Table of Contents

• 4 kbps improved pitch prediction CELP speech coding with 20 ms frameMasahiro Serizawa, Kazunori Ozawa. 1-4 [doi]
• A low-complexity toll-quality variable bit rate coder for CDMA cellular systemsPeter Kroon, Michael C. Recchione. 5-8 [doi]
• Toll-quality 16 kb/s CELP speech coding with very low complexityJuin-Hwey Chen. 9-12 [doi]
• CELP coding using trellis-coded vector quantization of the excitationAndrei Popescu 0004, Nicolas Moreau, Claude Lamblin. 13-16 [doi]
• Interpolating the history improved excitation coding for high quality CELP codingPer Hedelin, Thomas Eriksson. 17-20 [doi]
• Fast stochastic codebook search through the use of odd-symmetric crosscorrelation basis vectorsCheung-fat Chan. 21-24 [doi]
• Improvements of background sound coding in linear predictive speech codersTorbjörn Wigren, Anders Bergström, Susanne Harrysson, Fredrik Jansson, Hans Nilson. 25-28 [doi]
• Improved CS-CELP speech coding in a noisy environment using a trained sparse conjugate codebookAkitoshi Kataoka, Sachiko Hosaka, Jotaro Ikedo, Takehiro Moriya, Shinji Hayashi. 29-32 [doi]
• CELP coding based on mel-cepstral analysisKazuhito Koishida, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai. 33-36 [doi]
• An embedded scheme for regular pulse excited (RPE) linear predictive codingShude Zhang, Gordon Lockhart. 37-40 [doi]
• Performance of the IBM large vocabulary continuous speech recognition system on the ARPA Wall Street Journal taskLalit R. Bahl, S. Balakrishnan-Aiyer, Jerome R. Bellegarda, Martin Franz, P. S. Gopalakrishnan, David Nahamoo, Miroslav Novak, Mukund Padmanabhan, Michael A. Picheny, Salim Roukos. 41-44 [doi]
• New developments in the Lincoln stack-decoder based large-vocabulary CSR systemDouglas B. Paul. 45-48 [doi]
• Large vocabulary continuous speech recognition using word graphsXavier L. Aubert, Hermann Ney. 49-52 [doi]
• Reducing word error rate on conversational speech from the Switchboard corpusPhilippe Jeanrenaud, Ellen Eide, Upendra V. Chaudhari, John W. McDonough, Kenney Ng, Man-Hung Siu, Herbert Gish. 53-56 [doi]
• Golden Mandarin (III)-a user-adaptive prosodic-segment-based Mandarin dictation machine for Chinese language with very large vocabularyRen-Yuan Lyu, Lee-Feng Chien, Shiao-Hong Hwang, Hung-Yun Hsieh, Rung-Chiuan Yang, Bo-Ren Bai, Jia-Chi Weng, Yen-Ju Yang, Shi-Wei Lin, Keh-Jiann Chen, Chiu-yu Tseng, Lin-Shan Lee. 57-60 [doi]
• Complete recognition of continuous Mandarin speech for Chinese language with very large vocabulary but limited training dataHsin-Min Wang, Jia-lin Shen, Yen-Ju Yang, Chiu-yu Tseng, Lin-Shan Lee. 61-64 [doi]
• Developments in continuous speech dictation using the ARPA WSJ taskJean-Luc Gauvain, Lori Lamel, Martine Adda-Decker. 65-68 [doi]
• Recent improvements to the ABBOT large vocabulary CSR systemMichael M. Hochberg, Steve Renals, Anthony J. Robinson, Gary D. Cook. 69-72 [doi]
• The 1994 HTK large vocabulary speech recognition systemPhilip C. Woodland, C. J. Leggetter, Julian Odell, Valtcho Valtchev, Steve J. Young. 73-76 [doi]
• Tangerine: a large vocabulary Mandarin dictation systemYuqing Gao, Hsiao-Wuen Hon, Zhiwei Lin, Gareth Loudon, S. Yogananthan, Baosheng Yuan. 77-80 [doi]
• WSJCAMO: a British English speech corpus for large vocabulary continuous speech recognitionTony Robinson, Jeroen Fransen, David Pye, Jonathan Foote, Steve Renals. 81-84 [doi]
• Voice across Hispanic America: a telephone speech corpus of American SpanishYeshwant K. Muthusamy, Edward Holliman, Barbara Wheatley, Joseph Picone, John J. Godfrey. 85-88 [doi]
• Implementation of the POW (phonetically optimized words) algorithm for speech databaseYeonja Lim, YoungJik Lee. 89-92 [doi]
• Microsoft Windows highly intelligent speech recognizer: WhisperXuedong Huang, Alex Acero, Fil Alleva, Mei-Yuh Hwang, Li Jiang, Milind Mahajan. 93-96 [doi]
• Concept-based speech translationLaura Mayfield, Marsal Gavaldà, Wayne H. Ward, Alex Waibel. 97-100 [doi]
• PhoneBook: a phonetically-rich isolated-word telephone-speech databaseJohn F. Pitrelli, Cynthia Fong, Suk Wong, Judith Spitz, Hong C. Leung. 101-104 [doi]
• CTIMIT: a speech corpus for the cellular environment with applications to automatic speech recognitionKathy L. Brown, E. Bryan George. 105-108 [doi]
• Toward movement-invariant automatic lip-reading and speech recognitionPaul Duchnowski, Martin Hunke, Dietrich Büsching, Uwe Meier, Alex Waibel. 109-112 [doi]
• Some results with a trainable speech translation and understanding systemVíctor M. Jiménez, Antonio Castellanos, Enrique Vidal. 113-116 [doi]
• A continuous speech recognition system using finite state network and Viterbi beam search for the automatic interpretationNam-Yong Han, Hoi-Rin Kim, Kyu-Woong Hwang, Young-Mok Ahn, Joon-Hyung Ryoo. 117-120 [doi]
• Robust speech recognition based on stochastic matchingAnanth Sankar, Chin-Hui Lee. 121-124 [doi]
• On the robustness of linear discriminant analysis as a preprocessing step for noisy speech recognitionOlivier Siohan. 125-128 [doi]
• A maximum likelihood procedure for a universal adaptation method based on HMM compositionYasuhiro Minami, Sadaoki Furui. 129-132 [doi]
• A fast and flexible implementation of parallel model combinationMark John Francis Gales, Steve J. Young. 133-136 [doi]
• Multivariate-Gaussian-based cepstral normalization for robust speech recognitionPedro J. Moreno, Bhiksha Raj, Evandro B. Gouvêa, Richard M. Stern. 137-140 [doi]
• Robust speech recognition in noise using adaptation and mapping techniquesLeonardo Neumeyer, Mitchel Weintraub. 141-144 [doi]
• Noisy speech recognition using robust inversion of hidden Markov modelsSeokyong Moon, Jenq-Neng Hwang. 145-148 [doi]
• Rapid environment adaptation for robust speech recognitionKeizaburo Takagi, Hiroaki Hattori, Takao Watanabe. 149-152 [doi]
• Noise estimation techniques for robust speech recognitionHans-Günter Hirsch, Christoph Ehrlicher. 153-156 [doi]
• Pole-filtered cepstral mean subtractionDevang Naik. 157-160 [doi]
• Language model adaptation via minimum discrimination informationP. Srinivasa Rao, Michael D. Monkowski, Salim Roukos. 161-164 [doi]
• Clustering word category based on binomial posteriori co-occurrence distributionMasafumi Tamoto, Takeshi Kawabata. 165-168 [doi]
• Language modeling by variable length sequences: theoretical formulation and evaluation of multigramsSabine Deligne, Frédéric Bimbot. 169-172 [doi]
• An integrated grammar/bigram language model using path scoresHarvey Lloyd-Thomas, Jerry H. Wright, Gareth J. F. Jones. 173-176 [doi]
• Discourse structure for multi-speaker spontaneous spoken dialogs: incorporating heuristics into stochastic RTNsSheryl R. Young. 177-180 [doi]
• Improved backing-off for M-gram language modelingReinhard Kneser, Hermann Ney. 181-184 [doi]
• QWI: a method for improved smoothing in language modellingGermán Bordel, M. Inés Torres, Enrique Vidal. 185-188 [doi]
• Using a stochastic context-free grammar as a language model for speech recognitionDaniel Jurafsky, Chuck Wooters, Jonathan Segal, Andreas Stolcke, Eric Fosler, Gary Tajchman, Nelson Morgan. 189-192 [doi]
• Improved language modelling by unsupervised acquisition of structureKlaus Ries, Finn Dag Buø, Ye-Yi Wang. 193-196 [doi]
• Understanding referring expressions in a person-machine spoken dialogueClaudia Pateras, Gregory Dudek, Renato de Mori. 197-200 [doi]
• Analysis of acoustic-phonetic variations in fluent speech using TIMITDon X. Sun, Li Deng. 201-204 [doi]
• Analysing weaknesses of language models for speech recognitionJoerg P. Ueberla. 205-208 [doi]
• A hidden Markov model with optimized inter-frame dependenceF. Jack Smith, Ji Ming, Peter O'Boyle, A. D. Irvine. 209-212 [doi]
• On the use of scalar quantization for fast HMM computationShigeki Sagayama, Satoshi Takahashi. 213-216 [doi]
• Large-vocabulary speech recognition in specialized domainsHaakon Chevalier, Chuck Ingold, Carol Kunz, Chip Moore, Crispen Roven, Jon Yamron, Bradley Baker, Paul G. Bamberg, Sarah Bridle, Tracy Bruce, Amy Weader. 217-220 [doi]
• Understanding and improving speech recognition performance through the use of diagnostic toolsEllen Eide, Herbert Gish, Philippe Jeanrenaud, Angela Mielke. 221-224 [doi]
• Phrase bigrams for continuous speech recognitionEgidio P. Giachin. 225-228 [doi]
• Using explicit segmentation to improve HMM phone recognitionCarl D. Mitchell, Mary P. Harper, Leah H. Jamieson. 229-232 [doi]
• Viterbi algorithm for acoustic vectors generated by a linear stochastic differential equation on each stateMarco Saerens. 233-236 [doi]
• Non-deterministic stochastic language models for speech recognitionGiuseppe Riccardi, Enrico Bocchieri, Roberto Pieraccini. 237-240 [doi]
• Improving 16 kb/s G.728 LD-CELP speech coder for frame erasure channelsCraig R. Watkins, Juin-Hwey Chen. 241-244 [doi]
• Reconstruction of missing packets for CELP-based speech codersAamir Husain, Vladimir Cuperman. 245-248 [doi]
• A robust variable-rate speech coderAlbert Shen, Benjamim Tang, Abeer Alwan, Greg Pottie. 249-252 [doi]
• Wideband speech coding using multiple codebooks and glottal pulsesCiará McElroy, Brian P. Murray, Anthony D. Fagan. 253-256 [doi]
• Speech coding using ISI coded quantizationNam Phamdo, Cheng-Chieh Lee, Rajiv Laroia. 257-260 [doi]
• New techniques for multi-prototype waveform coding at 2.84 kb/sIan S. Burnett, G. J. Bradley. 261-264 [doi]
• Quantization of non-linear predictors in speech codingJes Thyssen, Henrik Nielsen, Steffen Duus Hansen. 265-268 [doi]
• A fast robust stochastic algorithm for vector quantizer design for nonstationary channelsBalázs Kövesi, Samir Saoudi, Jean-Marc Boucher, Z. Reguly. 269-272 [doi]
• Voice quality of interconnected PCS, Japanese cellular, and public switched telephone networksSpiros Dimolitsas, Franklin L. Corcoran, Channasandra Ravishankar, Marion Baraniecki. 273-276 [doi]
• Objective speech measure for Chinese in wireless environmentKai Hung Lam, Oscar C. Au, C. C. Chan, K. F. Hui, S. F. Lau. 277-280 [doi]
• A training procedure for verifying string hypotheses in continuous speech recognitionRichard C. Rose, Biing-Hwang Juang, Chin-Hui Lee. 281-284 [doi]
• Robust utterance verification for connected digits recognitionMazin G. Rahim, Chin-Hui Lee, Biing-Hwang Juang. 285-288 [doi]
• A hybrid wordspotting method for spontaneous speech understanding using word-based pattern matching and phoneme-based HMMHiroshi Kanazawa, Mitsuyoshi Tachimori, Yoichi Takebayashi. 289-292 [doi]
• Acoustic and language modeling of human and nonhuman noises for human-to-human spontaneous speech recognitionTanja Schultz, Ivica Rogina. 293-296 [doi]
• LVCSR log-likelihood ratio scoring for keyword spottingMitchel Weintraub. 297-300 [doi]
• Keyword spotting using supervised/unsupervised competitive learningChakib Tadj, Franck Poirier. 301-304 [doi]
• A continuous density neural tree network word spotting systemStephen V. Kosonocky, Richard J. Mammone. 305-308 [doi]
• Video mail retrieval: the effect of word spotting accuracy on precisionGareth J. F. Jones, J. T. Foote, Karen Spärck Jones, Steve J. Young. 309-312 [doi]
• Improved topic spotting through statistical modelling of keyword dependenciesJerry H. Wright, Michael J. Carey 0002, Eluned S. Parris. 313-316 [doi]
• Topic focusing mechanism for speech recognition based on probabilistic grammar and topic Markov modelTakeshi Kawabata. 317-320 [doi]
• The influence of noise on the speaker recognition performance using the higher frequency bandShoji Hayakawa, Fumitada Itakura. 321-324 [doi]
• Measuring fine structure in speech: application to speaker identificationCharles R. Jankowski Jr., Thomas F. Quatieri, Douglas A. Reynolds. 325-328 [doi]
• The effects of telephone transmission degradations on speaker recognition performanceDouglas A. Reynolds, Marc A. Zissman, Thomas F. Quatieri, Gerald C. O'Leary, Beth A. Carlson. 329-332 [doi]
• Covariance estimation methods for channel robust text-independent speaker identificationMichael Schmidt, Herbert Gish, Angela Mielke. 333-336 [doi]
• Channel and noise compensation for text dependent speaker verification over telephoneWilliam Y. Hueng, Bhaskar D. Rao. 337-340 [doi]
• Testing with the YOHO CD-ROM voice verification corpusJoseph P. Campbell. 341-344 [doi]
• An orthogonal polynomial representation of speech signals and its probabilistic model for text independent speaker verificationChi-shi Liu, Hsiao-Chuan Wang, Frank K. Soong, Chao-Shih Huang. 345-348 [doi]
• Text-dependent speaker verification using data fusionKevin R. Farrell. 349-352 [doi]
• Neural net approaches to speaker verification: comparison with second order statistic measuresM. Mehdi Homayounpour, Gérard Chollet. 353-356 [doi]
• A subword neural tree network approach to text-dependent speaker verificationHan-Sheng Liou, Richard J. Mammone. 357-360 [doi]
• Statistical modeling of speech feature vector trajectories based on a piecewise continuous mean pathMark M. Thomson. 361-364 [doi]
• Trace-segmentation of isolated utterances for speech recognitionEuvaldo F. Cabral Jr., Graham Tattersall. 365-368 [doi]
• Optimal linear feature transformations for semi-continuous hidden Markov modelsErnst Günter Schukat-Talamazzini, Joachim Hornegger, Heinrich Niemann. 369-372 [doi]
• Use of generalized dynamic feature parameters for speech recognition: maximum likelihood and minimum classification error approachesRathinavelu Chengalvarayan, Li Deng. 373-376 [doi]
• A statistical pattern recognition approach to robust recursive identification of non-stationary AR model of speech production systemMilan Z. Markovic, Branko D. Kovacevic, Milan M. Milosavljevic. 377-380 [doi]
• The NP speech activity detection algorithmJoseph Pencak, Douglas J. Nelson. 381-384 [doi]
• Improved speech modeling and recognition using multi-dimensional articulatory states as primitive speech unitsLi Deng, J. Nu, Hossein Sameti. 385-388 [doi]
• Speech analysis based on Malvar wavelet transformChristophe Ris, Vincent Fontaine, Henri Leich. 389-392 [doi]
• Magnitude spectral estimation via Poisson moments with application to speech recognitionSamel Çelebi, Jose C. Principe. 393-396 [doi]
• Stochastic perceptual models of speechNelson Morgan, Hervé Bourland, Steven Greenberg, Hynek Hermansky, Su-Lin Wu. 397-400 [doi]
• Auditory scene analysis and hidden Markov model recognition of speech in noisePhil D. Green, Martin P. Cooke, M. D. Crawford. 401-404 [doi]
• Speech enhancement based on temporal processingHynek Hermansky, Eric A. Wan, Carlos Avendaño. 405-408 [doi]
• A comparative study of mel cepstra and EIH for phone classification under adverse conditionsSumeet Sandhu, Oded Ghitza. 409-412 [doi]
• Supplementary orthogonal cepstral featuresKhaled T. Assaleh. 413-416 [doi]
• Subband analysis for robust speech recognition in the presence of car noiseEngin Erzin, A. Enis Çetin, Yasemin Yardimci. 417-420 [doi]
• Robust speech feature extraction using SBCOR analysisShoji Kajita, Fumitada Itakura. 421-424 [doi]
• New HOS-based parameter estimation methods for speech recognition in noisy environmentsAsunción Moreno, Sergio Tortola, Josep Vidal, José A. R. Fonollosa. 429-432 [doi]
• Noise compensation for speech recognition in car noise environmentsRuikang Yang, Petri Haavisto. 433-436 [doi]
• Speech recognition in impulsive noiseSaeed Vaseghi, Ben P. Milner. 437-440 [doi]
• Speaker-independent phone modeling based on speaker-dependent HMMs' composition and clusteringTetsuo Kosaka, Shoichi Matsunaga, Mikio Kuraoka. 441-444 [doi]
• Using morphology towards better large-vocabulary speech recognition systemsPetra Geutner. 445-448 [doi]
• Optimal splitting of HMM Gaussian mixture components with MMIE trainingYves Normandin. 449-452 [doi]
• Dictionary learning: performance through consistencyTilo Sloboda. 453-456 [doi]
• Incremental MAP estimation of HMMs for efficient training and improved performanceYoshihoko Goto, Michael M. Hochberg, Daniel J. Mashao, Harvey F. Silverman. 457-460 [doi]
• Discrete MMI probability models for HMM speech recognitionJ. T. Foote. 461-464 [doi]
• Global discrimination for neural predictive systems based on N-best algorithmAbdelhamid Mellouk, Patrick Gallinari. 465-468 [doi]
• Enhancement of discriminative capabilities of HMM based recognizer through modification of Viterbi algorithmJianming Song. 469-472 [doi]
• A generalization of the Baum algorithm to functions on non-linear manifoldsDimitri Kanevsky. 473-476 [doi]
• Data-driven codebook adaptation in phonetically tied SCHMMsThomas Kemp. 477-479 [doi]
• NATO STANAG 4479: a standard for an 800 bps vocoder and channel coding in HF-ECCM systemB. Mouy, P. De La Noue, G. Goudezeune. 480-483 [doi]
• Harmonic and noise coding of LPC residuals with classified vector quantizationMasayuki Nishiguchi, Jun Matsumoto. 484-487 [doi]
• Progress towards a new government standard 2400 bps voice coderM. A. Kohler, L. M. Supplee, T. E. Tremain. 488-491 [doi]
• Variable dimension spectral coding of speech at 2400 bps and below with phonetic classificationAmitava Das, Allen Gersho. 492-495 [doi]
• Spectral excitation coding of speech at 2.4 kb/sV. Cuperman, P. Lupini, B. Bhattacharya. 496-499 [doi]
• A robust 2400 bps subband LPC vocoderP. A. Laurent, P. De La Noue. 500-503 [doi]
• Band-widened harmonic vocoder at 2 to 4 kbpsGao Yang, G. Zanellato, H. Leich. 504-507 [doi]
• A speech coder based on decomposition of characteristic waveformsW. Bastiaan Kleijn, Jesper Haagen. 508-511 [doi]
• Speech compression using pitch synchronous interpolationR. Taori, R. J. Sluijter, E. Kathmann. 512-515 [doi]
• Pitch synchronous multi-band (PSMB) speech codingHaiyun Yang, Soo Ngee Koh, Pratab Sivaprakasapillai. 516-519 [doi]
• Four-level tied-structure for efficient representation of acoustic modelingSatoshi Takahashi, Shigeki Sagayama. 520-523 [doi]
• Application of clustering techniques to mixture density modelling for continuous-speech recognitionChristian Dugast, Peter Beyerlein, Reinhold Haeb-Umbach. 524-527 [doi]
• Context dependent phonetic duration models for decoding conversational speechMichael D. Monkowski, Michael A. Picheny, P. Srinivasa Rao. 528-531 [doi]
• A unified way in incorporating segmental feature and segmental model into HMMJun He, Henri Leich. 532-535 [doi]
• Experimental evaluation of segmental HMMsWendy J. Holmes, Martin J. Russell. 536-539 [doi]
• Improved acoustic modeling for speech recognition using 2D Markov random fieldsHelmut Lucke. 540-543 [doi]
• Structured Markov models for speech recognitionF. Wolfertstetter, G. Ruske. 544-547 [doi]
• Robust parametric modeling of durations in hidden Markov modelsDavid Burshtein. 548-551 [doi]
• Improved decision trees for phonetic modelingRoland Kuhn, Ariane Lazaridès, Yves Normandin, Julie Brousseau. 552-555 [doi]
• High speed speech recognition using tree-structured probability density functionTakao Watanabe, Koichi Shinoda, Keizaburo Takagi, Ken-ichi Iso. 556-559 [doi]
• A fast segmental Viterbi algorithm for large vocabulary recognitionP. Laface, C. Vair, L. Fissore. 560-563 [doi]
• Searching with a transcription graphZ. Li, P. Kenny, Douglas O'Shaughnessy. 564-567 [doi]
• On the use of stochastic inference networks for representing multiple word pronunciationsRenato de Mori, Charles Snow, Michael Galler. 568-571 [doi]
• A tree search strategy for large-vocabulary continuous speech recognitionP. S. Gopalakrishnan, L. R. Bahl, R. L. Mercer. 572-575 [doi]
• Lattice-based search strategies for large vocabulary speech recognitionF. Richardson, M. Ostendorf, J. R. Rohlicek. 576-579 [doi]
• On using a priori segmentation of the speech signal in an N-best solutions post-processingT. Moudenc, D. Jouvet, J. Monne. 580-583 [doi]
• Time-synchronous continuous speech recognizer driven by a context-free grammarTohru Shimizu, Seikou Monzen, Harald Singer, Shoichi Matsunaga. 584-587 [doi]
• Language model representations for beam-search decodingGiuliano Antoniol, Fabio Brugnara, Mauro Cettolo, Marcello Federico. 588-591 [doi]
• A lower-complexity Viterbi algorithmSarvar Patel. 592-595 [doi]
• Efficient search using posterior phone probability estimatesSteve Renals, Mike Hochberg. 596-599 [doi]
• Timing patterns in fluent and disfluent spontaneous speechDouglas O'Shaughnessy. 600-603 [doi]
• Stochastic modeling of pause insertion using context-free grammarShigeru Fujio, Yoshinori Sagisaka, Norio Higuchi. 604-607 [doi]
• Automatic classification of pitch movements via MLP-based estimation of class probabilitiesLouis F. M. ten Bosch. 608-611 [doi]
• On the effects of speech rate in large vocabulary speech recognition systemsMatthew A. Siegler, Richard M. Stern. 612-615 [doi]
• A prosodic model of Mandarin speech and its application to pitch level generation for text-to-speechShaw-Hwa Hwang, Sin-Horng Chen. 616-619 [doi]
• Prosodic cues to word usageKaren Ward, David G. Novick. 620-623 [doi]
• 0 clustering using superpositional modelingMitsuru Nakai, Harald Singer, Yoshinori Sagisaka, Hiroshi Shimodaira. 624-627 [doi]
• Duration modeling in large vocabulary speech recognitionAnastasios Anastasakos, Richard Schwartz, Han Shu. 628-631 [doi]
• Speaker-independent automatic classification of Thai tones in connected speech by analysis-synthesis methodSiripong Potisuk, Mary P. Harper, Jackson T. Gandour. 632-635 [doi]
• Speech synthesis system based on a variable decimation/interpolation factorF. M. Gimenez de los Galanes, M. H. Savoji, J. M. Pardo. 636-639 [doi]
• Automatic speech synthesiser parameter estimation using HMMsR. E. Donovan, P. C. Woodland. 640-643 [doi]
• Speaker modification with LPC pole analysisJanet Slifka, Timothy R. Anderson. 644-647 [doi]
• Synthesizing styled speech using the Klatt synthesizerJanet C. Rutledge, Kathleen E. Cummings, Daniel A. Lambert, Mark A. Clements. 648-651 [doi]
• Acoustic measurements of the vocal-tract area function: sensitivity analysis and experimentsHani Yehia, Masaaki Honda, Fumitada Itakura. 652-655 [doi]
• Shape-invariant pitch-synchronous text-to-speech conversionEduardo R. Banga, Carmen Garcia-Mateo. 656-659 [doi]
• Speech parameter generation from HMM using dynamic featuresKeiichi Tokuda, Takao Kobayashi, Satoshi Imai. 660-663 [doi]
• A source generator based modeling framework for synthesis of speech under stressSahar E. Bou-Ghazale, John H. L. Hansen. 664-667 [doi]
• MBE synthesis of speech coded in LPC formatK. F. Lam, C. F. Chan. 668-671 [doi]
• Modelling speech production using Yee's finite difference methodKathleen E. Cummings, James G. Maloney, Mark A. Clements. 672-675 [doi]
• Batch, incremental and instantaneous adaptation techniques for speech recognitionG. Zavaliagkos, R. Schwartz, J. Makhoul. 676-679 [doi]
• Speaker adaptation using combined transformation and Bayesian methodsVassilios Digalakis, Leonardo Neumeyer. 680-683 [doi]
• Rapid speaker adaptation using model predictionS. M. Ahadi, P. C. Woodland. 684-687 [doi]
• Speaker adaptation based on transfer vector field smoothing using maximum a posteriori probability estimationMasahiro Tonomura, Tetsuo Kosaka, Shoichi Matsunaga. 688-691 [doi]
• Experiments using data augmentation for speaker adaptationJerome R. Bellegarda, Peter V. de Souza, David Nahamoo, Mukund Padmanabhan, Michael A. Picheny, Lalit R. Bahl. 692-695 [doi]
• Vector-field-smoothed Bayesian learning for incremental speaker adaptationJun-ichi Takahashi, Shigeki Sagayama. 696-699 [doi]
• A speaker adaptation technique using linear regressionS. J. Cox. 700-703 [doi]
• Speaker adaptation based on spectral normalization and dynamic HMM parameter adaptationMing-Whei Feng. 704-707 [doi]
• On-line Bayes adaptation of SCHMM parameters for speech recognitionQiang Huo, Chorkin Chan. 708-711 [doi]
• Iterative self-learning speaker and channel adaptation under various initial conditionsYunxin Zhao. 712-715 [doi]
• Fast and low-complexity LSF quantization using algebraic vector quantizerMinjie Xie, Jean-Pierre Adoul. 716-719 [doi]
• Low cost vector quantization methods for spectral coding in low rate speech codersH. R. Sadegh Mohammadi, W. H. Holmes. 720-723 [doi]
• Matrix product quantization for very-low-rate speech codingStefan Bruhn. 724-727 [doi]
• An intrinsically reliable and fast algorithm to compute the line spectrum pairs (LSP) in low bit rate CELP codingA. Goalic, S. Saoudi. 728-731 [doi]
• Spectral dynamics is more important than spectral distortionH. Petter Knagenhjelm, W. Bastiaan Kleijn. 732-735 [doi]
• Efficient quantization of LSF parameters using classified SVQ combined with conditional splittingDong-Il Chang, Young-kwon Cho, SouGuil Ann. 736-739 [doi]
• Efficient coding of LSP parameters using split matrix quantisationC. S. Xydeas, C. Papanastasiou. 740-743 [doi]
• How good is your β?-observations on VQ training ratiosJohn S. Collura, Thomas E. Tremain. 744-747 [doi]
• Variable rate spectral quantization for phonetically classified CELP codingRoar Hagen, Erdal Paksoy, Allen Gersho. 748-751 [doi]
• Optimal distortion measures for the high rate vector quantization of LPC parametersWilliam R. Gardner, Bhaskar D. Rao. 752-755 [doi]
• Harmonics tracking and pitch extraction based on instantaneous frequencyToshihiko Abe, Takao Kobayashi, Satoshi Imai. 756-759 [doi]
• Decomposition of speech signals into deterministic and stochastic componentsC. D'Alessandro, B. Yegnanarayana, V. Darsinos. 760-763 [doi]
• Modeling and processing speech with sums of AM-FM formant modelsShan Lu, Peter C. Doerschuk. 764-767 [doi]
• On the statistical properties of line spectrum pairsJ. S. Erkelens, P. M. T. Broersen. 768-771 [doi]
• Individual variations in glottal characteristics of female speakersHelen M. Hanson. 772-775 [doi]
• A robust method for determining instants of major excitations in voiced speechB. Yegnanarayana, R. L. H. M. Smits. 776-779 [doi]
• Interpolation of LPC spectra via pole shiftingVladimir Goncharoff, Maureen Kaine-Krolak. 780-783 [doi]
• Speech formant frequency and bandwidth tracking using multiband energy demodulationAlexandros Potamianos, Petros Maragos. 784-787 [doi]
• Nonlinear prediction for speech coding using radial basis functionsFernando Diaz-de-Maria, Fernando R. Figueiras-Vidal. 788-791 [doi]
• Recognition of unvoiced stops from their time-frequency representationMaria Rangoussi, Anastasios Delopoulos. 792-795 [doi]
• Speech enhancement based on masking properties of the auditory systemNathalie Virag. 796-799 [doi]
• Optimizing speech enhancement by exploiting masking properties of the human earA. Akbari Azirani, R. Le Bouquin Jeannes, G. Faucon. 800-803 [doi]
• A spectrally-based signal subspace approach for speech enhancementYariv Ephraim, Harry L. van Trees. 804-807 [doi]
• Real-time implementation of HMM-based MMSE algorithm for speech enhancement in hearing aid applicationsH. Sheikhzadeh, R. L. Brennan, H. Sameti. 808-811 [doi]
• New methods for adaptive noise suppressionLevent Arslan, Alan McCree, Vishu Viswanathan. 812-815 [doi]
• Single-sensor speech enhancement using a soft-decision/variable attenuation algorithmE. Bryan George. 816-819 [doi]
• Speech enhancement using a ternary-decision based filterT. S. Sun, S. Nandkumar, J. Carmody, J. Rothweiler, A. Goldschen, N. Russell, S. Mpasi, P. Green. 820-823 [doi]
• Signal modeling enhancements for automatic speech recognitionZaki B. Nossair, Peter L. Silsbee, Stephen A. Zahorian. 824-827 [doi]
• Co-channel speaker separationDavid P. Morgan, E. Bryan George, Leonard T. Lee, Stephen M. Kay. 828-831 [doi]
• Speech enhancement based on the generalized dual excitation model with adaptive analysis windowChang D. Yoo, Jae S. Lim. 832-835 [doi]
• Foreign accent classification using source generator based prosodic featuresJohn H. L. Hansen, Levent M. Arslan. 836-839 [doi]
• Automatic transcription of unknown words in a speech recognition systemR. Haeb-Umbach, P. Beyerlein, E. Thelen. 840-843 [doi]
• An evaluation of an adaptive multichannel system for speech enhancement with automatic phase alignmentSilvana L. do N. Cunha Costa, Benedito G. Aguiar Neto. 844-847 [doi]
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• Set estimation via ellipsoidal approximationsAshutosh Sabharwal, Lee C. Potter. 897-900 [doi]
• 2-D phase retrieval by partitioning into coupled 1-D problems using discrete Radon transformsHyunduk Ahn. 901-904 [doi]
• Fast minimum variance resamplingTodd Findley Brennan, Paul H. Milenkovic. 905-908 [doi]
• On the kernel MUSIC algorithmHiroshi Shimotahira. 909-912 [doi]
• A new kind of adaptive frequency shift filterJian-Kang Zhang, Kon Max Wong, Qu Jin, Qiang Wu. 913-916 [doi]
• Dynamic tracking filters for decomposing nonstationary sinusoidal signalsAshwin Rao, Ramdas Kumaresan. 917-920 [doi]
• The DRFT-a rotation in time-frequency spaceBalu Santhanam, James H. McClellan. 921-924 [doi]
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• On the dynamics of the LRE algorithm: a distribution learning approach to adaptive equalizationTülay Adali, M. Kemal Sönmez, Kartik Patel. 929-932 [doi]
• Stability analysis of robust adaptive filter used in feedforward and feedback compensationMakoto Kajiki, Jingmin Xin, Hiromitsu Ohmori, Akira Sano. 933-936 [doi]
• Optimal filtering in fractional Fourier domainsM. Alper Kutay, Haldun M. Özaktas, Levent Onural, Orhan Arikan. 937-940 [doi]
• A time-varying AR modeling of heart wall vibrationHiroshi Kanai, Noriyoshi Chubachi, Yoshiro Koiwa. 941-944 [doi]
• Fast projection algorithm and its step size controlMasashi Tanaka, Yutaka Kaneda, Shoji Makino, Junji Kojima. 945-948 [doi]
• ∞ adaptive filteringBabak Hassibi, Thomas Kailath. 949-952 [doi]
• A fast robust LMS algorithm utilizing the dynamics of a damped pendulumTariq Haddad, Mohammed A. Khasawneh. 953-956 [doi]
• Adaptive Volterra filters using orthogonal structuresV. John Mathews. 957-960 [doi]
• Partition-based adaptive estimation of single response evoked potentialsAhmad M. Sarhan, Russell C. Hardie, Kenneth E. Barner. 961-964 [doi]
• Output-error LMS bilinear filters with stability monitoringJunghsi Lee, V. John Mathews. 965-968 [doi]
• Acoustic echo cancellation using a fast QR-RLS algorithm and multirate schemesFrançois Capman, Jérôme Boudy, Philip Lockwood. 969-972 [doi]
• On the convergence enhancment of the wavelet transform based LMSSumir Attallah, Mohamed Najim. 973-976 [doi]
• Adaptive nonlinear Wiener-Laguerre-lattice modelsZoran Fejzo, Hanoch Lev-Ari. 977-980 [doi]
• Memoryless polynomial adaptive predistortion [TV transmitters]John T. Stonick, Virginia L. Stonick, José M. F. Moura, R. Sam Zborowski. 981-984 [doi]
• Adaptive filtering in subbands using a weighted criterionMarc de Courville, Pierre Duhamel. 985-988 [doi]
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• A subband adaptive filter with the optimum analysis filter bankHiroshi Ochi, Yoshito Higa, Shigenori Kinjo. 993-996 [doi]
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• Time-frequency representations with multiform, tiltable Chebyshev kernelsAntonio H. Costa, Gloria Faye Boudreaux-Bartels. 1001-1004 [doi]
• Quadralinear time-frequency representationsJeffrey C. O'Neill, William J. Williams. 1005-1008 [doi]
• Fast approximations to positive time-frequency distributions, with applicationsPatrick J. Loughlin, James W. Pitton, Blake Hannaford. 1009-1012 [doi]
• Instantaneous frequency and time-frequency distributionsBerkant Tacer, Patrick J. Loughlin. 1013-1016 [doi]
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• Marginals vs. covariance in joint distribution theoryRichard G. Baraniuk. 1021-1024 [doi]
• Displacement-covariant time-frequency energy distributionsFranz Hlawatsch, Helmut Bölcskei. 1025-1028 [doi]
• Discrete-time, discrete-frequency time-frequency representationsMichael S. Richman, Thomas W. Parks, Ramachandra G. Shenoy. 1029-1032 [doi]
• Blind quadratic and time-frequency based detectors from training dataDouglas L. Jones, Akbar M. Sayeed. 1033-1036 [doi]
• A linearly constrained blind equalization scheme based on Bussgang type algorithmsSantiago Zazo, José Manuel Páez-Borrallo, Iván A. Pérez-Álvarez. 1037-1040 [doi]
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• On the choice of wavelet filters for audio compressionPierrick Philippe, François Moreau de Saint-Martin, Laurent Mainard. 1045-1048 [doi]
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• Blind linear recursive equalizer with decorrelation algorithmT. S. Castelein, Yeheskel Bar-Ness, Ramjee Prasad. 1069-1072 [doi]
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• New sampling theorems for MRA subspacesIgor Djokovic, Palghat P. Vaidyanathan. 1085-1088 [doi]
• Gabor-type matrices and discrete huge Gabor transformsSigang Qiu, Hans G. Feichtinger. 1089-1092 [doi]
• Wavelet interpolation: from orthonormal to the oversampled wavelet transformNurgun Erdol, Feng Bao, Zajing Chen. 1093-1096 [doi]
• Nearly shiftable scaling functionsSteven A. Benno, José M. F. Moura. 1097-1100 [doi]
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• Wavelet analysis in soliton detectionMark Allan Coffey, Delores M. Etter. 1109-1112 [doi]
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• An efficient block Newton-type algorithmKostas Berberidis, Sergios Theodoridis. 1133-1136 [doi]
• On programs for prime length FFTs and circular convolutionC. Sidney Burrus, Ivan W. Selesnick. 1137-1140 [doi]
• Structured total least norm method for Toeplitz problemsHaesun Park, J. Ben Rosen, John Glick. 1141-1144 [doi]
• Fast algorithms for matrix multiplication using pseudo-number-theoretic transformsAndrew E. Yagle. 1145-1148 [doi]
• New split algorithms for linear least squares prediction filters with linear phaseWen-Hsien Fang, Yang-Lung Hwang. 1149-1152 [doi]
• Subspace estimation using unitary Schur-type methodsJürgen Götze, Martin Haardt, Josef A. Nossek. 1153-1156 [doi]
• Fast algorithms for systems of equations in wavelet-based solution of integral equations with Toeplitz kernelsAmy E. Bell, Rajashri Joshi. 1157-1160 [doi]
• Fast short-time orthogonal wavelet packet transform algorithmsBenito Carnero, Andrzej Drygajlo. 1161-1164 [doi]
• Fast continuous wavelet transformMichael J. Vrhel, Chulhee Lee, Michael Unser. 1165-1168 [doi]
• n discrete cosine transform algorithm using recursive filter structureWan-Chi Siu, Yuk-Hee Chan, Lap-Pui Chau. 1169-1172 [doi]
• Digital signal processing aids cholesterol plaque detectionMichael I. Savic, Mahesh Chugani, Keith Peabody, Atiya Husain, Ming-Hong Tang, Zlatko Macek. 1173-1176 [doi]
• Prony residual analysis for the identification of cardiac arrhythmiasSzi-Wen Chen, Peter M. Clarkson. 1177-1180 [doi]
• A robust sequential detection algorithm for cardiac arrhythmia classificationPeter M. Clarkson, Szi-Wen Chen, Qi Fan. 1181-1184 [doi]
• Prony analysis based parameter estimation of an NMR signal of blood plasma for cancer detectionP. K. Gale, John W. Pierre. 1185-1188 [doi]
• Combined spatial and temporal equalization using an adaptive antenna array and a decision feedback equalization schemeErik Lindskog, Anders Ahlén, Mikael Sternad. 1189-1192 [doi]
• Geometric interpretation of multiaccess joint detection and the alternating projection algorithmRachel E. Learned, Stéphane Mallat, Bernhard Claus, Alan S. Willsky. 1193-1196 [doi]
• Error sensitivity analysis of a bandwidth expansion technique in spectral analysisWilliam S. McCormick. 1197-1200 [doi]
• The combination of finite-length geometric equalization and bandwidth optimization for multicarrier transceiversNaofal Al-Dhahir, John M. Cioffi. 1201-1204 [doi]
• Joint blind signal detection and carrier recovery over fading channelsLang Tong. 1205-1208 [doi]
• A smart time-frequency exciser for spread spectrum communicationsMehmet V. Tazebay, Ali N. Akansu. 1209-1212 [doi]
• Blind identification of FIR channels carrying multiple finite alphabet signalsAlle-Jan van der Veen, Shilpa Talwar, Arogyaswami Paulraj. 1213-1216 [doi]
• Lossless data compression using adaptive filtersNeeraj Magotra, Wes McCoy, Frank Livingston, Samuel D. Stearns. 1217-1220 [doi]
• Optimum: minimax estimation of quadratic functionals for quadratically constrained signal classesNihal I. Wijeyesekera, Ramachandra G. Shenoy. 1221-1224 [doi]
• Interpolation of lowpass signals at half the Nyquist rateFarrokh Marvasti. 1225-1228 [doi]
• RLS design of polyphase components for the interpolation of periodically nonuniformly sampled signalsLuc Vandendorpe, Benoît Maison, Laurent Cuvelier. 1229-1232 [doi]
• A multiband exponential rate operator for musical transient analysisRamamurthy Mani, S. Hamid Nawab. 1233-1236 [doi]
• Nonlinear recovery of sparse signals from narrowband dataRamesh A. Gopinath. 1237-1239 [doi]
• Successive projections-like algorithms for signal approximation/zero-error modellingStéphane Chretien, Ioannis Dologlou. 1240-1243 [doi]
• Regularized extrapolation of noisy data with a wavelet signal modelLi-Chien Lin, C. C. Jay Kuo. 1244-1247 [doi]
• Enforcing a minimum-phase condition on an arbitrary one-dimensional signal with application to a two-dimensional phase retrieval problemWooshik Kim. 1248-1251 [doi]
• A closed-form least squares solution to the discrete frequency domain design problem of two-dimensional FIR filtersMagdy T. Hanna. 1252-1255 [doi]
• Weighted least-squares design of linear-phase and arbitrary 2-D complex FIR filtersAmin G. Jaffer, William E. Jones, Theagenis J. Abatzoglou. 1256-1259 [doi]
• Constrained least square design of FIR filters without specified transition bandsIvan W. Selesnick, Markus Lang, C. Sidney Burrus. 1260-1263 [doi]
• Allpass filter design and applicationsMarkus Lang. 1264-1267 [doi]
• Efficient genetic algorithm design for power-of-two FIR filtersPaolo Gentili, Francesco Piazza, Aurelio Uncini. 1268-1271 [doi]
• A new WLS method for the design of two-dimensional FIR digital filtersChaur-Heh Hsieh, Ying-Luan Han, Chung Ming Kuo, Yue-Dar Jou. 1272-1275 [doi]
• Offset windowing for FIR fractional-sample delayGerald D. Cain, Anush Yardim, P. Henry. 1276-1279 [doi]
• th power design of complex FIR 2-D filters using the complex Newton methodTheagenis J. Abatzoglou, Amin G. Jaffer. 1280-1283 [doi]
• Laguerre digital filter designMohammad Ali Masnadi-Shirazi, M. Ghasemi. 1284-1287 [doi]
• Iterative algorithms for envelope constrained filter designBa-Ngu Vo, Antonio Cantoni, Kok Lay Teo. 1288-1291 [doi]
• Optimization of filter banks using cyclostationary spectral analysisShuichi Ohno, Hideaki Sakai. 1292-1295 [doi]
• Optimal orthogonal boundary filter banksRicardo L. de Queiroz, K. R. Rao. 1296-1299 [doi]
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• Tunable downsampling using fractional delay filters with applications to digital TV transmissionTimo I. Laakso, Vesa Välimäki, Jukka Henriksson. 1304-1307 [doi]
• Design of a class of multirate systemsWilliam M. Campbell, Thomas W. Parks. 1308-1311 [doi]
• A new look at signal-adapted QMF bank designPierre Moulin. 1312-1315 [doi]
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• 2(Z)Zoran Cvetkovic. 1456-1459 [doi]
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• The multiple lag process and its estimationM. Kemal Sönmez, John S. Baras. 1697-1700 [doi]
• Generalized linear prediction based on analytic signalsUnto K. Laine. 1701-1704 [doi]
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• A segmentation criterion for digital image compressionMahmoud R. El-Sakka, Mohamed S. Kamel. 2551-2554 [doi]
• SVEX: a knowledge-based tool for image segmentationDaniel Hernández-Sosa, Jorge Cabrera-Gámez, Antonio Falcón-Martel, Mario Hernández-Tejera. 2555-2558 [doi]
• Graph-theoretical analysis of the fractal transformJaroslaw Domaszewicz, Vinay A. Vaishampayan. 2559-2562 [doi]
• Performance bounds for fractal codingBernd Hürtgen. 2563-2566 [doi]
• A modified fractal transformHui Zhang, Xiqi Gao, Zhenya He. 2567-2570 [doi]
• Optimum parameters for hybrid fractal image codingS. J. Woolley, Donald M. Monro. 2571-2574 [doi]
• Fast image synthesis using an extended self-similar (ESS) modelLance M. Kaplan, C. C. Jay Kuo. 2575-2578 [doi]
• Finite state residual vector quantization using tree-structured competitive neural networkSyed A. Rizvi, Nasser M. Nasrabadi. 2579-2582 [doi]
• Image coding using FSVQ with dynamic states according to the edges in the DCT domainYoung-Tae Kim, Hyung Hwa Ko. 2583-2586 [doi]
• Permutative vector quantization-application to image compressionJ. Skowronski, Ioannis Dologlou. 2587-2590 [doi]
• Codebook adaptation algorithm for a scene adaptive video coderDongmei Wang, John Hartung. 2591-2594 [doi]
• The fractal pyramid with applications to image codingAlexandru Bogdan. 2595-2598 [doi]
• Point spread function estimation using the mean field approximationDilip G. Warrier, Uday B. Desai. 2599-2602 [doi]
• Design of zero-phase, multiform, tiltable two-dimensional filtersGloria Faye Boudreaux-Bartels, Antonio H. Costa. 2603-2606 [doi]
• A regularization approach to blind restoration of images degraded by shift-variant blursYu-Li You, Mostafa Kaveh. 2607-2610 [doi]
• ∞ deconvolution filter design and its application in image restorationXianggang Yu, Chin-Shung Hsu, Roberto H. Bamberger, Stanley J. Reeves. 2611-2614 [doi]
• Reconstruction of vector field in fluid measurementY. Zhang, Thomas S. Huang, Ronald J. Adrian. 2615-2618 [doi]
• Real-time on-line unconstrained handwriting recognition using statistical methodsKrishna S. Nathan, Homayoon S. M. Beigi, Jayashree Subrahmonia, Gregory J. Clary, Hiroshi Maruyama. 2619-2622 [doi]
• Multilevel HMM for handwritten word recognitionMou-Yen Chen, Amlan Kundu. 2623-2626 [doi]
• Modeling human facial expressions at multiple resolutionsAntai Peng, Monson H. Hayes III. 2627-2629 [doi]
• A discrete parameter HMM approach to on-line handwriting recognitionEveline J. Bellegarda, Jerome R. Bellegarda, David Nahamoo, Krishna S. Nathan. 2631-2634 [doi]
• Image quality criterion based on the cancellation of the masked noiseSerge Comes, Olivier Bruyndonckx, Benoit M. Macq. 2635-2638 [doi]
• Power minimization in DSP application specific systems using algorithm selectionMiodrag Potkonjak, Jan M. Rabaey. 2639-2642 [doi]
• Hierarchical static scheduling of dataflow graphs onto multiple processorsJosé Luis Pino, Edward A. Lee. 2643-2646 [doi]
• Fine grain code synthesis within a block diagram oriented code generation environmentMarkus Willems, Matthias Pankert, Sebastian Ritz. 2647-2650 [doi]
• Scheduling for optimum data memory compaction in block diagram oriented software synthesisSebastian Ritz, Markus Willems, Heinrich Meyr. 2651-2654 [doi]
• Design of an MPEG-based set-top box for video on demand servicesYin-Hwa Huang, Yuo-Chuan Chang, Chi-Shyan Wu, Chien-Hsing Wu. 2655-2658 [doi]
• Efficient throughput optimization of feedback linear computations using generalized Horner's schemeJan M. Rabaey, Miodrag Potkonjak, Kazutoshi Wakabayashi. 2659-2662 [doi]
• Rapid prototyping for simulation debugging environment: an enhanced developing method for embedded computer softwareZhongmin Yu, Yoshinao Aoki. 2663-2666 [doi]
• ASIC implementation of vector quantization for speaker recognitionM. Li, J. T. Proudfoot, Graham A. King. 2667-2670 [doi]
• A computational model of sound stream segregation with multi-agent paradigmTomohiro Nakatani, Takeshi Kawabata, Hiroshi G. Okuno. 2671-2674 [doi]
• Perturbation effects on filters having localised polesAnthony G. Place, Gregory H. Allen. 2675-2678 [doi]
• ATCURE: a heterogeneous processor for image recognitionJohn G. Ackenhusen, Quentin A. Holmes, Paul A. Kortesoja, David L. McCubbrey, Paul L. Mohan, Jeremy A. Salinger, Thomas N. Wessling, Lester J. Witter, Herbert Stopper. 2679-2682 [doi]
• Sensor signal processing for IVHS applicationsWai Ying Kan, James V. Krogmeier, Peter C. Doerschuk. 2683-2686 [doi]
• A real-time speech recognition architecture for a multi-channel interactive voice response systemAgus Trihandoyo, Adam Belloum, Kun Mean Hou. 2687-2690 [doi]
• Real-time RLS-MLSE equalizer implementation for application to PDC (Personal Digital Cellular)Masashi Naitoh, Mitsuo Kubo, Keno Urabe. 2691-2694 [doi]
• Introduction to ARPA's RASSP initiative and education/facilitation programJack H. Corley, Vijay K. Madisetti, Mark A. Richards. 2695-2698 [doi]
• VHDL modeling for signal processor developmentCory S. Myers, Ray Dreiling. 2699-2702 [doi]
• Architectures for rapid prototyping of embedded signal processorsGerry Caracciolo, Jeff Pridmore. 2703-2706 [doi]
• Assessing and improving current practice in the design of application-specific signal processorsGary A. Shaw, James C. Anderson, Vijay K. Madisetti. 2707-2710 [doi]
• Virtual prototyping of embedded DSP systemsVijay K. Madisetti, Thomas Egolf, Shahram Famorzadeh, Lan-Rong Dung. 2711-2714 [doi]
• A C++ software environment for the development of embedded signal processing systemsJoseph M. Winograd, S. Hamid Nawab. 2715-2718 [doi]
• A real time software-only H.261 codecKatherine S. Wang, James O. Normile, Hsi-Jung Wu, Dulce B. Ponceleon, Ken Chu, Kah Kay Sung. 2719-2722 [doi]
• Real-time implementation of an IIR acoustic echo canceller on ADSP21020Mansour Tahernezhadi, L. Liu. 2723-2726 [doi]
• Prediction of sound pressure fields by Picard-iterative BEM based on holographic interferometryH. Klingele, H. Steinbichler. 2727-2730 [doi]
• Power system disturbance monitoring using spectral analysis techniquesPeter O'Shea, Edward W. Palmer, Gordon Frazer. 2731-2734 [doi]
• An adaptive discrete cosine transformed videophone communicator for mobile applicationsJürgen Streit, Lajos Hanzo. 2735-2738 [doi]
• Comparison of microphone array designs for hearing aidDaniel Korompis, Arthur Wang 0002, Kung Yao. 2739-2742 [doi]
• A new Viterbi decoder design for code rate k/nHsiang-Ling Li, Chaitali Chakrabarti. 2743-2746 [doi]
• m) multiplier and squarer architecturesS. K. Jain, Keshab K. Parhi. 2747-2750 [doi]
• Unsupervised pattern recognition for digital waveform classification from radiation detectorsJianwei Miao, Mark A. Clements. 2751-2754 [doi]
• Comparison of 2-D FFT implementations on the Intel Paragon massively parallel supercomputerM. An, N. Anupindi, Michail Bletsas, George Kechriotis, C. Lu, Elias S. Manolakos, Richard Tolimieri. 2755-2758 [doi]
• Iterative least squares approach to the mixture modeling problemMurari Srinivasan, R. DeFries. 2759-2762 [doi]
• Cramer-Rao bounds for matched field tomography and ocean acoustic tomographyArthur B. Baggeroer, Henrik Schmidt. 2763-2766 [doi]
• Signals and signal processing for acoustic monitoring of ocean processesTheodore G. Birdsall. 2767-2770 [doi]
• A Cramer-Rao bound on acoustic measurement of ocean climate change in the presence of mesoscale sound-speed variabilityJeffrey L. Krolik, Sunil Narasimhan. 2771-2774 [doi]
• Environmental limits to source localizationJosé M. F. Moura, Maria-João Rendas, Georges Bienvenu. 2775-2778 [doi]
• Ocean acoustic tomography experiment using a vertical arrayClaire Noël, Christophe Viala, François Evennou, Yann Stéphan, François-Regis Martin-Lauzer. 2779 [doi]
• Multiresolution statistical analysis and assimilation of large ocean data setsPaul W. Fieguth, William Clement Karl, Alan S. Willsky. 2781-2784 [doi]
• Kalman filtering of large-scale geophysical flows by approximations based on Markov random field and waveletToshio Mike Chin, Arthur J. Mariano. 2785-2788 [doi]
• Assimilation of satellite data in beta-plane ocean circulation modelsAmir Asif, José M. F. Moura. 2789-2792 [doi]
• High frequency acoustic imaging in the oceanJules S. Jaffe, Girish Chandran, Edward Reuss. 2793-2796 [doi]
• Improved analysis of remote sensed data using combined physical and engineering approachesJames J. Simpson. 2797-2800 [doi]
• Interferometric synthetic aperture radar for terrain mappingJohn C. Curlander. 2801-2804 [doi]
• Robust material identification in hyperspectral data via multiresolution wavelet techniquesJohn D. Gorman, Nikola S. Subotic, Brian J. Thelen. 2805-2808 [doi]
• Space-time adaptive processing for airborne radarJames Ward. 2809-2812 [doi]
• Ground penetrating radar tomographyJohn E. Molyneux. 2813-2816 [doi]
• Seismic methods for imaging physical properties of the EarthGary Mavko, Nathalie Lucet, Tapan Mukerji. 2817-2820 [doi]
• 2David J. Thomson. 2821-2824 [doi]
• An approach for random media parameter estimation using seismic reflection dataMatthias G. Imhof, M. Nafi Toksoz. 2825-2828 [doi]
• RASSP: methods and tools for rapid signal processor development, upgrading and life cycle supportMark A. Richards, Randolph Reitmeyer Jr., A. Bard, Gerald T. Michael. 2829-2832 [doi]
• Managing complexity in heterogeneous system specification, simulation, and synthesisAsawaree Kalavade, José Luis Pino, Edward A. Lee. 2833-2836 [doi]
• Design guidance in the power dimensionJan M. Rabaey, Lisa M. Guerra, Renu Mehra. 2837-2840 [doi]
• MAT2DSP-A tool for evaluating implementation complexity of signal processing algorithmsBenjamin Friedlander. 2841-2844 [doi]
• Syst