Abstract is missing.
- Feature parameter curve method for high performance NN-based speech recognitionSan Zhu, Dao Wen Chen, Taiyi Huang. 1-4 [doi]
- On the use of residual cepstrum in speech recognitionJialong He, Li Liu, Günther Palm. 5-8 [doi]
- HMM-based speech recognition using state-dependent, linear transforms on Mel-warped DFT featuresChengalv Rathinavelu, Li Deng. 9-12 [doi]
- Mixed Malvar-wavelets for non-stationary signal representationJ. A. Thripuraneni, Wei Lou, Victor E. DeBrunner. 13-16 [doi]
- Experiments on a parametric nonlinear spectral warping for an HMM-based speech recognizerDaniel J. Mashao. 17-20 [doi]
- Robust distant-talking speech recognitionJohn C. Pearson, Qiguang Lin, Chiwei Che, Dong-Suk Yuk, Limin Jin, Bert de Vries, James L. Flanagan. 21-24 [doi]
- Time-frequency representation based cepstral processing for speech recognitionAdam B. Fineberg, Kevin C. Yu. 25-28 [doi]
- Knowledge-based parameters for HMM speech recognitionNabil N. Bitar, Carol Y. Espy-Wilson. 29-32 [doi]
- A phoneme-similarity based ASR front-endTed H. Applebaum, Philippe Morin, Brian A. Hanson. 33-36 [doi]
- A model of dynamic auditory perception and its application to robust speech recognitionBrian Strope, Abeer Alwan. 37-40 [doi]
- Independent calculation of power parameters on PMC methodHiroki Yamamoto, Masayuki Yamada, Tetsuo Kosaka, Yasuhiro Komori, Yasunori Ohora. 41-44 [doi]
- Noisy speech recognition using variance adapted likelihood measureJen-Tzung Chien, Lee-Min Lee, Hsiao-Chuan Wang. 45-48 [doi]
- An improved noise compensation algorithm for speech recognition in noiseRuikang Yang, Petri Haavisto. 49-52 [doi]
- Improved speech recognition via speaker stress directed classificationBrian D. Womak, John H. L. Hansen. 53-56 [doi]
- High-accuracy connected digit recognition for mobile applicationsSunil K. Gupta, Frank K. Soong, Raziel Haimi-Cohen. 57-60 [doi]
- Feature extraction based on zero-crossings with peak amplitudes for robust speech recognition in noisy environmentsDoh-Suk Kim, Jae-Hoon Jeong, Jae-Weon Kim, Soo-Young Lee. 61-64 [doi]
- Improving environmental robustness in large vocabulary speech recognitionPhilip C. Woodland, Mark John Francis Gales, David Pye. 65-68 [doi]
- Noise and room acoustics distorted speech recognition by HMM compositionSatoshi Nakamura, Tetsuya Takiguchi, Kiyohiro Shikano. 69-72 [doi]
- Developments in continuous speech dictation using the 1995 ARPA NAB news taskJean-Luc Gauvain, Lori Lamel, Gilles Adda, Driss Matrouf. 73-76 [doi]
- Evaluation of root-normalised front-end (RN LFCC) for speech recognition in wireless GSM network environmentsSandra Dufour, Catherine Glorion, Philip Lockwood. 77-80 [doi]
- Speaker background models for connected digit password speaker verificationAaron E. Rosenberg, Sarangarajan Parthasarathy. 81-84 [doi]
- Cohort selection and word grammar effects for speaker recognitionJohn M. Colombi, Dennis W. Ruck, Timothy R. Anderson 0001, Steven K. Rogers, Mark E. Oxley. 85-88 [doi]
- Discriminative training of GMM for speaker identificationCesar Martín del Alamo, Francisco Javier Caminero Gil, Celinda de la Torre-Munilla, Luis A. Hernández Gómez. 89-92 [doi]
- Subword-based text-dependent speaker verification system with user-selectable passwordsManish Sharma, Richard J. Mammone. 93-96 [doi]
- Robust methods of updating model and a priori threshold in speaker verificationTomoko Matsui, Takashi Nishitani, Sadaoki Firui. 97-100 [doi]
- A further investigation on AR-vector models for text-independent speaker identificationIvan Magrin-Chagnolleau, Joachim Wilke, Frédéric Bimbot. 101-104 [doi]
- Speaker identification via support vector classifiersMichael Schmidt, Herbert Gish. 105-108 [doi]
- Speaker verification using mixture likelihood profiles extracted from speaker independent hidden Markov modelsAnand R. Setlur, Rafid A. Sukkar, Malan B. Gandhi. 109-112 [doi]
- The effects of handset variability on speaker recognition performance: experiments on the Switchboard corpusDouglas A. Reynolds. 113-116 [doi]
- Speaker recognition in reverberant enclosuresPierre J. Castellano, Sridha Sridharan, David R. Cole. 117-120 [doi]
- Using a transcription graph for large vocabulary continuous speech recognitionZili Li 0001, Douglas D. O'Shaughnessy. 121-124 [doi]
- Fast and accurate recognition of very-large-vocabulary continuous Mandarin speech for Chinese language with improved segmental probability modelingJia-lin Shen, Lin-Shan Lee. 125-128 [doi]
- Decoding optimal state sequence with smooth state likelihoodsIlija Zeljkovic. 129-132 [doi]
- Improvements on the pronunciation prefix tree search organizationFil Alleva, Xuedong Huang, Mei-Yuh Hwang. 133-136 [doi]
- Minimizing search errors due to delayed bigrams in real-time speech recognition systemsMonika Woszczyna, Michael Finke. 137-140 [doi]
- Real-time recognition of broadcast radio speechGary D. Cook, James Christie, Philip Clarkson, Michael M. Hochberg, Beth T. Logan, Anthony J. Robinson, Carl W. Seymour. 141-144 [doi]
- Spontaneous dialogue speech recognition using cross-word context constrained word graphsTohru Shimizu, Hirofumi Yamamoto, Hirokazu Masataki, Shoichi Matsunaga, Yoshinori Sagisaka. 145-148 [doi]
- Efficient evaluation of the LVCSR search space using the NOWAY decoderSteve Renals, Mike Hochberg. 149-152 [doi]
- Developments in large vocabulary, continuous speech recognition of GermanMartine Adda-Decker, Gilles Adda, Lori Lamel, Jean-Luc Gauvain. 153-156 [doi]
- Speech recognition on Mandarin Call Home: a large-vocabulary, conversational, and telephone speech corpusFu-Hua Liu, Michael Picheny, Patibandla Srinivasa, Michael D. Monkowski, Julian Chen. 157-160 [doi]
- Multilingual stochastic n-gram class language modelsMichèle Jardino. 161-163 [doi]
- A variable-length category-based n-gram language modelThomas Niesler, Philip C. Woodland. 164-167 [doi]
- Improving n-gram models by incorporating enhanced distributionsPeter O'Boyle, Ji Ming, John G. McMahon, F. Jack Smith. 168-171 [doi]
- A novel word clustering algorithm based on latent semantic analysisJerome R. Bellegarda, John W. Butzberger, Yen-Lu Chow, Noah B. Coccaro, Devang Naik. 172-175 [doi]
- Statistical natural language understanding using hidden clumpingsMark Epstein, Kishore Papineni, Salim Roukos, Todd Ward, Stephen Della Pietra. 176-179 [doi]
- Clustering words for statistical language models based on contextual word similarityAzarshid Farhat, Jean-Francois Isabelle, Douglas D. O'Shaughnessy. 180-183 [doi]
- Domain word translation by space-frequency analysis of context length histogramsPascale Fung. 184-187 [doi]
- Variable-order N-gram generation by word-class splitting and consecutive word groupingHirokazu Masataki, Yoshinori Sgisaka. 188-191 [doi]
- Back-off method for n-gram smoothing based on binomial posteriori distributionTakeshi Kawabata, Masafumi Tamoto. 192-195 [doi]
- Ergodic multigram HMM integrating word segmentation and class tagging for Chinese language modelingHubert Hin-Cheung Law, Chorkin Chan. 196-199 [doi]
- A 2.4 kbit/s MELP coder candidate for the new U.S. Federal StandardAlan McCree, Kwan K. Truong, E. Bryan George, Thomas P. Barnwell III, Vishu R. Viswanathan. 200-203 [doi]
- Harmonic-stochastic excitation (HSX) speech coding below 4 kbit/sClaude Laflamme, Redwan Salami, Ridha Matmti, Jean-Pierre Adoul. 204-207 [doi]
- A high quality MBE-LPC-FE speech coder at 2.4 kbps and 1.2 kbpsTian Wang, Kun Tang, Chongxi Feng. 208-211 [doi]
- A low-complexity waveform interpolation coderW. Bastiaan Kleijn, Yair Shoham, Deep Sen, Roar Hagen. 212-215 [doi]
- Mixed-domain coding of speech at 3 kb/sJuan Carlos De Martin, Allen Gersho. 216-219 [doi]
- Source driven variable bit rate prototype interpolation codingCostas Xydeas, Binshi Cao. 220-223 [doi]
- A new approach to very low-rate speech coding using temporal decompositionShahrokh Ghaemmaghami, Mohamed Deriche. 224-227 [doi]
- A variable frame pitch estimator and test resultsXiaoshu Qian, Ramdas Kumaresan. 228-231 [doi]
- Robust method of measurement of fundamental frequency by ACLOS: autocorrelation of log spectrumNobuyuki Kunieda, Tetsuya Shimamura, Jouji Suzuki. 232-235 [doi]
- Lag-indexed VQ for pitch filter codingStan McClellan, Jerry D. Gibson. 236-239 [doi]
- Embedded algebraic vector quantizers (EAVQ) with application to wideband speech codingMinjie Xie, Jean-Pierre Adoul. 240-243 [doi]
- The two-dimensional discrete cosine transform applied to speech dataLadan Baghai-Ravary, Steve W. Beet, M. O. Tokhi. 244-247 [doi]
- Real-time high accurate cell loss recovery technique for speech over ATM networksKiyoshi Matsumoto. 248-250 [doi]
- Predictive fractal interpolation mapping: differential speech coding at low bit ratesZhicheng Wang. 251-254 [doi]
- 16 kbit/s wideband speech coding based on unequal subbandsJürgen W. Paulus, Jürgen Schnitzler. 255-258 [doi]
- Low delay IIR QMF banks with high perceptive quality for speech processingThomas Kleinmann, Arild Lacroix. 259-262 [doi]
- Demodulators for AM-FM models of speech signals: a comparisonShan Lu, Peter C. Doerschuk. 263-266 [doi]
- Synthesis and coding of continuous speech with the nonlinear oscillator modelGernot Kubin. 267-270 [doi]
- Variable frame rate parameter encoding via adaptive frame selection using dynamic programmingE. Bryan George, Alan McCree, Vishu R. Viswanathan. 271-274 [doi]
- Transform predictive coding of wideband speech signalsJuin-Hwey Chen, Dongmei Wang. 275-278 [doi]
- A system for unrestricted topic retrieval from radio news broadcastsDavid A. James. 279-282 [doi]
- Automated generation of N-best pronunciations of proper nounsNeeraj Deshmukh, Mary Weber, Joe Picone. 283-286 [doi]
- An efficient voice retrieval system for very-large-vocabulary Chinese textual databases with a clustered language modelSung-Chien Lin, Lee-Feng Chien, Keh-Jiann Chen, Lin-Shan Lee. 287-290 [doi]
- Concept-based phrase spotting approach for spontaneous speech understandingTatsuya Kawahara, Norihide Kitaoka, Shuji Doshita. 291-298 [doi]
- A DICTIONARY BASED METHOD FOR DETERMINING TOPICS IN TEXT AND TRANSCRIBED SPEECHPatrick Schone, Douglas Nelson. 295 [doi]
- Keyword spotting for video soundtrack indexingPhilippe Gelin, Christian Wellekens. 299-302 [doi]
- Improvements in switchboard recognition and topic identificationBarbara Peskin, Sean Connolly, Lawrence Gillick, Steve Lowe, Don McAllaster, Venkatesh Nagesha. 303-306 [doi]
- Statistical models for topic identification using phoneme substringsJerry H. Wright, Michael J. Carey, Eluned S. Parris. 307-310 [doi]
- Robust talker-independent audio document retrievalGareth J. F. Jones, J. T. Foote, Karen Spark Jones, Steve J. Young. 311-314 [doi]
- Unsupervised topic clustering of switchboard speech messagesBeth A. Carlson. 315-318 [doi]
- Speaker recognition and speaker normalization by projection to speaker subspaceYasuo Ariki, Shigeaki Tagashira, Masayuki Nishijima. 319-322 [doi]
- Compensated mel frequency cepstrum coefficientsRivarol Vergin, Douglas D. O'Shaughnessy, Vishwa Gupta. 323-326 [doi]
- Adaptation method based on HMM composition and EM algorithmYasuhiro Minami, Sadaoki Furui. 327-330 [doi]
- SNR-normalisation for robust speech recognitionTom Claes, Dirk Van Compernolle. 331-334 [doi]
- Towards robustness to fast speech in ASRNikki Mirghafori, Eric Fosler-Lussier, Nelson Morgan. 335-338 [doi]
- Speaker normalization on conversational telephone speechSteven Wegmann, Don McAllaster, Jeremy Orloff, Barbara Peskin. 339-341 [doi]
- Speaker and gender normalization for continuous-density hidden Markov modelsAlex Acero, Xuedong Huang. 342-345 [doi]
- A parametric approach to vocal tract length normalizationEllen Eide, Herbert Gish. 346-348 [doi]
- A study of speech recognition for children and the elderlyJay G. Wilpon, Claus Jacobsen. 349-352 [doi]
- Speaker normalization using efficient frequency warping proceduresLi Lee, Richard C. Rose. 353-356 [doi]
- A fast stochastic parser for determining phrase boundaries for text-to-speech synthesisRichard A. Sharman, Jerry H. Wright. 357-360 [doi]
- Speech concatenation and synthesis using an overlap-add sinusoidal modelMichael W. Macon, Mark A. Clements. 361-364 [doi]
- Voice conversion using partitions of spectral feature spaceWerner Verhelst, Johan Mertens. 365-368 [doi]
- Determination of vocal-tract shapes from formant frequencies based on perturbation theory and interpolation methodZhenli Yu, P. C. Ching. 369-372 [doi]
- Unit selection in a concatenative speech synthesis system using a large speech databaseAndrew J. Hunt, Alan W. Black. 373-376 [doi]
- Parametric hybrid source models for voiced and voiceless fricative consonantsShrikanth S. Narayanan, Abeer Alwan. 377-380 [doi]
- High-quality speech synthesis using context-dependent syllabic unitsTakashi Saito, Yasuhide Hashimoto, Masaharu Sakamoto. 381-384 [doi]
- Articulatory copy synthesis using a nine-parameter vocal tract modelColin C. Goodyear, Dongbing Wei. 385-388 [doi]
- Speech synthesis using HMMs with dynamic featuresTakashi Masuko, Keiichi Tokuda, Takao Kobayashi, Satoshi Imai. 389-392 [doi]
- Interpolating V/UV mixture functions of a harmonic model for concatenative speech synthesisKing-fai Lam, Cheung-fat Chan. 393-396 [doi]
- An efficient top-down parsing algorithm for understanding speech by using stochastic syntactic and semantic modelsHolger Stahl, Johannes Müller, Manfred K. Lang. 397-400 [doi]
- Data-driven discourse modeling for semantic interpretationFrancisco Javier Caminero Gil, Jorge Alvarez-Cercadillo, Carlos Crespo-Casas, Daniel Tapias Merino. 401-404 [doi]
- Statistical language modeling for speech disfluenciesAndreas Stolcke, Elizabeth Shriberg. 405-408 [doi]
- JANUS-II-translation of spontaneous conversational speechAlex Waibel, Michael Finke, Donna Gates, Marsal Gavaldà, Thomas Kemp, Alon Lavie, Lori S. Levin, Martin Maier, Laura Mayfield, Arthur E. McNair, Ivica Rogina, Kaori Shima, Tilo Sloboda, Monika Woszczyna, Torsten Zeppenfeld, Puming Zhan. 409-412 [doi]
- Language model acquisition from a text corpus for speech understandingTatsuo Matsuoka, Robert Hasson, Michael Barlow, Sadaoki Furui. 413-15 [doi]
- A class based language model for speech recognitionWayne Ward, Sunil Issar. 416-418 [doi]
- An integrated model of acoustics and language using semantic classification treesElmar Nöth, Renato de Mori, Julia Fischer, Arnd Gebhard, Stefan Harbeck, Ralf Kompe, R. Kuhn, Heinrich Niemann, Marion Mast. 419-422 [doi]
- Combining stochastic and linguistic language models for recognition of spontaneous speechWieland Eckert, Florian Gallwitz, Heinrich Niemann. 423-426 [doi]
- Error correction via a post-processor for continuous speech recognitionEric K. Ringger, James F. Allen. 427-430 [doi]
- Integration of concept-driven semantic interpretation with speech recognitionAkito Nagai, Yasushi Ishikawa, Kunio Nakajima. 431-434 [doi]
- A second-order HMM for high performance word and phoneme-based continuous speech recognitionJean-Francois Mari, Dominique Fohr, Jean-Claude Junqua. 435-438 [doi]
- Evaluation of segmental unit input HMMSeiichi Nakagawa, Kazumasa Yamamoto. 439-442 [doi]
- Design of a speech recognition system based on acoustically derived segmental unitsMichiel Bacchiani, Mari Ostendorf, Yoshinori Sagisaka, Kuldip K. Paliwal. 443-446 [doi]
- Modeling speech variability with segmental HMMsWendy J. Holmes, Martin J. Russell. 447-450 [doi]
- Context-dependent units for vocabulary-independent Spanish speech recognitionLuis Villarrubia, Luis A. Hernández Gómez, Jose Maria Elvira, Juan Carlos Torrecilla. 451-454 [doi]
- Context-dependent acoustic models for Chinese speech recognitionBin Ma, Taiyi Huang, Bo Xu, Xijun Zhang, Fei Qu. 455-458 [doi]
- Automatic recognition of Danish natural numbers for telephone applicationsClaus Jacobsen, Jay G. Wilpon. 459-462 [doi]
- Explicit modeling of coarticulation in a statistical speech recognizerRuxin Chen, Leah H. Jamieson. 463-466 [doi]
- Tied-structure HMM based on parameter correlation for efficient model trainingSatoshi Takahashi, Shigeki Sagayama. 467-470 [doi]
- A semi-continuous stochastic trajectory model for phoneme-based continuous speech recognitionOlivier Siohan, Yifan Gong. 471-474 [doi]
- Automatic evaluation of speaker recognizability of coded speechKhaled T. Assaleh. 475-478 [doi]
- A perceptually-based objective measure for speech coders using abductive networkMohamed M. Meky, Tarek N. Saadawi. 479-482 [doi]
- Objectively measured descriptors applied to speaker characterizationBurhan Necioglu, Mark A. Clements, Thomas P. Barnwell III. 483-486 [doi]
- Objective speech quality measure for cellular phoneKai Hung Lam, Oscar C. Au, C. C. Chan, K. F. Hui, S. F. Lau. 487-490 [doi]
- Vector quantization techniques for output-based objective speech qualityChiyi Jin, Robert F. Kubichek. 491-494 [doi]
- Objective measures for speech quality assessment in wireless communicationsAruna Bayya, Marvin Vis. 495-498 [doi]
- Performance assessment of 4.8 kbit/s AMBE coding under aeronautical environmental conditionsSimão F. Campos Neto, Franklin L. Corcoran, J. G. Phipps Jr., Spiros Dimolitsas. 499-502 [doi]
- Robust recognition of cellular telephone speech by adaptive vector quantizationM. Kemal Sönmez, Raja Rajasekaran, John S. Baras. 503-506 [doi]
- Efficient decoding and training procedures for utterance verification in continuous speech recognitionEduardo Lleida, Richard C. Rose. 507-510 [doi]
- Confidence measures for the SWITCHBOARD databaseStephen J. Cox, Richard C. Rose. 511-514 [doi]
- A phone-dependent confidence measure for utterance rejectionZeév Rivlin, Michael Cohen, Victor Abrash, Thomas Chung. 515-517 [doi]
- Utterance verification of keyword strings using word-based minimum verification error (WB-MVE) trainingRafid A. Sukkar, Anand R. Setlur, Mazin G. Rahim, Chin-Hui Lee. 518-521 [doi]
- Fast implementation methods for Viterbi-based word-spottingKate Knill, Steve J. Young. 522-525 [doi]
- Improving wordspotting performance with artificially generated dataEric I. Chang, Richard P. Lippmann. 526-529 [doi]
- Modelling unknown words in spontaneous speechThomas Kemp, Andreas Jusek. 530-533 [doi]
- Improved modeling of OOV words in spontaneous speechPablo Fetter, Alfred Kaltenmeier, Thomas Kuhn, Peter Regel-Brietzmann. 534-537 [doi]
- Two-pass strategy for continuous speech recognition with detection and transcription of unknown wordsShoichi Matsunaga, Hiroyuki Sakamoto. 538-541 [doi]
- A modified generalised Lloyd algorithm for VQ codebook designChang-Qian Chen, Soo Ngee Koh, Pratab Sivaprakasapillai. 542-545 [doi]
- Robust classification of speech based on the dyadic wavelet transform with application to CELP codingJoachim Stegmann, Gerhard Schröder, Kyrill A. Fischer. 546-549 [doi]
- Optimal wavelet packets for low-delay audio codingPierrick Philippe, François Moreau de de Saint-Martin, Michel Lever, Joel Soumagne. 550-553 [doi]
- A fast VSELP speech coder based on mutually orthonormal regular pulse vectorsYong-Soo Choi, Hong-Goo Kang, Dae Hee Youn. 554-557 [doi]
- Dual-Pulse CS-CELP: a toll-quality low-complexity speech coder at 7.8 kbit/sHitoshi Ohmuro, Jotaro Ikedo, Takehiro Moriya, Akitoshi Kataoka, Shinji Hayashi, Kazunori Mano. 558-561 [doi]
- Low-delay CELP with multi-pulse VQ and fast search for GSM EFRShinichi Taumi, Kazunori Ozawa, Toshiyuki Nomura, Masahiro Serizawa. 562-565 [doi]
- Speech compression with cosine and wavelet packet near-best basesCarl Taswell. 566-568 [doi]
- An enhanced full rate speech coder for digital cellular applicationsWilfrid LeBlanc, Christine Liu, Vishu R. Viswanathan. 569-572 [doi]
- Optimum harmonics tracking filter for auditory scene analysisKazuki Nishi, Shigeru Ando, Shuhei Aida. 573-576 [doi]
- Selective error protection of ITU-T G.729 codec for digital cellular channelsKumar Swaminathan, A. Roger Hammons Jr., Mark Austin. 577-580 [doi]
- Discriminative training of stochastic Markov graphs for speech recognitionFranz Wolfertstetter, Günther Ruske. 581-584 [doi]
- Incremental ML estimation of HMM parameters for efficient trainingYoshihiko Gotoh, Harvey F. Silverman. 585-588 [doi]
- Improved HMM training and scoring strategies with application to accent classificationLevent M. Arslan, John H. L. Hansen. 589-592 [doi]
- Maximum mutual information codebook mapping for discrete hidden Markov modelsVincent Fontaine, Christophe Ris, Henri Leich. 593-596 [doi]
- Minimum classification error training for a small amount of data enhanced by vector-field-smoothed Bayesian learningJun-ichi Takahashi, Shigeki Sagayama. 597-600 [doi]
- Maximum likelihood successive state splittingHarald Singer, Mari Ostendorf. 601-604 [doi]
- Lattice-based discriminative training for large vocabulary speech recognitionValtcho Valtchev, Julian Odell, Philip C. Woodland, Steve J. Young. 605-608 [doi]
- Training algorithms for a predictive classifierP. V. S. Rao, R. Raveendran. 609-612 [doi]
- Discriminative training of Gaussian mixture models for large vocabulary speech recognition systemsLalit R. Bahl, Mukund Padmanabhan, David Nahamoo, P. S. Gopalakrishnan. 613-616 [doi]
- Stochastic observation hidden Markov modelsCarl D. Mitchell, Mary P. Harper, Leah H. Jamieson. 617-620 [doi]
- Recursive speech enhancement using the EM algorithm with initial conditions trained by HMM'sKi Yong Lee, Byung-Gook Lee, Iickho Song, Jisang Yoo. 621-624 [doi]
- Robust noise reduction for speech and audio signalsSimon J. Godsill, Peter J. W. Rayner. 625-628 [doi]
- Speech enhancement based on a priori signal to noise estimationPascal Scalart, Jozue Vieira Filho. 629-632 [doi]
- Noise model adaptation in model based speech enhancementBruce L. McKinley, Gary H. Whipple. 633-636 [doi]
- Analysis of two structures for combined acoustic echo cancellation and noise reductionYann Guelou, Abdelkrim Benamar, Pascal Scalart. 637-640 [doi]
- Selective all-pole modeling of degraded speech using M-band decompositionChang D. Yoo. 641-644 [doi]
- Text-directed speech enhancement using phoneme classification and feature map constrained vector quantizationBryan L. Pellom, John H. L. Hansen. 645-648 [doi]
- Multi-channel signal separationDominic C. B. Chan, Peter J. W. Rayner, Simon J. Godsill. 649-652 [doi]
- Localization by harmonic structure and its application to harmonic sound stream segregationTomohiro Nakatani, Masataka Goto, Hiroshi G. Okuno. 653-656 [doi]
- A kind of fuzzy-neural networks for text-independent speaker identificationYuan Zhing-Xuan, Bo-Ling Xu, Chong-Zhi Yu. 657-660 [doi]
- A high performance text independent speaker recognition system based on vowel spotting and neural netsNikos Fakotakis, John Sirigos. 661-664 [doi]
- Signal modeling for speaker identificationLi Liu, Jialong He, Günther Palm. 665-668 [doi]
- A general framework of feature extraction: application to speaker recognitionChi-shi Liu. 669-672 [doi]
- An HMM approach to text-prompted speaker verificationChiwei Che, Qiguang Lin, Dong-Suk Yuk. 673-676 [doi]
- Text-independent speaker identificationMartha Birnbaum, Kathy L. Brown, Steven Bardenhagen. 677-680 [doi]
- Normalized discriminant analysis with application to a hybrid speaker-verification systemQi Li, Sarangarajan Parthasarathy, Aaron E. Rosenberg, Donald W. Tufts. 681-684 [doi]
- Language independent gender identificationEluned S. Parris, Michael J. Carey 0002. 685-688 [doi]
- Fine structure features for speaker identificationCharles R. Jankowski Jr., Thomas F. Quatieri, Douglas A. Reynolds. 689-692 [doi]
- Speaker verification with low storage requirementsJohan Schalkwyk, Neena Jain, Etienne Barnard. 693-696 [doi]
- A Markov random field approach to Bayesian speaker adaptationBen Shahshahani. 697-700 [doi]
- Speaker clustering and transformation for speaker adaptation in large-vocabulary speech recognition systemsMukund Padmanabhan, Lalit R. Bahl, David Nahamoo, Michael A. Picheny. 701-704 [doi]
- A study of on-line quasi-Bayes adaptation for CDHMM-based speech recognitionQiang Huo, Chin-Hui Lee. 705-708 [doi]
- Speaker time-drifting adaptation using trajectory mixture hidden Markov modelsJian Su, Haizhou Li, Jean-Paul Haton, Kai Tat Ng. 709-712 [doi]
- An experimental study of acoustic adaptation algorithmsAnanth Sankar, Leonardo Neumeyer, Mitchel Weintroub. 713-716 [doi]
- Speaker adaptation with autonomous model complexity control by MDL principleKoichi Shinoda, Takao Watanabe. 717-720 [doi]
- An approach to speaker adaptation based on analytic functionsJohn W. McDonough, George Zavaliagkos, Herbert Gish. 721-724 [doi]
- Maximum a posteriori adaptation for large scale HMM recognizersGeorge Zavaliagkos, Richard M. Schwartz, John W. McDonough. 725-728 [doi]
- Acoustic adaptation using nonlinear transformations of HMM parametersVictor Abrash, Ananth Sankar, Horacio Franco, Michael Cohen. 729-732 [doi]
- A vector Taylor series approach for environment-independent speech recognitionPedro J. Moreno, Bhiksha Raj, Richard M. Stern. 733-736 [doi]
- Two-stage vector quantization-pyramidal lattice vector quantization and application to speech LSP codingJianping Pan. 737-740 [doi]
- Linked split-vector quantizer of LPC parametersMoo-young Kim, Nam Kyu Ha, Sang Ryong Kim. 741-744 [doi]
- Scalar quantization using vector measure with application to quantization of LSF parametersH. C. Ng, Shu Hung Leung. 745-748 [doi]
- A robust LSP encoding scheme for noisy channelJunchen Du, Seung P. Kim. 749-752 [doi]
- An efficient coding of LSP parameters using multiple type frame segmentationYun-Keun Lee, Ki-Chul Kim, Hwang Soo Lee. 753-756 [doi]
- Switched prediction and quantization of LSP frequenciesHouman Zarrinkoub, Paul Mermelstein. 757-760 [doi]
- Classified nonlinear predictive vector quantization of speech spectral parametersJames H. Y. Loo, Wai-Yip Chan, Peter Kabal. 761-764 [doi]
- Exploiting interframe correlation in spectral quantization: a study of different memory VQ schemesThomas Eriksson, Jan Linden, Jan Skoglund. 765-768 [doi]
- Low rate coding of the spectral envelope using channel gainsElliot Singer, Robert J. McAulay, Robert B. Dunn, Thomas F. Quatieri. 769-772 [doi]
- A hybrid input/output spectrum adaptation scheme for LD-CELP coding of speechSiu-pun Chui, Cheung-fat Chan. 773-776 [doi]
- Automatic dialect identification of extemporaneous conversational, Latin American Spanish speechMarc A. Zissman, Terry P. Gleason, Deborah Rekart, Beth L. Losiewicz. 777-780 [doi]
- LVCSR-based language identificationTanja Schultz, Ivica Rogina, Alex Waibel. 781-784 [doi]
- Automatic language identification using large vocabulary continuous speech recognitionSergio Mendoza, Larry Gillick, Yoshiko Ito, Stephen Lowe, Michael Newman. 785-788 [doi]
- Experiments for an approach to language identification with conversational telephone speechYonghong Yan 0002, Etienne Barnard. 789-792 [doi]
- Statistical language identification based on untranscribed trainingMichael A. Lund, Kristine W. Ma, Herbert Gish. 793-796 [doi]
- A model for efficient formant estimationLutz Welling, Hermann Ney. 797-800 [doi]
- Improved vocal tract model for the analysis of nasal speech soundsMinsheng Liu, Arild Lacroix. 801-804 [doi]
- A scalar homotopy method for parallel and robust tracking of line spectral pairsUsha Pillai, Virginia L. Stonick. 805-808 [doi]
- Detection of syntactic boundaries by partial analysis-by-synthesis of fundamental frequency contoursKeikichi Hirose, Atsuhiro Sakurai. 809-812 [doi]
- A robust method for the estimation of formant frequency modulation in speech signalsPreeti Rao. 813-816 [doi]
- Visual speech recognition using active shape models and hidden Markov modelsJuergen Luettin, Neil A. Thacker, Steve W. Beet. 817-820 [doi]
- Integrating audio and visual information to provide highly robust speech recognitionMichael J. Tomlinson, Martin J. Russell, N. M. Brooke. 821-824 [doi]
- An estimation of speaker sampling in Voice Across Japan databaseIkuo Kudo, Takao Nakama, Tomoko Watanabe. 825-828 [doi]
- Automatic articulatory annotation of multi-sensor speech databaseNathalie Parlangeau, Régine André-Obrecht, Alain Marchal. 829-832 [doi]
- Adaptive bimodal sensor fusion for automatic speechreadingUwe Meier, Wolfgang Hürst, Paul Duchnowski. 833-836 [doi]
- The bucket box intersection (BBI) algorithm for fast approximative evaluation of diagonal mixture GaussiansJürgen Fritsch, Ivica Rogina. 837-840 [doi]
- Multi channel HMMDongxin Xu, Craig L. Fancourt, Chuan Wang. 841-844 [doi]
- On-line garbage modeling for word and utterance verification in natural numbers recognitionCelinda de la Torre-Munilla, Luis A. Hernández Gómez, Francisco Javier Caminero Gil, Cesar Martín del Alamo. 845-848 [doi]
- Creation of two children's speech databasesJames D. Miller, Sungbok Lee, Rosalie M. Uchanski, Arnold F. Heidbreder, Barbara B. Richman, John Tadlock. 849-852 [doi]
- The contribution of consonants versus vowels to word recognition in fluent speechRonald A. Cole, YongHong Yan, Brian Mak, Mark A. Fanty, Troy Bailey. 853-856 [doi]
- A novel word pre-selection method based on phonetic set indexingRamesh R. Sarukkai, Dana H. Ballard. 857-860 [doi]
- DP-based wordgraph pruningThomas Kuhn, Pablo Fetter, Alfred Kaltenmeier, Peter Regel-Brietzmann. 861-864 [doi]
- A new hybrid system based on MMI-neural networks for the RM speech recognition taskGerhard Rigoll, Christoph Neukirchen, Jörg Rottland. 865-868 [doi]
- Automatic modeling of user specific words for a speaker independent recognition systemJacques Duchateau, Dirk Van Compernolle. 869-872 [doi]
- A dependence tree model of phone correlationOrith Ronen, Mari Ostendorf. 873-876 [doi]
- The use of syllable phonotactics for word hypothesizationRenato de Mori, Michael Galler. 877-880 [doi]
- Creating speaker-specific phonetic templates with a speaker-independent phonetic recognizer: implications for voice dialingNeena Jain, Ronald A. Cole, Etienne Barnard. 881-884 [doi]
- Deleted interpolation and density sharing for continuous hidden Markov modelsXuedong Huang, Mei-Yuh Hwang, Li Jiang, Milind Mahajan. 885-888 [doi]
- A time continuous model for speech recognitionStephan Euler. 889-892 [doi]
- Fast output probability computation using scalar quantization and independent dimension multi-mixtureMasayuki Yamada, Hiroki Yamamoto, Tetsuo Kosaka, Yasuhiro Komori, Yasunori Ohora. 893-896 [doi]
- Microphone-array speech recognition via incremental map trainingJohn E. Adcock, Yoshihiko Gotoh, Daniel J. Mashao, Harvey F. Silverman. 897-900 [doi]
- A localization-error-based method for microphone-array designMichael S. Brandstein, John E. Adcock, Harvey F. Silverman. 901-904 [doi]
- Nearfield broadband frequency invariant beamformingRodney A. Kennedy, Thushara D. Abhayapala, Darren B. Ward, Robert C. Williamson. 905-908 [doi]
- A source subspace tracking array of microphones for double talk situationsSofiène Affes, Yves Grenier. 909-912 [doi]
- The true duration of the impulse response used to estimate reverberation timeLaurent Faiget, Robert Ruiz, Claude Legros. 913-916 [doi]
- Sound capture from spatial volumes: matched-filter processing of microphone arrays having randomly-distributed sensorsEa-Ee Jan, James L. Flanagan. 917-920 [doi]
- Acoustic source location in noisy and reverberant environment using CSP analysisMaurizio Omologo, Piergiorgio Svaizer. 921-924 [doi]
- A robust adaptive beamformer for microphone arrays with a blocking matrix using constrained adaptive filtersOsamu Hoshuyama, Akihiko Sugiyama. 925-928 [doi]
- Voice-activated AGC for teleconferencingPeter L. Chu. 929-932 [doi]
- A two-microphone adaptive broadband array for hearing aidsElaine D. McKinney, Victor E. DeBrunner. 933-936 [doi]
- Adaptive spectral contrast enhancement based on masking effect for the hearing impairedZlatan Ribic, Jun Yang, Mathias Latzel. 937-940 [doi]
- Stereo echo cancellation algorithm using imaginary input-output relationshipsSuehiro Shimauchi, Shoji Makino. 941-944 [doi]
- SSB subband echo canceller using low-order projection algorithmShoji Makino, Josef Noebauer, Yoichi Haneda, Akira Nakagawa. 945-948 [doi]
- A fast two-channel projection algorithm for stereophonic acoustic echo cancellationFabrice Amand, Jacob Benesty, André Gilloire, Yves Grenier. 949-952 [doi]
- Optimization of a noise reduction preprocessing in an acoustic echo and noise controllerBeghdad Ayad, Gérard Faucon, Régine Le Bouquin-Jeannès. 953-956 [doi]
- Dynamically regularized fast RLS with application to echo cancellationSteven L. Gay. 957-960 [doi]
- Experimental results of subband acoustic echo cancelers under spherically invariant random processesPhillip L. De Leon, Delores M. Etter. 961-964 [doi]
- The effect of structured uncertainty in multichannel feedforward control systemsAkira Omoto, Stephen J. Elliott. 965-968 [doi]
- Automated placement of transducers for active noise control: performance measuresJulia A. Olkin, Larry P. Heck, Koorosh Naghshineh. 969-972 [doi]
- Optimisation of controlled acoustic shadowsStephen E. Wright, Branislav Vuksanovic. 973-976 [doi]
- AM-FM separation using auditory-motivated filtersThomas F. Quatieri, Thomas E. Hanna, Gerald C. O'Leary. 977-980 [doi]
- A `gammachirp' function as an optimal auditory filter with the Mellin transformIrino Toshio. 981-984 [doi]
- A binaural auditory model for sound quality measurements and spatial hearing studiesMatti Karajalainen. 985-988 [doi]
- A study on auditory resolution using Bark-FAMlet clicksUnto K. Laine, Minna Huotilainen. 989-992 [doi]
- Real-time discrimination of broadcast speech/musicJohn Saunders. 993-996 [doi]
- Analysis and resynthesis of musical instrument sounds using energy separationRobert B. Sussman, Mark Kahrs. 997-1000 [doi]
- Automatic audio morphingMalcolm Slaney, Michele Covell, Bud Lassiter. 1001-1004 [doi]
- Residual modeling in music analysis-synthesisMichael M. Goodwin. 1005-1008 [doi]
- Computationally efficient algorithm for time scale modification (GLS-TSM)Susan Yim, Basavaraj I. Pawate. 1009-1012 [doi]
- Multichannel dynamic range compression for music signalsJon C. Schmidt, Janet C. Rutledge. 1013-1016 [doi]
- A bi-dimensional coding scheme applied to audio bitrate reductionLaurent Mainard, Michel Lever. 1017-1020 [doi]
- Audio coding with a dynamic wavelet packet decomposition based on frequency-varying modulated lapped transformsMarcus Purat, Peter Noll. 1021-1024 [doi]
- A test of MPEG using time-inverted spoken audioThomas McLaughlin, John Cookson, Lloyd Rasmussen. 1025-1028 [doi]
- Extension and complexity reduction of TwinVQ audio coderTakehiro Moriya, Naoki Iwakami, Kazunaga Ikeda, Satoshi Miki. 1029-1032 [doi]
- Minimising the effects of subband quantisation of the time domain aliasing cancellation filter bankConrad Jakob, Alan B. Bradley. 1033-1036 [doi]
- Speech analysis and coding using a multi-resolution sinusoidal transformDavid V. Anderson. 1037-1040 [doi]
- Audio coding using the wavelet packet transform and a combined scalar-vector quantizationSimon Boland, Mohamed Deriche. 1041-1044 [doi]
- Low bit rate high quality audio coding with combined harmonic and wavelet representationsKhaled N. Hamdy, Murtaza Ali, Ahmed H. Tewfik. 1045-1048 [doi]
- A high performance software implementation of MPEG audio encoderManoj Kumar, Mohammad Zubair. 1049-1052 [doi]
- Audio compression at low bit rates using a signal adaptive switched filterbankDeepen Sinha, James D. Johnston. 1053-1056 [doi]
- Signal processing techniques for efficient use of transmit diversity in wireless communicationsGregory W. Wornell, Mitchell D. Trott. 1057-1060 [doi]
- Adaptive suppression of narrowband digital interferers from spread spectrum signalsH. Vincent Poor, Xiaodong Wang. 1061-1064 [doi]
- Signal processing for interference suppression in direct-sequence CDMA systemsUpamanyu Madhow. 1065-1068 [doi]
- Digital video in a fading interference wireless environmentLouis C. Yun, David G. Messerschmitt. 1069-1072 [doi]
- Singular value analysis of space-time equalization in the GSM mobile systemAlle-Jan van der Veen, Arogyaswami Paulraj. 1073-1076 [doi]
- Unconditional maximum likelihood approach for blind estimation of digital signalsBijit Halder, Boon Chong Ng, Arogyaswami Paulraj, Thomas Kailath. 1081-1084 [doi]
- Geometric properties of the blind digital co-channel communications problemLars K. Hansen, Guanghan Xu. 1085-1088 [doi]
- Low complexity optimal multiple access joint detection for linearly dependent user setsRachel E. Learned, Alan S. Willsky. 1089-1092 [doi]
- Estimation and equalization of fading channels with random coefficientsMichail K. Tsatsanis, Georgios B. Giannakis, Guotong Zhou. 1093-1096 [doi]
- Real-time DSP for sophomoresKenneth H. Chiang, Brian L. Evans, William T. Huang, Ferenc Kovac, Edward A. Lee, David G. Messerschmitt, H. John Reekie, S. Shankar Sastry. 1097-1100 [doi]
- Multi-media and World Wide Web resources for teaching DSPJames H. McClellan, Ronald W. Schafer, Jeffrey B. Schodorf, Mark A. Yoder. 1101-1104 [doi]
- Developing Internet-based VHDL course materialJ. Scott Calhoun, Jack Griffin, Allan Bullwinkel. 1105-1108 [doi]
- Distance teaming experiments in undergraduate DSPDelores M. Etter, Geoffrey C. Orsak, Don Herrick Johnson. 1109-1112 [doi]
- A system characterization/identification laboratory teaching tool for InternetSimon Chatfield, Douglas Cochran, Michael Sadaka, Dana Sinno. 1113-1116 [doi]
- The Telecomputing laboratory: a multipurpose facility used in DSP education at the University of OklahomaVictor E. DeBrunner, Linda DeBrunner, Sridhar Radhakrishnan, A. Kamal Khan. 1117-1120 [doi]
- An integrated environment for modeling, simulation, digital signal processing, and controlCurtis C. Crane, Richard J. Kozick. 1121-1124 [doi]
- Spectra-a hands-on DSP learning experienceFred J. Taylor, Jon Mellott, Michael Lewis. 1125-1128 [doi]
- A laboratory course for designing and testing spoken dialogue systemsDon Colton, Ronald A. Cole, David G. Novick, Stephen Sutton. 1129-1132 [doi]
- A MATLAB software tool for the introduction of speech coding fundamentals in a DSP courseEdward M. Painter, Andreas Spanias. 1133-1136 [doi]
- Philosophy and goals of the DoD 2400 bps vocoder selection processThomas Tremain, Mary A. Kohler, Terrence G. Champion. 1137-1140 [doi]
- Developing a test program for the DoD 2400 bps vocoder selection processMatthew R. Bielefeld, Lynn M. Supplee. 1141-1144 [doi]
- Vocoder intelligibility and quality test methodsJohn D. Tardelli, Elizabeth Woodard Kreamer. 1145-1148 [doi]
- Speaker recognizability testing for voice codersAstrid Schmidt-Nielsen, Derek P. Brock. 1149-1156 [doi]
- Communicability Testing for Voice CodersElizabeth Woodard Kreamer, John D. Tardelli. 1153 [doi]
- Host laboratory functions for the DoD 2400 bps vocoder selection processPaul D. Gatewood, Philip A. LaFollette. 1157-1160 [doi]
- Criteria for the DoD 2400 bps vocoder selectionMary A. Kohler, Philip A. LaFollette, Matthew R. Bielefeld. 1161-1164 [doi]
- The ARPA/NAVY Mountaintop Program: adaptive signal processing for airborne early warning radarGerard W. Titi, Daniel F. Marshall. 1165-1168 [doi]
- Evaluation of partially adaptive STAP algorithms on the Mountain Top data setYaron Seliktar, Douglas B. Williams, James H. McClellan. 1169-1172 [doi]
- Reduced rank space-time adaptive radar processingJ. Scott Goldstein, Peter A. Zulch, Irving S. Reed. 1173-1176 [doi]
- Beamspace techniques for hot clutter cancellationStephen M. Kogon, Douglas B. Williams, E. Jeff Holder. 1177-1180 [doi]
- Linear constraints in hot clutter cancellationLloyd J. Griffiths. 1181-1184 [doi]
- Robust matched-field beamforming with benchmark shallow-water acoustic array dataJeffrey L. Krolik. 1185-1188 [doi]
- A coherent approach to broadband matched-field processing: application in the Hudson CanyonM. B. Porter, Z.-H. Michalopoulou. 1189-1192 [doi]
- Matched-field localization with many uncertain environmental parameters: experimental data resultsBrian F. Harrison, Richard J. Vaccaro, Donald W. Tufts. 1193-1196 [doi]
- A posteriori probability source localization in an uncertain shallow water environment (Hudson Canyon)Loren W. Nolte, Jeffrey A. Shorey. 1197-1200 [doi]
- A new 3D segmentation method for region-based video codingHang-Bong Kang. 1201-1204 [doi]
- A very low bit rate video coding system using adaptive region-classified vector quantizationYee-Wen Chen, Liang-Gee Chen, Mei-Juan Chen. 1205-1208 [doi]
- A hierarchical video coder with cache motion informationKwan K. Truong, Craig H. Richardson. 1209-1212 [doi]
- Evaluation of a mosaic based approach to video compressionBruce Tannenbaum, Rajesh Suryadevara, Stephen C. Hsu. 1213-1215 [doi]
- JACOB: just a content-based query system for video databasesMarco La Cascia, Edoardo Ardizzone. 1216-1219 [doi]
- Nonlinear editing by Generative VideoRadu S. Jasinschi, José M. F. Moura. 1220-1223 [doi]
- Tracking areas of interest for content-based functionalities in segmentation-based video codingFerran Marques, Beatriz Marcotegui, Fernand Meyer. 1224-1227 [doi]
- Key frame selection by motion analysisWayne H. Wolf. 1228-1231 [doi]
- Executable requirements: opportunities and impedimentsGary A. Shaw, Allan H. Anderson. 1232-1235 [doi]
- Rapid prototyping of DSP chip-sets via functional reuseMohamed Ben-Romdhane, Vijay K. Madisetti. 1236-1239 [doi]
- Heuristic techniques for synthesis of hard real-time DSP application specific systemsMiodrag Potkonjak, Wayne H. Wolf. 1240-1243 [doi]
- A CAD tool for the optimization of video signal processor architecturesHolger Kropp, Markus Schwiegershausen, Peter Pirsch. 1244-1247 [doi]
- Hardware/software co-design for DSP applications via the HMS frameworkMichael Sheliga, Edwin Hsing-Mean Sha. 1248-1251 [doi]
- An architectural trade capability using the Ptolemy kernelEric K. Pauer, Jonathan B. Prime. 1252-1255 [doi]
- Rapid prototyping of DSP systems via system interface module generationShahram Famorzadeh, Vijay K. Madisetti. 1256-1259 [doi]
- Autocoding: an enabling technology for rapid prototypingChristopher B. Robbins. 1260-1263 [doi]
- Middleware for realtime multicomputer tool developmentBarry Isenstein, Michael Krueger, Arlan Pool. 1264-1267 [doi]
- Interface synthesis in heterogeneous system-level DSP design toolsJosé Luis Pino, Michael C. Williamson, Edward A. Lee. 1268-1271 [doi]
- Improved design of long FIR filters using the frequency masking techniqueMaurice G. Bellanger. 1272-1275 [doi]
- Optimal least-squares design of pipelined recursive filters in the time-domainSrikanth Pokala, Mohammed Imtiaz, Arnab K. Shaw. 1276-1279 [doi]
- Hardware efficient FIR filter structures from linear program design constraintsKen T. Sienski, Mike Legako. 1280-1282 [doi]
- GA-based design of multiplierless 2-D state-space digital filters with very small roundoff noiseYoung-Ho Lee, Masayuki Kawamata, Tatsuo Higuchi 0001. 1283-1286 [doi]
- Design of efficient FIR filters with cyclotomic polynomial prefilters using mixed integer linear programmingWoo-Jin Oh, Yong-Hoon Lee. 1287-1290 [doi]
- Convolution using the undecimated discrete wavelet transformHaitao Guo, C. Sidney Burrus. 1291-1294 [doi]
- DSP implementation and performance evaluation of sparse-tap adaptive FIR filters with tap-position controlAkihiro Hirano, Akihiko Sugiyama, Shigeji Ikeda. 1295-1298 [doi]
- A fast transversal filter for the numerical factorization of polynomialsAli A. Hasan, Mohammed A. Hasan. 1299-1302 [doi]
- Analysis of subband adaptive filtersHideaki Sakai, Noriyuki Hirayama. 1303-1306 [doi]
- Generalized adaptive polyphase algorithmUmashankar Iyer, Majid Nayeri. 1307-1310 [doi]
- Real-valued fast discrete Fourier transform and cyclic convolution algorithms of highly composite even lengthHideo Murakami. 1311-1314 [doi]
- DFT computation using shift and addition operationsNirdosh Bhatnagar. 1315-1318 [doi]
- Pipeline processor for fast architecture oriented regular DCT-IDCT algorithmDavid Akopian, Jaakko Astola. 1319-1322 [doi]
- An FFT for the 2-sphere and applicationsDennis M. Healy Jr., Daniel N. Rockmore, Sean S. B. Moore. 1323-1326 [doi]
- Fast algorithm for 2-D image moments via the Radon transformTak-Wai Shen, Daniel Pak-Kong Lun, Wan-Chi Siu. 1327-1330 [doi]
- A new algorithm for computing sparse solutions to linear inverse problemsGopal Harikumar, Yoram Bresler. 1331-1334 [doi]
- A Levinson-like fast algorithm for solving block-slanted Toeplitz systems of equations arising in wavelet-based solution of integral equationsRajashri R. Joshi, Andrew E. Yagle. 1335-1338 [doi]
- A fast M-D Chandrasekhar algorithm for second order Volterra adaptive filteringMounir Sayadi, Abdelkader Chaari, Farhat Fnaiech, Mohamed Najim. 1339-1342 [doi]
- A fast wavelet transform-domain LMS algorithmSumir Attallah, Mohamed Najim. 1343-1346 [doi]
- A polynomial-transform based computation of the 2-D DCT with minimum multiplicative complexityJacques Prado, Pierre Duhamel. 1347-1350 [doi]
- The digital prolate spheroidal windowTony S. Verma, Stefan Bilbao, Teresa H. Y. Meng. 1351-1354 [doi]
- A well-conditioned quadratic program for unique design of two-dimensional weighted Chebyshev FIR filtersStefan Nordebo, Ingvar Claesson. 1355-1358 [doi]
- Efficient design of families of FIR filters by transformationLina J. Karam, James H. McClellan. 1359-1362 [doi]
- Applications of discrete-time Laguerre networks to envelope constrained filter designZhuquan Zang, Ba-Ngu Vo, Antonio Cantoni, Kok Lay Teo. 1363-1366 [doi]
- Generalized digital Butterworth filter designIvan W. Selesnick, C. Sidney Burrus. 1367-1370 [doi]
- Roundoff noise minimization in a direct form delta operator structureJuha Kauraniemi, Timo I. Laakso, Iiro Hartimo, Seppo J. Ovaska. 1371-1374 [doi]
- Improved power-of-two sharpening filter design by genetic algorithmPaolo Gentili, Francesco Piazza, Aurelio Uncini. 1375-1378 [doi]
- Design of stable IIR filters in the complex domain by automatic delay selectionRudi Vuerinckx, Yves Rolain, Johan Schoukens, Rik Pintelon. 1379-1382 [doi]
- Decolored Ll filter lattice: a subclass of Ll filters useful in practiceYeong-Taeg Kim. 1383-1386 [doi]
- High-order error feedback for noise reduction in state-space digital filtersTakao Hinamoto, Shuji Karino. 1387-1390 [doi]
- Oversampled FIR and IIR DFT filter banks and Weyl-Heisenberg framesHelmut Bölcskei, Franz Hlawatsch, Hans G. Feichtinger. 1391-1394 [doi]
- Wigner-type a-b and time-frequency analysis based on conjugate operatorsFranz Hlawatsch, Teresa Twaroch, Helmut Bölcskei. 1395-1398 [doi]
- Local polynomial periodograms for signals with the time-varying frequency and amplitudeVladimir Katkovnik. 1399-1402 [doi]
- Evolutionary spectral analysis using a warped Gabor expansionAydin Akan, Luis F. Chaparro. 1403-1406 [doi]
- Discrete powers-of-two kernels for time-frequency distributionsGopal T. Venkatesan, Moeness G. Amin. 1407-1410 [doi]
- Time-frequency strip filtersBeth A. Weisburn, Ramachandra G. Shenoy. 1411-1414 [doi]
- Interactive wavelet processing and techniques applied to digital mammographyIztok Koren, Andrew F. Laine, Fred J. Taylor, Michael Lewis. 1415-1418 [doi]
- Optimal phase kernels for time-frequency analysisL. Fridtjof Wisur-Olsen, Richard G. Baraniuk. 1419-1422 [doi]
- Pseudo affine Wigner distributionsPaulo Gonçalves, Richard G. Baraniuk. 1423-1426 [doi]
- What are the joint time-frequency moments of a signal?Berkant Tacer, Patrick J. Loughlin. 1427-1430 [doi]
- Generalized joint signal representations and optimum detectionAkbar M. Sayeed, Douglas L. Jones. 1431-1434 [doi]
- Analysis of errors in quantization of Weyl-Heisenberg frame expansions and oversampled A/D conversionZ. Cvetkovic. 1435-1438 [doi]
- Time-varying characteristics of postural sway differ between balance impaired and unimpaired populationsPatrick J. Loughlin, Mark S. Redfern, Berkant Tacer. 1439-1442 [doi]
- Wavelet-based multi-carrier CDMA for personal communications systemsKyungHi Chang, XuDuan Lin, Huijuan Li. 1443-1446 [doi]
- Optimized wavelets for fingerprint compressionBarry G. Sherlock, Donald M. Monro. 1447-1450 [doi]
- SurfWave: an object-oriented class library for wavelet analysisJames M. Adler, Bhaskar D. Rao. 1451-1454 [doi]
- A fast technique for identifying zerotrees in the EZW algorithmJerome M. Shapiro. 1455-1458 [doi]
- Fast computation of the continuous wavelet transform through oblique projectionsMichael J. Vrhel, Chulhee Lee, Michael Unser. 1459-1462 [doi]
- The design of maximally smooth waveletsM. Lang, Peter N. Heller. 1463-1466 [doi]
- Discrete finite variation: a new measure of smoothness for the design of wavelet basisJan E. Odegard, C. Sidney Burrus. 1467-1470 [doi]
- Biorthogonal cosine-modulated filter bankTruong Q. Nguyen, Peter N. Heller. 1471-1474 [doi]
- A distortion analysis and optimal design of orthogonal basis for DMT transceiversXueming Lin, Ali N. Akansu. 1475-1478 [doi]
- Processing arbitrary-length signals with MDFT filter banksTanja Karp, Jörg Kliewer, Alfred Mertins, Norbert J. Fliege. 1479-1482 [doi]
- Cosine-modulated 2-dimensional perfect reconstruction FIR filter banks with linear phaseTakuyuki Nagai, Masaaki Ikehara. 1483-1486 [doi]
- Theory of optimal orthonormal filter banksPalghat P. Vaidyanathan. 1487-1490 [doi]
- Filter bank design for minimum distortion in presence of subband quantizationKarine Gosse, François Moreau de Saint-Martin, Pierre Duhamel. 1491-1494 [doi]
- Signal-based design of wavelet filtersF. Moreau de Saint-Martin, E. Gautier, J.-P. d'Ales de Corbet. 1495-1498 [doi]
- Two-dimensional orthogonal and symmetrical wavelets and filter-banksDavid Stanhill, Yehoshua Y. Zeevi. 1499-1502 [doi]
- Statistical behavior of wavelet transmultiplexers in arbitrary synchronization statesMichael J. Lyall, Richard S. Orr. 1503-1506 [doi]
- Wavelets in polar coordinatesOleg Andric, Carolyn Johnston, Nurgun Erdol. 1507-1510 [doi]
- Non-uniform filter banks based on a multi-prototype cosine modulationFabrizio Argenti, Enrico Del Re. 1511-1514 [doi]
- Parameterization and implementation of orthogonal wavelet transformsPeter Rieder, Kolja Gerganoff, Jürgen Götze, Josef A. Nossek. 1515-1518 [doi]
- Design of signal-adapted FIR paraunitary filter banksPierre Moulin, Mihai Anitescu, Kenneth O. Kortanek, Florian A. Potra. 1519-1522 [doi]
- A nonlinear subband decomposition with perfect reconstructionFrances Jane Hampson, Jean-Christophe Pesquet. 1523-1526 [doi]
- Optimal design of transmitter and receiver filters with mixed performance objectivesWilliam M. Campbell, Tom W. Parks. 1527-1529 [doi]
- 2(Z)Martin Vetterli, Zoran Cvetkovic. 1530-1533 [doi]
- Multistage multirate adaptive filtersGeoffrey A. Williamson, Soura Dasgupta, Minyue Fu. 1534-1537 [doi]
- 2D factorizable FIR principal component filter banksBo Xuan, Roberto H. Bamberger. 1538-1541 [doi]
- Reconstruction-optimized lapped orthogonal transformsSheila S. Hemami. 1542-1545 [doi]
- A new two dimensional nonseparable modulated filter banksShing-Chow Chan. 1546-1549 [doi]
- Discrete coefficients filter banks and applications in image codingChi-Wah Kok, Truong Q. Nguyen. 1550-1553 [doi]
- A polyphase approach to time-varying filter banksSee-May Phoong, Palghat P. Vaidyanathan. 1554-1557 [doi]
- A low probability of intercept signal detection receiver using quadrature mirror filter bank treesThomas C. Farrell, Glenn Prescott. 1558-1561 [doi]
- Adaptive spectrum exchange and modification using multirate filtersShigeo Wada. 1562-1565 [doi]
- Design of 2-dimensional linear phase perfect reconstruction FIR filter bank using Lagrange multiplier methodMasaaki Ikehara, Kenji Inose. 1566-1569 [doi]
- A pyramidal coder using a nonlinear filter bankRicardo L. de Queiroz, Dinei A. F. Florêncio. 1570-1573 [doi]
- Determination of a canonical lattice structure for odd length linear-phase PR filter banksDidier Pinchon, Pierre Siohan. 1574-1577 [doi]
- Why and how prefiltering for discrete multiwavelet transformsXiang-Gen Xia, Jeffrey S. Geronimo, Douglas P. Hardin, Bruce W. Suter. 1578-1581 [doi]
- Wavelet packet shrinkage based adaptive resamplingChia-Chun Huang, Ashutosh Sabharwal, Yuan F. Zheng, Lee C. Potter. 1581 [doi]
- Modulated filter banks and wavelets-a general unified theoryRamesh A. Gopinath. 1585-1588 [doi]
- Parallel-cascade realizations and approximations of truncated Volterra systemsThomas M. Panicker, V. John Mathews. 1589-1592 [doi]
- Piecewise Volterra filters based on the threshold decomposition operatorEdwin A. Heredia, Gonzalo R. Arce. 1593-1596 [doi]
- Theory and applications of adaptive second order IIR Volterra filtersE. Roy, Robert W. Stewart, Tariq S. Durrani. 1597-1600 [doi]
- Blind Volterra signal modelingTania Stathaki. 1601-1604 [doi]
- Channel equalization for self-synchronizing chaotic systemsKevin M. Cuomo, Alan V. Oppenheim, Richard J. Barron. 1605-1608 [doi]
- TLS parameter estimation for filtering chaotic time seriesSusan M. Schweizer, Virginia L. Stonick, J. L. Evans. 1609-1612 [doi]
- An improved quantitative measure of image restoration qualitySteven Sheung-On Choy, Yuk-Hee Chnn, Wan-Chi Siu. 1613-1616 [doi]
- A class of stochastic resonance systems for signal processing applicationsHaralabos C. Papadopoulos, Gregory W. Wornell. 1617-1620 [doi]
- Jacobi's method for massive parallel wave digital filter algorithmThomas Felderhoff. 1621-1624 [doi]
- Detection and estimation of soliton signalsAndrew C. Singer. 1625-1628 [doi]
- A novel adaptive IIR filter for active noise controlDavid H. Crawford, Robert W. Stewart, E. Toma. 1629-1632 [doi]
- An effective approach to adaptive IIR filteringMoritz Harteneck, Robert W. Stewart, John G. McWhirter, Ian K. Proudler. 1633-1636 [doi]
- Canonical correlations and canonical time seriesJohn K. Thomas, Louis L. Scharf. 1637-1640 [doi]
- A new interpretation of the misadjustment in adaptive filteringHans-Jürgen Butterweck. 1641-1643 [doi]
- Adaptive threshold nonlinear algorithm for adaptive filters with robustness against impulse noiseShin'ichi Koike. 1644-1647 [doi]
- Adaptive subspace estimation-application to moving sources localization and blind channel identificationChristian Riou, Thierry Chonavel, Pierre Y. Cochet. 1648-1651 [doi]
- Average quantizer adaptation rates for stable ADPCMSoura Dasgupta. 1652-1655 [doi]
- Efficient approximate implementations of the fast affine projection algorithm using orthogonal transformsScott C. Douglas. 1656-1659 [doi]
- A statistical reference criterion for adaptive filteringJosep Sala, Gregori Vazquez. 1660-1663 [doi]
- Extended anti-Hebbian adaptation for unsupervised source extractionZied Malouche, Odile Macchi. 1664-1667 [doi]
- Nonseparable sampling theorems for two-dimensional signalsYuan-Pei Lin, Palghat P. Vaidyanathan. 1668-1671 [doi]
- Non-iterative implementation of a class of iterative signal restoration algorithmsDavid O. Walsh, Pamela A. Delaney, Michael W. Marcelin. 1672-1675 [doi]
- New results on observation selection in signal reconstructionStanley J. Reeves, Zhe Zhao 0002. 1676-1679 [doi]
- A new inverse filter criterion for blind deconvolution of spiky signals using Gaussian mixturesIgnacio Santamaría-Caballero, Carlos Pantaleon-Prieto, Fernando Díaz-de-María, Antonio Artes-Rodriguez. 1680-1683 [doi]
- A new linear co-channel interference mitigation algorithmSam Heidari, Chrysostomos L. Nikias. 1684-1687 [doi]
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- Determination of data matrix dimensions for subspace-based parameter estimation algorithmsYinong Ding, Richard J. Vaccaro. 2547-2550 [doi]
- Positive-definite Toeplitz completion in DOA estimation for fully-augmentable nonuniform linear antenna arraysYuri I. Abramovich, Douglas A. Gray, Alexei Y. Gorokhov, Nicholas K. Spencer. 2551-2554 [doi]
- Robust adaptive subspace extraction for DOA trackingDekun Yang, Stuart J. Flockton. 2555-2558 [doi]
- Signal selective DOA tracking for multiple moving targetsJaume Riba, Jason Goldberg, Gregori Vázquez, Miguel Angel Lagunas. 2559-2562 [doi]
- A second order recursive algorithm for subspace trackingÅke Andersson, Holger Broman. 2563-2566 [doi]
- A self-recovery scheme in anti-jamming communicationsMurat Torlak, Guanghan Xu, Hui Liu. 2567-2570 [doi]
- Closed-form 2D angle estimation with arbitrary sampling lattices over a circular apertureJavier Ramos, Michael D. Zoltowski. 2571-2574 [doi]
- Are quadratic time-frequency representations really necessary?Stanley Lawrence Marple Jr.. 2575-2578 [doi]
- Wavelet analysis of fractal point processesAlberto Recio. 2579-2582 [doi]
- Cross term deleted Wigner representation (CDWR) based signal detection methodologiesShubha Kadambe, Richard S. Orr, Micheal J. Lyall. 2583-2586 [doi]
- Analysis of the time-frequency representation using the Gamma filterSamel Çelebi, José C. Príncipe. 2587-2590 [doi]
- Design of biorthogonal MRA generating wavelets matched to specified signalMysore R. Raghuveer, Joseph O. Chapa. 2591-2593 [doi]
- Frames and orthonormal bases for variable windowed Fourier transformsNeng-Tsann Ueng, Louis L. Scharf. 2594-2597 [doi]
- A pitch dependent time-frequency response for periodic signalsMichael L. Kramer, Douglas L. Jones. 2598-2601 [doi]
- Detection in dense signal environments via the Gabor transformRichard S. Orr, Cameron M. Pike. 2602-2605 [doi]
- MMSE parameter estimation of multiple chirp signalsHsiang-Tsun Li, Petar M. Djuric. 2606-2609 [doi]
- Aliasing in the AF-GDTFD and the discrete spectrogramJeffrey C. O'Neill, William J. Williams. 2610-2613 [doi]
- Signal cancellation effects in adaptive radar Mountaintop data-setMurat O. Berin, Alexander M. Haimovich. 2614-2617 [doi]
- Accuracy analysis for off-line array calibrationBenjamin Friedlander, Boaz Porat. 2618-2621 [doi]
- Joint estimation of time delays and directions of arrival of multiple reflections of a known signalMati Wax, Amir Leshem. 2622-2625 [doi]
- Direction finding and array calibration based on maximal set of nonredundant cumulantsHuan Wu, Yongkang Jia, Zheng Bao. 2626-2629 [doi]
- Forwards-backwards averaging for adaptive beamforming and STAPMichael A. Zatman, Daniel F. Marshall. 2630-2633 [doi]
- Bayesian approach to best basis selectionJean-Christophe Pesquet, Hamid Krim, David Leporini, E. Hamman. 2634-2637 [doi]
- System identification via singular value decompositionShih-Ho Wang, Tong-Fu Lee, Randy Zachery. 2638-2641 [doi]
- Stability criteria for Arnoldi-based model-order reductionIbrahim M. Elfadel, Luis Miguel Silveira, Jacob K. White. 2642-2645 [doi]
- Application of loudness/pitch/timbre decomposition operators to auditory scene analysisMototsugu Abe, Shigeru Ando. 2646-2649 [doi]
- Gap detector for multipathChuang He, José M. F. Moura, Steven A. Benno. 2650-2653 [doi]
- A maximum likelihood CDMA receiver using the EM algorithm and the discrete wavelet transformIlan Sharfer, Alfred O. Hero III. 2654-2657 [doi]
- Blind equalization for CDMA systems with aperiodic spreading sequencesHui Liu, Guanghan Xu. 2658-2661 [doi]
- Multiuser detection with antenna arrays in the presence of multipath fadingTyler A. Brown, Mostafa Kaveh. 2662-2665 [doi]
- A constrained adaptive diversity combiner for interference suppression in CDMA systemsJeffrey B. Schodorf, Douglas B. Williams. 2666-2669 [doi]
- An efficient propagation delay estimator for DS-CDMA signalsDunmin Zheng, Jian Li, Scott L. Miller. 2670-2673 [doi]
- Fractionally-spaced CMA under channel noiseAzzédine Touzni, Inbar Fijalkow, John R. Treichler. 2674-2677 [doi]
- Block Shanno constant modulus algorithm for wireless equalizationsZifei Wang, Eric M. Dowling. 2678-2681 [doi]
- A fast tracking adaptive array system for mobile communicationsHua Wang, K. J. Ray Liu. 2682-2685 [doi]
- Separation of cochannel signals using the parametric constant modulus algorithmAshwin Rao, Ramadas Kumaresan. 2686-2689 [doi]
- Sequential modulation classification of dependent samplesYu-Chuan Lin, C. C. Jay Kuo. 2690-2693 [doi]
- Blind maximum likelihood sequence detectionDavid J. Reader, William G. Cowley. 2694-2697 [doi]
- Blind adaptive separation of wide-band sourcesChristine Servière, Véronique Capdevielle. 2698-2701 [doi]
- POCS-based blind array processing in incoherent microwave radiometric image reconstructionKenbu Teramoto, Kohei Arai. 2702-2705 [doi]
- An unconstrained single stage criterion for blind source separationAdriana Dapena, Luis Castedo, Carlos J. Escudero. 2706-2709 [doi]
- Blind multi-user combining at the base station for asynchronous CDMA systemsOlga Muñoz-Medina, Juan A. Fernández-Rubio. 2710-2713 [doi]
- Spatio-temporal training-sequence based channel equalization and adaptive interference cancellationDirk T. M. Slock. 2714-2717 [doi]
- Blind identification of a linear-quadratic mixture of independent components based on joint diagonalization procedureKarim Abed-Meraim, Adel Belouchiani, Yingbo Hua. 2718-2721 [doi]
- Resolution limits of blind multi-user multi-channel identification schemes-the bandlimited caseAlle-Jan van der Veen. 2722-2725 [doi]
- Blind separation of voice modulated single-side-band using the multi-target variable modulus algorithmJohn D. Fite, Stephen P. Bruzzone, Brian G. Agee. 2726-2729 [doi]
- Blind multi-user access interference cancellation for CDMA based PCS/cellular using antenna arraysMichael D. Zoltowski, Javier Ramos. 2730-2733 [doi]
- Application of MVDR beamforming to reject turbulence noise in a ductDaniel J. Sebald, Barry D. Van Veen. 2734-2737 [doi]
- Detection and characterization of cracks for failure monitoring and diagnosticsKevin M. Buckley, Gopal T. Venkatesan, Dennis West, Mostafa Kaveh. 2738-2741 [doi]
- On the problem of detection and discrimination of double talk and change in the echo pathCatharina Carlemalm, Fredrik Gustafsson, Bo Wahlberg. 2742-2745 [doi]
- Application of time-frequency analysis for optimum non-equidistant sampling of automotive signals captured at knockDetlef König. 2746-2749 [doi]
- Best basis segmentation of ECG signals using novel optimality criteriaDana H. Brooks, Hamid Krim, Jean-Christophe Pesquet, Robert S. MacLeod. 2750-2753 [doi]
- Measuring the performance of implemented multirate systemsFrank Heinle, Hans W. Schuessler. 2754-2757 [doi]
- Reduced interference distributions for the detection and classification of outside sound source acoustic emissionsJohn C. Mossing, Theresa A. Tuthill. 2758-2761 [doi]
- On the processing of piecewise-constant signals by hierarchical models with application to single ion channel currentsPetar M. Djuric, Jong-Kae Fwu, Slobodan Jovanovic, Kelvin Lynn. 2762-2765 [doi]
- Stochastic modelling and identification of lubricated polymer friction dynamicsGeesern Hsu, Andrew E. Yagle, Kenneth C. Ludema, Joel A. Levitt. 2766-2768 [doi]
- Two-channel multifilter banks and multiwaveletsTodor Cooklev, Akinori Nishihara, Masaki Kato, Michael Sablatash. 2769-2772 [doi]
- Applying the minimum free energy spectral estimation algorithm to ISAR imagingAvraham Freedman, Bernard D. Steinberg, George Chkhenkeli. 2773-2776 [doi]
- A statistical array processing approach for elastic wavefield dataGregory S. Wagner. 2777-2780 [doi]
- Efficient space-time adaptive processing for airborne MTI-mode radarJoseph R. Guerci, S. Unnikrishna Pillai, Younglok Kim. 2781-2784 [doi]
- Resonant pole identification from the acoustic backscatter of small submerged multi-specular elastic targetsJoEllen Wilbur, Robert J. McDonald. 2785-2788 [doi]
- High accuracy 2D angle estimation with extended aperture vector sensor arraysKainam Thomas Wong, Michael D. Zoltowski. 2789-2792 [doi]
- Spectral analysis for Doppler radar: a parametric model for regularisationGwénélle Le Foll, Pascal Larzabal, Henri Clergeot. 2793-2796 [doi]
- Computationally efficient estimation of sinusoidal frequency at low SNRSrinivasan Umesh, Douglas Nelson. 2797-2800 [doi]
- Simple, effective estimation of frequency based on Prony's methodDonald W. Tufts, Paul D. Fiore. 2801-2804 [doi]
- Structured least squares to improve the performance of ESPRIT-type high-resolution techniquesMartin Haardt, Josef A. Nossek. 2805-2808 [doi]
- Efficient phase-only frequency estimationPaul D. Fiore, Stephen W. Lang. 2809-2812 [doi]
- Volterra filter identification using penalized least squaresRobert D. Nowak. 2813-2816 [doi]
- Time-delay estimation for compound point-processes using hidden Markov modelsJens Wohlers, Alfred Mertins, Norbert J. Fliege. 2817-2820 [doi]
- Multiple-prediction-horizon recursive identification of hidden Markov modelsIain B. Collings, John B. Moore. 2821-2824 [doi]
- On operators providing shift-invariant uncertainty limitMilos Doroslovacki. 2825-2828 [doi]
- Signal modeling with dynamic fuzzy setsBogdan R. Kosanovic, Luis F. Chaparro, Robert J. Sclabassi. 2829-2832 [doi]
- Weighted myriad filters: a robust filtering framework derived from alpha-stable distributionsJuan Guillermo Gonzalez, Gonzalo R. Arce. 2833-2836 [doi]
- Evolutionary ARMA modelling of time-varying impulse responseJamila Bakkoury, Francis Castanie, Daniel Roviras. 2837-2840 [doi]
- Signal modeling and parameter estimation for 1/f processes using scale stationary modelsBirsen Yazici,