Table of Contents

• Portability of syntactic structure for language modelingCiprian Chelba. 1 [doi]
• The IBM Personal Speech AssistantLiam Comerford, David Frank, Ponani Gopalakrishnan, Ramesh Gopinath, Jan Sedivý. 1-4 [doi]
• Speaker- and language-independent speech recognition in mobile communication systemsOlli Viikki, Imre Kiss, Jilei Tian. 5-8 [doi]
• MiPad: a multimodal interaction prototypeXuedong Huang, Alex Acero, Ciprian Chelba, Li Deng, Jasha Droppo, Doug Duchene, Joshua Goodman, Hsiao-Wuen Hon, Derek Jacoby, Li Jiang, Ricky Loynd, Milind Mahajan, Peter Mau, Scott Meredith, Salman Mughal, Salvado Neto, Mike Plumpe, Kuansan Steury, Gina Venolia, Kuansan Wang, Ye-Yi Wang. 9-12 [doi]
• Ubiquitous speech processingSadaoki Furui, Koji Iwano, Chiori Hori, Takahiro Shinozaki, Yohei Saito, Satoshi Tamura. 13-16 [doi]
• On the implementation of ASR algorithms for hand-held wireless mobile devicesRichard C. Rose, Sarangarajan Parthasarathy, Bojana Gajic, Aaron E. Rosenberg, Shrikanth S. Narayanan. 17-20 [doi]
• Acoustic synthesis of training data for speech recognition in living room environmentsVolker Stahl, Alexander Fischer, Rolf Bippus. 21-24 [doi]
• Automatic transcription of voicemail at AT&TMichiel Bacchiani. 25-28 [doi]
• Error corrective mechanisms for speech recognitionLidia Mangu, Mukund Padmanabhan. 29-32 [doi]
• Explicit word error minimization using word hypothesis posterior probabilitiesFrank Wessel, Ralf Schlüter, Hermann Ney. 33-36 [doi]
• From broadcast news to spontaneous dialogue transcription: portability issuesNicola Bertoldi, Fabio Brugnara, Mauro Cettolo, Marcello Federico, Diego Giuliani. 37-40 [doi]
• Generation and expansion of word graphs using long span context informationChristoph Neukirchen, Dietrich Klakow, Xavier L. Aubert. 41-44 [doi]
• Improved discriminative training techniques for large vocabulary continuous speech recognitionDaniel Povey, Philip C. Woodland. 45-48 [doi]
• Improvements in linear transform based speaker adaptationLuís Felipe Uebel, Philip C. Woodland. 49-52 [doi]
• Innovative approaches for large vocabulary name recognitionYuqing Gao, Bhuvana Ramabhadran, C. Julian Chen, Hakan Erdogan, Michael Picheny. 53-56 [doi]
• New features in the CU-HTK system for transcription of conversational telephone speechThomas Hain, Philip C. Woodland, Gunnar Evermann, Daniel Povey. 57-60 [doi]
• Recognize tone languages using pitch information on the main vowel of each syllableC. Julian Chen, Haiping Li, Liqin Shen, Guokang Fu. 61-64 [doi]
• The ISL evaluation system for Verbmobil-IIHagen Soltau, Thomas Schaaf, Florian Metze, Alex Waibel. 65-68 [doi]
• Gaussian mixture selection using context-independent HMMAkinobu Lee, Tatsuya Kawahara, Kiyohiro Shikano. 69-72 [doi]
• Computing Mel-frequency cepstral coefficients on the power spectrumSirko Molau, Michael Pitz, Ralf Schlüter, Hermann Ney. 73-76 [doi]
• PLP coefficients can be quantized at 400 bpsWira Gunawan, Mark Hasegawa-Johnson. 77-80 [doi]
• Robust classification of stop consonants using auditory-based speech processingAhmed M. Abdellaty Ali, Jan Van der Spiegel, Paul Mueller. 81-84 [doi]
• Robust feature extraction using subband spectral centroid histogramsBojana Gajic, Kuldip K. Paliwal. 85-88 [doi]
• Extraction of pitch information in noisy speech using wavelet transform with aliasing compensationShi-Huang Chen, Jhing-Fa Wang. 89-92 [doi]
• A novel syllable duration modeling approach for Mandarin speechWen-Hsing Lai, Sin-Horng Chen. 93-96 [doi]
• All-pole modelling of mixed excitation signalsPeter Kabal, W. Bastiaan Kleijn. 97-100 [doi]
• On the effect of stress on certain modulation parameters of speechK. Gopalan. 101-104 [doi]
• The statistical structures of male and female speech signalsTe-Won Lee, Gil-Jin Jang. 105-108 [doi]
• Pole zero estimation from speech signals by an iterative procedureKarl Schnell, Arild Lacroix. 109-112 [doi]
• An efficient and scalable 2D DCT-based feature coding scheme for remote speech recognitionQifeng Zhu, Abeer Alwan. 113-116 [doi]
• Evaluation of mel-LPC cepstrum in a large vocabulary continuous speech recognitionHiroshi Matsumoto, Masanori Moroto. 117-120 [doi]
• Integration of fixed and multiple resolution analysis in a speech recognition systemRoberto Gemello, Dario Albesano, Loreta Moisa, Renato de Mori. 121-124 [doi]
• Perceptual harmonic cepstral coefficients for speech recognition in noisy environmentLiang Gu, Kenneth Rose. 125-128 [doi]
• Peripheral features for HMM-based speech recognitionTakashi Fukuda, Masashi Takigawa, Tsuneo Nitta. 129-132 [doi]
• Using phase spectrum information for improved speech recognition performanceRalf Schlüter, Hermann Ney. 133-136 [doi]
• A study of two dimensional linear discriminants for ASRSachin S. Kajarekar, Bayya Yegnanarayana, Hynek Hermansky. 137-140 [doi]
• Formant weighted cepstral feature for LSP-based speech recognitionHo Young Hur, Hyung Soon Kim. 141-144 [doi]
• On the use of matrix derivatives in integrated design of dynamic feature parameters for speech recognitionRathinavelu Chengalvarayan. 145-148 [doi]
• Subband feature extraction using lapped orthogonal transform for speech recognitionZekeriya Tufekci, John N. Gowdy. 149-152 [doi]
• Visual speech synthesis using quadtree splinesXue-wen Chen, Jie Yang. 153-156 [doi]
• Noise compensation in a multi-modal verification systemConrad Sanderson, Kuldip K. Paliwal. 157-160 [doi]
• Optimal weighting of posteriors for audio-visual speech recognitionMartin Heckmann, Frédéric Berthommier, Kristian Kroschel. 161-164 [doi]
• Hierarchical discriminant features for audio-visual LVCSRGerasimos Potamianos, Juergen Luettin, Chalapathy Neti. 165-168 [doi]
• Asynchronous stream modeling for large vocabulary audio-visual speech recognitionJuergen Luettin, Gerasimos Potamianos, Chalapathy Neti. 169-172 [doi]
• Weighting schemes for audio-visual fusion in speech recognitionHervé Glotin, D. Vergyr, Chalapathy Neti, Gerasimos Potamianos, Juergen Luettin. 173-176 [doi]
• Application of affine-invariant Fourier descriptors to lipreading for audio-visual speech recognitionSabri Gurbuz, Zekeriya Tufekci, Eric K. Patterson, John N. Gowdy. 177-180 [doi]
• Measuring the relation between speech acoustics and 2D facial motionAdriano Vilela Barbosa, Hani C. Yehia. 181-184 [doi]
• Microphone array sub-band speech recognitionIain McCowan, Sridha Sridharan. 185-188 [doi]
• Speech enhancement by multiple beamforming with reflection signal equalizationTakanobu Nishiura, S. Nakanura, Kiyohiro Shikano. 189-192 [doi]
• A microphone array-based 3-D N-best search algorithm for the simultaneous recognition of multiple sound sources in real environmentsPanikos Heracleous, Satoshi Nakamura, Kiyohiro Shikano. 193-196 [doi]
• Multichannel filtering for optimum noise reduction in microphone arraysDinei A. F. Florêncio, Henrique S. Malvar. 197-200 [doi]
• Microphone array speech dereverberation using coarse channel modelingScott M. Griebel, Michael S. Brandstein. 201-204 [doi]
• A multi-microphone signal subspace approach for speech enhancementFiras Jabloun, Benoît Champagne. 205-208 [doi]
• Efficient on-line acoustic environment estimation for FCDCN in a continuous speech recognition systemJasha Droppo, Alex Acero, Li Deng. 209-212 [doi]
• Environmental adaptation based on first order approximationChristophe Cerisara, Luca Rigazio, Robert Boman, Jean-Claude Junqua. 213-216 [doi]
• Hierarchical stochastic feature matching for robust speech recognitionHui Jiang 0001, Frank K. Soong, Chin-Hui Lee. 217-220 [doi]
• Model-combination-based acoustic mappingMartin Westphal, Alex Waibel. 221-224 [doi]
• Recursive estimation of time-varying environments for robust speech recognitionYunxin Zhao, Shaojun Wang, Kuan-Chieh Yen. 225-228 [doi]
• Sequential noise estimation with optimal forgetting for robust speech recognitionMohamed Afify, Olivier Siohan. 229-232 [doi]
• Robust, real-time endpoint detector with energy normalization for ASR in adverse environmentsQi Li, Jinsong Zheng, Qiru Zhou, Chin-Hui Lee. 233-236 [doi]
• Robust speech/non-speech detection using LDA applied to MFCCArnaud Martin, Delphine Charlet, Laurent Mauuary. 237-240 [doi]
• Feature enhancement for a bitstream-based front-end in wireless speech recognitionHong Kook Kim, Richard V. Cox. 241-244 [doi]
• Continuous speech recognition without end-point detectionOsamu Segawa, Kazuya Takeda, Fumitada Itakura. 245-248 [doi]
• Robust end-of-utterance detection for real-time speech recognition applicationsRamalingam Hariharan, Juha Häkkinen, Kari Laurila. 249-252 [doi]
• Multi-stream ASR trained with heterogeneous reverberant environmentsMichael L. Shire. 253-256 [doi]
• Adaptive ML-weighting in multi-band recombination of Gaussian mixture ASRAstrid Hagen, Hervé Bourlard, Andrew C. Morris. 257-260 [doi]
• Robust speech recognition in burst-like packet lossBen Milner. 261-264 [doi]
• Automatic transcription of compressed broadcast audioClaude Barras, Lori Lamel, Jean-Luc Gauvain. 265-268 [doi]
• Soft-feature decoding for speech recognition over wireless channelsAlexandros Potamianos, Vijitha Weerackody. 269-272 [doi]
• Speech in Noisy Environments: robust automatic segmentation, feature extraction, and hypothesis combinationRita Singh, Michael L. Seltzer, Bhiksha Raj, Richard M. Stern. 273-276 [doi]
• Adaptive transition bias for robust low complexity speech recognitionKonstantinos Koumpis, Søren Kamaric Riis. 277-280 [doi]
• Experiments with an extended adaptive SVD enhancement scheme for speech recognition in noiseChristian Uhl, Markus Lieb. 281-284 [doi]
• Acoustic synthesis of training data for speech recognition in living room environmentsVolker Stahl, Alexander Fischer, Rolf Bippus. 285-288 [doi]
• Maximum-likelihood compensation of zero-memory nonlinearities in speech signalsRobert W. Morris, Mark A. Clements. 289-292 [doi]
• Improved noise robustness by corrective and rival trainingCarsten Meyer, Georg Rose. 293-296 [doi]
• Continuous speech recognition under non-stationary musical environments based on speech state transition modelMasakiyo Fujimoto, Yasuo Ariki. 297-300 [doi]
• High-performance robust speech recognition using stereo training dataLi Deng, Alex Acero, Li Jiang, Jasha Droppo, Xuedong Huang. 301-304 [doi]
• SNR-dependent waveform processing for improving the robustness of ASR front-endDusan Macho, Yan Ming Cheng. 305-308 [doi]
• MVDR based feature extraction for robust speech recognitionSatya Dharanipragada, Bhaskar D. Rao. 309-312 [doi]
• Duration normalization for improved recognition of spontaneous and read speech via missing feature methodsJon P. Nedel, Richard M. Stern. 313-316 [doi]
• Eigenspace-based maximum a posteriori linear regression for rapid speaker adaptationKuan-Ting Chen, Hsin-Min Wang. 317-320 [doi]
• EMAP-based speaker adaptation with robust correlation estimationEugene Jon, Dong Kook Kim, Nam Soo Kim. 321-324 [doi]
• Linear feature space projections for speaker adaptationGeorge Saon, Geoffrey Zweig, Mukund Padmanabhan. 325-328 [doi]
• Online speaker adaptation based on quasi-Bayes linear regressionJen-Tzung Chien, Chih-Hsien Huang. 329-332 [doi]
• Rapid adaptation using penalized-likelihood methodsHakan Erdogan, Yuqing Gao, Michael Picheny. 333-336 [doi]
• Towards non-stationary model-based noise adaptation for large vocabulary speech recognitionTrausti T. Kristjansson, Brendan J. Frey, Li Deng, Alex Acero. 337-340 [doi]
• Unsupervised speaker adaptation based on sufficient HMM statistics of selected speakersShinichi Yoshizawa, Akira Baba, Kanako Matsunami, Yuichiro Mera, Miichi Yamada, Kiyohiro Shikano. 341-344 [doi]
• Rapid speaker adaptation using a priori knowledge by eigenspace analysis of MLLR parametersNick J.-C. Wang, Sammy S.-M. Lee, Frank Seide, Lin-Shan Lee. 345-348 [doi]
• Hypothesis-driven adaptation (Hydra): a flexible eigenvoice architectureS. Douglas Peters. 349-352 [doi]
• Anisotropic MAP defined by eigenvoices for large vocabulary continuous speech recognitionHenrik Botterweck. 353-356 [doi]
• Confidence-measure-driven unsupervised incremental adaptation for HMM-based speech recognitionDelphine Charlet. 357-360 [doi]
• Multiple-cluster adaptive training schemesMark J. F. Gales. 361-364 [doi]
• Neural-network-based HMM adaptation for noisy speechSadaoki Furui, Daisuke Itoh. 365-368 [doi]
• Speaker compensation with sine-log all-pass transformsJohn W. McDonough, Florian Metze, Hagen Soltau, Alex Waibel. 369-372 [doi]
• Very fast adaptation with a compact context-dependent eigenvoice modelRoland Kuhn, Florent Perronnin, Patrick Nguyen, Jean-Claude Junqua, Luca Rigazio. 373-376 [doi]
• A one-pass strategy for keyword spotting and verificationChak Shun Lai, Bertram E. Shi. 377-380 [doi]
• A support vector machines-based rejection technique for speech recognitionChangxue Ma, Mark A. Randolph, Joe Drish. 381-384 [doi]
• On combining recognizers for improved recognition of spelled namesDenis Jouvet, S. Droguet. 385-388 [doi]
• Robust confidence annotation and rejection for continuous speech recognitionBenoît Maison, Ramesh A. Gopinath. 389-392 [doi]
• Confidence measures for spoken dialogue systemsRubén San Segundo, Bryan L. Pellom, Kadri Hacioglu, Wayne Ward, José M. Pardo. 393-396 [doi]
• A comparison and combination of methods for OOV word detection and word confidence scoringTimothy J. Hazen, Issam Bazzi. 397-400 [doi]
• Comparison of different objective functions for optimal linear combination of classifiers for speaker identificationHakan Altinçay, Mübeccel Demirekler. 401-404 [doi]
• Fractal dimension applied to speaker identificationAdriano Petry, Dante Augusto Couto Barone. 405-408 [doi]
• Source and system features for speaker recognition using AANN modelsB. Yegnanarayana, K. Sharat Reddy, S. P. Kishore. 409-412 [doi]
• Speaker change detection and speaker clustering using VQ distortion for broadcast news speech recognitionKazumasa Mori, Seiichi Nakagawa. 413-416 [doi]
• A hybrid GMM/SVM approach to speaker identificationShai Fine, Jiri Navratil, Ramesh A. Gopinath. 417-420 [doi]
• Developing usable speech criteria for speaker identification technologyJereme M. Lovekin, Robert E. Yantorno, Kasturi R. Krishnamachari, Daniel S. Benincasa, Stanley J. Wenndt. 421-424 [doi]
• Learning the decision function for speaker verificationSamy Bengio, Johnny Mariéthoz. 425-428 [doi]
• Speaker indexing in large audio databases using anchor modelsDouglas E. Sturim, Douglas A. Reynolds, Elliot Singer, Joseph P. Campbell. 429-432 [doi]
• Speaker identification using Gaussian mixture models based on multi-space probability distributionChiyomi Miyajima, Yosuke Hattori, Keiichi Tokuda, Takashi Masuko, Takao Kobayashi, Tadashi Kitamura. 433-436 [doi]
• Learning statistically efficient features for speaker recognitionGil-Jin Jang, Te-Won Lee, Yung-Hwan Oh. 437-440 [doi]
• Language dependency in text-independent speaker verificationRoland Auckenthaler, Michael J. Carey 0002, John S. D. Mason. 441-444 [doi]
• Language-independent, short-enrollment voice verification over a far-field microphoneJerome R. Bellegarda, Devang Naik, Matthias Neeracher, Kim E. A. Silverman. 445-448 [doi]
• Password-dependent speaker verification using quantized acoustic trajectoriesLuc Gagnon, Peter Stubley, Ghislain Mailhot. 449-452 [doi]
• A combination between VQ and covariance matrices for speaker recognitionMarcos Faúndez-Zanuy. 453-456 [doi]
• Text-dependent speaker verification under noisy conditions using parallel model combinationLit Ping Wong, Martin Russell. 457-460 [doi]
• Very large population text-independent speaker identification using transformation enhanced multi-grained modelsUpendra V. Chaudhari, Jiri Navratil, Ganesh N. Ramaswamy, Stéphane H. Maes. 461-464 [doi]
• An EKF-based algorithm for learning statistical hidden dynamic model parameters for phonetic recognitionRoberto Togneri, Li Deng. 465-468 [doi]
• Continuous speech recognition using a hierarchical Bayesian modelFériel Mouria-beji. 469-472 [doi]
• Indicator variable dependent output probability modelling via continuous posterior functionsAndreas Tuerk, S. J. Young. 473-476 [doi]
• Investigating lightly supervised acoustic model trainingLori Lamel, Jean-Luc Gauvain, Gilles Adda. 477-480 [doi]
• Multiple linear transformsNagendra K. Goel, Ramesh A. Gopinath. 481-484 [doi]
• Relax frame independence assumption for standard hmms by state dependent auto-regressive feature modelsYing Jia, Jinyu Li. 485-488 [doi]
• Speech recognition for DARPA CommunicatorA. Aaron, S. Chen, P. Cohen, Satya Dharanipragada, Ellen Eide, M. Franz, J. M. LeRoux, X. Luo, Benoît Maison, Lidia Mangu, T. Mathes, M. Novak, Peder A. Olsen, Michael Picheny, Harry Printz, Bhuvana Ramabhadran, Andrej Sakrajda, George Saon, Borivoj Tydlitát, Karthik Visweswariah, D. Yuk. 489-492 [doi]
• Efficient mixture Gaussian synthesis for decision tree based state tyingTsuneo Kato, Shingo Kuroiwa, Tohru Shimizu, Norio Higuchi. 493-496 [doi]
• Discriminative training of HMM using maximum normalized likelihood algorithmKonstantin Markov, Seiichi Nakagawa, Satoshi Nakamura. 497-500 [doi]
• Modeling uncertainty of data observationAndreas Wendemuth. 501-504 [doi]
• Modular neural networks exploit large acoustic context through broad-class posteriors for continuous speech recognitionChristos Antoniou. 505-508 [doi]
• Multiple mixture segmental HMM and its applicationsBing Xiang, Toby Berger. 509-512 [doi]
• Multiple-regression hidden Markov modelKatsuhisa Fujinaga, Mitsuru Nakai, Hiroshi Shimodaira, Shigeki Sagayama. 513-516 [doi]
• Tandem acoustic modeling in large-vocabulary recognitionDaniel P. W. Ellis, Rita Singh, Sunil Sivadas. 517-520 [doi]
• Towards task-independent speech recognitionFabrice Lefevre, Jean-Luc Gauvain, Lori Lamel. 521-524 [doi]
• Nonlinear dynamical system based acoustic modeling for ASRNarada D. Warakagoda, Magne Hallstein Johnsen. 525-528 [doi]
• Composite background models and score standardization for language identification systemsTerry P. Gleason, Marc A. Zissman. 529-532 [doi]
• Improving trigram language modeling with the World Wide WebXiaojin Zhu, Ronald Rosenfeld. 533-536 [doi]
• Dialog-context dependent language modeling combining n-grams and stochastic context-free grammarsKadri Hacioglu, Wayne Ward. 537-540 [doi]
• Use of non-negative matrix factorization for language model adaptation in a lecture transcription taskMiroslav Novak, Richard J. Mammone. 541-544 [doi]
• Efficient class-based language modelling for very large vocabulariesEdward W. D. Whittaker, Philip C. Woodland. 545-548 [doi]
• Data augmentation and language model adaptationDavid Janiszek, Renato de Mori, Frédéric Béchet. 549-552 [doi]
• Using semantic class information for rapid development of language models within ASR dialogue systemsEric Fosler-Lussier, Hong-Kwang Jeff Kuo. 553-556 [doi]
• On-line learning of language models with word error probability distributionsRoberto Gretter, Giuseppe Riccardi. 557-560 [doi]
• Classes for fast maximum entropy trainingJoshua Goodman. 561-564 [doi]
• Automatic generation and selection of multiple pronunciations for dynamic vocabulariesSabine Deligne, Benoît Maison, Ramesh A. Gopinath. 565-568 [doi]
• Automatic generation of pronunciation lexicons for Mandarin spontaneous speechWilliam Byrne, Veera Venkataramani, Terri Kamm, Thomas Fang Zheng, Zhanjiang Song, Pascale Fung, Yi Liu, Umar Ruhi. 569-572 [doi]
• Sub-lexical modelling using a finite state transducer frameworkXiaolong Mou, Victor Zue. 573-576 [doi]
• What kind of pronunciation variation is hard for triphones to model?Dan Jurafsky, Wayne Ward, Zhang Banping, Keith Herold, Xiuyang Yu, Zhang Sen. 577-580 [doi]
• Bootstrap method for Chinese new words extractionShan He, Jie Zhu. 581-584 [doi]
• Kanji-to-hiragana conversion based on a language modelWei-Bin Chang. 585-588 [doi]
• A dynamic semantic model for re-scoring recognition hypothesesCarmen Wai, Roberto Pieraccini, Helen M. Meng. 589-592 [doi]
• A matching algorithm between arbitrary sections of two speech data sets for speech retrievalYoshiaki Itoh. 593-596 [doi]
• Advances in automatic meeting record creation and accessAlex Waibel, Michael Bett, Florian Metze, Klaus Ries, Thomas Schaaf, Tanja Schultz, Hagen Soltau, Hua Yu, Klaus Zechner. 597-600 [doi]
• Experiments on speech tracking in audio documents using Gaussian mixture modelingMouhamadou Seck, Ivan Magrin-Chagnolleau, Frédéric Bimbot. 601-604 [doi]
• Multi-scale-audio indexing for translingual spoken document retrievalHsin-Min Wang, Helen M. Meng, Patrick Schone, Berlin Chen, Wai Kit Lo. 605-608 [doi]
• Simplifying design specification for automatic training of robust natural language call routerHong-Kwang Jeff Kuo, Chin-Hui Lee. 609-612 [doi]
• Speech-to-speech translation based on finite-state transducersFrancisco Casacuberta, David Llorens, Carlos Martinez, Sirko Molau, Francisco Nevado, Hermann Ney, Moisés Pastor, David Picó, Alberto Sanchís, Enrique Vidal, Juan Miguel Vilar. 613-616 [doi]
• Topic Forest: a plan-based dialog management structureXiaojun Wu, Fang Zheng, Mingxing Xu. 617-620 [doi]
• Speech enhancement using the sparse code shrinkage techniqueIlyas Potamitis, Nikos Fakotakis, George Kokkinakis. 621-624 [doi]
• STFT-based multi-channel acoustic interference suppressorCarlos Avendaño, Guillermo Garcia. 625-628 [doi]
• Estimation of speech embedded in a reverberant environment with multiple sources of noiseAllan Kardec Barros, Fumitada Itakura, Tomasz M. Rutkowski, Ali Mansour, Noboru Ohnishi. 629-632 [doi]
• Experimental investigation of delayed instantaneous demixer for speech enhancementYong Xiang, Yingbo Hua, Senjian An, Alex Acero. 633-636 [doi]
• Lattice-ladder decorrelation filters developed for co-channel speech separationKuan-Chieh Yen, Yunxin Zhao. 637-640 [doi]
• Single channel speech enhancement using MDL-based subspace approach in Bark domainRolf Vetter. 641-644 [doi]
• Codebook constrained iterative noise cancellation with applications to speech enhancementYan Gao, Jing Lu, Kai Yu, Bo-Ling Xu. 645-648 [doi]
• Use of local kurtosis measure for spotting usable speech segments in co-channel speechKasturi Rangan Krishnamachari, Robert E. Yantorno, Jereme M. Lovekin, Daniel S. Benincasa, Stanley J. Wenndt. 649-652 [doi]
• Recursively updated eigenfilterbank for speech enhancementMorten Jeppesen, Christoffer Asgaard Rødbro, Søren Holdt Jensen. 653-656 [doi]
• Statistical speech reconstruction at the phoneme levelMichael I. Savic, Michael D. Moore, Christopher Scoville. 657-660 [doi]
• On speech enhancement under signal presence uncertaintyIsrael Cohen. 661-664 [doi]
• Speech enhancement via frequency bandwidth extension using line spectral frequenciesSamir Chennoukh, Andreas Johannes Gerrits, G. Miet, Robert Johannes Sluijter. 665-668 [doi]
• Estimation of the excitation variances of speech and noise AR-models for enhanced speech codingMarcin Kuropatwinski, W. Bastiaan Kleijn. 669-672 [doi]
• A cross-correlation technique for enhancing speech corrupted with correlated noiseYi Hu, Mukul Bhatnagar, Philipos C. Loizou. 673-676 [doi]
• A 1200 bps speech coder with LSF matrix quantizationSelma Özaydin, Buyurman Baykal. 677-680 [doi]
• A candidate for the ITU-T 4 kbit/s speech coding standardJes Thyssen, Yang Gao, Adil Benyassine, Eyal Shlomot, Carlo Murgia, Huan-yu Su, Kazunori Mano, Yusuke Hiwasaki, Hiroyuki Ehara, Kazutoshi Yasunaga, Claude Lamblin, Balász Kövesi, Joachim Stegmann, Hong-Goo Kang. 681-684 [doi]
• A hybrid coder based on a new phase model for synchronization between harmonic and waveform coded segmentsNilantha Katugampala, Ahmet M. Kondoz. 685-688 [doi]
• eX-CELP: a speech coding paradigmYang Gao, Adil Benyassine, Jes Thyssen, Huan-yu Su, Eyal Shlomot. 689-692 [doi]
• Maximum-take-precedence ACELP: a low complexity search methodFu-Kun Chen, Jar-Ferr Yang. 693-696 [doi]
• SEW representation for low rate WI codingJason Lukasiak, Ian S. Burnett. 697-700 [doi]
• Spectral magnitude quantization based on linear transforms for 4 kb/s speech codingCagri Özgenc Etemoglu, Vladimir Cuperman. 701-704 [doi]
• Speech LSF quantization with rate independent complexity, bit scalability and learningAnand D. Subramaniam, Bhaskar D. Rao. 705-708 [doi]
• The SMV algorithm selected by TIA and 3GPP2 for CDMA applicationsYang Gao, Eyal Shlomot, Adil Benyassine, Jes Thyssen, Huan-yu Su, Carlo Murgia. 709-712 [doi]
• Universal successive refinement of CELP speech codersHui Dong, Jerry D. Gibson. 713-716 [doi]
• A speech spectrum distortion measure with interframe memoryFredrik Nordin, Thomas Eriksson. 717-720 [doi]
• Approximating and exploiting the residual redundancies-applications to efficient reconstruction of speech over noisy channelsFarshad Lahouti, Amir K. Khandani. 721-724 [doi]
• Channel optimized matrix quantization (COMQ) of LSP parameters over waveform channelsJosé L. Pérez-Córdoba, Antonio J. Rubio, Juan M. López-Soler, Victoria E. Sánchez. 725-728 [doi]
• Estimation of missing LSF parameters using Gaussian mixture modelsRainer Martin, Carsten Hoelper, Ingo Wittke. 729-732 [doi]
• Hybrid multi-mode/multi-rate CS-ACELP speech coding for adaptive voice over IPGiuseppe Ruggeri, Francesco Beritelli, Salvatore Casale. 733-736 [doi]
• Improved voice activity detection based on a smoothed statistical likelihood ratioYong Duk Cho, Khaldoon Al-Naimi, Ahmet M. Kondoz. 737-740 [doi]
• Multiplexed predictive coding of speechSøren Vang Andersen, Gernot Kubin. 741-744 [doi]
• Parameter interpolation to enhance the frame erasure robustness of CELP coders in packet networksJian Wang, Jerry D. Gibson. 745-748 [doi]
• Perceptual evaluation of speech quality (PESQ)-a new method for speech quality assessment of telephone networks and codecsAntony W. Rix, John G. Beerends, Michael P. Hollier, Andries P. Hekstra. 749-752 [doi]
• Source-driven packet marking for speech transmission over differentiated-services networksJuan Carlos De Martin. 753-756 [doi]
• A candidate proposal for a 3GPP adaptive multi-rate wideband speech codecChristoph Erdmann, Peter Vary, Kyrill Fischer, Wen Xu, Matthias Marke, Tim Fingscheidt, Imre Varga, Markus Kaindl, Catherine Quinquis, Balázs Kövesi, Dominique Massaloux. 757-760 [doi]
• An embedded adaptive multi-rate wideband speech coderAlan McCree, Takahiro Unno, Anand K. Anandakumar, Alexis Bernard, Erdal Paksoy. 761-764 [doi]
• Optimal estimation of subband speech from nonuniform non-recurrent signal-driven sparse samplesPenio S. Penev, Liubomire G. Iordanov. 765-768 [doi]
• Variable-size vector entropy coding of speech and audioYair Shoham. 769-772 [doi]
• Wideband speech and audio coding using gammatone filter banksEliathamby Ambikairajah, Julien Epps, Lee Lin. 773-776 [doi]
• Frequency selectivity via the SpEnt methodology for wideband speech compressionMark G. Kokes, Jerry D. Gibson. 777-780 [doi]
• Joint prosody prediction and unit selection for concatenative speech synthesisIvan Bulyko, Mari Ostendorf. 781-784 [doi]
• Selecting non-uniform units from a very large corpus for concatenative speech synthesizerMin Chu, Hu Peng, Hong-yun Yang, Eric Chang. 785-788 [doi]
• Quantitative method for modeling context in concatenative synthesis using large speech databaseWael Hamza, Mohsen Rashwan, Mohamed Afify. 789-792 [doi]
• Duration modeling in a restricted-domain female-voice synthesis in Spanish using neural networksRicardo de Córdoba, Juan Manuel Montero, Juana M. Gutiérrez-Arriola, José Manuel Pardo. 793-796 [doi]
• A functional articulatory dynamic model for speech productionLeo J. Lee, Paul W. Fieguth, Li Deng. 797-800 [doi]
• The effect of language model probability on pronunciation reductionDaniel Jurafsky, Alan Bell, Michelle Gregory, William D. Raymond. 801-804 [doi]
• Adaptation of pitch and spectrum for HMM-based speech synthesis using MLLRMasatsune Tamura, Takashi Masuko, Keiichi Tokuda, Takao Kobayashi. 805-808 [doi]
• An RNN-based algorithm to detect prosodic phrase for Chinese TTSZhiwei Ying, Xiaohua Shi. 809-812 [doi]
• Design and evaluation of a voice conversion algorithm based on spectral envelope mapping and residual predictionAlexander Kain, Michael W. Macon. 813-816 [doi]
• 0 contours using a model-constrained data-driven methodAtsuhiro Sakurai, Keikichi Hirose, Nobuaki Minematsu. 817-820 [doi]
• 0 model to a text-to-speech system in Castillian SpanishJuana M. Gutiérrez-Arriola, Juan Manuel Montero, D. Saiz, José Manuel Pardo. 821-824 [doi]
• Segmenting unrestricted Chinese text into prosodic words instead of lexical wordsYao Qian, Min Chu, Hu Peng. 825-828 [doi]
• Speech synthesis using stochastic Markov graphsMatthias Eichner, Matthias Wolff, Sebastian Ohnewald, Rüdiger Hoffmann. 829-832 [doi]
• Trainable speech synthesis with trended hidden Markov modelsJohn Dines, Sridha Sridharan. 833-836 [doi]
• Perceptual and objective detection of discontinuities in concatenative speech synthesisYannis Stylianou, Ann K. Syrdal. 837-840 [doi]
• Voice conversion algorithm based on Gaussian mixture model with dynamic frequency warping of STRAIGHT spectrumTomoki Toda, Hiroshi Saruwatari, Kiyohiro Shikano. 841-844 [doi]
• Signal modeling with non-uniform topology lattice filtersSacha Krstulovic, Frédéric Bimbot. 845-848 [doi]
• Experiments and modeling of perceived speech quality of long samplesXiang-Chun Tan, Stefan Wänstedt, Gunnar Heikkilä. 849-852 [doi]
• Suppression of phasiness for time-scale modifications of speech signals based on a shape invariance propertyJoseph di Martino, Yves Laprie. 853-856 [doi]
• Mixture Gaussian envelope chirp model for speech and audioBishwarup Mondal, Thippur V. Sreenivas. 857-860 [doi]
• Glottal flow derivative modeling with the wavelet smoothed excitationArthur P. Lobo. 861-864 [doi]
• Robust singing detection in speech/music discriminator designWu Chou, Liang Gu. 865-868 [doi]
• Avoiding over-estimation in bandwidth extension of telephony speechMattias Nilsson, W. Bastiaan Kleijn. 869-872 [doi]
• Acoustic-phonetic characteristics of hyperarticulated speech for different speaking stylesStefanie Köster. 873-876 [doi]
• A robust technique for harmonic analysis of speechNazih Abu-Shikhah, Mohamed Deriche. 877-880 [doi]
• Implementation of fixed DSP functions using the reduced coefficient multiplierRichard H. Turner, Tim Courtney, Roger Woods. 881-884 [doi]
• Gigaop DSP on FPGABrad L. Hutchings, Brent E. Nelson. 885-888 [doi]
• Maximum likelihood carrier phase synchronization in FPGA-based software defined radiosMichael Rice, Chris Dick, Fred Harris. 889-892 [doi]
• Reconfigurable platform design for wireless protocol processorsTim Tuan, Suet-Fei Li, Jan M. Rabaey. 893-896 [doi]
• Parameterised floating-point arithmetic on FPGAsAllan Jaenicke, Wayne Luk. 897-900 [doi]
• Dynamically parameterized algorithms and architectures to exploit signal variations for improved performance and reduced powerWayne Burleson, Russell Tessier, Dennis Goeckel, Sriram Swaminathan, Prashant Jain, Jeongseon Euh, Subramanian Venkatraman, Vidhya Thyagarajan. 901-904 [doi]
• Energy efficient system partitioning for distributed wireless sensor networksAlice Wang, Anantha Chandrakasan. 905-908 [doi]
• Decision feedback equalization for Bluetooth systemsMohammed Nafie, Alan Gatherer, Anand Dabak. 909-912 [doi]
• A DSP-based DS/CDMA modem for multimedia applications over geo-stationary satellite networksClaudio Sacchi, Luca Simone Ronga. 913-916 [doi]
• Nonlinear echo cancellation using decoupled A-B net structureGuillermo B. Sentoni, Juan E. Cousseau, Paulo S. R. Diniz. 917-920 [doi]
• A stable and efficient DSP implementation of a LSL algorithm for acoustic echo cancellingAndré H. C. Carezia, Phillip M. S. Burt, Max Gerken, Maria D. Miranda, Magno T. M. Silva. 921-924 [doi]
• Frequency estimation for sensorless control of induction motorsMickaël Hilairet, François Auger. 925-928 [doi]
• Design and implementation of a mixed-signal embedded DSP systemMark B. Yeary, David W. Guidry, Mark Burns. 929-932 [doi]
• Vector processing in scalar processors for signal processing algorithmsMichael T. Brady, Jennifer Q. Trelewicz, Joan L. Mitchell. 933-936 [doi]
• A programmable processor with 4096 processing units for media applicationsArgy Krikelis, Ian P. Jalowiecki, David L. Bean, Ryan Bishop, M. Facey, D. Boughton, S. Murphy, M. Whitaker. 937-940 [doi]
• A single-chip MPEG-2 MP@ML audio/video encoder/decoder with a programmable video interface unitC. T. Chen, T.-C. Chen, F.-C. Jeng, H. Cheng, Konstantinos Konstantinides. 941-944 [doi]
• Efficient implementation of video post-processing algorithms on the BOPS parallel architectureDoina Petrescu. 945-948 [doi]
• VSIPL: an object-based open standard API for vector, signal, and image processingRandall Janka, Randall Judd, James Lebak, Mark Richards, Dan Campbell. 949-952 [doi]
• Standardization of the selectable mode vocoderS. Craig Greer, Andrew DeJaco. 953-956 [doi]
• 2.4 kb/sec compressed domain teleconference bridge with universal transcoderRichard L. Zinser, Philip T. Choong, Steven R. Koch. 957-960 [doi]
• On integrating acoustic echo and noise cancellation systems for hands-free telephonySeon Joon Park, Chum Gun Cho, Chungyong Lee, Dae Hee Youn. 961-964 [doi]
• A new W3C markup standard for text-to-speech synthesisMark R. Walker, Jim Larson, Andrew Hunt. 965-968 [doi]
• Spoken word recognition with digital cochlea using 32 DSP-boardsMasao Namiki, Takayuki Hamamoto, Seiichiro Hangai. 969-972 [doi]
• Force XXI Land Warrior: a systems approach to speech recognitionCharles C. Broun, William M. Campbell. 973-976 [doi]
• Multiresolution MRF-based texture segmentation using the Wreath Product Transform phaseGagan Mirchandani, Xuling Luo. 977-980 [doi]
• Logo insertion in MPEG transcoderKrit Panusopone, Xuemin Chen, Fan Ling. 981-984 [doi]
• High-quality and processor-efficient implementation of an MPEG-2 AAC encoderYuichiro Takamizawa, Toshiyuki Nomura, Masao Ikekawa. 985-988 [doi]
• Design optimization of main-profile MPEG-2 AAC decoderKyoung Ho Bang, Nam Hun Jeong, Young-Cheol Park, Dae Hee Youn. 989-992 [doi]
• On-line signature verification algorithm incorporating pen position, pen pressure and pen inclination trajectoriesD. Sakamoto, Hikaru Morita, Tetsu Ohishi, Yoshimitsu Komiya, Takashi Matsumoto. 993-996 [doi]
• Chaotic algorithms for data encryptionMohamed I. Sobhy, Alaa-eldin R. Shehata. 997-1000 [doi]
• Methods of attacking chaotic encryption and countermeasuresMohamed I. Sobhy, Alaa-eldin R. Shehata. 1001-1004 [doi]
• DSP application in e-commerce securityJiankun Hu, Ziping Xi, Andrew Jennings, H. Y. J. Lee, D. Wahyudi. 1005-1008 [doi]
• Open multimedia application platform: enabling multimedia applications in third generation wireless terminals through a combined RISC/DSP architectureJamil Chaoui, Ken Cyr, Sébastien de Gregorio, Jean Pierre Giacalone, Jennifer Webb, Yves Masse. 1009-1012 [doi]
• A 333-MHz dual-MAC DSP architecture for next-generation wireless applicationsRavi K. Kolagotla, José Fridman, Marc M. Hoffman, William C. Anderson, Bradley C. Aldrich, David B. Witt, Michael S. Allen, Randy R. Dunton, Lawrence A. Booth Jr.. 1013-1016 [doi]
• A low-power programmable DSP core architecture for 3G mobile terminalsTakahiro Kumura, Daiji Ishii, Masao Ikekawa, Ichiro Kuroda, Makoto Yoshida. 1017-1020 [doi]
• A programmable co-porcessor for MPEG-4 videoMladen Berekovic, Hans-Joachim Stolberg, Peter Pirsch, Holger Runge. 1021-1024 [doi]
• Estimation of the velocity of mobile units in microcellular systems using the instantaneous frequency of the received signalsGhasem Azemi, Bouchra Senadji, Boualem Boashash. 1025-1028 [doi]
• Transmit beamforming combined with diversity techniques for cdma2000 systemsRobert A. Soni, R. Michael Buehrer, Roger D. Benning. 1029-1032 [doi]
• A new false positive reduction method for MCCs detection in digital mammographyL. Zhang, W. Qian, R. Sankar, D. Song, R. Clark. 1033-1036 [doi]
• Wavelet analysis of the timing of individual components of the ECG signalM. J. Burke, M. Nasor. 1037-1040 [doi]
• Newborn EEG seizure pattern characterisation using time-frequency analysisBoualem Boashash, Mostefa Mesbah, Paul B. Colditz. 1041-1044 [doi]
• Sequential methods for DNA sequencingNicholas M. Haan, Simon J. Godsill. 1045-1048 [doi]
• Detection and estimation of embolic Doppler signals using discrete wavelet transformNizamettin Aydin, Hugh S. Markus, Farokh Marvasti. 1049-1052 [doi]
• DSP in genomics: processing and frequency-domain analysis of character stringsDimitris Anastassiou. 1053-1056 [doi]
• A new algorithm for EEG feature selection using mutual informationMohamed Deriche, Ahmed Al-Ani. 1057-1060 [doi]
• On-line EEG classification and sleep spindles detection using an adaptive recursive bandpass filterReda R. Gharieb, Andrzej Cichocki. 1061-1064 [doi]
• Cost-efficient multiplier-less FIR filter structure based on modified DECOR transformationI.-Hsien Lee, Cheng-Shing Wu, An-Yeu Wu. 1065-1068 [doi]
• Improving the near-perfect hybrid filter bank performance in the presence of realization errorsMarcos Aurélio de Andrade Pinheiro, Paulo B. Batalheiro, Antonio Petraglia, Mariane R. Petraglia. 1069-1072 [doi]
• Parallelizable eigenvalue decomposition techniques via the matrix sector functionMohammed A. Hasan, Ali A. Hasan. 1073-1076 [doi]
• Rational and radical fixed point functions for the eigenvalue problem and polynomialsJawad A. K. Hasan, Mohammed A. Hasan. 1077-1080 [doi]
• Pipelined architectures for the TD-LMS adaptive filterGeorge-Othon Glentis. 1081-1084 [doi]
• Canonic look ahead: critical cycle relaxed IIR filtering with minimum multiplicative complexityJ. Rubio, J. Sala, F. Nunez. 1085-1088 [doi]
• A digital chip for robust speech recognition in noisy environmentChang-Min Kim, Soo-Young Lee. 1089-1092 [doi]
• FPGA implementation of a tunable band-pass filter using the "basic heterodyne block"Dhinesh Sasidaran, Asad Azam, Karl E. Nelson, Michael A. Soderstrand. 1093-1096 [doi]
• Generic scheduling methods for a linear QR array SoC processorZhaohui Liu, Gaye Lightbody, Richard L. Walke, Y. Hu, John V. McCanny. 1097-1100 [doi]
• Efficient implementation of a set of lifting based wavelet filtersKishore Andra, Chaitali Chakrabarti, Tinku Acharya. 1101-1104 [doi]
• An FPGA implementation of Walsh-Hadamard transforms for signal processingAbbes Amira, Ahmed Bouridane, Peter Milligan, Mohammed Ali Roula. 1105-1108 [doi]
• Architecture independent short vector FFTsFranz Franchetti, Herbert Karner, Stefan Kral, Christoph W. Ueberhuber. 1109-1112 [doi]
• Rapid prototyping of multi-DSP systems based on accurate performance estimationBernhard Rinner, Bernd Ruprechter, Martin Schmid. 1113-1116 [doi]
• A block priority based instruction caching scheme for multimedia processorsJiyang Kang, Wonyong Sung. 1117-1120 [doi]
• Variable partitioning for dual memory bank DSPsRainer Leupers, Daniel Kotte. 1121-1124 [doi]
• Low power showdown: comparison of five DSP platforms implementing an LPC speech codecDavid Hwang, Cimarron Mittelsteadt, Ingrid Verbauwhede. 1125-1128 [doi]
• Performance analysis of low bit rate H.26L video encoderAntti Hallapuro, Ville Lappalainen, Timo D. Hämäläinen. 1129-1132 [doi]
• Integer code generation for the TI TMS320C62XMartin Coors, Holger Keding, Olaf Lüthje, Heinrich Meyr. 1133-1136 [doi]
• A survey on modeling issues using the machine description language LISAAndreas Hoffmann, Achim Nohl, Gunnar Braun, Heinrich Meyr. 1137-1140 [doi]
• Discrete polynomial transform representation using binary matrices and flow diagramsMaurice F. Aburdene, Richard J. Kozick, Ryan S. Magargle, Justin D. Maloney-Hahn, Christian M. Coviello. 1141-1144 [doi]
• CODEF: a system level design space exploration toolMichel Auguin, Laurent Capella, Fernand Cuesta, Emmanuel Gresset. 1145-1148 [doi]
• Data and instruction memory exploration of embedded systems for multimedia applicationsM. M. Dasigenis, Nikolas Kroupis, Antonios Argyriou, Konstantinos Tatas, Dimitrios Soudris, Nikolaos D. Zervas. 1149-1152 [doi]
• An efficient timing model for hardware implementation of multirate dataflow graphsNitin Chandrachoodan, Shuvra S. Bhattacharyya, K. J. Ray Liu. 1153-1156 [doi]
• Implementing parallelism and scheduling data flow graphs on Java virtual machineJin Xu, Edwin Hsing-Mean Sha. 1157-1160 [doi]
• Analytical exploration of power efficient data-reuse transformations on multimedia applicationsStamatiki Kougia, Alexander Chatzigeorgiou, Nikolaos D. Zervas, Spiridon Nikolaidis. 1161-1164 [doi]
• DSP data path synthesis for low-power applicationsLih-Yih Chiou, Khurram Muhammad, Kaushik Roy. 1165-1168 [doi]
• Power efficient semi-automatic instruction encoding for application specific instruction set processorsTilman Glökler, Stefan Bitterlich. 1169-1172 [doi]
• Digit-serial modular multiplication using skew-tolerant domino CMOSSungwook Kim, Gerald E. Sobelman. 1173-1176 [doi]
• Multi-port interconnection networks for radix-R algorithmsJarmo Takala, Tuomas Järvinen, Perttu Salmela, David Akopian. 1177-1180 [doi]
• Integer sinusoidal transforms based on lifting factorizationYonghong Zeng, Guoan Bi, Zhiping Lin. 1181-1184 [doi]
• Partial-result-reuse architecture and its design technique for morphological operationsShao-Yi Chien, Shyh-Yih Ma, Liang-Gee Chen. 1185-1188 [doi]
• An integrated systolic array design for video compressionPol-Lin Tai, Chii-Tung Liu, Shih-Yu Huang, Jia-Shung Wang. 1189-1192 [doi]
• Design of RNS-based distributed arithmetic DWT filterbanksJavier Ramírez, Antonio García, Uwe Meyer-Bäse, Fred J. Taylor, Pedro G. Fernández, Antonio Lloris-Ruíz. 1193-1196 [doi]
• Quantization effect on VLSI implementations for the 9/7 DWT filtersVassilis Spiliotopoulos, Nikolaos D. Zervas, Yiannis Andreopoulos, Giorgos P. Anagnostopoulos, Costas E. Goutis. 1197-1200 [doi]
• A fast motion estimation algorithm equivalent to exhaustive searchMohammad Gharavi-Alkhansari. 1201-1204 [doi]
• Cache conscious Walsh-Hadamard transformNeungsoo Park, Viktor K. Prasanna. 1205-1208 [doi]
• An approach for enabling DCT/IDCT energy reduction scalability in MPEG-2 video codecsRussell Henning, Chaitali Chakrabarti. 1209-1212 [doi]
• A DHT-based FFT/IFFT processor for VDSL transceiversChin-Liang Wang, Ching-Hsien Chang. 1213-1216 [doi]
• A FPGA-based Viterbi algorithm implementation for speech recognition systemsFabian Luis Vargas, Ruben Dutra R. Fagundes, D. Barros Júnior. 1217-1220 [doi]
• Configurable hardware implementation of triple-DES encryption algorithm for wireless local area networkPanu Hämäläinen, Marko Hännikäinen, Timo D. Hämäläinen, Jukka Saarinen. 1221-1224 [doi]
• CORDIC realization of the transversal adaptive filter using a trigonometric LMS algorithmMrityunjoy Chakraborty, Anindya Sundar Dhar, Suraiya Pervin. 1225-1228 [doi]
• A novel trellis-based searching scheme for EEAS-based CORDIC algorithmCheng-Shing Wu, An-Yeu Wu. 1229-1232 [doi]
• A unified design framework for vector rotational CORDIC family based on angle quantization processAn-Yeu Wu, Cheng-Shing Wu. 1233-1236 [doi]
• High-speed CORDIC implementations using advanced circuit techniquesGunok Jung, Seonki Kim, Gerald E. Sobelman. 1237-1240 [doi]
• Time-shared TMR for fault-tolerant CORDIC processorsJae-Hyuck Kwak, Vincenzo Piuri, Earl E. Swartzlander Jr.. 1241-1244 [doi]
• Decision feedback equalizer with two's complement computation sharing multiplicationHunsoo Choo, Khurram Muhammad, Kaushik Roy. 1245-1248 [doi]
• DSP-based signal processing for OFDM transmissionMatthias Schöbinger, Stefan R. Meier. 1249-1252 [doi]
• MATLAB based codesign framework for wireless broadband communication DSPsVolker Aue, Johannes Kneip, Matthias Weiss, Michael Bolle, Gerhard Fettweis. 1253-1256 [doi]
• Impulses and stochastic arithmetic for signal processingJohn F. Keane, Les E. Atlas. 1257-1260 [doi]
• Fuzzy anisotropic diffusion for speckle filteringSantiago Aja-Fernández, Carlos Alberola, Juan Ruiz. 1261-1264 [doi]
• A new optimizing procedure for ν-support vector regressorFernando Pérez-Cruz, Antonio Artés-Rodriguez. 1265-1268 [doi]
• Dual ν-support vector machine with error rate and training size biasingHong-Gunn Chew, Robert E. Bogner, Cheng-Chew Lim. 1269-1272 [doi]
• Fast principal component extraction by a homogeneous neural networkShan Ouyang, Zheng Bao. 1273-1276 [doi]
• A new feedforward neural network hidden layer neuron pruning algorithmFarhat Fnaiech, Nader Fnaiech, Mohamed Najim. 1277-1280 [doi]
• Complex backpropagation neural network using elementary transcendental activation functionsTaehwan Kim, Tülay Adali. 1281-1284 [doi]
• Optimized neural networks for modeling of loudspeaker directivity diagramsEva Wilk, Jan Wilk. 1285-1288 [doi]
• Subbands audio signal recovering using neural nonlinear predictionGianandrea Cocchi, Aurelio Uncini. 1289-1292 [doi]
• Minimal resource allocation network (MRAN) for magnetic recording channel equalizationDeng Jianping, Narasimhan Sundararajan, Paramasivan Saratchandran. 1293-1296 [doi]
• A recurrent RBF network for non-linear channelMasayoshi Mimura, Toshihiro Furukawa. 1297-1300 [doi]
• Comparison of neural network natural and ordinary gradient algorithms for satellite down link identificationFabien Langlet, Hasan Abdulkader, Daniel Roviras, L. Lapierre, Francis Castanie. 1301-1304 [doi]
• Fuzzy recursive symbol-by-symbol detector for single user CDMA receiversJoan Bas, Ana I. Pérez-Neira, Miguel Angel Lagunas. 1305-1308 [doi]
• Automatic classification of QAM signals by neural networksShintaro Taira. 1309-1312 [doi]
• On-line order selection for communicationsHongmei Ni, Tülay Adali. 1313-1316 [doi]
• Optimization in companion search spaces: the case of cross-entropy and the Levenberg-Marquardt algorithmCraig L. Fancourt, Jose C. Principe. 1317-1320 [doi]
• On learning and computational complexity of FIR radial basis function networks. Part I. Learning of FIR RBFN'sKayvan Najarian. 1321-1324 [doi]
• On learning and computational complexity of FIR radial basis function networks. Part II. Complexity measuresKayvan Najarian. 1325-1328 [doi]
• Learning topographic representation for multi-view image patternsStan Z. Li, XiaoGuang Lv, HongJiang Zhang, QingDong Fu, Yimin Cheng. 1329-1332 [doi]
• Classification by probabilistic clusteringThomas M. Breuel. 1333-1336 [doi]
• Spread spectrum signaling for speech watermarkingQiang Cheng, Jeffrey Sorensen. 1337-1340 [doi]
• New echo embedding technique for robust and imperceptible audio watermarkingHyen O. Oh, Jong Won Seok, Jin Woo Hong, Dae Hee Youn. 1341-1344 [doi]
• Robust spread-spectrum audio watermarkingDarko Kirovski, Henrique S. Malvar. 1345-1348 [doi]
• Next generation techniques for robust and imperceptible audio data hidingJim Chou, Kannan Ramchandran, Antonio Ortega. 1349-1352 [doi]
• Audio watermarking by time-scale modificationMohamed F. Mansour, Ahmed H. Tewfik. 1353-1356 [doi]
• Robust algorithm for watermark recovery from cropped speechAparna Gurijala, John R. Deller Jr.. 1357-1360 [doi]
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• MP3 resistant oblivious steganographyLitao Gang, Ali N. Akansu, Mahalingam Ramkumar. 1365-1368 [doi]
• Analysis of attacks on SDMI audio watermarksMin Wu 0001, Scott Craver, Edward W. Felten, Bede Liu. 1369-1372 [doi]
• Study on the application of an AMR speech codec to VoIPJeong-Wook Seo, Se Jeong Woo, Keun-Sung Bae. 1373-1376 [doi]
• Joint source-channel coding for scalable video using models of rate-distortion functionsLisimachos P. Kondi, Aggelos K. Katsaggelos. 1377-1380 [doi]
• Accurate estimation of R-D characteristics for rate control in real-time video encodingJunfeng Bai, Chang Feng, Qingmin Liao, Xinggang Lin, Xinhua Zhuang. 1381-1384 [doi]
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• Motion adaptive error resilient encoding for MPEG-4Stewart T. Worrall, Abdul H. Sadka, Peter Sweeney, Ahmet M. Kondoz. 1389-1392 [doi]
• Subband based MPEG audio mixing for Internet streaming applicationsPatrick de Smet, Thomas Vander Stichele. 1393-1396 [doi]
• New approaches to audio-visual segmentation of TV news for automatic topic retrievalUri Iurgel, Ralf Meermeier, Stefan Eickeler, Gerhard Rigoll. 1397-1400 [doi]
• Speech retrieval with video parsing for television news programsHelen M. Meng, Xiaoou Tang, Pui-Yu Hui, Xinbo Gao, Yuk-Chi Li. 1401-1404 [doi]
• Video scouting: an architecture and system for the integration of multimedia information in personal TV applicationsRadu S. Jasinschi, Nevenka Dimitrova, Thomas McGee, Lalitha Agnihotri, John Zimmerman. 1405-1408 [doi]
• The effect of text in storyboards for video navigationMichael G. Christel, Adrienne Warmack. 1409-1412 [doi]
• Major cast detection in video using both audio and visual informationZhu Liu, Yao Wang. 1413-1416 [doi]
• Sequence families sets constructed from quadratic congruence codes for use in secure spread spectrum watermarking for multimediaC. S. Lim, Saman S. Abeysekera, S. K. Amarasinghe. 1417-1420 [doi]
• An audio virtual DSP for multimedia frameworksGiorgio Zoia, Claudio Alberti. 1421-1424 [doi]
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• Very quick audio searching: introducing global pruning to the Time-Series Active SearchAkisato Kimura, Kunio Kashino, Takayuki Kurozumi, Hiroshi Murase. 1429-1432 [doi]
• Hierarchical adaptive regularisation method for depth extraction from planar recording of 3D-integral imagesSilvia Manolache Cirstea, Malcolm McCormick, Sun-Yuan Kung. 1433-1436 [doi]
• Hierarchical segmentation using latent semantic indexing in scale spaceMalcolm Slaney, Dulce B. Ponceleon. 1437-1440 [doi]
• The cellular text telephone modem - the solution for supporting text telephone functionality in GSM networksMatthias Dörbecker, Karl Hellwig, Fredrik Jansson, Tomas Frankkila. 1441-1444 [doi]
• Adaptive playout scheduling using time-scale modification in packet voice communicationsYi J. Liang, Nikolaus Färber, Bernd Girod. 1445-1448 [doi]
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• A new optimization procedure for extracting the point-based lip contour using active shape modelK. L. Sum, W. H. Lau, S. H. Leung, Alan Wee-Chung Liew, K. W. Tse. 1485-1488 [doi]
• Recognition of face profiles from the mugshot database using a hybrid connectionist/HMM approachFrank Wallhoff, Stefan Müller, Gerhard Rigoll. 1489-1492 [doi]
• A real-time face tracker for color videoSascha Spors, Rudolf Rabenstein. 1493-1496 [doi]
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• Principal component analysis for facial animationKhalid Goudeaux, Tsuhan Chen, Shyue-Wu Wang, Jen-Duo Liu. 1501-1504 [doi]
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• Decoding of text lines in grayscale document imagesKris Popat. 1513-1516 [doi]
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• HMM topology optimization for handwriting recognitionDanfeng Li, Alain Biem, Jayashree Subrahmonia. 1521-1524 [doi]
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• Minimum classification error training of hidden Markov models for handwriting recognitionAlain Biem. 1529-1532 [doi]
• Blockiness detection in compressed dataGeorge A. Triantafyllidis, Dimitrios Tzovaras, Michael G. Strintzis. 1533-1536 [doi]
• Moving targets detection using sequential importance samplingGang Qian, Rama Chellappa. 1537-1540 [doi]
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• Shadow identification and classification using invariant color modelsElena Salvador, Andrea Cavallaro, Touradj Ebrahimi. 1545-1548 [doi]
• Texture classification with a biorthogonal directional filter bankJosé-Gerardo Rosiles, Mark J. T. Smith. 1549-1552 [doi]
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• Wavelet based halftone segmentation and descreening filter designChunghui Kuo, A. Ravishankar Rao, John G. Thompson. 1573-1576 [doi]
• A generic fuzzy rule based technique for image segmentationGour C. Karmakar, Laurence Dooley. 1577-1580 [doi]
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• Optimal training and redundant precoding for block transmissions with application to wireless OFDMShuichi Ohno, Georgios B. Giannakis. 2389-2392 [doi]
• Optimal embedding of known symbols for OFDMSrihari Adireddy, Lang Tong, Harish Viswanathan. 2393-2396 [doi]
• On the performance of multicarrier CDMA using multiple transmittersQuentin H. Spencer, A. Lee Swindlehurst. 2397-2400 [doi]
• Joint transmitter/receiver design for multicarrier data transmission with low latency timeTanja Karp, Gerald Schuller. 2401-2404 [doi]
• Optimal time-frequency signaling for rapidly time-varying channelsTamer Kadous, Ke Liu, Akbar M. Sayeed. 2405-2408 [doi]
• A multirate pilot-symbol-assisted channel estimator for OFDM transmitter diversity systemsKing F. Lee, Douglas B. Williams. 2409-2412 [doi]
• An alias-free subband adaptive equalizer for OFDM systemTomohide Miyagi, Hiroshi Ochi. 2413-2416 [doi]
• Specification and design of a prototype filter for filter bank based multicarrier transmissionMaurice G. Bellanger. 2417-2420 [doi]
• High throughput wideband space-time signaling using channel state informationEko N. Onggosanusi, Barry D. Van Veen, Akbar M. Sayeed. 2421-2424 [doi]
• Maximum-SNR space-time designs for MIMO channelsPetre Stoica, Girish Ganesan. 2425-2428 [doi]
• Space-time diversity systems based on unitary constellation-rotating precodersYan Xin 0001, Zhengdao Wang, Georgios B. Giannakis. 2429-2432 [doi]
• Performance of unitary space-time modulation in a continuously changing channelChristian B. Peel, A. Lee Swindlehurst. 2433-2436 [doi]
• Performance of spatial multiplexing in the presence of polarization diversityHelmut Bölcskei, Rohit U. Nabar, Vinko Erceg, David Gesbert, Arogyaswami Paulraj. 2437-2440 [doi]
• Space-time block coding with optimal antenna selectionDhananjay Gore, Arogyaswami Paulraj. 2441-2444 [doi]
• Space-time signaling and frame theoryRobert W. Heath Jr., Helmut Bölcskei, Arogyaswami Paulraj. 2445-2448 [doi]
• Walsh coding across multiple antennas for high data rate wireless servicesAndrew C. McCormick, Peter M. Grant, John S. Thompson. 2449-2452 [doi]
• Joint beamformer estimation and co-antenna interference cancellation for TURBO-BLASTMathini Sellathurai, Simon Haykin. 2453-2456 [doi]
• Differential space-time modulation schemes for DS-CDMA systemsJianhua Liu, Jian Liu. 2457-2460 [doi]
• High-rate linear space-time codesBabak Hassibi, Bertrand M. Hochwald. 2461-2464 [doi]
• Design of space-time trellis codes for full spatial diversityZoltan Safar, K. J. Ray Liu. 2465-2468 [doi]
• Rate adaptive space-time modulation techniques for combating cochannel interferenceQing Yan, Rick S. Blum. 2469-2472 [doi]
• Union bound on error probability of linear space-time block codesSumeet Sandhu, Arogyaswami Paulraj. 2473-2476 [doi]
• A class of efficient-encoding generalized low-density parity-check codesTong Zhang 0002, Keshab K. Parhi. 2477-2480 [doi]
• A space-time coding approach for systems employing four transmit antennasConstantinos B. Papadias, Gerard J. Foschini. 2481-2484 [doi]
• Mutual coupling effects on the capacity of multielement antenna systemsThomas Svantesson, Anders Ranheim. 2485-2488 [doi]
• Modeling and capacity of realistic spatial MIMO channelsAkbar M. Sayeed. 2489-2492 [doi]
• A wireless MIMO channel probing approach for arbitrary antenna arraysBrian D. Jeffs, Edward Pyper, Brandon Hunter. 2493-2496 [doi]
• Spatial characteristics of the MIMO wireless channel: experimental data acquisition and analysisJon W. Wallace, Michael A. Jensen. 2497-2500 [doi]
• Wireless indoor channel modeling: statistical agreement of ray tracing simulations and channel sounding measurementsGus German, Quentin H. Spencer, A. Lee Swindlehurst, Reinaldo A. Valenzuela. 2501-2504 [doi]
• Space-time correlation modeling of multielement antenna systems in mobile fading channelsAli Abdi, Mostafa Kaveh. 2505-2508 [doi]
• Asymptotic capacity of space-time coding for arbitrary fading: a closed form expression using Girko's lawAnna Scaglione, Unal Sakoglu. 2509-2511 [doi]
• Wide band channel characterisation in coloured noise using the reversible jump MCMCJean-René Larocque, James P. Reilly. 2513-2516 [doi]
• Estimation of network link loss rates via chaining in multicast treesAgisilaos-Georgios P. Ziotopoulos, Alfred O. Hero III, Kimberly M. Wasserman. 2517-2520 [doi]
• Performance analysis of moment-based estimators for the K parameter of the Rice fading distributionCihan Tepedelenlioglu, Ali Abdi, Georgios B. Giannakis, Mostafa Kaveh. 2521-2524 [doi]
• Markovian model of the error probability density and application to the error propagation probability computation of the weighted, decision feedback equalizerAlban Goupil, Jacques Palicot. 2525-2528 [doi]
• Convergence analysis of a new blind equalization algorithm with M-ary PSK channel inputsPing Liu, Zhengyuan Xu. 2529-2532 [doi]
• Expression of the capacity for the Gilbert channel in presence of interleavingThierry Rancurel, Daniel Roviras, Francis Castanie. 2533-2536 [doi]
• Importance sampling evaluation of digital phase detectors based on extended Kalman-Bucy filtersFrancisco A. S. Silva, José M. N. Leitão. 2537-2540 [doi]
• Optimal signal design and detection for fast fading channelsChet Lo, Todd K. Moon. 2541-2544 [doi]
• Joint blind equalization and estimation of the symbol period: a contrast function approachSébastien Houcke, Antoine Chevreuil, Philippe Loubaton. 2545-2548 [doi]
• Subspace detectors for stochastic process shift keyingArnt-Børre Salberg, Alfred Hanssen. 2549-2552 [doi]
• Classification of digital modulations by MCMC samplingStéphane Lesage, Jean-Yves Tourneret, Petar M. Djuric. 2553-2556 [doi]
• Cramer-Rao lower bound for frequency estimation in multipath Rayleigh fading channelsVladimir M. Baronkin, Yuriy V. Zakharov, Tim C. Tozer. 2557-2560 [doi]
• Parameter estimation of binary CPM signalsJaume Riba, Gregori Vázquez. 2561-2564 [doi]
• Soft-input source decoding for robust transmission of compressed images using two-dimensional optimal estimationJörg Kliewer, Norbert Görtz. 2565-2568 [doi]
• Bounds on the channel distortion of vector quantizersGal Ben-David, David Malah. 2569-2572 [doi]
• Locally optimal joint encoding of image transform coefficientsPiya Bunyaratavej, David J. Miller 0001. 2573-2576 [doi]
• Sequential signal encoding and estimation for distributed sensor networksMohamed M. Abdallah, Haralabos C. Papadopoulos. 2577-2580 [doi]
• Joint source-channel coding using structured oversampled filters banks applied to image transmissionAbraham Gabay, Olivier Rioul, Pierre Duhamel. 2581-2584 [doi]
• Comparison of structures for joint equalization and decodingDavid Declercq. 2585-2588 [doi]
• An algorithm for transform coding for lossy packet networksFrancesco Palmieri, Dario Petriccione. 2589-2592 [doi]
• Joint and tandem source-channel coding with delay constraintsJongtae Lim, David L. Neuhoff. 2593-2596 [doi]
• Vectorial DPCM coding and application to wideband speech codingDavid L. Mary, Dirk T. M. Slock. 2597-2600 [doi]
• Low-complexity transform coding with integer-to-integer transformsCiprian Doru Giurcaneanu, Ioan Tabus. 2601-2604 [doi]
• Indexing and entropy coding of lattice codevectorsAdriana Vasilache, Ioan Tabus. 2605-2608 [doi]
• Decoding of half-rate wavelet codes; Golay code and moreFaramarz Fekri, Steven W. McLaughlin, Russell M. Mersereau, Ronald W. Schafer. 2609-2612 [doi]
• Source and channel coding for remote speech recognition over error-prone channelsAlexis Bernard, Abeer Alwan. 2613-2616 [doi]
• Rate distortion optimal signal compression using second order polynomial approximationRanveig Nygaard, Aggelos K. Katsaggelos. 2617-2620 [doi]
• A stable adaptive structure for delta modulation with improved performanceMansour A. Aldajani, Ali H. Sayed. 2621-2624 [doi]
• The mismatch-noise PSD from a tree-structured DAC in a second-order ΔΣ modulator with a midscale inputJared Welz, Ian Galton. 2625-2628 [doi]
• A Lagrangian formulation of high rate quantizationJoyce Shih, Anuradha K. Aiyer, Robert M. Gray. 2629-2632 [doi]
• Area-efficient high speed decoding schemes for turbo/MAP decodersZhongfeng Wang, Zhipei Chi, Keshab K. Parhi. 2633-2636 [doi]
• A study on the performance, power consumption tradeoffs of short frame turbo decoder designZhipei Chi, Zhongfeng Wang, Keshab K. Parhi. 2637-2640 [doi]
• Turbo equalization for GMSK signaling over multipath channelsXiaodong Wang, Zigang Yang. 2641-2644 [doi]
• Turbo-equalization: convergence analysisAline Roumy, Alex J. Grant, Inbar Fijalkow, Paul D. Alexander, Didier Pirez. 2645-2648 [doi]
• Iterative source and channel decoding for GSMRalf Perkert, Markus Kaindl, Thomas Hindelang. 2649-2652 [doi]
• Iterative source-channel decoder using extrinsic information from softbit-source decodingMarc Adrat, Peter Vary, Julian Spittka. 2653-2656 [doi]
• Joint source-channel turbo decoding of VLC-coded Markov sourcesEric Fabre, Arnaud Guyader, Christine Guillemot. 2657-2660 [doi]
• Turbo space-time equalization of TCM for broadband wireless channelsMutlu Koca, Bernard C. Levy. 2661-2664 [doi]
• Linear RF power amplifier design for TDMA signals: a spectrum analysis approachChunming Liu, Heng Xiao, Qiang Wu, Fu Li. 2665-2668 [doi]
• Compensation of amplifier nonlinearities on wavelet packet division multiplexingKin-Fai To, P. C. Ching, Kon Max Wong. 2669-2672 [doi]
• Analyzing spectral regrowth of QPSK and OQPSK signalsRaviv Raich, G. Tong Zhou. 2673-2676 [doi]
• Optimisation of the interference cost generated by the random access channel of the UMTS FDD systemMohamed Siala, Hatem Boujemaa. 2677-2680 [doi]
• Design of a totally passive wireless digital micro-transceiver for picocellular systemsAnna Scaglione, Judd A. Rohwer. 2681-2684 [doi]
• Use of Kalman filtering in data detection in optical communication systems with multiplicative noiseHarris R. Burris, Anne E. Reed, Nader M. Namazi, M. J. Vilcheck, Mike S. Ferraro. 2685-2688 [doi]
• Nice experiences teaching SP in ArgentinaDaniel Jacoby, Roxana Saint-Nom. 2689-2692 [doi]
• Demonstration platform for real-time beamformi