Table of Contents

• Source-dependent channel coding for CELPW. Bastiaan Kleijn. 1-4 [doi]
• A real-time implementation of the improved MBE speech coderMichael S. Brandstein, Peter A. Monta, John C. Hardwick, Jae S. Lim. 5-8 [doi]
• A 2400 bps multi-band excitation vocoderPaul C. Meuse. 9-12 [doi]
• Multi-band vector excitation coding of speech at 4.8 kbpsCarmen García-Mateo, Francisco Javier Casajús-Quirós, Luis A. Hernández Gómez. 13-16 [doi]
• Harmonic coding at 4.8 kb/sJorge S. Marques, Luís B. Almeida, José M. Tribolet. 17-20 [doi]
• 4.8 kbit/s delayed decision CELP coder using tree codingKazunori Mano, Takehiro Moriya. 21-24 [doi]
• Excitation modeling in a homomorphic vocoderJae H. Chung, Ronald W. Schafer. 25-28 [doi]
• CELP coding with an adaptive density pulse excitation modelMasami Akamine, Kimio Miseki. 29-32 [doi]
• Vector quantization and perceptual criteria in SVD based CELP codersRosario Drogo Deiacovo, Roberto Montagna, Daniele Sereno. 33-36 [doi]
• Finite state CELP for variable rate speech codingSaeed Vaseghi. 37-40 [doi]
• ORION: a two pass hybrid system for isolated-words automatic speech recognitionJean-Claude Junqua. 41-44 [doi]
• Spoken language interaction in a goal-directed taskAlexander I. Rudnicky, Michelle Sakamoto, Joseph Polifroni. 45-48 [doi]
• The SUMMIT speech recognition system: phonological modelling and lexical accessVictor Zue, James R. Glass, David Goodine, Michael Philips, Stephanie Seneff. 49-52 [doi]
• ATR HMM-LR continuous speech recognition systemToshiyuki Hanazawa, Kenji Kita, Satoshi Nakamura 0001, Takeshi Kawabata, Kiyohiro Shikano. 53-56 [doi]
• A 10000-word continuous-speech recognition systemVolker Steinbiss, Andreas Noll, A. Peaseler, Hermann Ney, Henning Bergmann, Christian Dugast, Hans-Hermann Hamer, H. Piotrowski, Horst Tomaschewski, A. Zielinski. 57-60 [doi]
• 100000-word recognition using acoustic-segment networksShinta Kimura. 61-64 [doi]
• A real-time Mandarin dictation machine for Chinese language with unlimited texts and very large vocabularyLin-Shan Lee, Chiu-yu Tseng, Hung-Yan Gu, F. H. Liu, C. H. Chang, S.-H. Hsieh, C.-H. Chen. 65-68 [doi]
• The ARM continuous speech recognition systemM. J. Russel, Keith Ponting, S. M. Peeling, Sue Browning, John S. Bridle, Roger K. Moore, Isabel Galiano, P. Howell. 69-72 [doi]
• The VOYAGER speech understanding system: preliminary development and evaluationVictor Zue, James R. Glass, David Goodine, Hong C. Leung, Michael S. Phillips 0001, Joseph Polifroni, Stephanie Seneff. 73-76 [doi]
• The decipher speech recognition systemMichael Cohen, Hy Murveit, Jared Bernstein, Patti J. Price, Mitch Weintraub. 77-80 [doi]
• The N-best algorithms: an efficient and exact procedure for finding the N most likely sentence hypothesesRichard M. Schwartz, Yen-Lu Chow. 81-84 [doi]
• Constructing groups of acoustically confusable wordsLalit R. Bahl, Peter V. de Souza, P. S. Gopalakrishnan, Dimitri Kanevsky, David Nahamoo. 85-88 [doi]
• Classification of coarse phonetic categories in continuous speech: statistical classifiers vs. temporal flow connectionist networkAbdulmesih Aktas, Otto Schmidbauer, K. H. Maier, W. H. Feix. 89-92 [doi]
• Continuous speech recognition from a phonetic transcriptionStephen E. Levinson, Andrej Ljolje, Laura G. Miller. 93-96 [doi]
• Tools for the analysis of benchmark speech recognition testsDavid S. Pallett, William M. Fisher, Jonathan G. Fiscus. 97-100 [doi]
• An English language speech database at the University of Western AustraliaEdmund Lai, G. A. Carrijo, R. Bennett, Roberto Togneri, Michael D. Alder, Yianni Attikiouzel. 101-104 [doi]
• The demographics of speaker independent digit recognitionJoseph Picone. 105-108 [doi]
• NTIMIT: a phonetically balanced, continuous speech, telephone bandwidth speech databaseCharles R. Jankowski Jr., Ashok Kalyanswamy, Sara Basson, Judith Spitz. 109-112 [doi]
• A keyword spotter which incorporates neural networks for secondary processingDavid P. Morgan, Christopher L. Scofield, Theresa M. Lorenzo, Edward C. Real, David P. Loconto. 113-116 [doi]
• Probabilistic vector mapping of noisy speech parameters for HMM word spottingHerbert Gish, Yen-Lu Chow, Jan Robin Rohlicek. 117-120 [doi]
• Continuous speech recognition using large vocabulary word spotting and CV syllable spottingNoboru Sugamura. 121-124 [doi]
• Automatic detection of new words in a large vocabulary continuous speech recognition systemAyman Asadi, Richard M. Schwartz, John Makhoul. 125-128 [doi]
• A hidden Markov model based keyword recognition systemRichard C. Rose, Douglas B. Paul. 129-132 [doi]
• Fast speaker adaptation for speech recognition systemsFritz Class, Alfred Kaltenmeier, Peter Regel-Brietzmann, Karl Trottler. 133-136 [doi]
• Speaker adaptation from a speaker-independent training corpusFrancis Kubala, Richard M. Schwartz, Chris Barry. 137-140 [doi]
• Baseform adaptation for large vocabulary hidden Markov model based speech recognition systemsGerhard Rigoll. 141-144 [doi]
• A study on speaker adaptation of continuous density HMM parametersChin-Hui Lee, Chih-Heng Lin, Biing-Hwang Juang. 145-148 [doi]
• Speaker hierarchical clustering for improving speaker-independent HMM word recognitionLuc Mathan, Laurent Miclet. 149-152 [doi]
• Supplementation of HMM for articulatory variation in speaker adaptationHiroaki Hattori, Satoshi Nakamura 0001, Kiyohiro Shikano. 153-156 [doi]
• A comparative study of spectral mapping for speaker adaptationSatoshi Nakamura 0001, Kiyohiro Shikano. 157-160 [doi]
• Simultaneous speaker normalisation and utterance labelling using Bayesian/neural net techniquesS. J. Cox, John S. Bridle. 161-164 [doi]
• The Meta-Pi network: connectionist rapid adaptation for high-performance multi-speaker phoneme recognitionJohn B. Hampshire II, Alexander H. Waibel. 165-168 [doi]
• Speaker adaptive phoneme recognition by multi-level clustering based on mutual information criterionKatsuhiko Shirai, Naoki Hosaka, Eiichiro Kitagawa. 169-172 [doi]
• A real-time vector excited adaptive predictive coder at 9.6 kbit/sNatalie Fournier, Yves Grenier. 173-176 [doi]
• On reducing computational complexity of codebook search in CELP coder through the use of algebraic codesClaude Laflamme, Jean-Pierre Adoul, H.-Y. Su, Sarto Morissette. 177-180 [doi]
• Real-time implementation and performance of a 16 kb/s low-delay CELP speech coderJuin-Hwey Chen, Melvin J. Melchner, Richard V. Cox, Duane O. Bowker. 181-184 [doi]
• Optimal quantization of LSP parameters using delayed decisionsFrank K. Soong, Biing-Hwang Juang. 185-188 [doi]
• Low bit-rate spectral coding in CELP, a new LSP-methodRoar Hagen, Per Hedelin. 189-192 [doi]
• Coding of speech LSP parameters using TSVQ with interblock noiseless codingNam Phamdo, Nariman Farvardin. 193-196 [doi]
• Conditional entropy-constrained vector quantization of linear predictive coefficientsPhilip A. Chou, Tom D. Lookabaugh. 197-200 [doi]
• Global maximum likelihood voice decodingRichard A. Dean. 201-204 [doi]
• A finite states Markov quantizer for speech codingAlessandro Falaschi, Massimo Giustiniani, Piero Pierucci. 205-208 [doi]
• An analysis-by-synthesis linear predictive model for narrowband speech codingF. F. Tzeng. 209-212 [doi]
• Speech coding using waveform matching based on LPC residual phase equalizationMasaaki Honda. 213-216 [doi]
• On Bernoulli-Gaussian process modeling of speech excitation sourceKi Yong Lee, Byung-Gook Lee, Iickho Song, SouGuil Ann. 217-220 [doi]
• A neural network algorithm for enhancing delta modulation/LPC tandem connectionsJ. A. Naylor. 221-224 [doi]
• CELP coding with adaptive output-error model identificationMahesan Niranjan. 225-228 [doi]
• Error detection and control for the parametric information in CELP codersS. A. Atungsiri, Ahmet M. Kondoz, Barry G. Evans. 229-232 [doi]
• Variable rate speech coding with online segmentation and fast algebraic codesR. Di Francesco, Claude Lamblin, A. Leguyader, Dominique Massaloux. 233-236 [doi]
• A comparison of backward adaptive prediction algorithms in low delay speech codersJerry D. Gibson, Yoon-Chae Cheong, Hong Chae Woo, Wen-Whei Chang. 237-240 [doi]
• Principal axis extracting vector excitation coding: high quality speech at 8 kb/sTomohiko Taniguchi, Yoshinori Tanaka, A. Sasama, Yasuji Ohta. 241-244 [doi]
• Conjugate gradient methods for designing vector quantizersEyal Yair, Kenneth Zeger, Allen Gersho. 245-248 [doi]
• Pitch estimation and voicing detection based on a sinusoidal speech modelRobert J. McAulay, Thomas F. Quatieri. 249-252 [doi]
• Robust pitch determination via SVD based cepstral methodsM. Scott Andrews, Joseph Picone, Ronald D. Degroat. 253-256 [doi]
• Nonlinear multiplicative cepstral analysis for pitch extraction in speechL. Hodgson, M. E. Jernigan, Barry L. Wills. 257-260 [doi]
• Optimisation of neural models for speaker identificationJohn Oglesby, John S. Mason. 261-264 [doi]
• A connectionist approach for automatic speaker identificationYounès Bennani, Françoise Fogelman-Soulié, Patrick Gallinari. 265-268 [doi]
• Sub-word unit talker verification using hidden Markov modelsAaron E. Rosenberg, Chin-Hui Lee, Frank K. Soong. 269-272 [doi]
• A weighted distance measure based on the fine structure of feature space: application to speaker recognitionRen-Hua Wang, Lingshen He, Hiroya Fujisaki. 273-276 [doi]
• Study of line spectrum pair frequencies for speaker recognitionChi-shi Liu, Wern-Jun Wang, Min-Tau Lin, Hsiao-Chuan Wang. 277-280 [doi]
• Variable parameter speaker verification system based on hidden Markov modelingMichael I. Savic, Sunil K. Gupta. 281-284 [doi]
• Text-independent speaker recognition by trajectory space comparisonYifan Gong, Jean-Paul Haton. 285-288 [doi]
• Robust discrimination in automatic speaker identificationHerbert Gish. 289-292 [doi]
• Text independent speaker identification using automatic acoustic segmentationRichard C. Rose, Douglas A. Reynolds. 293-296 [doi]
• Neural network control for a cascade/parallel formant synthesizerMichael S. Scordilis, John N. Gowdy. 297-300 [doi]
• Extraction of control parameters for the voice source in a text-to-speech systemJohan de Veth, Bert Cranen, Helmer Strik, Lou Boves. 301-304 [doi]
• Robust ARMA analysis as an aid in developing parameter control rules for a pole-zero cascade speech synthesizerJohan de Veth, W. van Golstein Brouwers, H. Loman, Lou Boves. 305-308 [doi]
• A real-time French text-to-speech system generating high-quality synthetic speechEric Moulines, Françoise Emerard, Danielle Larreur, J. L. Le Saint-Milon, L. Le Faucheur, F. Marty, Francis Charpentier, Christel Sorin. 309-312 [doi]
• Extension of the bandwidth of the JSRU parallel-formant synthesizer for high quality synthesis of male and female speechWendy J. Holmes, John N. Holmes, Michael W. Judd. 313-316 [doi]
• Evaluation and development of the KTH text-to-speech system on the segmental levelRolf Carlson, Björn Granström, Lennart Nord. 317-320 [doi]
• Statistical properties of fluctuation of pitch intervals and its modeling for natural synthetic speechTetsunori Kobayashi, Hidetoshi Sekine. 321-324 [doi]
• 0 shape for Japanese text-to-speechYoshinori Sagisaka. 325-328 [doi]
• From interlingua to speech: generating prosodic information from conceptual representationShinsuke Sakai, Kazunori Muraki. 329-332 [doi]
• Modeling duration adjustment with dynamic time warpingMarian J. Macchi, Murray F. Spiegel, Karen L. Wallace. 333-336 [doi]
• Phoneme-level parameterization of speech using an articulatory modelS. Parthasarathy, Cecil H. Coker. 337-340 [doi]
• A psychologically-governed approach to novel-word pronunciation within a text-to-speech systemKirk P. H. Sullivan, Robert I. Damper. 341-344 [doi]
• Cross-language voice conversionMasanobu Abe, Kiyohiro Shikano, Hisao Kuwabara. 345-348 [doi]
• Application of Markov random fields to formant extractionLynn D. Wilcox, Francine R. Chen. 349-352 [doi]
• High-resolution characterization of formants in vowel-consonant transitionsKrishna S. Nathan, Harvey F. Silverman. 353-356 [doi]
• A perceptual pitch detectorMalcolm Slaney, Richard F. Lyon. 357-360 [doi]
• Pitch period determination of aperiodic speech signalsPer Hedelin, Dieter Huber. 361-364 [doi]
• A dynamical systems approach to speech processingNaftali Tishby. 365-368 [doi]
• Analysis of glottal waveforms across stress stylesKathleen E. Cummings, Mark A. Clements. 369-372 [doi]
• Modeling spectral processing in the central auditory systemZong Liang Wu, Jean-Luc Schwartz, Pierre Escudier. 373-376 [doi]
• Adaptive filtering based on cepstral representation-adaptive cepstral analysis of speechKeiichi Tokuda, Takao Kobayashi, Shoji Shiomoto, Satoshi Imai. 377-380 [doi]
• On a simple algorithm to calculate the 'energy' of a signalJames F. Kaiser. 381-384 [doi]
• New nonstationary techniques for the analysis and display of speech transientsLex Atlas, William Koolman, Patrick J. Loughlin, Randy Cole. 385-388 [doi]
• Speech analysis using the weighted recursive least squares algorithm with a variable forgetting factorY. T. Ting, D. G. Childers. 389-392 [doi]
• Evaluation of improved articulatory codebooks and codebook access distance measuresJuergen Schroeter, Peter Meyer, S. Parthasarathy. 393-396 [doi]
• Implementation of a nonlinear wave-digital-filter cochlear modelDavid H. Friedman. 397-400 [doi]
• An auditory model based on the analysis of envelope patternsJean-Pierre Martens, Luc Van Immerseel. 401-404 [doi]
• A linear periodically time-varying filter for time-frequency scaling of speechM. K. Asi, Bahaa E. A. Saleh. 405-408 [doi]
• High quality time scale modification of speech signals using fast synchronized-overlap-add algorithmsE. Hardam. 409-412 [doi]
• Continuous speech recognition using multilayer perceptrons with hidden Markov modelsNelson Morgan, Hervé Bourlard. 413-416 [doi]
• Combining hidden Markov model and neural network classifiersLes T. Niles, Harvey F. Silverman. 417-420 [doi]
• TDNN labeling for a HMM recognizerWeiye Ma, Dirk Van Compernolle. 421-423 [doi]
• Connectionist Viterbi training: a new hybrid method for continuous speech recognitionMichael A. Franzini, Kai-Fu Lee, Alex Waibel. 425-428 [doi]
• Neural net classifiers for robust speech recognition under noisy environmentsKuldip K. Paliwal. 429-432 [doi]
• Word recognition using hidden control neural architectureEsther Levin. 433-436 [doi]
• Large vocabulary recognition using linked predictive neural networksJoe Tebelskis, Alex Waibel. 437-440 [doi]
• Speaker-independent word recognition using a neural prediction modelKen-ichi Iso, Takao Watanabe. 441-444 [doi]
• Continuous speech recognition for the TIMIT database using neural networksFrank Fallside, H. Lucke, Tim Marsland, P. J. O'Shea, M. St. J. Owen, Richard W. Prager, Anthony J. Robinson, N. H. Russell. 445-448 [doi]
• Integrated training for spotting Japanese phonemes using large phonemic time-delay neural networksMasanori Miyatake, Hidelumi Sawai, Yasuhiro Minami, Kiyohiro Shikano. 449-452 [doi]
• High-quality 16 kb/s speech coding with a one-way delay less than 2 msJuin-Hwey Chen. 453-456 [doi]
• A 2 ms-delay adaptive code excited linear predictive coderJean E. Menez, Claude R. Galand, Michele M. Rosso. 457-460 [doi]
• Vector sum excited linear prediction (VSELP) speech coding at 8 kbpsIra A. Gerson, Mark A. Jasiuk. 461-464 [doi]
• CELP coding for land mobile radio applicationsDouglas J. Rahikka, Thomas E. Tremain, Vanoy C. Welch, Joseph P. Campbell Jr.. 465-468 [doi]
• CELP: a candidate for GSM half-rate codingIsabel Trancoso, C. Ribeiro, Luís B. Almeida, L. Oliveira, Jorge S. Marques. 469-472 [doi]
• Performance of error protected binary pulse excitation coders at 11.4 kb/s over mobile radio channelsRedwan Salami, K. H. H. Wong, Raymond Steele, D. G. Appleby. 473-476 [doi]
• Combined source and channel coding based on multimode codingTomohiko Taniguchi, Fumio Amano, Shigeyuki Unagami. 477-480 [doi]
• An efficient code structure and search strategy for stochastic coding at 8 kb/sDale E. Veeneman, Baruch Mazor. 481-484 [doi]
• Optimal and sub-optimal algorithms for selecting the excitation in linear predictive codersPrzemyslaw Dymarski, Nicolas Moreau, A. Vigier. 485-488 [doi]
• A hybrid speech recognition system using HMMs with an LVQ-trained codebookHitoshi Iwamida, Shigeru Katagiri, Erik McDermott, Y. Tohkura. 489-492 [doi]
• Neural networks, maximum mutual information training, and maximum likelihood training [speech recognition]Les T. Niles, Harvey F. Silverman, Marcia A. Bush. 493-496 [doi]
• Speaker-independent vowel classification using hidden Markov models and LVQ2Don Kimber, Marcia A. Bush, Gary N. Tajchman. 497-500 [doi]
• HMM-based warping in neural networksYuqing Gao, Tai-Yi Huang, Daowen Chen. 501-504 [doi]
• Combining phoneme identification neural networks into an expert system using spectrogram reading knowledgeYasuhiro Komori, Kaichiro Hatazaki, Takaharu Tanaka, Takeshi Kawabata. 505-508 [doi]
• Comparison among time-delay neural networks, LVQ2 discrete parameter HMM and continuous parameter HMMSeiichi Nakagawa, Yoshimitsu Hirata. 509-512 [doi]
• Large vocabulary speech recognition using neural-fuzzy and concept networksNobuo Hataoka, Akio Amano, Toshiyuki Aritsuka, Akira Ichikawa. 513-516 [doi]
• Mandarin tone recognition by multi-layer perceptronPao-Chung Chang, San-Wei Sun, Sin-Horng Chen. 517-520 [doi]
• Neural networks for voiced/unvoiced speech classificationAage Bendiksen, Kenneth Steiglitz. 521-524 [doi]
• Phonetic classification using multi-layer perceptronsHong C. Leung, Victor W. Zue. 525-528 [doi]
• A connectionist model for speaker-independent isolated word recognitionMing Zhu 0005, Klaus Fellbaum. 529-532 [doi]
• Speaker-independent vowel recognition: spectrograms versus cochleagramsYeshwant K. Muthusamy, Ronald A. Cole, Malcolm Slaney. 533-536 [doi]
• A hybrid coder for hidden Markov models using a recurrent neural networkYoshua Bengio, Régis Cardin, Renato de Mori, Yves Normandin. 537-540 [doi]
• A model block-training method for HMM-based speech recognition systemsYu Qing Gao, Jin-Hui Xie. 541-544 [doi]
• Some improvements in speech recognition algorithms based on HMMAbdelaziz Kriouile, Jean-François Mari, Jean-Paul Haton. 545-548 [doi]
• Improved training and recognition algorithms with VQ-based hidden Markov modelsMarkus Falkhausen, Stephan Euler, Dietrich Wolf. 549-552 [doi]
• Issues related to the estimation of time variant hidden Markov models [speech recognition]Solomon Lerner, Baruch Mazor. 553-556 [doi]
• Vector quantization of the HMM distributions: further resultsEleftherios D. Frangoulis. 557-560 [doi]
• Discriminant methods for improving the robustness of Mandarin syllables recognition based upon hidden Markov modelFu-Hua Liu, Jung-Kuei Chen, Eng-Fong Huang, Chi-shi Liu. 561-564 [doi]
• Template adaptation in a hypersphere word classifierL. A. Smith, Brian L. Scott, L. S. Lin, J. M. Newell. 565-568 [doi]
• Adapting probability-transitions in DP matching process for an oral task-oriented dialogueK. Matrouf, Jean-Luc Gauvain, Françoise D. Néel, Joseph Mariani. 569-572 [doi]
• Integrating natural language constraints into HMM-based speech recognitionHy Murveit, Robert C. Moore. 573-576 [doi]
• A morphological model for large vocabulary speech recognitionMarc El-Bèze, Anne-Marie Derouault. 577-580 [doi]
• HMM continuous speech recognition using stochastic language modelsKenji Kita, Takeshi Kawabata, Toshiyuki Hanazawa. 581-584 [doi]
• An augmented chart parsing algorithm integrating unification grammar and Markov language model for continuous speech recognitionLee-Feng Chien, Keh-Jiann Chen, Lin-Shan Lee. 585-588 [doi]
• A continuous speech recognition system based on a two-level grammar approachShoichi Matsunaga, Shigeki Sagayama, Shigeru Homma, Sadaoki Furui. 589-592 [doi]
• Robust connectionist parsing of spoken languageAjay N. Jain, Alexander H. Waibel. 593-596 [doi]
• Comparing a connectionist and a rule based model for assigning parts-of-speechKjell Elenius. 597-600 [doi]
• On adaptive acquisition of languageAllen L. Gorin, Stephen E. Levinson, Laura G. Miller, A. N. Gertner, Andrej Ljolje, E. R. Goldman. 601-604 [doi]
• Spoken word recognition for multiple speakers based on path-limited DP matching and a method for speaker normalizationKeikichi Hirose, Hiroya Fujisaki, Sumio Ohno, Haruhiko Mio. 605-608 [doi]
• Analysis and quantization procedures for a real-time implementation of a 4.8 kb/s CELP coderFrancisco Javier Casajús-Quirós, Luis Alfonso Hernández Gómez, Carmen García-Mateo. 609-612 [doi]
• Speaker recognition based on source coding approachesBiing-Hwang Juang, Frank K. Soong. 613-616 [doi]
• A system for synthesizing Japanese speech from orthographic textHiroya Fujisaki, Keikichi Hirose, Hisashi Kawai, Yasuharu Asano. 617-620 [doi]
• Classifying words for improved statistical language modelsFrederick Jelinek, Robert L. Mercer, Salim Roukos. 621-624 [doi]
• Improved interoperable 2.4 kb/s LPC using sinusoidal transform coder techniquesRobert J. McAulay, Terrence G. Champion. 641-643 [doi]
• A high-quality speech coder at 600 b.p.sY. J. Liu. 645-648 [doi]
• A 450 b.p.s. vocoder with natural-sounding speechYan Ming Cheng, Douglas D. O'Shaughnessy. 649-652 [doi]
• Improving intelligibility of a 300 b/s segment vocoderPatrick Peterson, Philippe Jeanrenaud, Jeffrey Vandegrift. 653-656 [doi]
• Speaker adaptation algorithms based on piecewise moving adaptive segment quantization methodYoshinao Shiraki, Masaaki Honda. 657-660 [doi]
• Pitch predictors with high temporal resolutionPeter Kroon, Bishnu S. Atal. 661-664 [doi]
• Improved pitch prediction with fractional delays in CELP codingJorge S. Marques, Isabel Trancoso, José M. Tribolet, Luís B. Almeida. 665-668 [doi]
• Fractal excitation signals for CELP speech codersPetros Maragos, Kenneth L. Young. 669-672 [doi]
• A 2.4 KBPS speech coding method based on fuzzy vector quantizationYoshiaki Asakawa, Katsuya Yamasaki, Akira Ichikawa. 673-676 [doi]
• Competitive training in hidden Markov models [speech recognition]Steve J. Young. 681-684 [doi]
• Discriminant analysis and supervised vector quantization for continuous speech recognitionGeorge Yu, William Russell, Richard M. Schwartz, John Makhoul. 685-688 [doi]
• On semi-continuous hidden Markov modelingXuedong Huang 0001, Kai-Fu Lee, Hsiao-Wuen Hon. 689-692 [doi]
• A probabilistic acoustic map based discriminative HMM trainingEng-Fong Huang, Frank K. Soong. 693-696 [doi]
• Supervised selection of prototypes for classification [speech recognition]Subrata K. Das. 697-700 [doi]
• Maximum mutual information estimation of HMM parameters for continuous speech recognition using the N-best algorithmYen-Lu Chow. 701-704 [doi]
• Joint quantizer design and parameter estimation for discrete hidden Markov modelsMari Ostendorf, Jan Robin Rohlicek. 705-708 [doi]
• Estimation of hidden Markov model parameters by minimizing empirical error rateAndrej Ljolje, Yariv Ephraim, Lawrence R. Rabiner. 709-712 [doi]
• Experiments on mixture-density phoneme-modelling for the speaker-independent 1000-word speech recognition DARPA taskHermann Ney. 713-716 [doi]
• Learning the structure of HMM's through grammatical inference techniquesFrancisco Casacuberta, Enrique Vidal 0001, B. Mas, Hector Rulot Segovia. 717-720 [doi]
• Acoustic modeling of subword units for speech recognitionChin-Hui Lee, Lawrence R. Rabiner, Roberto Pieraccini, Jay G. Wilpon. 721-724 [doi]
• On vocabulary-independent speech modelingHsiao-Wuen Hon, Kai-Fu Lee. 725-728 [doi]
• Lexicon-building methods for an acoustic sub-word based speech recognizerKuldip K. Paliwal. 729-732 [doi]
• Automatic generation of lexical and grammatical constraints for speech recognitionMichael K. Brown, Jay G. Wilpon. 733-736 [doi]
• Word juncture modeling using phonological rules for HMM-based continuous speech recognitionEgidio P. Giachin, Aaron E. Rosenberg, Chin-Hui Lee. 737-740 [doi]
• Acoustic recognition component of an 86000-word speech recognizerLi Deng, Vishwa Gupta, Matthew Lennig, Patrick Kenny, Paul Mermelstein. 741-744 [doi]
• Statistical segmentation and word modeling techniques in isolated word recognitionS. A. Euler, Biing-Hwang Juang, Chin-Hui Lee, Frank K. Soong. 745-748 [doi]
• Allophone clustering for continuous speech recognitionKai-Fu Lee, Satoru Hayamizu, Hsiao-Wuen Hon, Cecil Huang, Jonathan Swartz, Robert Weide. 749-752 [doi]
• Identification of contextual factors for pronunciation networksFrancine R. Chen. 753-756 [doi]
• Statistical feature selection for isolated word recognitionEduardo Lleida, Climent Nadeu, Enrique Monte, José B. Mariño. 757-760 [doi]
• Low bit quantization of the smoothed group delay spectrum for speech recognitionHarald Singer, Taizo Umezaki, Fumitada Itakura. 761-764 [doi]
• Transducer effect on ASR applications over the telephone networkHarry Chang. 765-768 [doi]
• Dependence of phonemic feature on contextTetsunori Kobayashi, Kazuhiro Watanabe. 769-772 [doi]
• Isolated word intonation recognition using hidden Markov modelsJohn W. Butzberger, Mari Ostendorf, Patti J. Price, Stefanie Shattuck-Hufnagel. 773-776 [doi]
• Markov modeling of prosodic phrase structureN. M. Veilleux, Mari Ostendorf, Patti J. Price, Stefanie Shattuck-Hufnagel. 777-780 [doi]
• Towards feature-based speech metricAruna Bayya, Hynek Hermansky. 781-784 [doi]
• Information-theoretic distortion measures for speech recognition: theoretical considerations and experimental resultsYi-Teh Lee, Daniel Kahn. 785-788 [doi]
• On the use of hierarchical spectral dynamics in speech recognitionSadaoki Furui. 789-792 [doi]
• Dynamic spectral shape features for speaker-independent automatic recognition of stop consonantsStephen A. Zahorian, Zaki B. Nossair. 793-796 [doi]
• Joint maximum likelihood estimation of pitch and AR parameters using the EM algorithmDavid Burshtein. 797-800 [doi]
• A study of LSF representation for speaker-dependent and speaker-independent HMM-based speech recognition systemsKuldip K. Paliwal. 801-804 [doi]
• A phoneme model of English speech for the design of a parallel speech understanding systemEdward C. Bronson, Michael R. Taaffe, Leah H. Jamieson. 805-808 [doi]
• Speech separation by means of stationary least-squares harmonic estimationFernando M. Silva, Luís B. Almeida. 809-812 [doi]
• Automatic talker activity labeling for co-channel talker interference suppressionMarc A. Zissman, Clifford J. Weinstein, Louis D. Braida. 813-816 [doi]
• Spectral resolution and noise robustness in auditory modelingThippur V. Sreenivas, Karamjeet Singh, Russell J. Niederjohn. 817-820 [doi]
• Noise reduction using a soft-decision sine-wave vector quantizerThomas F. Quatieri, Robert J. McAulay. 821-824 [doi]
• Improvements to the noise reduction neural networkShinichi Tamura, Masami Nakamura. 825-828 [doi]
• A minimum mean square error approach for speech enhancementYariv Ephraim. 829-832 [doi]
• Switching adaptive filters for enhancing noisy and reverberant speech from microphone array recordingsDirk Van Compernolle. 833-836 [doi]
• Experimental results showing the effects of optimal spacing between elements of a linear microphone arrayVictor M. Alvarado, Harvey F. Silverman. 837-840 [doi]
• Acoustic and perceptual studies of Lombard speech: application to isolated-words automatic speech recognitionJean-Claude Junqua, Yolande Anglade. 841-844 [doi]
• Hidden Markov model decomposition of speech and noiseAndrew P. Varga, Roger K. Moore. 845-848 [doi]
• Environmental robustness in automatic speech recognitionAlejandro Acero, Richard M. Stern. 849-852 [doi]
• Estimation using log-spectral-distance criterion for noise-robust speech recognitionAdoram Erell, Mitch Weintraub. 853-856 [doi]
• Robust speaker-independent word recognition using static, dynamic and acceleration features: experiments with Lombard and noisy speechBrian A. Hanson, Ted H. Applebaum. 857-860 [doi]
• A neural network learning algorithm for adaptive principal component extraction (APEX)Sun-Yuan Kung, Kostas I. Diamantaras. 861-864 [doi]
• A learning rule eliminating local minima in multilayer perceptronsPietro Burrascano, P. Lucci. 865-868 [doi]
• Analyses of the hidden units of the multi-layer perceptron and its application in acoustic-to-articulatory mappingQiuzhen Xue, Yu Hen Hu, Paul L. Milenkovic. 869-872 [doi]
• Articulator-based synthesis for conversational speechAndrew D. Girson, Ronald D. Williams. 873-876 [doi]
• Structured neural network topologies with application to acoustic transientsAndrea Maccato, Rui J. P. de Figueiredo. 877-880 [doi]
• Three-layer neural networks for spectral estimationChia-Jiu Wang, Mark A. Wickert, Chwan-Hwa Wu. 881-884 [doi]
• Complexity reduction in Volterra connectionist modelling by consideration of output mappingPeter J. W. Rayner, M. R. Lynch. 885-888 [doi]
• Bearing estimation using neural optimisation methodsSanjay K. Jha, Tariq S. Durrani. 889-892 [doi]
• A hardware efficient parallel Viterbi algorithmPeter J. Black, Teresa H. Y. Meng. 893-896 [doi]
• A formal high level synthesis approach for DSP architecturesKhaled M. Elleithy, Magdy A. Bayoumi. 897-990 [doi]
• Efficient and reliable VLSI algorithms and architectures for the discrete Fourier transformWilliam G. Bliss, Archie W. Julien. 901-904 [doi]
• A new parallel 2-D FFT architectureTeresa M. Pytosh, Alberto M. Magnani. 905-908 [doi]
• High sampling rate adaptive decision feedback equalizersAlan Gatherer, Teresa H. Y. Meng. 909-912 [doi]
• VLSI architectures for digital image codingM. Yan, John V. McCanny, Yi Hu. 913-916 [doi]
• Architecture of a highly reliable systolic correlator arrayRavi Ramaswamy, Gavin Brebner, David Aspinall 0002. 917-920 [doi]
• A new VLSI architecture for large kernel real time convolutionFrancis Jutand, Nicolas Demassieux, A. Artieri. 921-924 [doi]
• RNS approach to full parallel linear combinatorsMario Salerno, Gian-Carlo Cardarilli, Roberto Lojacono, F. Sargeni. 925-928 [doi]
• A high-speed adaptive equalizer for magnetic disk drivesWilliam L. Abbott, Philip S. Bednarz, Kevin D. Fisher, John M. Cioffi. 929-932 [doi]
• A VLSI architecture for two-dimensional Radon transform computationsPaul J. Hurst, K. Wayne Current, Iskender Agi, Eric Shieh. 933-936 [doi]
• CMOS VLSI implementation of the 2D-DCT with linear processor arraysU. Totzek, F. Matthiesen, S. Wohlleben, Tobias G. Noll. 937-940 [doi]
• Memory based digital signal processingBasavaraj I. Pawate, George R. Doddington, Shivaling S. Mahant-Shetti, Mark G. Harward, Derek Smith. 941-944 [doi]
• A two-dimensional, distributed logic processor for machine visionMary Jane Irwin, Robert Michael Owens. 945-948 [doi]
• OSPREY: a transputer based continuous speech recognition systemAndrew M. Sutherland, M. Campbell, Yasuo Ariki, Mervyn A. Jack. 949-952 [doi]
• Systolic implementation of counterpropagation networksHon Keung Kwan, Pang-Chung Tsang. 953-956 [doi]
• Implementation of a parallel Jacobi algorithm on multiple DSP96002 digital signal processorsGuy R. L. Sohie. 957-960 [doi]
• A transform coding chip set for image compressionImran Shah, Olu Akiwumi-Assani, Brian Johnson. 961-964 [doi]
• Programming DSP systems on multiprocessor architecturesM. Veiga, José Parera, Juan Santos. 965-968 [doi]
• A high speed programmable digital FIR filterJoseph B. Evans, Yong Ching Lim, Bede Liu. 969-971 [doi]
• The Motorola 16-bit DSP ASIC coreJose Chau Lee, Eric Cheval, Joe Gergen. 973-976 [doi]
• Parallel architecture for real-time speech recognition in SpanishS. Alexandres, J. Morán, J. Carazo, A. Santos. 977-980 [doi]
• A neural halftoning algorithm suiting VLSI implementationThierry M. Bernard, Patrick Garda, Bertrand Y. Zavidovique. 981-984 [doi]
• Hardware implementation of a color image decoder for remote database accessSchuyler Quackenbush. 985-988 [doi]
• A real-time implementation of half-band filters to obtain 18-20 bit resolution from the DSP56ADC16 sigma-delta A/D converterSangil Park. 989-992 [doi]
• Image processing performance evaluation for DSP based parallel computers with distributed frame buffersM. Reha Civanlar, Hayder Radha. 993-996 [doi]
• Parallel implementation and analysis of adaptive transform codingM. N. Chong, John J. Soraghan, Tariq S. Durrani. 997-1000 [doi]
• A fast implementation of the ESPRIT algorithmJ. S. McGarrity, John J. Soraghan, Tariq S. Durrani. 1001-1004 [doi]
• RAP: a ring array processor for multilayer perceptron applicationsNelson Morgan, James Beck, Eric Allman, Joachim Beer. 1005-1008 [doi]
• Nonlinear predictive image coding with a neural networkZhenya He, Haibo Li. 1009-1012 [doi]
• Real time application of the floating point pipeline CORDIC processor in massive-parallel pipelined DSP algorithmsAlfons A. J. de Lange, Ed F. Deprettere, Alle-Jan van der Veen, Jichun Bu. 1013-1016 [doi]
• Spectral decomposition via systolic triarray based on QR iterationKuoJuey Ray Liu, K. Yao. 1017-1020 [doi]
• The systolic phase rotation FFT-a new algorithm and parallel processor architectureJohn E. Whelchel, John O'Malley, William J. Rinard, James F. McArthur. 1021-1024 [doi]
• Systolic architectures for radar CFAR detectorsJenq-Neng Hwang, James A. Ritcey. 1025-1028 [doi]
• Parameterized dependence graph and its applications to multi-stage systolic mapping procedureY. T. Hwang, Y. H. Hu. 1029-1032 [doi]
• Pipelined two-port adaptor for wave digital filteringRajinder Jit Singh, John V. McCanny, Roger F. Woods. 1033-1036 [doi]
• Skew-pipeline and interleaved pipeline structures for 2-D recursive filteringTongxin Lu, Mahmood R. Azimi-Sadjadi, A. R. Rostampour. 1037-1040 [doi]
• High-speed architectures for dynamic programming problemsKeshab K. Parhi. 1041-1044 [doi]
• Vectorizing IIR filtersRobert B. Feinberg. 1045-1048 [doi]
• VLSI design of multi-rate arrays for DSP algorithmAihua Li, Sayfe Kiaei. 1049-1052 [doi]
• Automatic design of parallel implementations of DSP algorithmsByung Moo Kim, Larry P. Heck. 1053-1056 [doi]
• DSP specification using the Silage languageDominique Genin, Paul N. Hilfinger, Jan M. Rabaey, Chris J. Scheers, Hugo De Man. 1056-1060 [doi]
• An assembly source level global compacter for digital signal processorsStephen M. Kafka. 1061-1064 [doi]
• DSP/C: a standard high level language for DSP and numeric processingKevin W. Leary, William Waddington. 1065-1068 [doi]
• KINFIDA: an object-oriented model-based digital filter design assistantAmir Hekmatpour, Scott R. Powell, Paul M. Chau. 1069-1072 [doi]
• Code generation for the AT&T DSP32Kurt Baudendistel, James H. McClellan. 1073-1076 [doi]
• CADiSP-a graphical compiler for the programming of DSP in a completely symbolic wayAlois Knoll, Rupert C. Nieberle. 1077-1080 [doi]
• A high-level ASIC design tool for digital signal processingStewart G. Smith, Ralph W. Morgan, Julian G. Payne. 1081-1084 [doi]
• Interactive design environment of VLSI architecture for digital signal processingToshiyuki Takezawa, Katsuhiko Shirai. 1085-1088 [doi]
• Statistical error analysis of VLSI architectures: methodology and a case studyVijay K. Jain. 1089-1092 [doi]
• Adaptive transform coding with an adaptive block size (ATC-ABS)Akihiko Sugiyama, Fumie Hazu, Masahiro Iwadare, Takao Nishitani. 1093-1096 [doi]
• Digital audio carrying extra informationW. R. Th. ten Kate, L. M. van de Kerkhof, F. F. M. Zijderveld. 1097-1100 [doi]
• High quality audio coding using multipulse LPCSharad Singhal. 1101-1104 [doi]
• Audio codec for 64 kbit/sec (ISDN channel)-requirements and resultsErnst Eberlein, Heinz Gerhäuser, Stefan Krägeloh. 1105-1108 [doi]
• High fidelity audio transform coding with vector quantizationWai-Yip Chan, Allen Gersho. 1109-1112 [doi]
• Stereophonic ADPCM voice coding methodShigenobu Minami, Osamu Okada. 1113-1116 [doi]
• High-quality audio transform coding at 128 kbits/sGrant A. Davidson, Louis D. Fielder, Mike Antill. 1117-1120 [doi]
• Vowel articulation training aid for the deafStephen A. Zahorian, Subhashri Venkat. 1121-1124 [doi]
• Feedback cancellation in hearing aidsJames M. Kates. 1125-1128 [doi]
• Sensitivity to design parameters in an adaptive-beamforming hearing aidPatrick M. Zurek, Julie E. Greenberg, Patrick Peterson. 1129-1132 [doi]
• Acoustic echo canceller algorithm based on the variation characteristics of a room impulse responseShoji Makino, Yutaka Kaneda. 1133-1136 [doi]
• Robust adaptive algorithms for active noise and vibration controlHong Fan, Ram Vemuri. 1137-1140 [doi]
• Fast converging subband acoustic echo cancellation using RAP on the WE DSP16ASteven L. Gay, Richard J. Mammone. 1141-1144 [doi]
• Recursive least squares algorithms using multirate systems for cancellation of acoustical echoesBirger Hätty. 1145-1148 [doi]
• Microphone arrays for desktop computers and speech recognitionJames Parry. 1149-1151 [doi]
• A filter compiler for digital sound synthesisKenneth Steiglitz. 1153-1156 [doi]
• Pulsed noise in self-sustained oscillations of musical instrumentsChris Chafe. 1157-1160 [doi]
• Characterization of aperiodicity in nearly periodic signalsRobert T. Schumacher, Chris Chafe. 1161-1164 [doi]
• Nonlinear distortion analysis using digital higher-order coherence spectraYong Soo Cho, Sung Bae Kim, Elmer L. Hixson, Edward J. Powers. 1165-1168 [doi]
• An adaptive IIR phase measurement structure for estimation of multiple sinusoidsJohn Lane, Sangil Park. 1169-1172 [doi]
• Audio and ultrasonic transducers based on electrothermomechanical film (ETMF)Juha Backman, Matti Karjalainen. 1173-1776 [doi]
• A data-driven method for discovering and predicting allophonic variationMark A. Randolph. 1177-1180 [doi]
• Non-Euclidean locally monotonic regressionAlfredo Restrepo Palacios, Irwin W. Sandberg, Alan C. Bovik. 1201-1204 [doi]
• A parametric technique for superresolution image reconstructionYoram Bresler, Scott P. Litke. 1205-1208 [doi]
• Robust parametric reconstruction of signals by rational modelingVip Desai, K. S. Arun. 1209-1212 [doi]
• Ill-conditioned signal restoration and the conjugate gradient methodBarry J. Sullivan. 1213-1216 [doi]
• Inverse of linear periodically time-varying filteringHeping Ding, Moustafa M. Fahmy. 1217-1220 [doi]
• Instantaneous frequency estimation and automatic time-varying filteringBoualem Bouachache, Langford B. White. 1221-1224 [doi]
• Least-squares order statistic filters for signal restoration in dependent noiseLaith Naaman, Alan C. Bovik. 1225-1228 [doi]
• Adaptive recursive order statistic filtersFrancesco Palmieri. 1229-1232 [doi]
• Multi-channel signal reconstruction using noisy samplesJohn L. Brown Jr., Sergio D. Cabrera. 1233-1236 [doi]
• Superresolution of noisy band-limited data by data adaptive regularization and its application to seismic trace inversionNaoki Saito 0001. 1237-1240 [doi]
• Some new properties of the LMS FIR-ALEBhaskar D. Rao. 1241-1244 [doi]
• On the performance analysis of ALE using an IIR lattice notch filterNam Ik Cho, Sang Uk Lee. 1245-1248 [doi]
• A comparative analysis of the stationary points of the constant modulus algorithm based on Gaussian assumptionsJohn J. Shynk, Christina K. Chan. 1249-1252 [doi]
• Almost-sure sample stability of the continuous-time LMS algorithmPeter J. Voltz, Frank Kozin. 1253-1256 [doi]
• Mean square error analysis for the fast LMS-sine algorithmM. A. Khasawneh, Winser E. Alexander. 1257-1260 [doi]
• An informational approach to the convergence of output error adaptive IIR filter structuresVictor E. DeBrunner, A. A. (Louis) Beex. 1261-1264 [doi]
• Asymptotic stability of SMM and RGM for insufficient order IIR adaptive filtersMajid Nayeri, Hong Fan. 1265-1268 [doi]
• Analysis and generalization of a median adaptive filterTarek I. Haweel, Peter M. Clarkson. 1269-1272 [doi]
• Delayed adaptive LMS filtering: current resultsRaziel Haimi-Cohen, Hanan Herzberg, Yair Be'ery. 1273-1276 [doi]
• Statistical analysis of the soft constrained initialization of recursive least squares algorithmsNancy E. Hubing, S. Thomas Alexander. 1277-1280 [doi]
• Discrete cosine transform filteringBowonkoon Chitprasert, K. R. Rao. 1281-1284 [doi]
• A design technique for high order digital differentiatorsW. Gregory Medlin. 1285-1288 [doi]
• A minimum-order structure for fixed-point digital filtersAli Zilouchian, Nurgun Erdol. 1289-1292 [doi]
• An exact realization of 2-D IIR filters using separable 1-D modulesArnab K. Shaw, Pradeep Misra. 1293-1296 [doi]
• Optimum removal of LCOs in digital filtersY. Z. Zhang, T. W. Cole, Y. F. Yao. 1297-1300 [doi]
• Results on the application of simulated annealing algorithm for the design of digital filters with powers-of-two coefficientsNevio Benvenuto, Michele Marchesi, Aurelio Uncini. 1301-1304 [doi]
• Least squared error FIR filter design with spline transition functionsC. Sidney Burrus, Admadji W. Soewito, Ramesh A. Gopinath. 1305-1308 [doi]
• Laguerre approximation of nonrecursive discrete-time systemsMohammad A. Masnadi-Shirazi, Nasir Ahmed 0001. 1309-1312 [doi]
• On the design of allpasses with prescribed group delayHans Wilhelm Schüßler, Peter Steffen. 1313-1316 [doi]
• Design of three-dimensional spherically symmetric digital filtersWei-Ping Zhu, Zhenya He. 1317-1320 [doi]
• Complex Chebyshev approximation for IIR digital filters using an iterative WLS techniqueTakao Kobayashi, Satoshi Imai. 1321-1324 [doi]
• Global optimal IIR filter design and ARMA estimation using homotopy continuation methodsVirginia L. Stonick, S. Thomas Alexander. 1325-1328 [doi]
• A functional silicon compiler for high speed FIR digital filtersPaul T. Yang, Rajeev Jain, Toshiaki Yoshino, Wanda Gass, Ashwin Shah. 1329-1332 [doi]
• A simple Remez exchange algorithm to design IIR filters with zeros on the unit circleLeland B. Jackson, Gerald J. Lemay. 1333-1336 [doi]
• Design of efficient parallel hybrid FIR filters using dynamic programming and subset selection methodsRichard J. Hartnett, Gloria Faye Boudreaux. 1337-1340 [doi]
• Chebyshev polynomial based transfer functions for orthogonal lattice filtersRoy Chapman, M. A. Rahman. 1341-1344 [doi]
• A bound on mean square estimate errorJonathan S. Abel. 1345-1348 [doi]
• Error propagation and numerical recovery for a class of parametric DSP algorithmsConstantin Papaodysseus, Constantin Triantafillou, Elias Koukoutsis, George Carayannis. 1349-1352 [doi]
• Modeling and analyzing fractal point processesDon H. Johnson, Anand R. Kumar. 1353-1356 [doi]
• The performance analysis of a fourth-moment detectorBrian H. Maranda, John A. Fawcett. 1357-1360 [doi]
• A probabilistic approach to the understanding and training of neural network classifiersHerbert Gish. 1361-1364 [doi]
• An extended fourth order blind identification algorithm in spatially correlated noiseV. C. Soon, Lang Tong, Y. F. Huang, R. Liu. 1365-1368 [doi]
• Adaptive detection with diagonal loading [for adaptive antenna array]S. Kalson. 1369-1372 [doi]
• Linear and non-linear adaptive noise cancelersGeorgios B. Giannakis, Amod V. Dandawate. 1373-1376 [doi]
• Analysis of a perceptron learning algorithm with momentum updatingJohn J. Shynk, Sumit Roy. 1377-1380 [doi]
• Supervised learning process of multi-layer perceptron neural networks using fast least squaresMahmood R. Azimi-Sadjadi, Stuart Citrin, Sassan Sheedvash. 1381-1384 [doi]
• Stochastic gradient adaptive filters with gradient adaptive step sizesV. John Mathews, Zhenhua Xie. 1385-1388 [doi]
• LMS and RLS adaptive L-filtersIoannis Pitas, Anastasios N. Venetsanopoulos. 1389-1392 [doi]
• Least squares adaptive equalizer using analytic signalsTetsuya Shimamura, Sho Takahashi. 1393-1396 [doi]
• A row-action projection algorithm for adaptive filteringJohn F. Doherty, Richard J. Mammone. 1397-1400 [doi]
• A new 2-D fast RLS algorithmA. M. Sequeira, Charles W. Therrien. 1401-1404 [doi]
• Modified LMS algorithms for robust ADPCMJean-Christophe Pesquet, Georgios Tziritas, Odile Macchi. 1405-1408 [doi]
• A new numerically stable structure for fast RLS adaptive filteringXiao-Hu Yu. 1409-1412 [doi]
• Adaptive multichannel digital filter with lattice-escalator hybrid structureKi-Man Kim, Il Whan Cha, Dae Hee Youn. 1413-1416 [doi]
• Convergence speed up in adaptive time delay estimationK. C. Ho 0001, P. C. Ching, Yiu-Tong Chan. 1417-1420 [doi]
• Optimum error nonlinearities for LMS adaptationScott C. Douglas, Teresa H. Y. Meng. 1421-1424 [doi]
• A fast self-optimized LMS algorithm for non-stationary identification: application to underwater equalizationP. Bragard, Geneviève Jourdain. 1425-1428 [doi]
• Efficient order recursive algorithms for multichannel LS filteringGeorge-Othon Glentis, Nicholas Kalouptsidis. 1429-1432 [doi]
• A dual form adaptive filterLouis B. Fertig, James H. McClellan. 1433-1436 [doi]
• Transform domain adaptive filtering with the chirp Z transformAndrew W. Hull, W. Kenneth Jenkins. 1437-1440 [doi]
• Alternative implementations for the frequency-domain LMS adaptive filterTokunbo Ogunfunmi, Allen M. Peterson. 1441-1444 [doi]
• A novel blind adaptive equalization techniqueXiao-Hu Yu, Zhen-Yang Wu. 1445-1448 [doi]
• On the implementation of adaptive filters with adjustable linear constraintsChing-Yih Tseng, Lloyd J. Griffiths. 1449-1452 [doi]
• A variable-step LMS based on tap weight time variationM. A. Arnao, S. S. Rao. 1453-1456 [doi]
• A fast exact least mean square adaptive algorithmJacob Benesty, Pierre Duhamel. 1457-1460 [doi]
• Adaptive comb filters for enhancement of quasi periodic signalsNacer K. M'Sirdi, Jean-Luc Zarader. 1461-1464 [doi]
• Mean square error in an IIR adaptive echo cancellerJohn B. Kenney, Mark D. Peek. 1465-1468 [doi]
• Adaptive least square complex lattice clutter rejection filters applied to the radar detection of low altitude windshearB. M. Keel, Ernest G. Baxa Jr.. 1469-1472 [doi]
• Multi microphone techniques for noise reduction in speech recognition devicesDan Chazan, Yoav Medan, N. Sharir. 1473-1474 [doi]
• Blind adaptive equalization for digital storage systems using timing interpolationToshiyuki Shimada, Sanjit K. Mitra. 1475-1478 [doi]
• Convergence and error analysis of the fixed point RLS algorithm with correlated inputsTülay Adals, Sasan H. Ardalan. 1479-1482 [doi]
• Robustness to quantization errors in LMS adaptation via degree of excitationGeoffrey A. Williamson, William A. Sethares. 1483-1486 [doi]
• Effect of floating-point error reduction with recursive least square for parallel architectureHiroshi Tsubokawa, Hajime Kubota, Shigeo Tsujii. 1487-1490 [doi]
• Transient behavior of the LMS algorithm: a deterministic approachPaul F. Titchener. 1491-1494 [doi]
• A new nesting scheme of PFA [prime factor algorithm]Bin Nong, Zhongde Wang. 1495-1498 [doi]
• Yet a faster address generation scheme for the computation of prime factor algorithmsDaniel Pak-Kong Lun, Raymond Chan, Wan-Chi Siu. 1499-1502 [doi]
• FFT implementation on DSP-chips-theory and practiceR. Meyer, Karl Schwarz. 1503-1506 [doi]
• Efficient FFT algorithms for DSP processors using tensor product decompositionsHenrik V. Sorensen, Charles A. Katz, C. Sidney Burrus. 1507-1510 [doi]
• A fast recursive bit-reversal algorithmJechang Jeong, William J. Williams. 1511-1514 [doi]
• Polynomial transform computation of the 2-D DCTPierre Duhamel, Christine Guillemot. 1515-1518 [doi]
• Efficient signal coding with hierarchical lapped transformsHenrique S. Malvar. 1519-1522 [doi]
• Fourier analysis using adaptive AFTWeiping Li. 1523-1526 [doi]
• Residue computations in rings of algebraic integersGary A. Ray. 1527-1530 [doi]
• Fast RNS DSP algorithms implemented with binary arithmeticElio D. Di Claudio, Gianni Orlandi, Francesco Piazza. 1531-1534 [doi]
• New results on Yen's approach to interpolation from nonuniformly spaced samplesGiancarlo Calvagno, David C. Munson Jr.. 1535-1538 [doi]
• On the sampling and reconstruction of time-warped bandlimited signalsDouglas Cochran, James J. Clark. 1539-1541 [doi]
• A functional minimization interpretation of fast iterative reconstruction algorithmsMark A. Richards. 1543-1546 [doi]
• A new algorithm for iterative deconvolution of sparse spike trainsYves Goussard, Guy Demoment, Jérôme Idier. 1547-1550 [doi]
• Short-time signal representation by nonlinear difference equationsThomas F. Quatieri, Edward M. Hofstetter. 1551-1554 [doi]
• A wavenumber-domain Kalman filtering approach to one-dimensional phase retrievalHamid Soltanian-Zadeh, Andrew E. Yagle. 1555-1558 [doi]
• Use of penalty function or Lagrangian multipliers for constrained deconvolutionG. Thomas, N. Souilah. 1559-1562 [doi]
• Iterative phase retrieval using two Fourier transform intensitiesWooshik Kim, Monson H. Hayes. 1563-1566 [doi]
• Signal reconstruction from the phase of the bispectrumAthina P. Petropulu, Chrysostomos L. Nikias. 1567-1570 [doi]
• A new iterative algorithm for minimum-phase signal reconstruction from its autocorrelationMarco Fratti. 1571-1574 [doi]
• An efficient VLSI CORDIC array structure implementation of Toeplitz eigensystem solversYu Hen Hu, Homer H. M. Chern. 1575-1578 [doi]
• Multichannel coupled split algorithms for non-Hermitian block Toeplitz matricesAndrew E. Yagle. 1579-1582 [doi]
• Fast least squares algorithm based on partitioning techniqueKaushal K. Dhar. 1583-1586 [doi]
• Regularized fast recursive least squares algorithmsAmrane Houacine. 1587-1590 [doi]
• Reconciling fast RLS lattice and QR algorithmsDirk T. M. Slock. 1591-1594 [doi]
• A class of fast covariance least squares algorithmsRichard M. Todd, J. R. Cruz. 1595-1598 [doi]
• Efficient computation of the wavelet transformsRamesh A. Gopinath, C. Sidney Burrus. 1599-1601 [doi]
• Discrete Mellin transform for signal analysisJacqueline Bertrand, Pierre Bertrand, Jean Philippe Ovarlez. 1603-1606 [doi]
• Time-frequency subspaces and their application to time-varying filteringFranz Hlawatsch, Werner Kozek, Werner Krattenthaler. 1607-1610 [doi]
• General signal synthesis algorithms for smoothed versions of Wigner distributionWerner Krattenthaler, Franz Hlawatsch. 1611-1614 [doi]
• An orthogonal set of frequency and amplitude modulated (FAM) functions for variable resolution signal analysisUnto K. Laine, Toomas Altosaar. 1615-1618 [doi]
• The fast Householder filters-RLS adaptive filterJ. Cioffi. 1619-1622 [doi]
• Stable recursive least squares filtering using an inverse QR decompositionAvinash L. Ghirnikar, S. Thomas Alexander. 1623-1626 [doi]
• A hybrid lattice-QR fast algorithm for least squares adaptive filteringPhillip A. Regalia, Maurice G. Bellanger. 1627-1630 [doi]
• Recursive LS filtering using block Householder transformationsK. J. Ray Liu, Shih Fu Hsieh, Kung Yao. 1631-1634 [doi]
• Recursive Givens reduction using a second-order windowPeter Strobach. 1635-1638 [doi]
• Synthesis of bandwidth efficient OQAM and VSB transmultiplexersRavi P. Ramachandran, Peter Kabal. 1639-1642 [doi]
• Digital prediction with spectral noise shapingMadeleine Bonnet, Mamadou Mboup, Odile Macchi. 1643-1646 [doi]
• Speech encryption using discrete orthogonal transformsS. Sridharan, Ed Dawson, Bruce Goldburg. 1647-1650 [doi]
• Modem constellation identification: a performance comparison of two methodsSally L. Wood, Michael G. Larimore, John R. Treichler. 1651-1654 [doi]
• The joint optimization of transmitter and receiver filters for data transmission systems using mean field annealingRamin A. Nobakht, David E. van den Bout, J. Keith Townsend, Sasan H. Ardalan. 1655-1658 [doi]
• Phase synchronizer for fading channels with the modified method of least squaresHiroyasu Sano, Masafumi Hagiwara, Masao Nakagawa. 1659-1662 [doi]
• Binary and biphase sequences and arrays with low periodic autocorrelation sidelobesLeopold Bömer, Markus Antweiler. 1663-1666 [doi]
• A new nonlinear equalizer for mobile radio channelsJörn Thielecke. 1667-1670 [doi]
• A new complex system identification method and its application to echo canceller fast initializationGuozhu Long, Fuyun Ling. 1671-1674 [doi]
• Performance of the 2-2 constant modulus (CM) adaptive algorithm for Rayleigh fading sinusoids in Gaussian noiseNeil J. Bershad, S. Roy. 1675-1678 [doi]
• The DSP implementation of a new timing recovery technique for high-speed digital data transmissionAlbert M. Gottlieb, Pedro M. Crespo, James L. Dixon, T. Russell Hsing. 1679-1682 [doi]
• Determination of the modulation type of communication signalsStanton B. McMillan, Brian P. Flanagan, Tom K. Doong. 1683-1686 [doi]
• A median type matched filterJaakko Astola, Yrjö Neuvo. 1687-1690 [doi]
• An improved analysis of ΣΔ modulators with bandlimited Gaussian inputsJohn T. Stonick, Sasan H. Ardalan, J. Keith Townsend. 1691-1694 [doi]
• Mobile radio bit synchronisation and time delay measurement with a digital phase locked loopWalter Stammler, Frank Negretti. 1695-1698 [doi]
• Blind equalization based on radius directed adaptationMichael J. Ready, Richard P. Gooch. 1699-1702 [doi]
• Suppression of CW interference and filtered noise in QPRS systems using decision-feedback filtersMasayoshi Miyagi, Toshiaki Ogawa, Iwao Sasase, Shinsaku Mori. 1703-1706 [doi]
• On the admissibility of blind adaptive equalizersZhi Ding 0001, C. Richard Johnson Jr., Rodney A. Kennedy. 1707-1710 [doi]
• Simultaneous frequency and phase estimation of digital modulated signals in burst mode transmissionMargarita Cabrera, Miguel A. Lagunas. 1711-1714 [doi]
• Automatic bit pattern analysis of communication signalsJames E. Gaby, Stanton B. McMillan. 1715-1718 [doi]
• An alternative realization of the SVD-based FSEM. Barton. 1719-1722 [doi]
• Wavelets and filter banks: relationships and new resultsMartin Vetterli, Cormac Herley. 1723-1726 [doi]
• A new approach for subband processingJ. C. Feauveau. 1727-1730 [doi]
• Lagrange multiplier approaches to the design of two-channel perfect-reconstruction linear-phase FIR filter banksBor-Rong Horng, Alan N. Wilson Jr.. 1731-1734 [doi]
• The time domain analysis and design of exactly reconstructing FIR analysis/synthesis filter banksKambiz Nayebi, Thomas P. Barnwell III, Mark J. T. Smith. 1735-1738 [doi]
• Narrow band analysis of a filter bank for the directional decomposition of imagesRoberto H. Bamberger, Mark J. T. Smith. 1739-1742 [doi]
• Quantization effects in high-speed pipelined recursive filtersKeshab K. Parhi, Gregory S. Munson, Lai Q. Pham. 1743-1746 [doi]
• When is DPCM a stable system?C. Uhl, Odile Macchi. 1747-1750 [doi]
• Time domain analysis of sigma delta modulationWu Chou, Teresa H. Meng, Robert M. Gray. 1751-1754 [doi]
• FIR digital filter with sigma-delta modulated input and impulse responsePing Wah Wong. 1755-1758 [doi]
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• An eigenstructure based nonparametric algorithm to estimate the frequencies of two sinusoids in white noiseK. R. Raghavan, S. S. Rao, Gregory H. Wakefield, Purusottam Mookerjee. 2679-2682 [doi]
• Adaptive structured eigenvector methodsSophocles J. Orfanidis. 2683-2686 [doi]
• Effect of spatial smoothing on the performance of noise subspace methodsBhaskar D. Rao, K. V. S. Hari. 2687-2690 [doi]
• Multiple window based minimum variance broadband spatial spectrum estimationTsung-Ching Liu, Barry D. Van Veen. 2691-2694 [doi]
• On a wold-like decomposition of 2-D discrete random fieldsJoseph M. Francos, A. Zvi Meiri, Boaz Porat. 2695-2698 [doi]
• Multidimensional autoregressive parameter estimation using iteratively reweighted least squaresSteven D. Blostein, Haydn S. Richardson. 2699-2702 [doi]
• A weighted min-norm method with a recursive implementationManuel Duarte Ortigueira, Miguel Angel Lagunas. 2703-2706 [doi]
• Specific 2-D spectral estimation for wideband beamformingMiguel Angel Lagunas, Gregori Vázquez, J. F. Sitjar. 2707-2710 [doi]
• Statistically robust signal subspace identificationLarry P. Ammann. 2711-2714 [doi]
• Mode, maximum likelihood and Cramer-Rao bound: conditional and unconditional resultsPetre Stoica, Arye Nehorai. 2715-2718 [doi]
• Time-domain identification of spectral signal subspaces of rational type with application to the broadband sources localization problemPhilippe Loubaton, Phillip A. Regalia. 2719-2722 [doi]
• The performance of two classes of active sonar detectors in a shallow water environmentD. Almagor, William S. Hodgkiss. 2723-2726 [doi]
• Multiuser sonar properties for Costas array frequency hop coded signalsEdward L. Titlebaum, Svetislav V. Maric. 2727-2730 [doi]
• A hidden Markov model approach to the classification of acoustic transientsM. K. Shields, Charles W. Therrien. 2731-2734 [doi]
• The theoretical prediction, interpretation and computation of the Fourier-Mellin transform applied to sonar classification of shipsJ. S. Lee, Martin J. Burke, Joe K. Hammond. 2735-2738 [doi]
• A neural network approach to the detection problem using joint time-frequency distributionsDonald B. Malkoff, Leon Cohen. 2739-2742 [doi]
• Broadband source detection and range/depth localization via full-wavefield (matched field) processingWilliam S. Hodgkiss, Richard K. Brienzo. 2743-2746 [doi]
• Eigen-based DOA estimators for non-linear array configurationsPaul R. White. 2747-2750 [doi]
• Signal detection in the presence of inaccurate multipath time delay modellingIsabel M. G. Lourtie, G. Clifford Carter. 2751-2754 [doi]
• Frequency lines data-fusion for a multi-octaves towed arrayS. Sitbon, P. Blanc-Benon. 2755-2758 [doi]
• Simultaneous signal detection and classification under a false alarm constraintAlfred O. Hero III, J. K. Kim. 2759-2762 [doi]
• Time delay estimation for closely spaced echoesAriela Zeira, Peter M. Schultheiss. 2763-2766 [doi]
• Performance analysis of a high resolution time delay estimation algorithmIvars P. Kirsteins, Alex C. Kot. 2767-2770 [doi]
• Estimation of source separation with an array of arbitrary shapeA. Bhuyan, Peter M. Schultheiss. 2771-2774 [doi]
• Direct design of focusing filters for wideband direction findingS. Sivanand, Mostafa Kaveh. 2775-2778 [doi]
• Cramer-Rao bounds for passive range and depth in a vertically inhomogeneous mediumMaria-João Rendas, José M. F. Moura. 2779-2782 [doi]
• Ranging using spectral informationJonathan S. Abel, Greg Gibbons. 2783-2786 [doi]
• Source localization by summing multiple correlator outputsStephen W. Davies, Mariner A. Price. 2787-2790 [doi]
• Passive localization of an emitting source by parametric meansJ. J. Towers, Yiu-Tong Chan. 2791-2794 [doi]
• Asymptotic and empirical results on approximate maximum likelihood and least squares estimates for sensor array processingDieter Kraus, Johann F. Böhme. 2795-2798 [doi]
• Resolution of coherent sources using an arc arrayDejian Ren, Wei-Ping Zhu. 2799-2802 [doi]
• 1 beamformer in the presence of underwater burst noiseVictor A. N. Barroso, José M. F. Moura. 2803-2806 [doi]
• Adaptive linear-quadratic array for detectionPascal Chevalier 0001, Bernard Picinbono. 2807-2810 [doi]
• Soft constrained minimum variance beamformingBarry D. Van Veen. 2811-2814 [doi]
• A signal subspace beamformerDavid R. Farrier, L. R. Prosper. 2815-2818 [doi]
• Consistency of direction-of-arrival estimation with multipath and few snapshotsArye Nehorai, David Starer, Petre Stoica. 2819-2822 [doi]
• Analysis of enhanced covariance matrix technique for source direction estimation in non-coherent and coherent signal environmentB. L. Lim, Y. C. Lim, Siew Kok Hui. 2823-2826 [doi]
• The detection performance of coherent wideband focusing for a spatially resampled arrayJeffrey L. Krolik, David N. Swingler. 2827-2830 [doi]
• Time domain approach to wideband source locationY. D. Wang, J. J. Fuchs. 2831-2834 [doi]
• Direction-of-arrival estimation using eigenstructure methods for maneuvering arraysV. V. Krishna, A. Paulraj. 2835-2838 [doi]
• Comparative performance of eigenvector rotation and MUSIC algorithms for angle-of-arrival estimationQun Shi, Fred Haber. 2839-2842 [doi]
• An improved MUSIC algorithm for superresolution array processingMeng Hwa Er, Siew Kok Hui. 2843-2846 [doi]
• Analytical results on the performance of some eigen-assisted array processors: two correlated source caseDejian Ren, Wei-Ping Zhu. 2847-2850 [doi]
• Maximum-likelihood multiple source tracking utilizing multiple coupled PLLsR. Lynn Kirlin, Yiqun Su. 2851-2854 [doi]
• Nusty bias effects of the cross spectral matrix when detecting target in presence of strong jammersAlain Maguer. 2855-2858 [doi]
• Adaptive recursive algorithms for image restoration and array processingSiew Kok Hui, Meng Hwa Er. 2859-2862 [doi]
• A deterministic approach to adaptive beamforming with imperfect arraysWassim G. Najm. 2863-2866 [doi]
• Performance study of the optimal beamformer in the M-sequence receiving systemZhaohong Zhang, Theodore G. Birdsall. 2867-2870 [doi]
• A high speed super resolution DOA algorithmAntone Kusmanoff, Rao K. Yarlagadda. 2871-2874 [doi]
• An approximate maximum entropy solution for bearing estimation of coherent sourcesJenho Tsao. 2875-2878 [doi]
• Optimum space-time processing for semi-stationary signals in spatially correlated noiseBenoît Champagne, Moshe Eizenman, Subbarayan Pasupathy. 2879-2882 [doi]
• Multiple-channel detection using generalized coherenceDouglas Cochran, Herbert Gish. 2883-2886 [doi]
• Instantaneous frequency estimation using the cross Wigner-Ville distribution with application to nonstationary transient detectionPeter O'Shea, Boualem Boashash. 2887-2890 [doi]
• Interference removal using multiple detection hypothesesD. E. Ohlms, W. A. Bradford, D. N. Nitka. 2891-2893 [doi]
• A normalization technique for non-additive noiseDominique Garreau, Michel Bouvet. 2895-2898 [doi]
• A comparison of five algorithms for tracking frequency and frequency rate-of-changeMark McIntyre, Anthony Ashley. 2899-2902 [doi]
• Congruential frequency hop signals for multi-user environments: a comparative analysisJerome R. Bellegarda. 2903-2906 [doi]
• Acoustic detection of propeller cavitationMichel Grenié. 2907-2910 [doi]
• Target classification in a passive sonar-an expert system approachR. Rajagopal 0001, B. Sankaranarayanan, P. Ramakrishna Rao. 2911-2914 [doi]
• Efficient location of closely spaced wide-band sourcesF. Giannella, Peter M. Schultheiss. 2915-2918 [doi]
• An efficient algorithm and pipelined VLSI architecture for the maximum likelihood estimation of directions of arrivalJeng-Kuang Hwang, Yung-Chang Chen. 2919-2922 [doi]
• Time delay estimation for ocean acoustic transient signal extraction in a multipath environmentRichard J. Vaccaro, E. Maragakis, R. L. Field. 2923-2926 [doi]
• Optimal sensor placement for passive source localizationJonathan S. Abel. 2927-2930 [doi]
• A new approach to the optimal modeling of sound propagation in the random oceanRui J. P. de Figueiredo, Guanrong Chen. 2931-2934 [doi]
• Sonar reverberation synthesis using nonstationary autoregressive modelingMichel Barlaud, J. M. Bruneau, M. Le Dard. 2935-2938 [doi]
• Array sensor localization and calibration by cyclic regressionJames Ting-Ho Lo. 2939-2942 [doi]
• Sufficient conditions for array calibration using sources of mixed typesMoshe Levi, Hagit Messer. 2943-2946 [doi]
• ESPRIT with perturbed subarraysEdward E. Hines. 2947-2950 [doi]
• Direction finding using an interpolated arrayBenjamin Friedlander. 2951-2954 [doi]
• On a high resolution array processing method non-based on the eigenanalysis approachSylvie Marcos, Messaoud Benidir. 2955-2958 [doi]
• Analysis of algorithms for sensor arrays with invariance structureBjörn E. Ottersten, Mats Viberg, Thomas Kailath. 2959-2962 [doi]
• Comparison of two efficient ML algorithms for high resolution sources tracking-time recursive implementationHenri Clergeot, Sara Tressens. 2963-2966 [doi]
• A comparative study of robust parameter estimation methodsLúcia V. R. Arruda, Gérard Favier, Wagner Caradori do Amaral. 2967-2970 [doi]