimitrios A. Gaganelis, Eleftherios D. Frangoulis:
A novel approach to speaker verification. 373-376 - Tomoko Matsui, Sadaoki Furui:
A text-independent speaker recognition method robust against utterance variations. 377-380 - Aaron E. Rosenberg, Chin-Hui Lee, Sedat Gokcen:
Connected word talker verification using whole word hidden Markov models. 381-384 - Younès Bennani, Patrick Gallinari:
On the use of TDNN-extracted features information in talker identification. 385-388 - Laszlo Rudasi, Stephen A. Zahorian:
Text-independent talker identification with neural networks. 389-392 - John Oglesby, John S. D. Mason:
Radial basis function networks for speaker recognition. 393-396 - Michael J. Carey, Eluned S. Parris, John S. Bridle:
A speaker verification system using alpha-nets. 397-400 - Richard C. Rose, J. Fitzmaurice, Edward M. Hofstetter, Douglas A. Reynolds:
Robust speaker identification in noisy environments using noise adaptive speaker models. 401-404 - Alan L. Higgins, Lawrence G. Bahler:
Text-independent speaker verification by discriminator counting. 405-408 - Jesse W. Fussell:
Automatic sex identification from short segments of speech. 409-412 - Eleftherios D. Frangoulis:
Isolated word recognition in noisy environment by vector quantization of the HMM and noise distributions. 413-416 - Petros Maragos:
Fractal aspects of speech signals: dimension and interpolation. 417-420 - Petros Maragos, Thomas F. Quatieri, James F. Kaiser:
Speech nonlinearities, modulations, and energy operators. 421-424 - Brent Townshend:
Nonlinear prediction of speech. 425-428 - Kuldip K. Paliwal, Man Mohan Sondhi:
Recognition of noisy speech using cumulant-based linear prediction analysis. 429-432 - Les E. Atlas, Patrick J. Loughlin, James W. Pitton:
Truly nonstationary techniques for the analysis and display of voiced speech. 433-436 - Ronald P. Cohn:
Robust voiced/unvoiced speech classification using a neural net. 437-440 - Thea B. Ghiselli-Crippa, Amro El-Jaroudi:
A fast neural net training algorithm and its application to voiced-unvoiced-silence classification of speech. 441-444 - Krishna S. Nathan, Harvey F. Silverman:
Classification of unvoiced stops based on formant transitions prior to release. 445-448 - Shubha Kadambe, Gloria Faye Boudreaux-Bartels:
A comparison of a wavelet functions for pitch detection of speech signals. 449-452 - Alex I. C. Monaghan, D. Robert Ladd:
Manipulating synthetic intonation for speaker characterisation. 453-456 - Shiufun Cheung, Jae S. Lim:
Combined multi-resolution (wideband/narrowband) spectrogram. 457-460 - David Rainton:
Speech recognition-a time-frequency subspace filtering based approach. 461-464 - David J. Pepper, Mark A. Clements:
On the phonetic structure of a large hidden Markov model. 465-468 - Neri Merhav, Yariv Ephraim:
Hidden Markov modeling using the most likely state sequence. 469-472 - Andrej Ljolje, Michael Riley:
Automatic segmentation and labeling of speech. 473-476 - Gabriele C. Hegerl, Harald Höge:
Numerical simulation of the glottal flow by a model based on the compressible Navier-Stokes equations. 477-480 - Sunil K. Gupta, Juergen Schroeter:
Low update rate articulatory analysis/synthesis of speech. 481-484 - Mazin G. Rahim, Willem B. Kleijn, Juergen Schroeter, Colin C. Goodyear:
Acoustic to articulatory parameter mapping using an assembly of neural networks. 485-488 - Tetsunori Kobayashi, Masayuki Yagyu, Katsuhiko Shirai:
Application of neural networks to articulatory motion estimation. 489-492 - Shihua Wang, Andrew Sekey, Allen Gersho:
Auditory distortion measure for speech coding. 493-496 - Ulrich Halka:
A new objective quality measure for speech-coding systems based on the estimation of their nonlinear properties. 497-500 - Werner Verhelst:
On the quality of speech produced by impulse driven linear systems. 501-504 - Ajit L. Lalwani, D. G. Childers:
Modeling vocal disorders via formant synthesis. 505-508 - Simon M. Lucas, Robert I. Damper:
Syntactic neural networks for text-phonetics translation. 509-512 - Nathaniel Polish:
Mixed distance measures for synthetic speech evaluation. 513-516 - Peri Bhaskararao, Stephen J. Eady, John H. Esling:
Use of triphones for demisyllable-based speech synthesis. 517-520 - Horacio Martinez-Alfaro, José L. Contreras-Vidal:
A robust real-time pitch detector based on neural networks. 521-523 - Nickolas Yiourgalis, George K. Kokkinakis:
Text-to-speech system for Greek. 525-528 - Neri Merhav, Yariv Ephraim:
A Bayesian classification approach with application to speech recognition. 529-532 - Régis Cardin, Yves Normandin, Renato De Mori:
High performance connected digit recognition using maximum mutual information estimation. 533-536 - Yves Normandin, Salvatore D. Morgera:
An improved MMIE training algorithm for speaker-independent, small vocabulary, continuous speech recognition. 537-540 - Keh-Yih Su, Chin-Hui Lee:
Robustness and discrimination oriented speech recognition using weighted HMM and subspace projection approaches. 541-544 - Philip C. Woodland, David R. Cole:
Optimising hidden Markov models using discriminative output distributions. 545-548 - Pao-Chung Chang, Sin-Horng Chen, Biing-Hwang Juang:
Discriminative analysis of distortion sequences in speech recognition. 549-552 - Hitoshi Iwamida, Shigeru Katagiri, Erik McDermott:
Speaker-independent large vocabulary word recognition using an LVQ/HMM hybrid algorithm. 553-556 - Tatsuya Kawahara, Shuji Doshita:
Phoneme recognition by combining discriminant analysis and HMM. 557-560 - Stephen A. Zahorian, Danming Qian, Amir Jalali Jagharghi:
Acoustic-phonetic transformations for improved speaker-independent isolated word recognition. 561-564 - Arthur Nádas, David Nahamoo, Michael Picheny, J. Powell:
An iterative 'flip-flop' approximation of the most informative split in the construction of decision trees. 565-568 - Douglas B. Paul, James K. Baker, Janet M. Baker:
On the interaction between true source, training, and testing language models. 569-572 - Hermann Ney:
Speech recognition in a neural network framework: discriminative training of Gaussian models and mixture densities as radial basis functions. 573-576 - Robert J. McAulay, Thomas F. Quatieri:
Sine-wave phase coding at low data rates. 577-580 - Holger Carl, Bernd W. Kolpatzik:
Speech coding using nonstationary sinusoidal modelling and narrow-band basis functions. 581-584 - Luis A. Hernández Gómez, Francisco Javier Casajús-Quirós, Carmen García-Mateo, Javier Ortega-Garcia:
Real-time implementation and evaluation of variable rate CELP coders. 585-588 - Jesper Haagen, Henrik Nielsen, Steffen Duus Hansen:
A 2.4 kbps high-quality speech coder. 589-592 - Alan McCree, Thomas P. Barnwell III:
A new mixed excitation LPC vocoder. 593-596 - Jose Antonio Martins, Fábio Violaro:
Low bit rate LPC vocoders using vector quantization and interpolation. 597-600 - Y. J. Liu:
A robust 400-bps speech coder against background noise. 601-604 - Philippe Jeanrenaud, Patrick Peterson:
Segment vocoder based on reconstruction with natural segments. 605-608 - David P. Kemp, John S. Collura, Thomas E. Tremain:
Multi-frame coding of LPC parameters at 600-800 bps. 609-612 - Yuhang Wu, Henrik B. Hansen, Knud J. Larsen, Henrik Nielsen, John Aasted Sørensen:
High performance coder: a possible candidate for the GSM half-rate system. 613-616 - C. Gruet, F. Pommaret, M. Delprat:
Experiments with a regular pulse CELP coder for the pan European half rate channel. 617-620 - Giovanni Fausto Andreotti, V. Maiorano, Luigi Vetrano:
A 6.3 kb/s CELP codec suitable for half-rate system. 621-624 - Hans Hermansson, Tor Björn Minde, Jonas Ahlberg, P. Lundqvist:
A speech codec for cellular radio at a gross bit rate of 11.4 kb/s. 625-628 - Siu-Wah Wong:
An evaluation of 6.4 kbit/s speech codecs for Inmarsat-M system. 629-632 - Yoav Medan:
Using super resolution pitch in waveform speech coders. 633-636 - Frank H. Wu, Keshab K. Parhi, Kalyan Ganesan:
Neural network vector quantizer design using sequential and parallel learning techniques. 637-640 - Rajiv Laroia, Nam Phamdo, Nariman Farvardin:
Robust and efficient quantization of speech LSP parameters using structured vector quantizers. 641-644 - R. L. Joshi, P. G. Poonacha:
A new MMSE encoding algorithm for vector quantization. 645-648 - Franck Poirier:
Improving the training and testing speed and the ability of generalization in learning vector quantization-DVQ. 649-652 - D. H. Lee, David L. Neuhoff, Kuldip K. Paliwal:
Cell-conditioned multistage vector quantization. 653-656 - John Grass, Peter Kabal:
Methods of improving vector-scalar quantization of LPC coefficients. 657-660 - Kuldip K. Paliwal, Bishnu S. Atal:
Efficient vector quantization of LPC parameters at 24 bits/frame. 661-664 - Fritz Class, Alfred Kaltenmeier, Peter Regel:
Soft-decision vector quantization based on the Dempster/Shafer theory. 665-668 - Lu Chang, M. M. Bayoumi:
An economical binary tree structure for vector quantization. 669-672 - Ngoc-Ai Lu, Darryl Morrell:
VQ codebook design using improved simulated annealing algorithms. 673-676 - François J. Malassenet, Russell M. Mersereau:
Wavelet representations and coding of self-affine signals. 677-680 - Rosario Drogo de Iacovo, Daniele Sereno:
Embedded CELP coding for variable bit-rate between 6.4 and 9.6 kbit/s. 681-684 - M. G. Easton, Colin C. Goodyear:
A CELP codebook and search technique using a Hopfield net. 685-688 - Patrick Kenny, Rene Hollan, Vishwa Gupta, Matthew Lennig, Paul Mermelstein, Douglas D. O'Shaughnessy:
A*-admissible heuristics for rapid lexical access. 689-692 - Douglas B. Paul:
Algorithms for an optimal A* search and linearizing the search in the stack decoder. 693-696 - Steve Austin, Richard M. Schwartz, Paul Placeway:
The forward-backward search algorithm. 697-700 - Richard M. Schwartz, Steve Austin:
A comparison of several approximate algorithms for finding multiple (N-best) sentence hypotheses. 701-704 - Frank K. Soong, Eng-Fong Huang:
A tree-trellis based fast search for finding the N-best sentence hypotheses in continuous speech recognition. 705-708 - Eduardo Lleida, José B. Mariño, Climent Nadeu, Joan Salavedra:
Demisyllable-based HMM spotting for continuous speech recognition. 709-712 - Victor Zue, James R. Glass, David Goodine, Hong C. Leung, Michael S. Phillips, Joseph Polifroni, Stephanie Seneff:
Integration of speech recognition and natural language processing in the MIT VOYAGER system. 713-716 - Shinta Kimura, Yasushi Yamazaki:
Boosting the accuracy of a 100000-word Japanese speech recognition system. 717-720 - Michio Okada:
A unification-grammar-directed one-pass search algorithm for parsing spoken language. 721-724 - Yifan Gong, Jean-Paul Haton, Feriel Mouria:
Continuous speech recognition based on high plausibility regions. 725-728 - Roberto Pieraccini, Chin-Hui Lee, Egidio P. Giachin, Lawrence R. Rabiner:
Complexity reduction in a large vocabulary speech recognizer. 729-732 - Evaggelos Dermatas, Nikolaos D. Fakotakis, George K. Kokkinakis:
Fast endpoint detection algorithm for isolated word recognition in office environment. 733-736 - Michael D. Riley:
A statistical model for generating pronunciation networks. 737-740 - Robert W. P. Luk, Robert I. Damper:
A novel approach to inferring letter-phoneme correspondences. 741-744 - Bert Van Coile:
Inductive learning of pronunciation rules with the Depes system. 745-748 - Christine Tuerk, Peter Monaco, Tony Robinson:
The development of a connectionist multiple-voice text-to-speech system. 749-752 - V. Vinod Kumar, Stanley C. Ahalt, Ashok K. Krishnamurthy:
Phonetic to acoustic mapping using recurrent neural networks. 753-756 - Rijk Willemse, Louis Boves:
Context free wild card parsing in a text-to-speech system. 757-760 - Kirk P. H. Sullivan, Robert I. Damper:
Speech synthesis by analogy: recent advances and results. 761-764 - Masanobu Abe:
A segment-based approach to voice conversion. 765-768 - Kenji Matsui, Stephen D. Pearson, Kazue Hata, Takahiro Kamai:
Improving naturalness in text-to-speech synthesis using natural glottal source. 769-772 - H. Loman, Johan de Veth, Lou Boves:
The analysis of Dutch nasal consonants using robust ARMA analysis: consequences for rule development in speech synthesis. 773-776 - Ajit L. Lalwani, D. G. Childers:
A flexible formant synthesizer. 777-780 - Hugo C. van Leeuwen, Enrico te Lindert:
Speech Maker: text-to-speech synthesis based on a multi-level, synchronized data structure. 781-784 - Ye-Yi Wang, Alex Waibel:
A connectionist model for dialog processing. 785-788 - Natividad Prieto, Enrique Vidal:
Automatic learning of structural language models. 789-792 - Alex Waibel, Ajay N. Jain, Arthur E. McNair, Hiroaki Saito, Alexander G. Hauptmann, Joe Tebelskis:
JANUS: a speech-to-speech translation system using connectionist and symbolic processing strategies. 793-796 - Renato De Mori, Roland Kuhn, Gianni Lazzari:
A probabilistic approach to person-robot dialogue. 797-800 - Toshiyuki Takezawa, Kenji Kita, Junko Hosaka, Tsuyoshi Morimoto:
Linguistic constraints for continuous speech recognition in goal-directed dialogue. 801-804 - Allen L. Gorin, Stephen E. Levinson, A. N. Gertner:
Adaptive acquisition of spoken language. 805-808 - Bernard Mérialdo:
Tagging text with a probabilistic model. 809-812 - Masahide Sugiyama:
Automatic language recognition using acoustic features. 813-816 - Michael I. Savic, Elena Acosta, Sunil K. Gupta:
An automatic language identification system. 817-820 - Shigeru Ono:
Syllable structure parsing for continuous speech recognition. 821-824 - Hermann Ney, Ute Essen:
On smoothing techniques for bigram-based natural language modelling. 825-828 - Alexander I. Rudnicky, Jean-Michel Lunati, Alexander M. Franz:
Spoken language recognition in an office management domain. 829-832 - Francis Kubala, Richard M. Schwartz:
A new paradigm for speaker-independent training. 833-836 - David J. B. Pearce, Lynn C. Wood:
Adaptation of reference speaker HMM word models for speaker-independent recognition. 837-840 - Akihiro Imamura:
Speaker-adaptive HMM-based speech recognition with a stochastic speaker classifier. 841-844 - Hiroaki Hattori:
Speaker adaptation based on Markov modeling of speakers in speaker-independent speech recognition. 845-848 - Marco Ferretti, Anna Maria Mazza:
Fast speaker adaptation: some experiments on different techniques for codebook and HMM parameters estimation. 849-852 - Satoshi Nakamura, Toshio Akabane:
A neural speaker model for speaker clustering. 853-856 - Koichi Shinoda, Ken-ichi Iso, Takao Watanabe:
Speaker adaptation for demi-syllable based continuous density HMM. 857-860 - Eleftherios D. Frangoulis, Victoria Sgardoni:
A novel speaker adaptation approach for continuous densities HMM's. 861-864 - William A. Rozzi, Richard M. Stern:
Speaker adaptation in continuous speech recognition via estimation of correlated mean vectors. 865-868 - Douglas A. Reynolds, Larry P. Heck:
Integration of speaker and speech recognition systems. 869-872 - Herbert Gish, Man-Hung Siu, Jan Robin Rohlicek:
Segregation of speakers for speech recognition and speaker identification. 873-876 - X. D. Huang, Kai-Fu Lee:
On speaker-independent, speaker-dependent, and speaker-adaptive speech recognition. 877-880 - Melvyn J. Hunt, Stephen M. Richardson, D. Charles Bateman, Alain Piau:
An investigation of PLP and IMELDA acoustic representations and of their potential for combination. 881-884 - Yasuhiro Komori, Kaichiro Hatazaki:
Robustness of a feature based phoneme segmentation system to speaker independent and continuous speech. 885-888 - Hsiao-Wuen Hon, Kai-Fu Lee:
CMU robust vocabulary-independent speech recognition system. 889-892 - Alejandro Acero, Richard M. Stern:
Robust speech recognition by normalization of the acoustic space. 893-896 - Maha Kadirkamanathan, Andrew P. Varga:
Simultaneous model re-estimation from contaminated data by "composed hidden Markov modeling". 897-900 - John H. L. Hansen:
Speech enhancement employing adaptive boundary detection and morphological based spectral constraints. 901-904 - Yoichi Takebayashi, Hiroyuki Tsuboi, Hiroshi Kanazawa:
A robust speech recognition system using word-spotting with noise immunity learning. 905-908 - Adoram Erell, Mitch Weintraub:
Pitch-aided spectral estimation for noise-robust speech recognition. 909-912 - Alberto D. Berstein, Ilan D. Shallom:
An hypothesized Wiener filtering approach to noisy speech recognition. 913-916 - V. L. Beattie, Steve J. Young:
Noisy speech recognition using hidden Markov model state-based filtering. 917-920 - Beth A. Carlson, Mark A. Clements:
Application of a weighted projection measure for robust hidden Markov model based speech recognition. 921-924 - Chafic Mokbel, Gérard Chollet:
Word recognition in the car-speech enhancement/spectral transformations. 925-928 - Kazumi Ohkura, Masahide Sugiyama:
Speech recognition in a noisy environment using a noise reduction neural network and a codebook mapping technique. 929-932 - Helge B. D. Sørensen:
A cepstral noise reduction multi-layer neural network. 933-936 - J. A. Naylor, J. Porter:
An effective speech separation system which requires no a priori information. 937-940 - Régine Le Bouquin, Gérard Faucon:
Maximum likelihood noise cancellation with spectral constraints. 941-944 - B. Yegnanarayana, Hema A. Murthy, V. R. Ramachandran:
Processing of noisy speech using modified group delay functions. 945-948 - Y. H. Gu, W. M. G. van Bokhoven:
Co-channel speech separation using frequency bin non-linear adaptive filtering. 949-952 - Hong Wang, Fumitada Itakura:
An approach to dereverberation using multi-microphone sub-band envelope estimation. 953-956 - Chung Wu, Vien Nguyen, H. Sabrin, William M. Kushner, John N. Damoulakis:
Fast self-adapting broadband noise removal in the cepstral domain. 957-960 - Yan Ming Cheng, Douglas D. O'Shaughnessy:
Speech enhancement based conceptually on auditory evidence. 961-964 - Walter Etter, George S. Moschytz, Daniel Graupe:
Adaptive noise reduction using discrimination functions. 965-968 - Hideaki Yamada, Hong Wang, Fumitada Itakura:
Recovering of broadband reverberant speech signal by sub-band MINT method. 969-972 - Hidefumi Kobatake, Kaoru Gyoutoku, Sheng Li:
Enhancement of noisy speech by maximum likelihood estimation. 973-976 - Duncan Bees, Maier Blostein, Peter Kabal:
Reverberant speech enhancement using cepstral processing. 977-980 - Luc Gagnon, William F. McGee:
Speech enhancement using resonator filterbanks. 981-984 - Ted H. Applebaum, Brian A. Hanson:
Regression features for recognition of speech in quiet and in noise. 985-988 - Jeroen G. van Velden, Guido F. Smoorenburg:
Vowel recognition in noise for male, female and child voices. 989-992 - Arthur Schaub, Peter Straub:
Spectral sharpening for speech enhancement/noise reduction. 993-996 - Yariv Ephraim:
On minimum mean square error speech enhancement. 997-1000 - Edwin Hsing-Mean Sha, Kenneth Steiglitz:
Reconfigurability and reliability of systolic/wavefront arrays. 1001-1004 - Lih-Gwo Jeng, Liang-Gee Chen:
A globally static rate optimal scheduling for recursive DSP algorithms. 1005-1008 - Shih-Fu Hsieh, K. J. Ray Liu:
A residual-based selective window for robust recursive least squares estimation. 1009-1012 - Marc Moonen, Joos Vandewalle:
A systolic array for recursive least squares computations. 1013-1016 - Shih-Fu Hsieh, Kuo J. Ray Liu, Kung Yao:
A unified SQRT-free rank-1 up/down-dating approach for recursive least-squares problems. 1017-1020 - Stephen P. S. Lam:
A systolic implementation of the Jacobi algorithm. 1021-1024 - Domingo Rodríguez:
Tensor product algebra as a tool for VLSI implementation of the discrete Fourier transform. 1025-1028 - Heonchul Park, V. K. Prasanna Kumar:
VLSI architectures for computing the arithmetic Fourier transform. 1029-1032 - C. F. T. Tang, K. J. Ray Liu, Steven A. Tretter:
On systolic arrays for recursive complex Householder transformations with applications to array processing. 1033-1036 - William G. Bliss:
The VLSI area of high throughput linear filters of moderate order using binary weighted arithmetic. 1037-1040 - Keqin Gao, M. Omair Ahmad, M. N. S. Swamy:
Neural LS estimator with a non-quadratic energy function. 1041-1044 - Thomas H. Reiss:
Learning image invariants with the Volterra connectionist model. 1045-1048 - Kostas I. Diamantaras, Sun-Yuan Kung:
An unsupervised neural model for oriented principal component extraction. 1049-1052 - Martin D. Emmerson, Robert I. Damper, Anthony J. G. Hey, Colin Upstill:
Fault tolerance and redundancy of neural nets for the classification of acoustic data. 1053-1056 - Anargyros Krikelis:
A novel massively parallel associative processing architecture for the implementation of artificial neural networks. 1057-1060 - Yu Hen Hu, Qiuzhen Xue, Willis J. Tompkins:
Structural simplification of a feed-forward, multilayer perceptron artificial neural network. 1061-1064 - James Ting-Ho Lo:
Backestimation for training multilayer perceptrons. 1065-1068 - Jin-Shiuh Taur, Sun-Yuan Kung:
Comparison of several learning subspace methods for classification. 1069-1072 - Sun-Yuan Kung, W. D. Mao:
Competition-based supervised learning algorithm for nonlinear discriminant functions. 1073-1076 - Kevin W. Leary, Chris Cavigioli:
The ADSP-21020: an IEEE floating point and fixed point DSP for HLL programming. 1077-1080 - Fred J. Taylor, Glenn S. Zelniker, Jeremy C. Smith, Jon Mellott:
The Gauss machine-a DSP processor with a high RNS content. 1081-1084 - R. Bhatia, M. Furuta, J. Ponce:
A quasi radix-16 VLSI processor. 1085-1088 - Ray Simar Jr.:
The TMS320C40: a DSP for parallel processing. 1089-1092 - Paul H. Siegel, C. Bernard Shung, Thomas D. Howell, Hemant K. Thapar:
Exact bounds for Viterbi detector path metric differences. 1093-1096 - Gennady Feygin, Patrick Glenn Gulak, Paul Chow:
Generalized cascade Viterbi decoder-a locally connected multiprocessor with linear speed-up. 1097-1100 - B. K. Min, Nicolas Demassieux:
A versatile architecture for VLSI implementation of the Viterbi algorithm. 1101-1104 - M. Ott, Liyanage C. De Silva, L. Satou, Kiyoharu Aizawa, Mitsutoshi Hatori:
VIWOB: an interactive programming environment for distributed simulations. 1105-1108 - Steve Beling, Kevin W. Leary, Greg Yukna:
An in-circuit signal analyzer for mixed signal digital signal processor. 1109-1112 - Brian Barrera, Edward A. Lee:
Multirate signal processing in Comdisco's SPW. 1113-1116 - Brian T. Kelley, Vijay K. Madisetti:
High speed migration of multidimensional seismic data. 1117-1120 - James O. Normile, Dan Wright:
Image compression using coarse grain parallel processing. 1121-1124 - Mary Jane Irwin, Robert Michael Owens, Thomas P. Kelliher, Kin-Ki Leung, Mohan Vishwanath:
The arithmetic cube II: a second generation VLSI DSP processor. 1125-1128 - G. Adams, A. M. Finn, M. F. Griffin:
A fast implementation of the complex singular value decomposition on the Connection Machine. 1129-1132 - Keshab K. Parhi, Joosang Lee:
Register allocation for design of data format converters. 1133-1136 - Francisco Argüello, Ramon Doallo, Javier D. Bruguera, Emilio L. Zapata:
Design of a constant geometry fast Hartley transformer. 1137-1140 - Kenneth Zeger, Allen Gersho:
A parallel processing algorithm for vector quantizer design based on subpartitioning. 1141-1143 - Hon Keung Kwan:
Multilayer feedbackward neural networks. 1145-1148 - Shu-Jen Yeh, Henry Stark:
A new learning rule for multilayer neural net. 1149-1152 - Zhi-Jian (Alex) Mou, Francis Jutand:
A high-speed low-cost DCT architecture for HDTV applications. 1153-1156 - N. Ranganathan, Rajiv Mehrotra, Kameswara Rao Namuduri:
An architecture to implement multiresolution. 1157-1160 - Peter Dean Schuler, R. H. S. Hardy, Vladimir Cuperman:
A custom VLSI architecture for low-delay speech coding. 1161-1164 - Steven C. Bass, G. M. Butler, R. L. Williams, F. Barlos, D. R. Miller:
A bit-serial, floating point CORDIC processor in VLSI. 1165-1168 - Yukio Kadowaki, Syogo Nakamura, Akiyoshi Kawahashi:
An implementation of a hierarchical generalized transversal digital filter. 1169-1172 - Bing J. Sheu, Wai-Chi Fang:
Real-time high-ratio image compression using adaptive VLSI neuroprocessors. 1173-1176 - Marios D. Dikaiakos, Kenneth Steiglitz:
Comparison of tree and straight-line clocking for long systolic arrays. 1177-1180 - Takeshi Oto, Kazukuni Kitagaki, Tatsuhiko Demura, Yoshitsugu Araki, Tomoji Takada:
A new DSP architecture suited for image analysis. 1181-1184 - Fang Lu, Henry Samueli:
A reconfigurable decision-feedback equalizer chip set architecture for high bit-rate QAM digital modems. 1185-1188 - O. C. McNally, William P. Marnane, John V. McCanny:
Design and test of a bit parallel 2nd order IIR filter structure. 1189-1192 - Hiroshi Miyanaga, Hironori Yamauchi, Kazuhiro Matsuda:
A real-time 256×256 point two-dimensional FFT single-chip processor. 1193-1196 - Hironori Yamauchi, Yutaka Tashiro, Toshihiro Minami, Yutaka Suzuki:
A highly-parallel single-chip DSP architecture for video signal processing. 1197-1200 - Vijay K. Jain, Gibert E. Perez, Jeffrey M. Wills:
Novel reciprocal and square-root VLSI cell: architecture and application to signal processing. 1201-1204 - Stephen E. McQuillan, John V. McCanny:
A VLSI architecture for multiplication, division and square root. 1205-1208 - Willem Steenaart, J. Y. Zhang:
Mapping recursive algorithms onto systolic architectures. 1209-1212 - Chen-Yi Lee, Francky Catthoor, Hugo De Man:
Breaking the bottleneck of sequential decoding for high-speed digital communication. 1213-1216 - Wayne P. Burleson:
The partitioning problem on VLSI arrays: I/O and local memory complexity. 1217-1220 - Tung-Hao Huang, Chi-Min Liu, Chein-Wei Jen:
A high-level synthesizer for VLSI array architectures dedicated to digital signal processing. 1221-1224 - Julian G. Payne, Stephen M. Kruse, Stewart G. Smith, Ralph W. Morgan:
An ASIC DSP engine compiler: philosophy and applications. 1225-1228 - Lynette Liu, Toshiaki Yoshino, Steven Sprouse, Rajeev Jain:
Computer-aided design of high-speed lattice wave digital filter integrated circuits. 1229-1232 - Kwok K. Chau, I-Fay Wang, Creighton L. Eldridge:
VLSI implementation of a 2-D DCT in a compiler. 1233-1236 - Sanjay V. Rajopadhye, Sayfe Kiaei:
A folding transformation for VLSI IIR filter array design. 1237-1240 - Xiaoxiong Zhong, Sanjay V. Rajopadhye:
Synthesizing fully efficient systolic arrays. 1241-1244 - Joseph T. Buck, Soonhoi Ha, Edward A. Lee, David G. Messerschmitt:
Multirate signal processing in Ptolemy. 1245-1248 - C. P. Hong, Thomas P. Barnwell III:
Compilation for interprocessor communication in clock-skewed parallel processing system. 1249-1252 - Ching-Yi Wang, Keshab K. Parhi:
Dedicated DSP architecture synthesis using the MARS design system. 1253-1256 - Michael Takefman, Paul Chow:
A streamlined DSP microprocessor architecture. 1257-1260 - Ariela Zeira, Peter M. Schultheiss:
Thresholds and related problems in time delay estimation. 1261-1264 - Jitendra K. Tugnait:
Time delay estimation with unknown spatially correlated Gaussian noise. 1265-1268 - Benoît Champagne, Moshe Eizenman, Subbarayan Pasupathy:
Maximum likelihood estimation of time-varying delays in the presence of directional interference. 1269-1272 - Billur Barshan, Roman Kuc:
Active sonar for obstacle localization using envelope shape information. 1273-1276 - Olivier Michel, Henri Clergeot:
Multiple source tracking using a high resolution method. 1277-1280 - Jeffrey L. Krolik, George Henry Niezgoda, David N. Swingler:
A bootstrap approach for evaluating source localization performance on real sensor array data. 1281-1284 - Maria-João Rendas, José M. F. Moura:
Ambiguity analysis in source localization with unknown signals. 1285-1288 - David Jaarsma, R. Alan Davis:
Joint estimation of time delay and Doppler ratio from broadband data using discrete estimation techniques. 1289-1292 - Yaakov Bar-Shalom, Francesco Palmieri, Anil Kumar, Hemchandra M. Shertukde:
Analysis of wide-band cross-correlation for target detection and time delay estimation. 1293-1296 - Isabel M. G. Lourtie, G. Clifford Carter:
Likelihood ratio detection in a random multipath channel. 1297-1300 - Christian J. Musso, Claude Jauffret:
Linear maximum likelihood estimator. 1301-1304 - Doron Kletter, Peter M. Schultheiss, Hagit Messer:
Optimal detection of non-Gaussian random signals in Gaussian noise. 1305-1308 - Larry Marple, Tom Brotherton:
Detection and classification of short duration underwater acoustic signals by Prony's method. 1309-1312 - Mordechai Frisch, Hagit Messer:
Detection of a transient signal of unknown scaling and arrival time using the discrete wavelet transform. 1313-1316 - Bülent Baygün, Alfred O. Hero III:
Tradeoffs between detection and estimation for multiple signals. 1317-1320 - Marc Solal, Denis Pillon, Stéphane Brasseur:
Simultaneous detection and target motion analysis from conventional passive beamforming outputs. 1321-1324 - Jean-Michel Passerieux:
Bearings-only tracking with non Gaussian measurement errors. 1325-1328 - Bernard C. Picinbono, M. P. Boyer:
A new approach of decentralized detection. 1329-1332 - John D. Gorman, Alfred O. Hero III:
On the application of Cramer-Rao type lower bounds for constrained estimation. 1333-1336 - Sanjay K. Mehta, Edward L. Titlebaum:
A new method for measurement of the target and channel scattering functions using Costas arrays and other frequency hop signals. 1337-1340 - Steven D. Blostein, Yong Liu:
On sequentially detecting an abrupt change in a stochastic process. 1341-1344 - Mati Wax:
Detection and localization of multiple sources in noise with unknown covariance. 1345-1348 - James C. Preisig:
A robust adaptive matched field processor based upon a minmax criterion. 1349-1352 - Yoram Bresler:
On the resolution capacity of wideband sensor arrays: further results. 1353-1356 - G. T. Zunich, Lloyd J. Griffiths:
A robust method in adaptive array processing for random phase errors. 1357-1360 - Keith Blanton, James H. McClellan:
New search algorithm for minimum redundancy linear arrays. 1361-1364 - J. Pierre, Mostafa Kaveh:
Experimental performance of calibration and direction-finding algorithms. 1365-1368 - Daniel R. Fuhrmann:
Estimation of sensor gain and phase using known field covariance. 1369-1372 - Barry D. Van Veen:
Convergence of the SMI algorithm in partially adaptive linearly constrained beamformers. 1373-1376 - Anthony J. Weiss, Benjamin Friedlander:
Performance analysis of spatial smoothing with interpolated arrays. 1377-1380 - David D. Feldman, Lloyd J. Griffiths:
A constraint projection approach for robust adaptive beamforming. 1381-1384 - G. Adams, M. F. Griffin, G. W. Stewart:
Direction-of-arrival estimation using the rank-revealing URV decomposition. 1385-1388 - R. Lynn Kirlin, Weixiu Du:
Design of transformation matrices for array processing and spectral analysis. 1389-1392 - Robert C. Turner:
Resolving line-like targets using MUSIC in active sonar. 1393-1396 - George Henry Niezgoda, Jeffrey L. Krolik:
The application of adaptive multichannel spectral estimation for broadband array processing. 1397-1400 - Norman A. Lehtomaki:
Implementation of a robustness constraint in adaptive beamforming. 1401-1404 - Xinping Huang, James P. Reilly, Kon Max Wong:
Optimal design of linear array of sensors. 1405-1408 - Victor A. N. Barroso, José M. F. Moura:
Maximum likelihood beamforming in the presence of outliers. 1409-1412 - Fu Li, Richard J. Vaccaro:
Performance degradation of DOA estimators due to unknown noise fields. 1413-1416 - Constantino Rago, Carlos H. Muravchik:
Estimation of the direction of arrival of water waves. 1417-1420 - Salah Bourennane, B. Faure:
Localization of underwater wide band acoustic sources. 1421-1423 - Qiang Wu, Kon Max Wong:
A robust estimator for direction finding. 1425-1428 - Chun-Chao Wang, James A. Cadzow:
Direction-finding with sensor gain, phase and location uncertainty. 1429-1432 - Siew Kok Hui, Meng Hwa Er:
Fast iterative algorithm for harmonics retrieval. 1433-1436 - Serge Prosperi:
New passive tracking algorithms using power information with application to passive sonar. 1437-1440 - Jean-Pierre Le Cadre, Olivier Zugmeyer:
Temporal integration for array processing. 1441-1444 - James G. Kelly, Robert N. Carpenter, John A. Tague, Nicholas K. Haddad:
Optimum classification with active sonar: new theoretical results. 1445-1448 - Zoran Kostic, M. Ibrahim Sezan, Edward L. Titlebaum:
Estimation of the parameters of a multipath channel using set-theoretic deconvolution. 1449-1452 - Richard Lee Culver, Suzanne T. McDaniel:
Bistatic ocean surface reverberation simulation. 1453-1456 - Ömür Bozma, Roman Kuc:
Wide-band analysis of reflections from rough surfaces. 1457-1460 - K. C. Ho, P. C. Ching, Yiu Tong Chan:
Adaptive time delay estimation in noisy environments. 1461-1464 - Yiu Tong Chan, J. J. Towers:
Passive localization from Doppler shifted frequency measurements. 1465-1468 - Zhen-Ya He, Li-Hua Li:
High-resolution multipath time delay estimation using a neural network. 1469-1472 - Shaolin Li, Peter M. Schultheiss:
Source localization in underwater acoustic channels using random or deterministic signals. 1473-1476 - Xiaodong Gu, Max Wong:
A modified Ziv-Zakai lower bound and its application in array processing. 1477-1480 - Ian J. Morrison, Peter J. W. Rayner:
The application of Volterra series to signal estimation. 1481-1484 - Chintana Griffin:
A comparison study on the Wigner and Choi-Williams distributions for detection. 1485-1488 - Cameron M. Pike, John A. Tague, Edmund J. Sullivan:
Transient signal detection in multipath: a bispectral analysis approach. 1489-1492 - David Burton:
Acoustic transient classification of passive sonar signals by using vector quantization. 1493-1496 - Guanghan Xu, Seth D. Silverstein, Richard H. Roy III, Thomas Kailath:
Parallel implementation and performance analysis of beamspace ESPRIT. 1497-1500 - Preeti Rao:
A robust narrowband transient signal detector. 1501-1504 - Oscar C. Au, John B. Thomas:
On transformation noise as a model for correlated noise. 1505-1508 - W. Y. Chen:
Adaptive pilot filtering for LMS algorithm. 1509-1512 - Bernard Mulgrew:
An adaptive whitened matched filter. 1513-1516 - Graham W. Pulford, Rodney A. Kennedy, Brian D. O. Anderson:
Neural network structure for emulating decision feedback equalisers. 1517-1520 - Nevio Benvenuto, Michele Marchesi, Francesco Piazza, Aurelio Uncini:
Non linear satellite radio links equalized using blind neural networks. 1521-1524 - Shigeji Ikeda, Akihiko Sugiyama:
A fast convergence algorithm for adaptive FIR filters with coarsely located taps. 1525-1528 - William A. Sethares, Rodney A. Kennedy, Zhenguo Gu:
An approach to blind equalization of non-minimum phase systems. 1529-1532 - Zhi Ding, C. Richard Johnson Jr., Rodney A. Kennedy:
Local convergence of 'globally convergent' blind adaptive equalization algorithms. 1533-1536 - Dimitrios Hatzinakos:
Carrier phase recovery issues in polyspectra-based equalizers. 1537-1540 - Sharlene Gee, Markus Rupp:
A comparison of adaptive IIR echo canceller hybrids. 1541-1544 - Farouk Laichi, Tyseer Aboulnasr, Willem J. D. Steenaart:
A new efficient implementation of the AIFIR echo tail canceller for ISDN applications. 1545-1548 - Francisco Lorenz Speranzini, José Manuel Páez-Borrallo:
Convergence analysis of Lk-norm echo cancellers for PSKL and QAM data transmission. 1549-1552 - Jonathon A. Chambers, Anthony G. Constantinides:
Adaptive notch filters from lossless bounded real all-pass functions for frequency tracking and line enhancing. 1553-1556 - Monica Murphy, Fred J. Taylor, Luke J. Smithwick:
Computer-aided DSP instruction. 1557-1560 - Michael Ansorge, U. Sjöström, I. Defilippis, Peter Balsiger, Fausto Pellandini:
On the automated symbolic design of wave digital filters. 1561-1564 - Sau-Gee Chen, Jih-Feng Lin:
Efficient implementation of the normalized recursive least-square lattice filter. 1565-1568 - Ferenc Kocsis:
A fully pipelined, high speed DFT architecture. 1569-1572 - Daniel Chein-Chung Chen, Russell H. Price:
A real-time TMS320C40 based parallel system for high rate digital signal processing. 1573-1576 - Horng-Dar Lin, C. Bernard Shung:
General in-place scheduling for the Viterbi algorithm. 1577-1580 - Duen-Jeng Wang, Yu Hen Hu:
Optimal scheduling of linear recurrence equations on a multiprocessor array. 1581-1584 - King-Wai Chow, Bede Liu:
On mapping signal processing algorithms to a heterogeneous multiprocessor system. 1585-1588 - Willard L. Eastman, Richard A. Games, Michael J. Sousa:
Multiple mainbeam sidelobe cancellation architectures for constrained and/or excess degrees of freedom processing. 1589-1592 - Alan Gatherer, Teresa H.-Y. Meng:
A robust parallel DFE using extended LMS. 1593-1596 - James L. Dixon, W. Y. Chen:
The implementation of an adaptive echo canceller for high-speed digital data transmission. 1597-1600 - Rom-Shen Kao:
A multiplier-free fast transform with efficient VLSI implementation for polynomial RNS processors. 1601-1604 - David B. Chester, Callie A. Riley, David H. Damerow, Clay Olmstead:
VLSI implementation of a wide band, high dynamic range digital drop receiver. 1605-1608 - Srinivasa R. Malladi, Seshagiri R. Myneni, Pardha V. Pothana, Magdy A. Bayoumi:
A high speed pipelined FFT processor. 1609-1612 - Callie Riley, David B. Chester, Abbas Razavi, Fred J. Taylor, William Ricker:
High-decimation digital filters. 1613-1616 - David G. Long:
Finding the global minima of objective functions using modified Sturm sequences. 1617-1620 - C. W. Kim, Henrik V. Sorensen:
Approximately linear phase IIR polyphase filter design. 1621-1624 - Sachin P. Ghanekar, Sawasd Tantaratana, Lewis E. Franks:
Implementation of recursive filters using highly quantized periodically time-varying coefficients. 1625-1628 - Hon Keung Kwan, Ying Chun Lui:
Minimal normalized lattice structure for ARMA digital filter realization. 1629-1632 - Derek C. Cowley, Bahram Shafai:
A direct method for minimizing the roundoff noise in digital filter design. 1633-1636 - Jan Radecki, Janusz Konrad, Eric Dubois:
Design of finite wordlength IIR filters with prescribed magnitude, group delay and stability properties using simulated annealing. 1637-1640 - S. Radhakrishnan Pillai, William Robertson, William J. Phillips:
Subband filters using allpass structures. 1641-1644 - Ferenc Leeb:
Lattice wave digital filters with simultaneous conditions on amplitude and phase. 1645-1648 - Ramdas Kumaresan, C. Sidney Burrus:
Fitting a pole-zero filter model to arbitrary frequency response samples. 1649-1652 - N. K. Bose:
Vertex implication results for analysis of coefficient quantization errors in digital filters. 1653-1654 - Soo-Chang Pei, Jong-Jy Shyu:
Relationships among digital one/half band filters, low/high order differentiators and discrete/differentiating Hilbert transformers. 1657-1660 - David Baez-López:
Linear phase IIR digital filters using the Bernstein polynomial. 1661-1664 - Constantine Kotropoulos, Ioannis Pitas:
Constrained adaptive LMS L-filters. 1665-1668 - Thomas G. Marshall Jr.:
Error control coding of television signals by multidimensional filtering. 1669-1672 - Don H. Johnson, P. Srinivasa Rao:
On the existence of Gaussian noise [signal modelling]. 1673-1676 - Fredrik Gustafsson:
Optimal segmentation of signals in a linear regression framework. 1677-1680 - Vinay P. Sathe, P. P. Vaidyanathan:
Analysis of the effects of multirate filters on stationary random inputs, with application in adaptive filtering. 1681-1684 - Q. T. Zhang, H. S. Miao:
Sample distributions of the partial correlation coefficients of a complex autoregressive process. 1685-1688 - Arnab K. Shaw, Pradeep Misra:
Time domain identification of proper discrete systems from measured impulse response data. 1689-1692 - Mark L. Fowler, Leon H. Sibul:
Signal detection using group transforms. 1693-1696 - Larry P. Heck, James H. McClellan:
Mechanical system monitoring using hidden Markov models. 1697-1700 - Salvatore D. Morgera, Frédéric Simard:
Parameter estimation for a burst-noise channel. 1701-1704 - Richard M. Todd, J. R. Cruz:
Statistical analysis of a fast frequency estimator using a symbolic algebra system. 1705-1708 - Kenneth C. Chou, Stuart A. Golden, Alan S. Willsky:
Modeling and estimation of multiscale stochastic processes. 1709-1712 - Mahesan Niranjan, Visakan Kadirkamanathan:
A nonlinear model for time series prediction and signal interpolation. 1713-1716 - Don M. Raymond, Moustafa M. Fahmy:
On the multivariate distributions of order statistics of discrete-state processes. 1717-1720 - Ramachandra G. Shenoy, Nihal I. Wijeyesekera, Thomas W. Parks:
Interpolation of transients based on training data. 1721-1724 - Daniel M. Beyerbach, Hamid Nawab:
Principal components analysis of the short-time Fourier transform. 1725-1728 - Barry J. Sullivan, Hung-Chi Chang:
A generalized Landweber iteration for ill-conditioned signal restoration. 1729-1732 - Sergio D. Cabrera, John L. Brown Jr.:
Input reconstruction from oversampled multi-channel data. 1733-1736 - Joseph P. Noonan, B. Achour:
Iterative minimization of entropic stabilizing functionals in signal restoration. 1737-1740 - M. R. Bhatt, Uday B. Desai:
Orthogonal l1 technique for image restoration. 1741-1744 - Jonathan S. Abel, Julius O. Smith III:
Restoring a clipped signal. 1745-1748 - Akira Sano, Hiroyuki Tsuji, Hiromitsu Ohmori:
Signal extrapolation based on generalized singular value decomposition using prior information. 1749-1752 - Fu-Chun Zheng, Steve McLaughlin, Bernard Mulgrew:
Blind deconvolution algorithms based on 3rd- and 4th-order cumulants. 1753-1756 - Athina P. Petropulu, Chrysostomos L. Nikias:
Blind deconvolution based on signal reconstruction from partial information using higher-order spectra. 1757-1760 - Alfredo Restrepo Palacios, Alan C. Bovik:
Windowed locally monotonic regression. 1761-1764 - Wooshik Kim, Monson H. Hayes:
The phase retrieval problem in X-ray crystallography. 1765-1768 - Richard Schreier, W. Martin Snelgrove:
Stability in a general ΣΔ modulator. 1769-1772 - C. Cenker, Hans G. Feichtinger, H. Steier:
Fast iterative and non-iterative reconstruction of band-limited functions from irregular sampling values. 1773-1776 - P. P. Vaidyanathan:
The role of Smith-form decomposition of integer-matrices, in multidimensional multirate systems. 1777-1780 - Kambiz Nayebi, Thomas P. Barnwell III, Mark J. T. Smith:
The design of perfect reconstruction nonuniform band filter banks. 1781-1784 - Jelena Kovacevic, Martin Vetterli:
Perfect reconstruction filter banks with rational sampling rate changes. 1785-1788 - Tor A. Ramstad, Joar P. Tanem:
Cosine-modulated analysis-synthesis filterbank with critical sampling and perfect reconstruction. 1789-1792 - Ravinder David Koilpillai, P. P. Vaidyanathan:
New results on cosine-modulated FIR filter banks satisfying perfect reconstruction. 1793-1796 - Henrique S. Malvar:
Extended lapped transforms: fast algorithms and applications. 1797-1800 - Truong Q. Nguyen:
On the problem of reconstructing a segment of a wideband signal using a digital filter bank. 1801-1804 - Bennett Levitan, Gershon Buchsbaum:
Architecture-dependent properties of analysis multirate filter banks. 1805-1808 - Mukund Padmanabhan, Ken Martin:
Digital filter-banks for transform computations and other applications. 1809-1812 - Henry Samueli, Thu-ji Lin:
A VLSI architecture for a universal high-speed multirate FIR digital filter with selectable power-of-two decimation/interpolation ratios. 1813-1816 - Guoan Bi:
Minimisation of delay requirements for rational sampling rate alternating systems. 1817-1820 - G. Zhang, William F. McGee:
Windowing techniques in the design of resonator based frequency interpolation filter banks. 1821-1824 - Amir Averbuch, Samuel Itzikowitz, Tal Kapon:
Multifilters for radar tracking of a maneuvering target-Viterbi vs. IMM. 1825-1828 - William F. McGee, Genzao Zhang:
All-pass analysis-synthesis frequency interpolation filterbanks. 1829-1832 - Maurice G. Bellanger:
A survey of QR based fast least squares adaptive filters: from principles to realization. 1833-1836 - Mushtaq A. Syed:
QRD-based fast RLS multichannel adaptive algorithms. 1837-1840 - Hervé Dedieu, M. Hasler:
Error propagation in the recursive QRD LS filter. 1841-1844 - Peter Strobach:
The square-root Schur RLS adaptive filter. 1845-1848 - Karl Zhao, Fuyun Ling, John G. Proakis:
Multichannel Givens lattice adaptive algorithm. 1849-1852 - Carlos E. Davila:
Recursive total least squares algorithms for adaptive filtering. 1853-1856 - Terence R. Albert, Hana Abusalem, Michael D. Juniper:
Experimental results: detection and tracking of low SNR sinusoids using real-time LMS and RLS lattice adaptive line enhancers. 1857-1860 - Ki-Ho Kim, Edward J. Powers:
Analysis of initialization and numerical instability of fast RLS algorithms. 1861-1864 - Tülay Adali, Sasan H. Ardalan:
Fixed-point roundoff error analysis of the RLS algorithm with time-varying channels. 1865-1868 - Kay-Cheung Chew, James R. Zeidler, Walter H. Ku:
Tracking model of an adaptive lattice filter for a linear chirp signal in noise. 1869-1872 - André Gilloire:
Response of accelerated fast RLS adaptive filters to linear chirps. 1873-1876 - Demetrios G. Lainiotis, Pavlos K. Giannakopoulos, Sokratis K. Katsikas:
Optimal state estimation for uncertain, time varying systems with non-Gaussian initial state. 1877-1880 - Louis-Jérôme Leclerc, Peter H. Bauer:
A computer-aided test for global asymptotic stability of digital filters. 1881-1884 - Timo I. Laakso, Paulo S. R. Diniz, Iiro Hartimo:
General criterion for the absence of limit cycles in filter structures with a single quantizer. 1885-1888 - Tamal Bose, Mei-Qin Chen, Frank Brammer:
Stability of normal form digital filters with two's complement quantization. 1889-1892 - Bhaskar D. Rao:
Analysis of roundoff noise in floating point digital filters. 1893-1896 - José L. Sanz-González, Juan C. Herranz-Torés, Eloísa Calero-Pérez:
Optimal structures for high speed and low roundoff noise digital filters. 1897-1900 - Wai-Man Lam, Amy R. Reibman:
Asymptotic quantization for signal estimation with noisy channels. 1901-1904 - Newell D. Stacey, Richard L. Frost, Gene A. Ware:
Error spectrum shaping quantizers with non-ideal reconstruction filters and saturating quantizers. 1905-1908 - Hatice Örün Öztürk, Winser E. Alexander:
Sampling error analysis in perfect reconstruction transmultiplexers. 1909-1912 - Yunbiao Wang, Hari Krishna:
Error analysis of algorithms in linear prediction. 1913-1916 - Richard C. North, James R. Zeidler, Walter H. Ku, Terence R. Albert:
Arithmetic effects in adaptive lattice filters. 1917-1920 - Eric B. Hall, Gary L. Wise:
On a failure of multidimensional quantization. 1921-1924 - Bing Zeng, Timo I. Laakso, Iiro Hartimo, Yrjö Neuvo:
Error feedback for floating-point cascade-form digital filters. 1925-1928 - Aníbal R. Figueiras-Vidal, Antonio Artés-Rodríguez, Juan Antonio A. Hernández-Méndez:
Phase constraining algorithms for data blind equalization. 1929-1932 - Francesco Palmieri, R. Edward Croteau:
Adaptive channel equalization using generalized order statistic filters. 1933-1936 - Wen Tong, Eugene I. Plotkin, Dov Wulich, M. N. S. Swamy:
Self-synchronized signal controlled constrained notch filter for rejection of nonstationary interference. 1937-1940 - John W. Cook:
Linearity testing of multi-level transmitters for communication systems. 1941-1944 - Gregory W. Wornell:
Communication over fractal channels. 1945-1948 - Michael J. W. Hall, Ian G. Fuss, Langford B. White:
Digital communication and quantum phase detection. 1949-1951 - Wu Chou:
Sigma delta and multi-stage sigma delta modulation with inside loop dithering. 1953-1956 - Sang Ju Park, Robert M. Gray, Wu Chou:
Analysis of a sigma delta modulator with a multi-level quantizer and single-bit feedback. 1957-1960 - P. J. Fernandes, Mohsin M. Jamali, Subhash C. Kwatra, J. Budinger:
A reconfigurable pipelined transmultiplexer architecture. 1961-1964 - Sangil Park, Dion Messer:
Analysis and digital implementation of jointly adapting echo canceller, transversal equalizer and decision feedback equalizer for the V.32 modem. 1965-1968 - Takis Kasparis:
Frequency independent sinusoidal suppression using median filters. 1969-1972 - Richard J. Hartnett, Gloria Faye Boudreaux:
Design of efficient FIR filters using cyclotomic polynomial prefilters. 1973-1976 - Gregory W. Medlin, James F. Kaiser:
Bandpass digital differentiator design using quadratic programming. 1977-1980 - Truong Q. Nguyen:
The eigenfilter for the design of linear-phase filters with arbitrary magnitude response. 1981-1984 - Ashraf Alkhairy, Kevin G. Christian, Jae S. Lim:
Design of FIR filters by complex Chebyshev approximation. 1985-1988 - Peter J. Kootsookos, Robert R. Bitmead, Michael Green:
FIR (q) filter design without the linear phase constraint. 1989-1992 - Akiyoshi Kawahashi, Shogo Nakanura, Yoshihiko Horio, Yukio Kadowaki:
A simple method for designing a hierarchical structure transversal filter. 1993-1996 - Giancarlo Calvagno, Guido M. Cortelazzo, Gian Antonio Mian:
A comparison between complex approximation and multiple criterion optimization in FIR filter design. 1997-2000 - Sze Fong Yau:
FIR digital filtering of delta modulated signals. 2001-2004 - Kenneth E. Barner:
C-stack filters. 2005-2008 - Bing Zeng, Moncef Gabbouj, Yrjö Neuvo:
Optimal stack filtering and classical Bayes decision. 2009-2012 - Michael E. Deisher, Andreas S. Spanias:
Real-time implementation of a frequency-domain adaptive filter on a fixed-point signal processor. 2013-2016 - Cormac Herley, Martin Vetterli:
Linear phase wavelets: theory and design. 2017-2020 - Ramesh A. Gopinath, Wayne M. Lawton, C. Sidney Burrus:
Wavelet-Galerkin approximation of linear translation invariant operators. 2021-2023 - Ahmed H. Tewfik, Paul E. Jorgensen:
On the choice of a wavelet for signal coding and processing. 2025-2028 - Christian Dorize, Lars F. Villemoes:
Optimizing time-frequency resolution of orthonormal wavelets. 2029-2032 - Klaus Gram-Hansen:
A bandwidth concept for CPB time-frequency analysis. 2033-2036 - Ali N. Akansu, Yipeng Liu:
Tree structures for time-frequency signal analysis. 2037-2040 - Michele Covell, John Richardson:
A new, efficient structure for the short-time Fourier transform, with an application in code-division sonar imaging. 2041-2044 - Franz Hlawatsch, Werner Kozek:
Time-frequency analysis of linear signal spaces. 2045-2048 - Werner Kozek, Franz Hlawatsch:
Time-frequency filter banks with perfect reconstruction. 2049-2052 - Langford B. White:
Signal synthesis from Cohen's class of bilinear time-frequency signal representations using convex projections. 2053-2056 - D. Adler, Shlomit Raz:
Synthesis of legal Wigner filter. 2057-2060 - Jean-Marc Vesin, Murat Kunt:
Parametric probability density estimation based on an approximation by a discretized stochastic differential equation. 2061-2064 - Ramin A. Nobakht, Sasan H. Ardalan, David E. van den Bout:
Adaptive filtering of nonlinear systems with memory using quantized mean field annealing. 2065-2068 - Michael R. Lynch, Peter J. W. Rayner, Sean B. Holden:
Removal of degeneracy in adaptive Volterra networks by dynamic structuring. 2069-2072 - Hassan M. Ahmed, Muzaffar U. Khurram:
A new nonlinear adaptive lattice filter. 2073-2076 - Heung Ki Baik, V. John Mathews:
Adaptive algorithms for identifying recursive nonlinear systems. 2077-2080 - Stuart J. Flockton:
Error surfaces of adaptive active noise controllers. 2081-2084 - Delores M. Etter, Bock-Sim Chia:
Adaptive IIR filtering using stochastic filters. 2085-2088 - Andrew W. Hull, W. Kenneth Jenkins:
Low computational complexity adaptive algorithms for IIR digital filters. 2089-2092 - Dennis R. Morgan, James C. Thi:
A multi-tone pseudo-cascade filtered-x LMS adaptive notch filter. 2093-2096 - Gernot Kubin:
Adaptation in rapidly time-varying environments using coefficient filters. 2097-2100 - Marcel Lapointe, Paul Fortier, Huu Tuê Huynh:
A very fast digital realization of a time-domain block LMS filter. 2101-2104 - Joseph B. Evans:
A new variable step size method suitable for efficient VLSI implementation. 2105-2108 - David B. Chester, W. Ronald Young, Mike Petrowski:
A fully systolic adaptive filter implementation. 2109-2112 - Makoto Ohki, Sumihisa Hashiguchi:
A new 2-D LMS adaptive algorithm. 2113-2116 - S. Thomas Alexander, Virginia L. Stonick:
Analytical probability density functions for LMS adaptive filters using the Fokker-Planck equation. 2117-2120 - Xuan Kong, Victor Solo:
Effects of attenuation factor on adaptive time delay estimation. 2121-2124 - Scott C. Douglas, Teresa H.-Y. Meng:
An optimum NLMS algorithm: performance improvement over LMS. 2125-2128 - Peter J. Voltz:
Sample convergence of the normed LMS algorithm with feedback delay. 2129-2132 - Behrouz Farhang-Boroujeny, Saeed Gazor:
Performance analysis of transform domain normalized LMS algorithm. 2133-2136 - Woon-Seng Gan, John J. Soraghan, Robert W. Stewart, Tariq S. Durrani:
The non-canonical LMS algorithm (NCLMS): characteristics and analysis. 2137-2140 - Saul B. Gelfand, C. S. Ravishankar, Edward J. Delp:
A tree-structured piecewise linear adaptive filter. 2141-2144 - Michael J. Reed, Bede Liu:
An analysis of LMS adaptive two-sided transversal filters. 2145-2148 - Geoffrey A. Williamson, Peter M. Clarkson:
On convergence and steady state behavior in the median LMS adaptive filter. 2149-2152 - S. C. Huang, Y. F. Huang, I.-C. Jou:
Analysis of perceptron training algorithms and applications to hand-written character recognition. 2153-2156 - Neil J. Bershad, John J. Shynk, Paul L. Feintuch:
Statistical analysis of the single-layer backpropagation algorithm. 2157-2160 - Thomas G. Xydis:
A new IIR adaptive filter design using Tellegen's theorem. 2161-2164 - Anthony Celia, Satyanarayan S. Rao:
Parallel implementation of a second-order adaptive Volterra filter. 2165-2168 - Ioannis N. M. Papadakis, Stelios C. A. Thomopoulos:
A duality approach to model reference adaptive control and identification. 2169-2172 - Stelios C. A. Thomopoulos, Ioannis N. M. Papadakis:
Model reference adaptive control and identification via the modified output error method. 2173-2176 - Visakan Kadirkamanathan, Mahesan Niranjan:
Nonlinear adaptive filtering in nonstationary environments. 2177-2180 - Mahmood R. Azimi-Sadjadi, Sassan Sheedvash:
Recursive node creation in back-propagation neural networks using orthogonal projection method. 2181-2184 - Chao Lu, James W. Cooley, Richard Tolimieri:
Variants of the Winograd multiplicative FFT algorithms and their implementation on IBM RS/6000. 2185-2188 - Domingo Rodríguez:
A new FFT algorithm and its implementation on the DSP96002. 2189-2192 - Raimund Meyer, Richard Reng, Karl Schwarz:
Convolution algorithms on DSP processors. 2193-2196 - Nam Ik Cho, Il Dong Yun, Sang Uk Lee:
A fast algorithm for 2-D DCT. 2197-2200 - Elliot N. Linzer, Ephraim Feig:
New scaled DCT algorithms for fused multiply/add architectures. 2201-2204 - Rolf Gluth:
Regular FFT-related transform kernels for DCT/DST-based polyphase filter banks. 2205-2208 - Pierre Duhamel, Yannick Mahieux, Jean-Pierre Petit:
A fast algorithm for the implementation of filter banks based on 'time domain aliasing cancellation'. 2209-2212 - Olivier Rioul:
Fast algorithms for the continuous wavelet transform. 2213-2216 - Richard S. Orr:
Computational assessment of Gabor representations. 2217-2220 - Paulo S. R. Diniz, Luiz W. P. Biscainho:
Optimal convergence factor for the LMS/Newton algorithm. 2221-2224 - Masahichi Kishi:
Fast processing for the short time DFT Hilbert transformer. 2225-2228 - K. V. S. Babu:
A fast algorithm for adaptive estimation of root-MUSIC polynomial coefficients. 2229-2232 - Jyh-Chern Ho, Jar-Ferr Yang:
Parallel Rayleigh quotient iterative algorithm for rooting nonstationary spectral polynomials. 2233-2236 - Christopher J. Zarowski:
A QR algorithm for the delta AR model assuming autocorrelation windowed data. 2237-2240 - Guy Le Besnerais, Yves Goussard:
Improved square-root forms of fast linear least squares estimation algorithms. 2241-2244 - Edward J. Baranoski:
Triangular factorization of inverse data covariance matrices. 2245-2247 - K. J. Ray Liu, Shih-Fu Hsieh:
Fast Toeplitz-Hankel orthogonalization for high-resolution spectral estimation. 2249-2252 - Andrew E. Yagle:
New analogues of split algorithms for Toeplitz-plus-Hankel matrices. 2253-2256 - Radomir T. Sokolov, James C. Rogers:
Realizing homomorphic systems for convolution by time domain cepstral transformations. 2257-2260 - Jorge S. Marques, Luís B. Almeida:
A fast algorithm for generating sinusoids with polynomial phase. 2261-2264 - A. Liavas, Sergios Theodoridis:
Efficient Levinson type algorithm for block ρ-Toeplitz systems. 2265-2267 - Thomas R. Fischer:
Entropy-constrained geometric vector quantization for transform image coding. 2269-2272 - Marc Antonini, Michel Barlaud, Pierre Mathieu:
Image coding using lattice vector quantization of wavelet coefficients. 2273-2276 - David L. Neuhoff, Don H. Lee:
On the performance of tree-structured vector quantization. 2277-2280 - Ruey-Feng Chang, Wen-Tsuen Chen, Jia-Shung Wang:
Image sequence coding using adaptive tree-structured vector quantization with multipath searching. 2281-2284 - Shei-Zein Kiang, Gary J. Sullivan, Chung-Yen Chiu, Richard L. Baker:
Recursive optimal pruning of tree-structured vector quantizers. 2285-2288 - Eve A. Riskin, Robert M. Gray:
Lookahead in growing tree-structured vector quantizers. 2289-2292 - Karen L. Oehler, Eve A. Riskin, Robert M. Gray:
Unbalanced tree-growing algorithms for practical image compression. 2293-2296 - Michael T. Orchard:
A fast nearest-neighbor search algorithm. 2297-2300 - Yunus Hussain, Nariman Farvardin:
Variable-rate finite-state vector quantization of images. 2301-2304 - Limin Wang, Morris Goldberg, Seymour Shlien:
Interleaved image adaptive vector quantization. 2305-2308 - Yushu Feng:
An adaptive multi-layer address-VQ scheme. 2309-2312 - L. Zhang, Morris Goldberg, Sethuraman Panchanathan:
Combined source channel frame replenishment coding. 2313-2316 - Eric B. Hall, Gary L. Wise:
Some aspects of multidimensional convolution. 2317-2320 - Mark D. Plumbley, Frank Fallside:
The effect of receptor signal-to-noise levels on optimal filtering in a sensory system. 2321-2324 - Gerasimos Potamianos, John Goutsias:
A novel method for computing the partition function of Markov random field images using Monte Carlo simulations. 2325-2328 - Michael P. Clark, Louis L. Scharf, Clifford T. Mullis:
Quadratic estimators of the frequency-wavenumber spectrum. 2329-2332 - Thomas G. Xydis, Andrew E. Yagle:
Estimation of locations of multiple sources in a multi-layer volume conductor using medium filters. 2333-2336 - Masato Abe, Kiyohito Fujii, Toshio Sone, Ken'iti Kido:
Estimation of position and waveform of a specified sound source decreasing the effect of other sound sources and reflection. 2337-2340 - Ta-Kang Ku, C.-C. Jay Kuo:
Preconditioned iterative methods for block Toeplitz systems. 2341-2344 - Evangelos E. Milios, S. Hamid Nawab:
Combined numeric and symbolic processing of multidimensional signals for source tracking. 2345-2348 - P. Lie Chin Cheong, Salvatore D. Morgera:
Matching schemes using the steepest-ascent/descent methods. 2349-2352 - Izidor Gertner, Yehoshua Y. Zeevi:
Image representation with position-frequency localization. 2353-2356 - Vito Cappellini, Alessandro Mecocci:
Intelligent perception: an approach to multidimensional data interpretation. 2357-2360 - Rathika Rajaravivarma, P. Karivaratha Rajan:
Fast computation of DFT of 2-D data with symmetry about two diagonals. 2361-2364 - T. N. Daskalakis, C. N. Daskalakis:
Knapsack formulation of image matching. 2365-2368 - Margaret Varga, John Radford:
Automatic car model classification. 2369-2372 - Yang He, Amlan Kundu:
Shape classification using hidden Markov model. 2373-2376 - Mark J. Paulik, N. Mohankrishnan:
Shape recognition using a nonstationary autoregressive hidden Markov model. 2377-2380 - Ioannis Pitas, Anastasios N. Venetsanopoulos:
Morphological shape representation. 2381-2384 - P. Robert Hsu, Hiroshi Harashima:
Recovery of contour deformation using steerable filters. 2385-2388 - Philippe Salembier:
Multiresolution decomposition and adaptive filtering with rank order based filters-application to defect detection. 2389-2392 - A. Ravichandran, B. Yegnanarayana:
A two-stage neural network for translation, rotation and size-invariant visual pattern recognition. 2393-2396 - Jagath C. Rajapakse, Raj Acharya:
Neuromorphic model for information fusion. 2397-2400 - Sebastiano B. Serpico:
Image classification by integration of neural networks and machine learning. 2401-2404 - Majid Foodeei, Peter Kabal:
Backward adaptive prediction: high-order predictors and formant-pitch configurations. 2405-2408 - Chiou-Shann Fuh, Petros Maragos:
Affine models for image matching and motion detection. 2409-2412 - Wai-Chi Fang, Bing J. Sheu, Ji-Chien Lee:
A VLSI neuroprocessor for real-time image flow computing. 2413-2416 - Nader M. Namazi:
A new iterative scheme for waveform estimation with application to frame-to-frame image motion estimation. 2417-2420 - Jianzhong Huang, Russell M. Mersereau:
Displacement estimation along contours in image sequence analysis. 2421-2424 - C. R. Moloney, Eric Dubois:
Estimation of motion fields from image sequences with illumination variation. 2425-2428 - M. Ibrahim Sezan, Mehmet K. Özkan, Sergei V. Fogel:
Temporally adaptive filtering of noisy image sequences using a robust motion estimation algorithm. 2429-2432 - Hiroyuki Morikawa, Hiroshi Harashima:
Structure and motion of deformable objects from image sequences. 2433-2436 - G. S. Young, Rama Chellappa, Ting-Hu Wu:
Monocular motion estimation using a long sequence of noisy images. 2437-2440 - Jerry L. Prince, Elliot R. McVeigh:
Optical flow for tagged MR images. 2441-2444 - Xiaoping Hu, Narendra Ahuja:
Sufficient conditions for double or unique solution of motion and structure. 2445-2448 - Gordon W. Braudaway, Fred Mintzer:
Techniques for locating the stereo axis of stereo-pair images. 2449-2452 - M. Armon Rahgozar, Jan P. Allebach:
Motion estimation and object tracking based on time-sequentially sampled imagery. 2453-2456 - Michael Pyeron, Amir Waks, George K. Gregoriou, Oleh J. Tretiak, Izhak Bar-Kana:
A robust hierarchical probabilistic framework for visual target tracking. 2457-2460 - Ivo Tchoukanov, Reza Safaee-Rad, Beno Benhabib, Kenneth C. Smith:
The angle-of-sight signature for 2D shape analysis. 2461-2464 - Reza Safaee-Rad, Kenneth C. Smith, Beno Benhabib, Ivo Tchoukanov:
Application of moment and Fourier descriptors to the accurate estimation of elliptical shape parameters. 2465-2468 - Feng Lu, Evangelos E. Milios:
Optimal local spline approximation of planar shape. 2469-2472 - Basil G. Mertzios, Kostas D. Tsirikolias:
Fast shape discrimination using one-dimensional moments. 2473-2475 - Fabio Arduini, R. Cabri, Gian Luca Foresti, Vittorio Murino, Carlo S. Regazzoni:
A numerical and symbolic fusion method for interpretation of image sequence. 2477-2480 - Chang Wen Chen, Thomas S. Huang, Yung-Chang Chen:
Model based estimation of left ventricle motion. 2481-2484 - M. Ibrahim Sezan, Gordana Pavlovic, A. Murat Tekalp, A. Tanju Erdem:
On modeling the focus blur in image restoration. 2485-2488 - Gordana Pavlovic, A. Murat Tekalp:
Maximum likelihood parametric blur identification based on a continuous spatial domain model. 2489-2492 - H. Joel Trussell, Andreas E. Savakis:
Blur identification by statistical analysis. 2493-2496 - Reginald L. Lagendijk, Jan Biemond:
Block-adaptive image identification and restoration. 2497-2500 - Michalis E. Zervakis, Anastasios N. Venetsanopoulos:
A new regularized approach and its applications in image restoration. 2501-2504 - Eveline J. Bellegarda, Tarek M. Habashy:
Constrained least-squares reconstruction of multidimensional objects buried in inhomogeneous elastic media. 2505-2508 - Nikolas P. Galatsanos, Aggelos K. Katsaggelos, Roland T. Chin, Allen D. Hillery:
Two methods for least squares multi-channel image restoration. 2509-2512 - H. Joel Trussell, Michael J. Vrhel:
Estimation of illumination for color correction. 2513-2516 - Georgios Angelopoulos, Ioannis Pitas:
Multichannel Wiener filters in color image restoration based on AR color image modelling. 2517-2520 - Mohamed Chellali, Vinay K. Ingle:
Adaptive image restoration using the multichannel recursive least squares algorithm. 2521-2524 - Sanjit K. Mitra, Hui Li, Ing-Song Lin, Tian-Hu Yu:
A new class of nonlinear filters for image enhancement. 2525-2528 - Lin Yin, Jaakko Astola, Yrjö Neuvo:
Optimal weighted order statistic filters under the mean absolute error criterion. 2529-2532 - Bing Zeng, Hongbing Zhou, Yrjö Neuvo:
Synthesis of optimal detail-restoring stack filters for image processing. 2533-2536 - Lale Akarun, Richard A. Haddad:
Performance measures for rank order filters. 2537-2540 - Xinhua Zhuang, Yunxin Zhao, Ming-Yee Chiu:
Morphological structuring image decomposition. 2541-2544 - Ramakrishna Kakarala, Alfred O. Hero III:
On achievable accuracy in edge localization. 2545-2548 - William E. Higgins, C. M. Hsu:
Edge detection using 2-D local structural information. 2549-2552 - Michel Barlaud, Thierry Gaidon, Pierre Mathieu, Jean-Christophe Feauveau:
Edge detection using recursive biorthogonal wavelet transform. 2553-2556 - Roberto H. Bamberger, Mark J. T. Smith:
A multirate filter bank based approach to the detection and enhancement of linear features in images. 2557-2560 - Thulasinath G. Manickam, Pradeep Misra:
Application of recursive filters in edge detection of images. 2561-2564 - Dongming Zhao, David G. Daut:
Shape recognition using morphological transformations. 2565-2568 - Yao Wang, Sanjit K. Mitra:
Edge detection based on orientation distribution of gradient images. 2569-2572 - J. Patrick Fitch:
The single antenna interferometer. 2573-2576 - Mehrdad Soumekh:
Bistatic synthetic aperture radar inversion with application in dynamic object imaging. 2577-2580 - Yongjian Bao:
A model self-correction approach to reconstruction of binary image from two orthogonal projections. 2581-2584 - Robin N. Strickland, David W. Chandler:
Reconstruction of an axisymmetric image from its blurred and noisy projection. 2585-2588 - Der-Shan Luo, Andrew E. Yagle:
A Kalman filtering approach to stochastic tomography. 2589-2592 - Ken D. Sauer, Charles A. Bouman:
Bayesian estimation from projections with low photon dosages. 2593-2596 - Brian K. Jennison, Jan P. Allebach:
Maximum likelihood image reconstruction from Fourier-offset data using the expectation-maximization algorithm. 2597-2600 - Daniel E. Kreithen, William W. Irving, Stephen M. Crooks:
Generating correlated gamma random fields with application to synthesis of simulated SAR imagery. 2601-2604 - James K. E. Tunaley, Timothy Sibbald, Maria T. Rey-Cousins:
Use of the Dempster-Shafer algorithm in the detection of ship wakes from synthetic aperture radar images. 2605-2608 - Neeraj Magotra, Wes McCoy, Byong Kun Chang:
Narrow band synthetic aperture radar image formation. 2609-2612 - Osamu Ikeda:
Synthetic-aperture optical imaging system using digital phase conjugation. 2613-2616 - A. Agostinelli, C. Dambra, Sebastiano B. Serpico, Gianni Vernazza:
Multisensor data fusion by a region-based approach. 2617-2620 - Andy C. Hung, Teresa H.-Y. Meng:
Optimal quantizer step sizes for transform coders. 2621-2624 - André Zaccarin, Bede Liu:
Transform coding of color images with limited palette size. 2625-2628 - Yunming Huang, Nikolas P. Galatsanos, Howard M. Dreizen:
Priority DCT coding for image sequences. 2629-2632 - Philippe Delsarte, Benoît Macq, Dirk T. M. Slock:
Efficient multiresolution signal coding via a signal-adapted perfect reconstruction filter pyramid. 2633-2636 - Isao Furukawa, Mitsuru Nomura, Sadayasu Ono:
Hierarchical coding of super high definition images with subband+multistage VQ. 2637-2640 - Takahiro Saito, Hirofumi Higuchi, Takashi Komatsu:
Sub-band coding with improved permutation codes for high fidelity image compression. 2641-2644 - Kai-Kuang Ma, Sarah A. Rajala:
Subband coding of digital images using absolute moment block truncation. 2645-2648