Abstract is missing.
- Speech and language processing: where have we been and where are we going?Kenneth Ward Church. 1-4 [doi]
- Auditory principles in speech processing - do computers need silicon ears ?Birger Kollmeier. 5-8 [doi]
- A speech processing front-end with eigenspace normalization for robust speech recognition in noisy automobile environmentsKaisheng Yao, Erik M. Visser, Oh-Wook Kwon, Te-Won Lee. 9-12 [doi]
- Maximum likelihood normalization for robust speech recognitionYiu-Pong Lai, Man-Hung Siu. 13-16 [doi]
- Robust speech recognition using model-based feature enhancementVeronique Stouten, Hugo Van Hamme, Kris Demuynck, Patrick Wambacq. 17-20 [doi]
- Several HKU approaches for robust speech recognition and their evaluation on Aurora connected digit recognition tasksJian Wu, Qiang Huo. 21-24 [doi]
- Average instantaneous frequency (AIF) and average log-envelopes (ALE) for ASR with the Aurora 2 databaseYadong Wang, Jesse Hansen, Gopi Krishna Allu, Ramdas Kumaresan. 25-28 [doi]
- Adaptation of acoustic model using the gain-adapted HMM decomposition methodAkira Sasou, Futoshi Asano, Kazuyo Tanaka, Satoshi Nakamura. 29-32 [doi]
- Person authentication by voice: a need for cautionJean-François Bonastre, Frédéric Bimbot, Louis-Jean Boë, Joseph P. Campbell, Douglas A. Reynolds, Ivan Magrin-Chagnolleau. 33-36 [doi]
- ISCA special session: hot topics in speech synthesisGérard Bailly, Nick Campbell, Bernd Möbius. 37-40 [doi]
- Perceiving emotions by ear and by eyeBéatrice de Gelder. 41-44 [doi]
- Strategies for automatic multi-tier annotation of spoken language corporaSteven Greenberg. 45-48 [doi]
- Why is the special structure of the language important for Chinese spoken language processing? - examples on spoken document retrieval, segmentation and summarizationLin-Shan Lee, Yuan Ho, Jia-Fu Chen, Shun-Chuan Chen. 49-52 [doi]
- Speech analysis with the short-time chirp transformLuis Weruaga, Marián Képesi. 53-56 [doi]
- Glottal spectrum based inverse filteringIxone Arroabarren, Alfonso Carlosena. 57-60 [doi]
- A novel method of analysing and comparing responses of hearing aid algorithms using auditory time-frequency representationG. V. Kiran, Thippur V. Sreenivas. 61-64 [doi]
- Frequency-related representation of speechKuldip K. Paliwal, Bishnu S. Atal. 65-68 [doi]
- Tracking a moving speaker using excitation source informationVikas C. Raykar, Ramani Duraiswami, B. Yegnanarayana, S. R. Mahadeva Prasanna. 69-72 [doi]
- Tracking vocal tract resonances using an analytical nonlinear predictor and a target-guided temporal constraintLi Deng, Issam Bazzi, Alex Acero. 73-76 [doi]
- Features of contracted syllables of spontaneous MandarinShu-Chuan Tseng. 77-80 [doi]
- Durational characteristics of hindi stop consonantsK. Samudravijaya. 81-84 [doi]
- Quantity comparison of Japanese and finnish in various word structuresToshiko Isei-Jaakkola. 85-88 [doi]
- Broad focus across sentence types in greekMary Baltazani. 89-92 [doi]
- Analysis and modeling of syllable duration for Thai speech synthesisChatchawarn Hansakunbuntheung, Virongrong Tesprasit, Rungkarn Siricharoenchai, Yoshinori Sagisaka. 93-96 [doi]
- Reaction time as an indicator of discrete intonational contrasts in EnglishAoju Chen. 97-100 [doi]
- Transforming F0 contoursBen Gillett, Simon King. 101-104 [doi]
- Evaluation of the affect of speech intonation using a model of the perception of interval dissonance and harmonic tensionNorman D. Cook, Takeshi Fujisawa, Kazuaki Takami. 105-108 [doi]
- A new pitch modeling approach for Mandarin speechWen-Hsing Lai, Yih-Ru Wang, Sin-Horng Chen. 109-112 [doi]
- Bayesian induction of intonational phrase breaksPanagiotis Zervas, Manolis Maragoudakis, Nikos Fakotakis, George Kokkinakis. 113-116 [doi]
- Predicting the perceptive judgment of voices in a telecom context: selection of acoustic parametersThibaut Ehrette, Noël Chateau, Christophe d Alessandro, Valérie Maffiolo. 117-120 [doi]
- Stress-based speech segmentation revisitedSven L. Mattys. 121-124 [doi]
- Emotion recognition by speech signalsOh-Wook Kwon, Kwokleung Chan, Jiucang Hao, Te-Won Lee. 125-128 [doi]
- Automatic prosodic prominence detection in speech using acoustic features: an unsupervised systemFabio Tamburini. 129-132 [doi]
- Improved emotion recognition with large set of statistical featuresVladimir Hozjan, Zdravko Kacic. 133-136 [doi]
- Recognition of intonation patterns in Thai utterancePatavee Charnvivit, Nuttakorn Thubthong, Ekkarit Maneenoi, Sudaporn Luksaneeyanawin, Somchai Jitapunkul. 137-140 [doi]
- Use of linguistic information for automatic extraction of f_0 contour generation process model parametersKeikichi Hirose, Yusuke Furuyama, Shuichi Narusawa, Nobuaki Minematsu, Hiroya Fujisaki. 141-144 [doi]
- Potential audiovisual correlates of contrastive focus in FrenchMarion Dohen, Hélène Loevenbruck, Marie-Agnès Cathiard, Jean-Luc Schwartz. 145-148 [doi]
- How does human segment the speech by prosody ?Toshie Hatano, Yasuo Horiuchi, Akira Ichikawa. 149-152 [doi]
- Language-reconfigurable universal phone recognitionBrenton D. Walker, Bradley C. Lackey, J. S. Muller, Patrick John Schone. 153-156 [doi]
- Emotion recognition using a data-driven fuzzy inference systemChul-Min Lee, Shrikanth Narayanan. 157-160 [doi]
- Effects of voice prosody by computers on human behaviorsNoriko Suzuki, Yohei Yabuta, Yugo Takeuchi, Yasuhiro Katagiri. 161-164 [doi]
- An investigation of intensity patterns for GermanOliver Jokisch, Marco Kühne. 165-168 [doi]
- Segmental durations predicted with a neural networkJoão Paulo Teixeira, Diamantino Freitas. 169-172 [doi]
- Generation and perception of f_0 markedness in conversational speech with adverbs expressing degreesTakumi Yamashita, Yoshinori Sagisaka. 173-176 [doi]
- Quantitative analysis and synthesis of syllabic tones in vietnameseHansjörg Mixdorff, Nguyen Hung Bach, Hiroya Fujisaki, Luong Chi Mai. 177-180 [doi]
- Japanese prosodic labeling support system utilizing linguistic informationShinya Kiriyama, Yoshifumi Mitsuta, Yuta Hosokawa, Yoshikazu Hashimoto, Toshihiko Itoh, Shigeyoshi Kitazawa. 181-184 [doi]
- Why and how to control the authentic emotional speech corporaVéronique Aubergé, Nicolas Audibert, Albert Rilliard. 185-188 [doi]
- Prosodic cues for emotion characterization in real-life spoken dialogsLaurence Devillers, Ioana Vasilescu. 189-192 [doi]
- Towards the automatic generation of mixed-initiative dialogue systems from web contentJoseph Polifroni, Grace Chung, Stephanie Seneff. 193-196 [doi]
- A context resolution server for the galaxy conversational systemsEdward Filisko, Stephanie Seneff. 197-200 [doi]
- Semantic and dialogic annotation for automated multilingual customer serviceHilda Hardy, Kirk Baker, Hélène Bonneau-Maynard, Laurence Devillers, Sophie Rosset, Tomek Strzalkowski. 201-204 [doi]
- Disfluency under feedback and time-pressureH. B. M. Nicholson, Ellen Gurman Bard, Anne H. Anderson, María L. Flecha-García, D. Kenicer, Lucy Smallwood, Jim Mullin, Robin J. Lickley, Y. Chen. 205-208 [doi]
- Control in task-oriented dialoguesPeter A. Heeman, Fan Yang, Susan E. Strayer. 209-212 [doi]
- The 300k LIMSI German broadcast news transcription systemKevin McTait, Martine Adda-Decker. 213-216 [doi]
- Weighted entropy training for the decision tree based text-to-phoneme mappingJilei Tian, Janne Suontausta, Juha Häkkinen. 217-220 [doi]
- Word class modeling for speech recognition with out-of-task words using a hierarchical language modelYoshihiko Ogawa, Hirofumi Yamamoto, Yoshinori Sagisaka, Gen-ichiro Kikui. 221-224 [doi]
- Compound decomposition in dutch large vocabulary speech recognitionRoeland Ordelman, Arjan van Hessen, Franciska de Jong. 225-228 [doi]
- Designing for errors: similarities and differences of disfluency rates and prosodic characteristics across domainsGuergana K. Savova, Joan Bachenko. 229-232 [doi]
- Syllable classification using articulatory-acoustic featuresMirjam Wester. 233-236 [doi]
- Hierarchical class n-gram language models: towards better estimation of unseen events in speech recognitionImed Zitouni, Olivier Siohan, Chin-Hui Lee. 237-240 [doi]
- Incremental and iterative monolingual clustering algorithmsSergio Barrachina, Juan Miguel Vilar. 241-244 [doi]
- Techniques for effective vocabulary selectionAnand Venkataraman, Wen Wang. 245-248 [doi]
- Recognition of out-of-vocabulary words with sub-lexical language modelsLucian Galescu. 249-252 [doi]
- A semantic representation for spoken dialogsHélène Bonneau-Maynard, Sophie Rosset. 253-256 [doi]
- A corpus-based decompounding algorithm for German lexical modeling in LVCSRMartine Adda-Decker. 257-260 [doi]
- Modeling cross-morpheme pronunciation variations for korean large vocabulary continuous speech recognitionKyong-Nim Lee, Minhwa Chung. 261-264 [doi]
- Unit selection based on voice recognitionYi Zhou, Yiqing Zu. 265-268 [doi]
- On unit analysis for Cantonese corpus-based TTSJun Xu, Thomas Choy, Minghui Dong, Cuntai Guan, Haizhou Li. 269-272 [doi]
- Unit selection in concatenative TTS synthesis systems based on mel filter bank amplitudes and phonetic contextTanya Lambert, Andrew P. Breen, Barry Eggleton, Stephen J. Cox, Ben P. Milner. 273-276 [doi]
- Text design for TTS speech corpus building using a modified greedy selectionBaris Bozkurt, Özlem Öztürk, Thierry Dutoit. 277-280 [doi]
- Discriminative weight training for unit-selection based speech synthesisSeung Seop Park, Chong Kyu Kim, Nam Soo Kim. 281-284 [doi]
- The application of interactive speech unit selection in TTS systemsPeter Rutten, Justin Fackrell. 285-288 [doi]
- On the design of cost functions for unit-selection speech synthesisFrancisco Campillo Díaz, Eduardo Rodríguez Banga. 289-292 [doi]
- Kalman-filter based join cost for unit-selection speech synthesisJithendra Vepa, Simon King. 293-296 [doi]
- Optimizing integrated cost function for segment selection in concatenative speech synthesis based on perceptual evaluationsTomoki Toda, Hisashi Kawai, Minoru Tsuzaki. 297-300 [doi]
- Automatic segmentation for czech concatenative speech synthesis using statistical approach with boundary-specific correctionJindrich Matousek, Daniel Tihelka, Josef Psutka. 301-304 [doi]
- Automatic speech segmentation and verification for concatenative synthesisChih-Chung Kuo, Chi-Shiang Kuo, Jau-Hung Chen, Sen-Chia Chang. 305-308 [doi]
- DTW-based phonetic alignment using multiple acoustic featuresSérgio Paulo, Luís C. Oliveira. 309-312 [doi]
- Evaluating and correcting phoneme segmentation for unit selection synthesisJohn Kominek, Christina L. Bennett, Alan W. Black. 313-316 [doi]
- Control and prediction of the impact of pitch modification on synthetic speech qualityEsther Klabbers, Jan P. H. van Santen. 317-320 [doi]
- My voice, your prosody: sharing a speaker specific prosody model across speakers in unit selection TTSMatthew P. Aylett, Justin Fackrell, Peter Rutten. 321-324 [doi]
- Learning phrase break detection in Thai text-to-speechVirongrong Tesprasit, Paisarn Charoenpornsawat, Virach Sornlertlamvanich. 325-328 [doi]
- A speech model of acoustic inventories based on asynchronous interpolationAlexander Kain, Jan P. H. van Santen. 329-332 [doi]
- Corpus-based synthesis of fundamental frequency contours of Japanese using automatically-generated prosodic corpus and generation process modelKeikichi Hirose, Takayuki Ono, Nobuaki Minematsu. 333-336 [doi]
- Analysis of the Aurora large vocabulary evaluationsNaveen Parihar, Joseph Picone. 337-340 [doi]
- Evaluation of quantile based histogram equalization with filter combination on the Aurora 3 and 4 databasesFlorian Hilger, Hermann Ney. 341-344 [doi]
- Large vocabulary noise robustness on Aurora4Luca Rigazio, Patrick Nguyen, David Kryze, Jean-Claude Junqua. 345-348 [doi]
- Evaluation of model-based feature enhancement on the AURORA-4 taskVeronique Stouten, Hugo Van Hamme, Jacques Duchateau, Patrick Wambacq. 349-352 [doi]
- Improved feature extraction based on spectral noise reduction and nonlinear feature normalizationJosé C. Segura, Javier Ramírez, M. Carmen Benítez, Ángel de la Torre, Antonio J. Rubio. 353-356 [doi]
- Feature compensation technique for robust speech recognition in noisy environmentsYoung-Joon Kim, Hyun Woo Kim, Woohyung Lim, Nam Soo Kim. 357-360 [doi]
- The statistical approach to machine translation and a roadmap for speech translationHermann Ney. 361-364 [doi]
- Coupling vs. unifying: modeling techniques for speech-to-speech translationYuqing Gao. 365-368 [doi]
- Speechalator: two-way speech-to-speech translation on a consumer PDAAlex Waibel, Ahmed Badran, Alan W. Black, Robert E. Frederking, Donna Gates, Alon Lavie, Lori S. Levin, Kevin A. Lenzo, Laura Mayfield Tomokiyo, Jürgen Reichert, Tanja Schultz, Dorcas Wallace, Monika Woszczyna, Jing Zhang. 369-372 [doi]
- Development of phrase translation systems for handheld computers: from concept to fieldHoracio Franco, Jing Zheng, Kristin Precoda, Federico Cesari, Victor Abrash, Dimitra Vergyri, Anand Venkataraman, Harry Bratt, Colleen Richey, Ace Sarich. 373-376 [doi]
- Evaluation frameworks for speech translation technologiesMarcello Federico. 377-380 [doi]
- Creating corpora for speech-to-speech translationGen-ichiro Kikui, Eiichiro Sumita, Toshiyuki Takezawa, Seiichi Yamamoto. 381-384 [doi]
- Prosodic analysis and modeling of the NAGAUTA singing to synthesize its prosodic patterns from the standard notationNobuaki Minematsu, Bungo Matsuoka, Keikichi Hirose. 385-388 [doi]
- Statistical evaluation of the influence of stress on pitch frequency and phoneme durations in farsi languageDavood Gharavian, Seyed Mohammad Ahadi. 389-392 [doi]
- Prosody dependent speech recognition with explicit duration modelling at intonational phrase boundariesKen Chen, Sarah Borys, Mark Hasegawa-Johnson, Jennifer Cole. 393-396 [doi]
- Prediction of fujisaki model s phrase commandsJoão Paulo Teixeira, Diamantino Freitas, Hiroya Fujisaki. 397-400 [doi]
- Corpus-based modeling of naturalness estimation in timing control for non-native speechMakiko Muto, Yoshinori Sagisaka, Takuro Naito, Daiju Maeki, Aki Kondo, Katsuhiko Shirai. 401-404 [doi]
- Perceptually-related acoustic-prosodic features of phrase finals in spontaneous speechCarlos Toshinori Ishi, Parham Mokhtari, Nick Campbell. 405-408 [doi]
- Efficient linear combination for distant n-gram modelsDavid Langlois, Kamel Smaïli, Jean-Paul Haton. 409-412 [doi]
- Improving a connectionist based syntactical language modelAhmad Emami. 413-416 [doi]
- Using untranscribed user utterances for improving language models based on confidence scoringMikio Nakano, Timothy J. Hazen. 417-420 [doi]
- Improved Chinese broadcast news transcription by language modeling with temporally consistent training corpora and iterative phrase extractionPi-Chuan Chang, Shuo-Peng Liao, Lin-Shan Lee. 421-424 [doi]
- Language model adaptation using word clusteringShinsuke Mori, Masafumi Nishimura, Nobuyasu Itoh. 425-428 [doi]
- Hierarchical topic classification for dialog speech recognition based on language model switchingIan R. Lane, Tatsuya Kawahara, Tomoko Matsui, Satoshi Nakamura. 429-432 [doi]
- Linear predictive method with low-frequency emphasisPaavo Alku, Tomas Bäckström. 433-436 [doi]
- Beyond a single critical-band in TRAP based ASRPratibha Jain, Hynek Hermansky. 437-440 [doi]
- Variational Bayesian GMM for speech recognitionFabio Valente, Christian Wellekens. 441-444 [doi]
- Time alignment for scenario and sounds with voice, music and BGMYamato Wada, Masahide Sugiyama. 445-448 [doi]
- Efficient quantization of speech excitation parameters using temporal decompositionPhu Chien Nguyen, Masato Akagi. 449-452 [doi]
- Distributed genetic algorithm to discover a wavelet packet best basis for speech recognitionRobert van Kommer, Béat Hirsbrunner. 453-456 [doi]
- New model-based HMM distances with applications to run-time ASR error estimation and model tuningChao-Shih Huang, Chin-Hui Lee, Hsiao-Chuan Wang. 457-460 [doi]
- Analysis of voice source characteristics using a constrained polynomial modelTokihiko Kaburagi, Koji Kawai. 461-464 [doi]
- Tone pattern discrimination combining parametric modeling and maximum likelihood estimationJinfu Ni, Hisashi Kawai. 465-468 [doi]
- Feature selection for the classification of crosstalk in multi-channel audioStuart N. Wrigley, Guy J. Brown, Vincent Wan, Steve Renals. 469-472 [doi]
- A DTW-based DAG technique for speech and speaker feature analysisJingwei Liu. 473-476 [doi]
- Feature transformations and combinations for improving ASR performancePanu Somervuo, Barry Y. Chen, Qifeng Zhu. 477-480 [doi]
- On the role of intonation in the organization of Mandarin Chinese speech prosodyChiu-yu Tseng. 481-484 [doi]
- An optimized multi-duration HMM for spontaneous speech recognitionYuichi Ohkawa, Akihiro Yoshida, Motoyuki Suzuki, Akinori Ito, Shozo Makino. 485-488 [doi]
- Speaker recognition using MPEG-7 descriptorsHyoung-Gook Kim, Edgar Berdahl, Nicolas Moreau, Thomas Sikora. 489-492 [doi]
- A comparative study on maximum entropy and discriminative training for acoustic modeling in automatic speech recognitionWolfgang Macherey, Hermann Ney. 493-496 [doi]
- Extraction methods of voicing feature for robust speech recognitionAndrás Zolnay, Ralf Schlüter, Hermann Ney. 497-500 [doi]
- Use of a CSP-based voice activity detector for distant-talking ASRLuca Armani, Marco Matassoni, Maurizio Omologo, Piergiorgio Svaizer. 501-504 [doi]
- Maximum conditional mutual information projection for speech recognitionMohamed Kamal Omar, Mark Hasegawa-Johnson. 505-508 [doi]
- A semi-blind source separation method for hands-free speech recognition of multiple talkersPanikos Heracleous, Satoshi Nakamura, Kiyohiro Shikano. 509-512 [doi]
- Influence of the waveguide propagation on the antenna performance in a car cabinLeonid G. Krasny, Ali S. Khayrallah. 513-516 [doi]
- Multi-speaker DOA tracking using interactive multiple models and probabilistic data associationIlyas Potamitis, George Tremoulis, Nikos Fakotakis. 517-520 [doi]
- Speech enhancement using weighting function based on the variance of wavelet coefficientsChing-Ta Lu, Hsiao-Chuan Wang. 521-524 [doi]
- Microphone array voice activity detection and noise suppression using wideband generalized likelihood ratioIlyas Potamitis, Eran Fishler. 525-528 [doi]
- Adaptive beamforming in room with reverberationZoran Saric, Slobodan Jovicic. 529-532 [doi]
- Perceptually-constrained generalized singular value decomposition-based approach for enhancing speech corrupted by colored noiseGwo-Hwa Ju, Lin-Shan Lee. 533-536 [doi]
- Blind separation and deconvolution for convolutive mixture of speech using SIMO-model-based ICA and multichannel inverse filteringHiroaki Yamajo, Hiroshi Saruwatari, Tomoya Takatani, Tsuyoki Nishikawa, Kiyohiro Shikano. 537-540 [doi]
- Quality enhancement of CELP coded speech by using an MFCC based Gaussian mixture modelD. G. Raza, C. F. Chan. 541-544 [doi]
- Enhancement of noisy speech for noise robust front-end and speech reconstruction at back-end of DSR systemHyoung-Gook Kim, Markus Schwab, Nicolas Moreau, Thomas Sikora. 545-548 [doi]
- Improved kalman filter-based speech enhancementJianqiang Wei, Limin Du, Zhaoli Yan, Hui Zeng. 549-552 [doi]
- Speech segregation based on fundamental event information using an auditory vocoderToshio Irino, Roy D. Patterson, Hideki Kawahara. 553-556 [doi]
- Time delay estimation based on hearing characteristicZhaoli Yan, Limin Du, Jianqiang Wei, Hui Zeng. 557-560 [doi]
- Parametric multi-band automatic gain control for noisy speech enhancementMikhail Stolbov, Serguei Koval, Mikhail Khitrov. 561-564 [doi]
- Neural networks versus codebooks in an application for bandwidth extension of speech signalsBernd Iser, Gerhard Schmidt. 565-568 [doi]
- Wavelet-based perceptual speech enhancement using adaptive threshold estimationEssa Jafer, Abdulhussain E. Mahdi. 569-572 [doi]
- A trainable speech enhancement technique based on mixture models for speech and noiseIlyas Potamitis, Nikos Fakotakis, George Kokkinakis. 573-576 [doi]
- Perceptual wavelet adaptive denoising of speechQiang Fu, Eric A. Wan. 577-580 [doi]
- Enhancement of speech in multispeaker environmentB. Yegnanarayana, S. R. Mahadeva Prasanna, Mathew Magimai-Doss. 581-584 [doi]
- Noise reduction using paired-microphones on non-equally-spaced microphone arrangementMitsunori Mizumachi, Satoshi Nakamura. 585-588 [doi]
- Two studies of open vs. directed dialog strategies in spoken dialog systemsSilke M. Witt, Jason D. Williams. 589-592 [doi]
- The queen s communicator: an object-oriented dialogue managerIan M. O Neill, Philip Hanna, Xingkun Liu, Michael F. McTear. 593-596 [doi]
- Ravenclaw: dialog management using hierarchical task decomposition and an expectation agendaDan Bohus, Alexander I. Rudnicky. 597-600 [doi]
- Features for tree based dialogue course managementKlaus Macherey, Hermann Ney. 601-604 [doi]
- Development of a stochastic dialog manager driven by semanticsFrancisco Torres, Emilio Sanchis, Encarna Segarra. 605-608 [doi]
- Generation of natural response timing using decision tree based on prosodic and linguistic informationMasashi Takeuchi, Norihide Kitaoka, Seiichi Nakagawa. 609-612 [doi]
- Child and adult speaker adaptation during error resolution in a publicly available spoken dialogue systemLinda Bell, Joakim Gustafson. 613-616 [doi]
- Conceptual decoding for spoken dialog systemsYannick Estève, Christian Raymond, Frédéric Béchet, Renato de Mori. 617-620 [doi]
- Sentence verification in spoken dialogue systemHuei-Ming Wang, Yi-Chung Lin. 621-624 [doi]
- Detection and recognition of correction utterance in spontaneously spoken dialogNorihide Kitaoka, Naoko Kakutani, Seiichi Nakagawa. 625-628 [doi]
- Topic-specific parser design in an air travel natural language understanding applicationChaitanya Ekanadham, Juan M. Huerta. 629-632 [doi]
- The use of confidence measures in vector based call-routingStephen J. Cox, Gavin Cawley. 633-636 [doi]
- Multi-channel sentence classification for spoken dialogue language modelingFrédéric Béchet, Giuseppe Riccardi, Dilek Z. Hakkani-Tür. 637-640 [doi]
- Automatic induction of n-gram language models from a natural language grammarStephanie Seneff, Chao Wang, Timothy J. Hazen. 641-644 [doi]
- Connectionist classification and specific stochastic models in the understanding process of a dialogue systemDavid Vilar, María José Castro, Emilio Sanchis. 645-648 [doi]
- Robust parsing of utterances in negotiative dialogueJohan Boye, Mats Wirén. 649-652 [doi]
- Flexible speech act identification of spontaneous speech with disfluencyChung-Hsien Wu, Gwo-Lang Yan. 653-656 [doi]
- Efficient spoken dialogue control depending on the speech recognition rate and system s databaseKohji Dohsaka, Norihito Yasuda, Kiyoaki Aikawa. 657-660 [doi]
- Robust speech understanding based on expected discourse planShinya Takahashi, Tsuyoshi Morimoto, Sakashi Maeda, Naoyuki Tsuruta. 661-664 [doi]
- Normalization of time-derivative parameters using histogram equalizationYasunari Obuchi, Richard M. Stern. 665-668 [doi]
- Tree-structured noise-adapted HMM modeling for piecewise linear-transformation-based adaptationZhipeng Zhang, Kiyotaka Otsuji, Sadaoki Furui. 669-672 [doi]
- Maximum likelihood sub-band weighting for robust speech recognitionDonglai Zhu, Satoshi Nakamura, Kuldip K. Paliwal, Ren-Hua Wang. 673-676 [doi]
- Feature compensation scheme based on parallel combined mixture modelWooil Kim, Sungjoo Ahn, Hanseok Ko. 677-680 [doi]
- A comparison of three non-linear observation models for noisy speech featuresJasha Droppo, Li Deng, Alex Acero. 681-684 [doi]
- A new supervised-predictive compensation scheme for noisy speech recognitionKhalid Daoudi, Murat Deviren. 685-688 [doi]
- Statistical methods and Bayesian interpretation of evidence in forensic automatic speaker recognitionAndrzej Drygajlo, Didier Meuwly, Anil Alexander. 689-692 [doi]
- Robust likelihood ratio estimation in Bayesian forensic speaker recognitionJoaquin Gonzalez-Rodriguez, Daniel Garcia-Romero, Marta Garcia-Gomar, Daniel Ramos-Castro, Javier Ortega-Garcia. 693-696 [doi]
- Automated speaker recognition in real world conditions: controlling the uncontrollableHirotaka Nakasone. 697-700 [doi]
- Estimating the weight of evidence in forensic speaker verificationBeat Pfister, René Beutler. 701-704 [doi]
- Auditory-instrumental forensic speaker recognitionStefan G. Gfrörer. 705-708 [doi]
- Earwitness line-ups: effects of speech duration, retention interval and acoustic environment on identification accuracyJose H. Kerstholt, E. J. M. Jansen, A. G. van Amelsvoort, A. P. A. Broeders. 709-712 [doi]
- Characteristics of authentic anger in hebrew speechNoam Amir, Shirley Ziv, Rachel Cohen. 713-716 [doi]
- Prosody-based classification of emotions in spoken finnishTapio Seppänen, Eero Väyrynen, Juhani Toivanen. 717-720 [doi]
- Frequency distribution based weighted sub-band approach for classification of emotional/stressful content in speechMandar A. Rahurkar, John H. L. Hansen. 721-724 [doi]
- Classifying subject ratings of emotional speech using acoustic featuresJackson Liscombe, Jennifer J. Venditti, Julia Hirschberg. 725-728 [doi]
- Recognition of emotions in interactive voice response systemsSherif M. Yacoub, Steven J. Simske, Xiaofan Lin, John Burns 0003. 729-732 [doi]
- We are not amused - but how do you know? user states in a multi-modal dialogue systemAnton Batliner, Viktor Zeißler, Carmen Frank, Johann Adelhardt, Rui P. Shi, Elmar Nöth. 733-736 [doi]
- On-line user modelling in a mobile spoken dialogue systemNiels Ole Bernsen. 737-740 [doi]
- Towards dynamic multi-domain dialogue processingBotond Pakucs. 741-744 [doi]
- User modeling in spoken dialogue systems for flexible guidance generationKazunori Komatani, Shinichi Ueno, Tatsuya Kawahara, Hiroshi G. Okuno. 745-748 [doi]
- Empowering end users to personalize dialogue systems through spoken interactionStephanie Seneff, Grace Chung, Chao Wang. 749-752 [doi]
- LET s GO: improving spoken dialog systems for the elderly and non-nativesAntoine Raux, Brian Langner, Alan W. Black, Maxine Eskenazi. 753-756 [doi]
- Agents for integrated tutoring in spoken dialogue systemsJaakko Hakulinen, Markku Turunen, Esa-Pekka Salonen. 757-760 [doi]
- Corpus-based syntax-prosody tree matchingDafydd Gibbon. 761-764 [doi]
- A new approach to segment and detect syllables from high-speed speechD. W. Ying, W. Gao, W. Q. Wang. 765-768 [doi]
- Information structure and efficiency in speech productionR. J. J. H. van Son, Louis C. W. Pols. 769-772 [doi]
- Learning rule ranking by dynamic construction of context-free grammars using AND/OR graphsAnna Corazza, Louis ten Bosch. 773-776 [doi]
- The effect of surrounding phrase lengths on pause durationElena Zvonik, Fred Cummins. 777-780 [doi]
- Statistical estimation of phoneme s most stable point based on universal constraintShigeki Okawa, Katsuhiko Shirai. 781-784 [doi]
- Independent automatic segmentation by self-learning categorial pronunciation rulesNicole Beringer. 785-788 [doi]
- Prosodic correlates of contrastive and non-contrastive themes in GermanBettina Braun, D. Robert Ladd. 789-792 [doi]
- Accentual lengthening in standard Chinese: evidence from four-syllable constituentsYiya Chen. 793-796 [doi]
- Syllable structure based phonetic units for context-dependent continuous Thai speech recognitionSupphanat Kanokphara. 797-800 [doi]
- An acoustic phonetic analysis of diphthongs in ningbo ChineseFang Hu. 801-804 [doi]
- Latent ability to manipulate phonemes by Japanese preliterates in roman alphabetTakashi Otake, Yoko Sakamoto. 805-808 [doi]
- The /i/-/a/-/u/-ness of spoken vowelsHartmut R. Pfitzinger. 809-812 [doi]
- A computational model of arm gestures in conversationDafydd Gibbon, Ulrike Gut, Benjamin Hell, Karin Looks, Alexandra Thies, Thorsten Trippel. 813-816 [doi]
- Nonlinear analysis of speech signals: generalized dimensions and lyapunov exponentsVassilis Pitsikalis, Iasonas Kokkinos, Petros Maragos. 817-820 [doi]
- Time-domain based temporal processing with application of orthogonal transformationsPetr Motlícek, Jan Cernocký. 821-824 [doi]
- Recognition of phoneme strings using TRAP techniquePetr Schwarz, Pavel Matejka, Jan Cernocký. 825-828 [doi]
- Comparative study on hungarian acoustic model sets and training methodsTibor Fegyó, Péter Mihajlik, Péter Tatai. 829-832 [doi]
- F_0 estimation of one or several voicesAlain de Cheveigné, Alexis Baskind. 833-836 [doi]
- In search of target class definition in tandem feature extractionSunil Sivadas, Hynek Hermansky. 837-840 [doi]
- Segmentation of speech for speaker and language recognitionAndré Gustavo Adami, Hynek Hermansky. 841-844 [doi]
- Feature generation based on maximum classification probability for improved speech recognitionXiang Li, Richard M. Stern. 845-848 [doi]
- Speech recognition with a generative factor analyzed hidden Markov modelKaisheng Yao, Kuldip K. Paliwal, Te-Won Lee. 849-852 [doi]
- Learning discriminative temporal patterns in speech: development of novel TRAPS-like classifiersBarry Y. Chen, Shuangyu Chang, Sunil Sivadas. 853-856 [doi]
- Using mutual information to design class-specific phone recognizersPatricia Scanlon, Daniel P. W. Ellis, Richard B. Reilly. 857-860 [doi]
- Estimation of GMM in voice conversion including unaligned dataHelenca Duxans, Antonio Bonafonte. 861-864 [doi]
- Trajectory modeling based on HMMs with the explicit relationship between static and dynamic featuresKeiichi Tokuda, Heiga Zen, Tadashi Kitamura. 865-868 [doi]
- On the advantage of frequency-filtering features for speech recognition with variable sampling frequencies. experiments with speechdatcar databasesHermann Bauerecker, Climent Nadeu, Jaume Padrell. 869-872 [doi]
- Towards the automatic extraction of fujisaki model parameters for MandarinHansjörg Mixdorff, Hiroya Fujisaki, Gao Peng Chen, Yu Hu. 873-876 [doi]
- Product of Gaussians as a distributed representation for speech recognitionS. S. Airey, M. J. F. Gales. 877-880 [doi]
- Harmonic weighting for all-pole modeling of the voiced speechDavor Petrinovic. 881-884 [doi]
- Estimation of resonant characteristics based on AR-HMM modeling and spectral envelope conversion of vowel soundsNobuyuki Nishizawa, Keikichi Hirose, Nobuaki Minematsu. 885-888 [doi]
- Utterance verification under distributed detection and fusion frameworkTaeyoon Kim, Hanseok Ko. 889-892 [doi]
- Joint estimation of thresholds in a bi-threshold verification problemSimon Ho, Brian Mak. 893-896 [doi]
- Confidence measures for phonetic segmentation of continuous speechSamir Nefti, Olivier Boëffard, Thierry Moudenc. 897-900 [doi]
- Using confidence measures and domain knowledge to improve speech recognitionPascal Wiggers, Léon J. M. Rothkrantz. 901-904 [doi]
- Isolated word verification using cohort word-level verificationKishan Thambiratnam, Sridha Sridharan. 905-908 [doi]
- A new approach to minimize utterance verification error rate for a specific operating pointWing-Hei Au, Man-Hung Siu. 909-912 [doi]
- Continuous speech recognition and verification based on a combination scoreBinfeng Yan, Rui Guo, Xiaoyan Zhu. 913-916 [doi]
- Impact of word graph density on the quality of posterior probability based confidence measuresTibor Fábián, Robert Lieb, Günther Ruske, Matthias Thomae. 917-920 [doi]
- An efficient keyword spotting technique using a complementary language for filler models trainingPanikos Heracleous, Tohru Shimizu. 921-924 [doi]
- Context-sensitive evaluation and correction of phone recognition outputMichael Levit, Hiyan Alshawi, Allen L. Gorin, Elmar Nöth. 925-928 [doi]
- Estimating speech recognition error rate without acoustic test dataYonggang Deng, Milind Mahajan, Alex Acero. 929-932 [doi]
- Multigram-based grapheme-to-phoneme conversion for LVCSRMaximilian Bisani, Hermann Ney. 933-936 [doi]
- Integrating statistical and rule-based knowledge for continuous German speech recognitionRené Beutler, Beat Pfister. 937-940 [doi]
- A fast, accurate and stream-based speaker segmentation and clustering algorithmAn Vandecatseye, Jean-Pierre Martens. 941-944 [doi]
- A sequential metric-based audio segmentation method via the Bayesian information criterionShih-Sian Cheng, Hsin-Min Wang. 945-948 [doi]
- Sentence boundary detection in arabic speechAmit Srivastava, Francis Kubala. 949-952 [doi]
- Automated transcription and topic segmentation of large spoken archivesMartin Franz, Bhuvana Ramabhadran, Todd Ward, Michael Picheny. 953-956 [doi]
- Automatic disfluency identification in conversational speech using multiple knowledge sourcesYang Liu, Elizabeth Shriberg, Andreas Stolcke. 957-960 [doi]
- Topic segmentation and retrieval system for lecture videos based on spontaneous speech recognitionNatsuo Yamamoto, Jun Ogata, Yasuo Ariki. 961-964 [doi]
- Hybrid HMM/BN ASR system integrating spectrum and articulatory featuresKonstantin Markov, Jianwu Dang, Yosuke Iizuka, Satoshi Nakamura. 965-968 [doi]
- Context-dependent output densities for hidden Markov models in speech recognitionGeorg Stemmer, Viktor Zeißler, Christian Hacker, Elmar Nöth, Heinrich Niemann. 969-972 [doi]
- Time adjustable mixture weights for speaking rate fluctuationTakahiro Shinozaki, Sadaoki Furui. 973-976 [doi]
- A switching linear Gaussian hidden Markov model and its application to nonstationary noise compensation for robust speech recognitionJian Wu, Qiang Huo. 977-980 [doi]
- On factorizing spectral dynamics for robust speech recognitionVivek Tyagi, Iain McCowan, Hervé Bourlard, Hemant Misra. 981-984 [doi]
- Joint model and feature based compensation for robust speech recognition under non-stationary noise environmentsChuan Jia, Peng Ding, Bo Xu. 985-988 [doi]
- Weighted automata kernels - general framework and algorithmsCorinna Cortes, Patrick Haffner, Mehryar Mohri. 989-992 [doi]
- Large margin methods for label sequence learningYasemin Altun, Thomas Hofmann. 993-996 [doi]
- Robust multi-class boostingGunnar Rätsch. 997-1000 [doi]
- Statistical signal processing with nonnegativity constraintsLawrence K. Saul, Fei Sha, Daniel D. Lee. 1001-1004 [doi]
- Inline updates for HMMsAshutosh Garg, Manfred K. Warmuth. 1005-1008 [doi]
- Factorial models and refiltering for speech separation and denoisingSam T. Roweis. 1009-1012 [doi]
- Band-independent speech-event categories for TRAP based ASRHynek Hermansky, Pratibha Jain. 1013-1016 [doi]
- Local averaging and differentiating of spectral plane for TRAP-based ASRFrantisek Grézl, Hynek Hermansky. 1017-1020 [doi]
- Minimum variance distortionless response on a warped frequency scaleMatthias Wölfel, John W. McDonough, Alex Waibel. 1021-1024 [doi]
- Improving the efficiency of automatic speech recognition by feature transformation and dimensionality reductionXuechuan Wang, Douglas D. O Shaughnessy. 1025-1028 [doi]
- Distributed speech recognition on the WSJ taskJan Stadermann, Gerhard Rigoll. 1029-1032 [doi]
- Integrating multilingual articulatory features into speech recognitionSebastian Stüker, Florian Metze, Tanja Schultz, Alex Waibel. 1033-1036 [doi]
- Using corpus-based methods for spoken access to news texts on the webAlexandra Klein, Harald Trost. 1037-1040 [doi]
- Cross-modal informational masking due to mismatched audio cues in a speechreading taskDouglas Brungart, Brian D. Simpson, Alexander J. Kordik. 1041-1044 [doi]
- Audiovisual speech enhancement based on the association between speech envelope and video featuresFrédéric Berthommier. 1045-1048 [doi]
- Robust speech interaction in a mobile environment through the use of multiple and different media input typesRainer Wasinger, Christoph Stahl, Antonio Krüger. 1049-1052 [doi]
- Speech-based, manual-visual, and multi-modal interaction with an in-car computer - evaluation of a pilot studyRogier Woltjer, Wah Jin Tan, Fang Chen. 1053-1056 [doi]
- Bayesian networks for spoken dialogue management in multimodal systems of tour-guide robotsPlamen J. Prodanov, Andrzej Drygajlo. 1057-1060 [doi]
- Optimization of window and LSF interpolation factor for the ITU-t g.729 speech coding standardWai C. Chu, Toshio Miki. 1061-1064 [doi]
- Likelihood ratio test with complex laplacian model for voice activity detectionJoon-Hyuk Chang, Jong Won Shin, Nam Soo Kim. 1065-1068 [doi]
- Multi-mode quantization of adjacent speech parameters using a low-complexity prediction schemeJani Nurminen. 1069-1072 [doi]
- Multi-mode matrix quantizer for low bit rate LSF quantizationUlpu Sinervo, Jani Nurminen, Ari Heikkinen, Jukka Saarinen. 1073-1076 [doi]
- Voicing controlled frame loss concealment for adaptive multi-rate (AMR) speech frames in voice-over-IPFrank Mertz, Hervé Taddei, Imre Varga, Peter Vary. 1077-1080 [doi]
- Perceptual irrelevancy removal in narrowband speech codingMarja Lahdekorpi, Jani Nurminen, Ari Heikkinen, Jukka Saarinen. 1081-1084 [doi]
- Very-low-rate speech compression by indexation of polyphonesCharles du Jeu, Maurice Charbit, Gérard Chollet. 1085-1088 [doi]
- Entropy-optimized channel error mitigation with application to speech recognition over wirelessVictoria E. Sánchez, Antonio M. Peinado, Angel M. Gomez, José L. Pérez-Córdoba. 1089-1092 [doi]
- Robust jointly optimized multistage vector quantization for speech codingVenkatesh Krishnan, David V. Anderson. 1093-1096 [doi]
- Polar quantization of sinusoids from speech signal blocksHarald Pobloth, Renat Vafin, W. Bastiaan Kleijn. 1097-1100 [doi]
- Transcoding algorithm for g.723.1 and AMR speech coders: for interoperability between voIP and mobile networksSung-Wan Yoon, Jin-Kyu Choi, Hong-Goo Kang, Dae Hee Youn. 1101-1104 [doi]
- Quality-complexity trade-off in predictive LSF quantizationDavorka Petrinovic, Davor Petrinovic. 1105-1108 [doi]
- Variable bit rate control with trellis diagram approximationKei Kikuiri, Nobuhiko Naka, Tomoyuki Ohya. 1109-1112 [doi]
- Towards optimal encoding for classification with applications to distributed speech recognitionNaveen Srinivasamurthy, Antonio Ortega, Shrikanth Narayanan. 1113-1116 [doi]
- Multi-rate extension of the scalable to lossless PSPIHT audio coderMohammed Raad, Ian S. Burnett, Alfred Mertins. 1117-1120 [doi]
- Entropy constrained quantization of LSP parametersTuraj Zakizadeh Shabestary, Per Hedelin, Fredrik Norden. 1121-1124 [doi]
- Named entity extraction from Japanese broadcast newsAkio Kobayashi, Franz Josef Och, Hermann Ney. 1125-1128 [doi]
- Morpheme-based lexical modeling for korean broadcast news transcriptionYoung Hee Park, Dong-Hoon Ahn, Minhwa Chung. 1129-1132 [doi]
- Data driven example based continuous speech recognitionMathias De Wachter, Kris Demuynck, Dirk Van Compernolle, Patrick Wambacq. 1133-1136 [doi]
- Large vocabulary speaker independent isolated word recognition for embedded systemsSergey Astrov, Bernt Andrassy. 1137-1140 [doi]
- Low-latency incremental speech transcription in the synface projectAlexander Seward. 1141-1144 [doi]
- Multilingual acoustic modeling using graphemesStephan Kanthak, Hermann Ney. 1145-1148 [doi]
- A cross-media retrieval system for lecture videosAtsushi Fujii, Katunobu Itou, Tomoyosi Akiba, Tetsuya Ishikawa. 1149-1152 [doi]
- Confidence measure driven scalable two-pass recognition strategy for large list grammarsMiroslav Novak, Diego Ruiz. 1157-1160 [doi]
- An efficient, fast matching approach using posterior probability estimates in speech recognitionSherif Abdou, Michael S. Scordilis. 1161-1164 [doi]
- On lexicon creation for turkish LVCSRKadri Hacioglu, Bryan L. Pellom, Tolga Çiloglu, Özlem Öztürk, Mikko Kurimo, Mathias Creutz. 1165-1168 [doi]
- Compiling large-context phonetic decision trees into finite-state transducersStanley F. Chen. 1169-1172 [doi]
- Automatic summarization of broadcast news using structural featuresSameer Maskey, Julia Hirschberg. 1173-1176 [doi]
- A dynamic cross-reference pruning strategy for multiple feature fusion at decoder run timeYongHong Yan, Chengyi Zheng, Jianping Zhang, Jielin Pan, Jiang Han, Jian Liu. 1177-1180 [doi]
- Design of the CMU sphinx-4 decoderPaul Lamere, Philip Kwok, William Walker, Evandro B. Gouvêa, Rita Singh, Bhiksha Raj, Peter Wolf. 1181-1184 [doi]
- A new decoder design for large vocabulary turkish speech recognitionOnur Cilingir, Mübeccel Demirekler. 1185-1188 [doi]
- Automatic speech recognition with sparse training data for dysarthric speakersPhil Green, James Carmichael, Athanassios Hatzis, Pam Enderby, Mark S. Hawley, Mark Parker. 1189-1192 [doi]
- Prediction of sentence importance for speech summarization using prosodic parametersAkira Inoue, Takayoshi Mikami, Yoichi Yamashita. 1193-1196 [doi]
- An automatic singing transcription system with multilingual singing lyric recognizer and robust melody trackerChong-kai Wang, Ren-Yuan Lyu, Yuang-Chin Chiang. 1197-1200 [doi]
- Speech shift: direct speech-input-mode switching through intentional control of voice pitchMasataka Goto, Yukihiro Omoto, Katunobu Itou, Tetsunori Kobayashi. 1201-1204 [doi]
- Evaluating multiple LVCSR model combination in NTCIR-3 speech-driven web retrieval taskMasahiko Matsushita, Hiromitsu Nishizaki, Takehito Utsuro, Yasuhiro Kodama, Seiichi Nakagawa. 1205-1208 [doi]
- Semantic object synchronous understanding in SALT for highly interactive user interfaceKuansan Wang. 1209-1212 [doi]
- Information retrieval based call classificationJan Kneissler, Anne K. Kienappel, Dietrich Klakow. 1213-1216 [doi]
- Using syllable-based indexing features and language models to improve German spoken document retrievalMartha Larson, Stefan Eickeler. 1217-1220 [doi]
- An empirical text transformation method for spontaneous speech synthesizersShiva Sundaram, Shrikanth Narayanan. 1221-1224 [doi]
- A new approach to reducing alarm noise in speechYilmaz Gul, Aladdin M. Ariyaeeinia, Oliver Dewhirst. 1225-1228 [doi]
- Improved name recognition with user modelingDong Yu, Kuansan Wang, Milind Mahajan, Peter Mau, Alex Acero. 1229-1232 [doi]
- Speech recognition over bluetooth wireless channelsZiad Al Bawab, Ivo Locher, Jianxia Xue, Abeer Alwan. 1233-1236 [doi]
- Speech starter: noise-robust endpoint detection by using filled pausesKoji Kitayama, Masataka Goto, Katunobu Itou, Tetsunori Kobayashi. 1237-1240 [doi]
- Automatic segmentation of film dialogues into phonemes and graphemesGilles Boulianne, Jean-Francois Beaumont, Patrick Cardinal, Michel Comeau, Pierre Ouellet, Pierre Dumouchel. 1241-1244 [doi]
- Automated closed-captioning of live TV broadcast news in FrenchJulie Brousseau, Jean-Francois Beaumont, Gilles Boulianne, Patrick Cardinal, Claude Chapdelaine, Michel Comeau, Frédéric Osterrath, Pierre Ouellet. 1245-1248 [doi]
- Automatic construction of unique signatures and confusable sets for natural language directory assistance applicationsE. E. Jan, Benoît Maison, Lidia Mangu, Geoffrey Zweig. 1249-1252 [doi]
- Recent enhancements in CU VOCAL for Chinese TTS-enabled applicationsHelen M. Meng, Yuk-Chi Li, Tien Ying Fung, Man Cheuk Ho, Chi-Kin Keung, Tin Hang Lo, Wai Kit Lo, P. C. Ching. 1253-1256 [doi]
- Evaluation of an alert system for selective dissemination of broadcast newsIsabel Trancoso, João Paulo Neto, Hugo Meinedo, Rui Amaral. 1257-1260 [doi]
- Low complexity joint optimization of excitation parameters in analysis-by-synthesis speech codingUdar Mittal, James P. Ashley, Edgardo M. Cruz-Zeno. 1261-1264 [doi]
- Named entity extraction from word latticesJames Horlock, Simon King. 1265-1268 [doi]
- A topic classification system based on parametric trajectory mixture modelsWilliam Belfield, Herbert Gish. 1269-1272 [doi]
- Model based noisy speech recognition with environment parameters estimated by noise adaptive speech recognition with priorKaisheng Yao, Kuldip K. Paliwal, Satoshi Nakamura. 1273-1276 [doi]
- A harmonic-model-based front end for robust speech recognitionMichael L. Seltzer, Jasha Droppo, Alex Acero. 1277-1280 [doi]
- A new perspective on feature extraction for robust in-vehicle speech recognitionUmit H. Yapanel, John H. L. Hansen. 1281-1284 [doi]
- Speech recognition of double talk using SAFIA-based audio segregationToshiyuki Sekiya, Tetsuji Ogawa, Tetsunori Kobayashi. 1285-1288 [doi]
- CFA-BF: a novel combined fixed/adaptive beamforming for robust speech recognition in real car environmentsXianxian Zhang, John H. L. Hansen. 1289-1292 [doi]
- Audio-visual speech recognition in challenging environmentsGerasimos Potamianos, Chalapathy Neti. 1293-1296 [doi]
- SYNFACE - a talking face telephoneInger Karlsson, Andrew Faulkner, Giampiero Salvi. 1297-1300 [doi]
- A voice-driven web browser for blind peopleBostjan Vesnicer, Janez Zibert, Simon Dobrisek, Nikola Pavesic, France Mihelic. 1301-1304 [doi]
- Exploiting speech for recognizing elderly users to respond to their special needsChristian A. Müller, Frank Wittig, Jörg Baus. 1305-1308 [doi]
- Spoken language and e-inclusionAlan F. Newell. 1309-1312 [doi]
- Acoustic normalization of children s speechGeorg Stemmer, Christian Hacker, Stefan Steidl, Elmar Nöth. 1313-1316 [doi]
- Unit size in unit selection speech synthesisS. P. Kishore, Alan W. Black. 1317-1320 [doi]
- Restricted unlimited domain synthesisAntje Schweitzer, Norbert Braunschweiler, Tanja Klankert, Bernd Möbius, Bettina Säuberlich. 1321-1324 [doi]
- Evaluation of units selection criteria in corpus-based speech synthesisHélène Francois, Olivier Boëffard. 1325-1328 [doi]
- Combining non-uniform unit selection with diphone based synthesisMichael Pucher, Friedrich Neubarth, Erhard Rank, Georg Niklfeld, Qi Guan. 1329-1332 [doi]
- Evolutionary weight tuning based on diphone pairs for unit selection speech synthesisFrancesc Alías, Xavier Llorà. 1333-1336 [doi]
- Keeping rare events rareOve Andersen, Charles Hoequist. 1337-1340 [doi]
- NIST 2003 language recognition evaluationAlvin F. Martin, Mark A. Przybocki. 1341-1344 [doi]
- Acoustic, phonetic, and discriminative approaches to automatic language identificationElliot Singer, Pedro A. Torres-Carrasquillo, Terry P. Gleason, William M. Campbell, Douglas A. Reynolds. 1345-1348 [doi]
- Using place name data to train language identification modelsStanley F. Chen, Benoît Maison. 1349-1352 [doi]
- Use of trajectory models for automatic accent classificationPongtep Angkititrakul, John H. L. Hansen. 1353-1356 [doi]
- Language identification using parallel sub-word recognition - an ergodic HMM equivalenceV. Ramasubramanian, A. K. V. Sai Jayram, T. V. Sreenivas. 1357-1360 [doi]
- On the combination of speech and speaker recognitionMohamed Faouzi BenZeghiba, Hervé Bourlard. 1361-1364 [doi]
- Improving speech intelligibility by steady-state suppression as pre-processing in small to medium sized hallsNao Hodoshima, Takayuki Arai, Tsuyoshi Inoue, Keisuke Kinoshita, Akiko Kusumoto. 1365-1368 [doi]
- Enhancement of hearing-impaired Mandarin speechChen-Long Lee, Ya-Ru Yang, Wen-Whei Chang, Yuan-Chuan Chiang. 1369-1372 [doi]
- Speech enhancement for a car environment using LP residual signal and spectral subtractionA. Alvarez, Victor Nieto Lluis, Pedro Gómez Vilda, Rafael Martínez. 1373-1376 [doi]
- Speech enhancement and improved recognition accuracy by integrating wavelet transform and spectral subtraction algorithmGwo-Hwa Ju, Lin-Shan Lee. 1377-1380 [doi]
- Multi-referenced correction of the voice timbre distortions in telephone networksGaël Mahé, André Gilloire. 1381-1384 [doi]
- Efficient speech enhancement based on left-right HMM with state sequence detection using LRTJ.-J. Lee, J. H. Lee, K. Y. Lee. 1385-1388 [doi]
- Introduction of the CELP structure of the GSM coder in the acoustic echo canceller for the GSM networkH. Gnaba, M. Turki-Hadj Alouane, Meriem Jaïdane-Saïdane, Pascal Scalart. 1389-1392 [doi]
- Extracting an AV speech source from a mixture of signalsDavid Sodoyer, Laurent Girin, Christian Jutten, Jean-Luc Schwartz. 1393-1396 [doi]
- Speech enhancement for hands-free car phones by adaptive compensation of harmonic engine noise componentsHenning Puder. 1397-1400 [doi]
- Enhance low-frequency suppression of GSC beamformingZhaorong Hou, Ying Jia. 1401-1404 [doi]
- Speech enhancement using a-priori informationSriram Srinivasan, Jonas Samuelsson, W. Bastiaan Kleijn. 1405-1408 [doi]
- Blind inversion of multidimensional functions for speech enhancementJohn Hogden, Patrick Valdez, Shigeru Katagiri, Erik McDermott. 1409-1412 [doi]
- Convergence improvement for oversampled subband adaptive noise and echo cancellationHamid Reza Abutalebi, Hamid Sheikhzadeh, Robert L. Brennan, George Freeman. 1413-1416 [doi]
- A speech dereverberation method based on the MTF conceptMasashi Unoki, Keigo Sakata, Masato Akagi. 1417-1420 [doi]
- Accuracy improved double-talk detector based on state transition diagramSang-Gyun Kim, Jong-Uk Kim, Chang D. Yoo. 1421-1424 [doi]
- Perceptual based speech enhancement for normal-hearing and hearing-impaired individualsAjay Natarajan, John H. L. Hansen, Kathryn Hoberg Arehart, Jessica Rossi-Katz. 1425-1428 [doi]
- Residual echo power estimation for speech reinforcement systems in vehiclesAlfonso Ortega, Eduardo Lleida, Enrique Masgrau. 1429-1432 [doi]
- Dual-mode wideband speech recovery from narrowband speechYasheng Qian, Peter Kabal. 1433-1436 [doi]
- A robust noise and echo cancellerKhaldoon Al-Naimi, Christian Sturt, Ahmet M. Kondoz. 1437-1440 [doi]
- Computational auditory scene analysis by using statistics of high-dimensional speech dynamics and sound source directionJohannes Nix, Michael Kleinschmidt, Volker Hohmann. 1441-1444 [doi]
- Vocal tract normalization as linear transformation of MFCCMichael Pitz, Hermann Ney. 1445-1448 [doi]
- Non-native spontaneous speech recognition through polyphone decision tree specializationZhirong Wang, Tanja Schultz. 1449-1452 [doi]
- Live speech recognition in sports games by adaptation of acoustic model and language modelYasuo Ariki, Takeru Shigemori, Tsuyoshi Kaneko, Jun Ogata, Masakiyo Fujimoto. 1453-1456 [doi]
- Speaker adaptation using regression classes generated by phonetic decision tree-based successive state splittingSe-Jin Oh, Kwang-Dong Kim, Duk-Gyoo Roh, Woo-Chang Sung, Hyun-Yeol Chung. 1457-1460 [doi]
- Reduction of dimension of HMM parameters using ICA and PCA in MLLR framework for speaker adaptationJiun Kim, Jaeho Chung. 1461-1464 [doi]
- Geometric constrained maximum likelihood linear regression on Mandarin dialect adaptationHuayun Zhang, Bo Xu. 1465-1468 [doi]
- Adapting language models for frequent fixed phrases by emphasizing n-gram subsetsTomoyosi Akiba, Katunobu Itou, Atsushi Fujii. 1469-1472 [doi]
- Learning intra-speaker model parameter correlations from many short speaker segmentsAnne K. Kienappel. 1473-1476 [doi]
- Modeling Cantonese pronunciation variation by acoustic model refinementPatgi Kam, Tan Lee, Frank K. Soong. 1477-1480 [doi]
- Performance improvement of rapid speaker adaptation based on eigenvoice and bias compensationJong Se Park, Hwa Jeon Song, Hyung Soon Kim. 1481-1484 [doi]
- Training data optimization for language model adaptationXiaoshan Fang, Jianfeng Gao, Jianfeng Li, Huanye Sheng. 1485-1488 [doi]
- Approaches to foreign-accented speaker-independent speech recognitionStefanie Aalburg, Harald Höge. 1489-1492 [doi]
- Unsupervised speaker adaptation based on HMM sufficient statistics in various noisy environmentsShingo Yamade, Akinobu Lee, Hiroshi Saruwatari, Kiyohiro Shikano. 1493-1496 [doi]
- Using genetic algorithms for rapid speaker adaptationFabrice Lauri, Irina Illina, Dominique Fohr, Filipp Korkmazsky. 1497-1500 [doi]
- Structural state-based frame synchronous compensationVincent Barreaud, Irina Illina, Dominique Fohr, Filipp Korkmazsky. 1501-1504 [doi]
- Effect of foreign accent on speech recognition in the NATO n-4 corpusAaron D. Lawson, David M. Harris, John J. Grieco. 1505-1508 [doi]
- Duration normalization and hypothesis combination for improved spontaneous speech recognitionJon P. Nedel, Richard M. Stern. 1509-1512 [doi]
- Maximum a posteriori linear regression (MAPLR) variance adaptation for continuous density HMMSWu Chou, Xiaodong He. 1513-1516 [doi]
- On divergence based clustering of normal distributions and its application to HMM adaptationTor André Myrvoll, Frank K. Soong. 1517-1520 [doi]
- Fast incremental adaptation using maximum likelihood regression and stochastic gradient descentSreeram V. Balakrishnan. 1521-1524 [doi]
- Tfarsdat - the telephone farsi speech databaseMahmood Bijankhan, Javad Sheykhzadegan, Mahmood R. Roohani, Rahman Zarrintare, Seyyed Z. Ghasemi, Mohammad E. Ghasedi. 1525-1528 [doi]
- Large lexica for speech-to-speech translation: from specification to creationElviira Hartikainen, Giulio Maltese, Asunción Moreno, Shaunie Shammass, Ute Ziegenhain. 1529-1532 [doi]
- A pronunciation lexicon for turkish based on two-level morphologyKemal Oflazer, Sharon Inkelas. 1533-1536 [doi]
- Using both global and local hidden Markov models for automatic speech unit segmentationHong Zheng, Yiqing Lu. 1537-1540 [doi]
- Quality control of language resources at ELRAHenk van den Heuvel, Khalid Choukri, Harald Höge, Bente Maegaard, Jan Odijk, Valérie Mapelli. 1541-1544 [doi]
- Validation of phonetic transcriptions based on recognition performanceChristophe Van Bael, Diana Binnenpoorte, Helmer Strik, Henk van den Heuvel. 1545-1548 [doi]
- The basque speech_dat (II) database: a description and first test recognition resultsInmaculada Hernáez, Iker Luengo, Eva Navas, Maria Luisa Zubizarreta, Iñaki Gaminde, J. Sanchez. 1549-1552 [doi]
- Towards an evaluation standard for speech control concepts in real-world scenariosJens Maase, Diane Hirschfeld, Uwe Koloska, Timo Westfeld, Jörg Helbig. 1553-1556 [doi]
- Orientel: recording telephone speech of turkish speakers in GermanyChristoph Draxler. 1557-1560 [doi]
- Spanish broadcast news transcriptionGerhard Backfried, Roser Jaquemot Caldes. 1561-1564 [doi]
- Large vocabulary continuous speech recognition in greek: corpus and an automatic dictation systemVassilios Digalakis, Dimitris Oikonomidis, Dimitris Pratsolis, Nikos Tsourakis, Christos Vosnidis, Nikos Chatzichrisafis, Vassilios Diakoloukas. 1565-1568 [doi]
- The LIUM-AVS database : a corpus to test lip segmentation and speechreading systems in natural conditionsPhilippe Daubias, Paul Deléglise. 1569-1572 [doi]
- Implementation and evaluation of a text-to-speech synthesis system for turkishÖzgül Salor, Bryan L. Pellom, Mübeccel Demirekler. 1573-1576 [doi]
- The czech speech and prosody database both for ASR and TTS purposesJáchym Kolár, Jan Romportl, Josef Psutka. 1577-1580 [doi]
- Construction of an advanced in-car spoken dialogue corpus and its characteristic analysisItsuki Kishida, Yuki Irie, Yukiko Yamaguchi, Shigeki Matsubara, Nobuo Kawaguchi, Yasuyoshi Inagaki. 1581-1584 [doi]
- Measuring the readability of automatic speech-to-text transcriptsDouglas A. Jones, Florian Wolf, Edward Gibson, Elliott Williams, Evelina Fedorenko, Douglas A. Reynolds, Marc A. Zissman. 1585-1588 [doi]
- The NESPOLE! voIP multilingual corpora in tourism and medical domainsNadia Mana, Susanne Burger, Roldano Cattoni, Laurent Besacier, Victoria MacLaren, John W. McDonough, Florian Metze. 1589-1592 [doi]
- Lexica and corpora for speech-to-speech translation: a trilingual approachDavid Conejero, Jesús Giménez, Victoria Arranz, Antonio Bonafonte, Neus Pascual, Núria Castell, Asunción Moreno. 1593-1596 [doi]
- From switchboard to fisher: telephone collection protocols, their uses and yieldsChristopher Cieri, David Miller, Kevin Walker. 1597-1600 [doi]
- Development of the estonian speechdat-like databaseEinar Meister, Jürgen Lasn, Lya Meister. 1601-1604 [doi]
- Towards a repository of digital talking booksAntónio Joaquim Serralheiro, Isabel Trancoso, Diamantino Caseiro, Teresa Chambel, Luís Carriço, Nuno Guimarães. 1605-1608 [doi]
- Shared resources for robust speech-to-text technologyStephanie Strassel, David Miller, Kevin Walker, Christopher Cieri. 1609-1612 [doi]
- Large vocabulary conversational speech recognition with a subspace constraint on inverse covariance matricesScott Axelrod, Vaibhava Goel, Brian Kingsbury, Karthik Visweswariah, Ramesh A. Gopinath. 1613-1616 [doi]
- Speaker adaptation based on confidence-weighted trainingGyucheol Jang, Minho Jin, Chang D. Yoo. 1617-1620 [doi]
- Jacobian adaptation based on the frequency-filtered spectral energiesAlberto Abad, Climent Nadeu, Javier Hernando, Jaume Padrell. 1621-1624 [doi]
- Structural linear model-space transformations for speaker adaptationDriss Matrouf, Olivier Bellot, Pascal Nocera, Georges Linares, Jean-François Bonastre. 1625-1628 [doi]
- Minimum classification error (MCE) model adaptation of continuous density HMMSXiaodong He, Wu Chou. 1629-1632 [doi]
- Adapting acoustic models to new domains and conditions using untranscribed dataAsela Gunawardana, Alex Acero. 1633-1636 [doi]
- Towards synthesising expressive speech; designing and collecting expressive speech dataNick Campbell. 1637-1640 [doi]
- Is there an emotion signature in intonational patterns? and can it be used in synthesis?Tanja Bänziger, Michel Morel, Klaus R. Scherer. 1641-1644 [doi]
- Multilayered extensions to the speech synthesis markup language for describing expressivenessEllen Eide, Raimo Bakis, Wael Hamza, John F. Pitrelli. 1645-1648 [doi]
- Unit selection and emotional speechAlan W. Black. 1649-1652 [doi]
- Voice quality modification for emotional speech synthesisChristophe d Alessandro, Boris Doval. 1653-1656 [doi]
- Applications of computer generated expressive speech for communication disordersJan P. H. van Santen, Lois M. Black, Gilead Cohen, Alexander Kain, Esther Klabbers, Taniya Mishra, Jacques de Villiers, Xiaochuan Niu. 1657-1660 [doi]
- Speaker verification systems and security considerationsDavid A. van Leeuwen. 1661-1664 [doi]
- Phonetic class-based speaker verificationMatthieu Hébert, Larry P. Heck. 1665-1668 [doi]
- An evaluation of VTS and IMM for speaker verification in noiseSuhadi Suhadi, Sorel Stan, Tim Fingscheidt, Christophe Beaugeant. 1669-1672 [doi]
- Locally recurrent probabilistic neural network for text-independent speaker verificationTodor Ganchev, Dimitris K. Tasoulis, Michael N. Vrahatis, Nikos Fakotakis. 1673-1676 [doi]
- Learning to boost GMM based speaker verificationStan Z. Li, Dong Zhang, Chengyuan Ma, Heung-Yeung Shum, Eric Chang. 1677-1680 [doi]
- Speaker verification based on g.729 and g.723.1 coder parameters and handset mismatch compensationEric W. M. Yu, Man-Wai Mak, Chin-Hung Sit, Sun-Yuan Kung. 1681-1684 [doi]
- Should i tell all?: an experiment on conciseness in spoken dialogueStephen Whittaker, Marilyn A. Walker, Preetam Maloor. 1685-1688 [doi]
- Natural language response generation in mixed-initiative dialogs using task goals and dialog actsHelen M. Meng, Wing Lin Yip, Oi Yan Mok, Shuk Fong Chan. 1689-1692 [doi]
- Speech generation from concept for realizing conversation with an agent in a virtual roomKeikichi Hirose, Junji Tago, Nobuaki Minematsu. 1693-1696 [doi]
- A trainable generator for recommendations in multimodal dialogMarilyn A. Walker, Rashmi Prasad, Amanda Stent. 1697-1700 [doi]
- Spoken dialogue system for queries on appliance manuals using hierarchical confirmation strategyTatsuya Kawahara, Ryosuke Ito, Kazunori Komatani. 1701-1704 [doi]
- SAG: a procedural tactical generator for dialog systemsDalina Kallulli. 1705-1708 [doi]
- Optimization of the CELP model in the LSP domainKhosrow Lashkari, Toshio Miki. 1709-1712 [doi]
- Transforming voice qualityBen Gillett, Simon King. 1713-1716 [doi]
- DOA estimation of speech signal using equilateral-triangular microphone arrayYusuke Hioka, Nozomu Hamada. 1717-1720 [doi]
- Multi-array fusion for beamforming and localization of moving speakersIlyas Potamitis, George Tremoulis, Nikos Fakotakis, George Kokkinakis. 1721-1724 [doi]
- Integrated pitch and MFCC extraction for speech reconstruction and speech recognition applicationsXu Shao, Ben P. Milner, Stephen J. Cox. 1725-1728 [doi]
- Exploiting time warping in AMR-NB and AMR-WB speech codersLasse Laaksonen, Sakari Himanen, Ari Heikkinen, Jani Nurminen. 1729-1732 [doi]
- A new approach to voice activity detection based on self-organizing mapsStephan Grashey. 1733-1736 [doi]
- Estimating the spectral envelope of voiced speech using multi-frame analysisYoshinori Shiga, Simon King. 1737-1740 [doi]
- Adaptive noise estimation using second generation and perceptual wavelet transformsEssa Jafer, Abdulhussain E. Mahdi. 1741-1744 [doi]
- A clustering approach to on-line audio source separationJulien Bourgeois. 1745-1748 [doi]
- Estimation of voice source and vocal tract characteristics based on multi-frame analysisYoshinori Shiga, Simon King. 1749-1752 [doi]
- A new method for pitch prediction from spectral envelope and its application in voice conversionTaoufik En-Najjary, Olivier Rosec, Thierry Chonavel. 1753-1756 [doi]
- Maximum likelihood endpoint detection with time-domain featuresMarco Orlandi, Alfiero Santarelli, Daniele Falavigna. 1757-1760 [doi]
- Unified analysis of glottal source spectrumIxone Arroabarren, Alfonso Carlosena. 1761-1764 [doi]
- A hidden Markov model-based missing data imputation approachYu Luo, Limin Du. 1765-1768 [doi]
- Integration of noise reduction algorithms for Aurora2 taskTakeshi Yamada, Jiro Okada, Kazuya Takeda, Norihide Kitaoka, Masakiyo Fujimoto, Shingo Kuroiwa, Kazumasa Yamamoto, Takanobu Nishiura, Mitsunori Mizumachi, Satoshi Nakamura. 1769-1772 [doi]
- Classification with free energy at raised temperaturesRita Singh, Manfred K. Warmuth, Bhiksha Raj, Paul Lamere. 1773-1776 [doi]
- Flooring the observation probability for robust ASR in impulsive noisePei Ding, Bertram E. Shi, Pascale Fung, Zhigang Cao. 1777-1780 [doi]
- Combination of temporal domain SVD based speech enhancement and GMM based speech estimation for ASR in noise - evaluation on the AURORA2 task -Masakiyo Fujimoto, Yasuo Ariki. 1781-1784 [doi]
- Additive noise and channel distortion-robust parametrization tool - performance evaluation on Aurora 2 & 3Petr Fousek, Petr Pollák. 1785-1788 [doi]
- Robust feature extraction and acoustic modeling at multitel: experiments on the Aurora databasesStéphane Dupont, Christophe Ris. 1789-1792 [doi]
- Noise robust speech parameterization based on joint wavelet packet decomposition and autoregressive modelingBojan Kotnik, Zdravko Kacic, Bogomir Horvat. 1793-1796 [doi]
- Database adaptation for ASR in cross-environmental conditions in the SPEECON projectChristophe Couvreur, Oren Gedge, Klaus Linhard, Shaunie Shammass, Johan Vantieghem. 1797-1800 [doi]
- Autoregressive modeling based feature extraction for Aurora3 DSR taskPetr Motlícek, Jan Cernocký. 1801-1804 [doi]
- Evaluation on the Aurora 2 database of acoustic models that are less noise-sensitiveEdmondo Trentin, Marco Matassoni, Marco Gori. 1805-1808 [doi]
- Revisiting scenarios and methods for variable frame rate analysis in automatic speech recognitionJavier Macías Guarasa, J. Ordonez, Juan Manuel Montero, Javier Ferreiros, Ricardo de Córdoba, Luis Fernando D Haro. 1809-1812 [doi]
- Multitask learning in connectionist robust ASR using recurrent neural networksShahla Parveen, Phil Green. 1813-1816 [doi]
- Confusion matrix based entropy correction in multi-stream combinationHemant Misra, Andrew C. Morris. 1817-1820 [doi]
- Large vocabulary ASR for spontaneous czech in the MALACH projectJosef Psutka, Pavel Ircing, Josef V. Psutka, Vlasta Radová, William J. Byrne, Jan Hajic, Jirí Mírovský, Samuel Gustman. 1821-1824 [doi]
- Active and unsupervised learning for automatic speech recognitionGiuseppe Riccardi, Dilek Z. Hakkani-Tür. 1825-1828 [doi]
- Perceptual MVDR-based cepstral coefficients (PMCCs) for high accuracy speech recognitionUmit H. Yapanel, Satya Dharanipragada, John H. L. Hansen. 1829-1832 [doi]
- A discriminative decision tree learning approach to acoustic modelingSheng Gao, Chin-Hui Lee. 1833-1836 [doi]
- Large corpus experiments for broadcast news recognitionPatrick Nguyen, Luca Rigazio, Jean-Claude Junqua. 1837-1840 [doi]
- Performance evaluation of phonotactic and contextual onset-rhyme models for speech recognition of Thai languageSomchai Jitapunkul, Ekkarit Maneenoi, Visarut Ahkuputra, Sudaporn Luksaneeyanawin. 1841-1844 [doi]
- Overlapped di-tone modeling for tone recognition in continuous Cantonese speechYao Qian, Tan Lee, Yujia Li. 1845-1848 [doi]
- Speaker model selection using Bayesian information criterion for speaker indexing and speaker adaptationMasafumi Nishida, Tatsuya Kawahara. 1849-1852 [doi]
- Automatic transcription of football commentaries in the MUMIS projectJanienke Sturm, Judith M. Kessens, Mirjam Wester, Febe de Wet, Eric Sanders, Helmer Strik. 1853-1856 [doi]
- On the limits of cluster-based acoustic modelingS. Douglas Peters. 1857-1860 [doi]
- Large vocabulary taiwanese (min-nan) speech recognition using tone features and statistical pronunciation modelingDau-Cheng Lyu, Min-Siong Liang, Yuang-Chin Chiang, Chun-Nan Hsu, Ren-Yuan Lyu. 1861-1864 [doi]
- A new spectral transformation for speaker normalizationPierre L. Dognin, Amro El-Jaroudi. 1865-1868 [doi]
- Enhanced tree clustering with single pronunciation dictionary for conversational speech recognitionHua Yu, Tanja Schultz. 1869-1872 [doi]
- Fitting class-based language models into weighted finite-state transducer frameworkPavel Ircing, Josef Psutka. 1873-1876 [doi]
- Multi-source training and adaptation for generic speech recognitionFabrice Lefevre, Jean-Luc Gauvain, Lori Lamel. 1877-1880 [doi]
- Toward domain-independent conversational speech recognitionBrian Kingsbury, Lidia Mangu, George Saon, Geoffrey Zweig, Scott Axelrod, Vaibhava Goel, Karthik Visweswariah, Michael Picheny. 1881-1884 [doi]
- Comparative study of boosting and non-boosting training for constructing ensembles of acoustic modelsRong Zhang, Alexander I. Rudnicky. 1885-1888 [doi]
- A study on domain recognition of spoken dialogue systemsT. Isobe, Shoji Hayakawa, Hiroya Murao, T. Mizutani, Kazuya Takeda, Fumitada Itakura. 1889-1892 [doi]
- Domain adaptation augmented by state-dependence in spoken dialog systemsWei He, Honglian Li, Baozong Yuan. 1893-1896 [doi]
- Smartkom-home - an advanced multi-modal interface to home entertainmentThomas Portele, Silke Goronzy, Martin C. Emele, Andreas Kellner, Sunna Torge, Jürgen te Vrugt. 1897-1900 [doi]
- Methods to improve its portability of a spoken dialog system both on task domains and languagesYunbiao Xu, Fengying Di, Masahiro Araki, Yasuhisa Niimi. 1901-1904 [doi]
- Voxenter^TM - intelligent voice enabled call center for hungarianTibor Fegyó, Péter Mihajlik, Mate Szarvas, Péter Tatai, Gábor Tatai. 1905-1908 [doi]
- Automatic call-routing without transcriptionsQiang Huang, Stephen J. Cox. 1909-1912 [doi]
- Jaspis^2 - an architecture for supporting distributed spoken dialoguesMarkku Turunen, Jaakko Hakulinen. 1913-1916 [doi]
- Development of a bilingual spoken dialog system for weather information retrievalJanez Zibert, Sanda Martincic-Ipsic, Melita Hajdinjak, Ivo Ipsic, France Mihelic. 1917-1920 [doi]
- Improving how may i help you? systems using the output of recognition latticesJames Allen, David Attwater, Peter J. Durston, Mark Farrell. 1921-1924 [doi]
- Incremental learning of new user formulations in automatic directory assistanceM. Andorno, Luciano Fissore, Pietro Laface, M. Nigra, Cosmin Popovici, Franco Ravera, Claudio Vair. 1925-1928 [doi]
- Dialog systems for automotive environmentsJulie Baca, Feng Zheng, Hualin Gao, Joseph Picone. 1929-1932 [doi]
- The development of a multi-purpose spoken dialogue systemJoão Paulo Neto, Nuno J. Mamede, Renato Cassaca, Luís C. Oliveira. 1933-1936 [doi]
- The dynamic, multi-lingual lexicon in smartkomSilke Goronzy, Zica Valsan, Martin C. Emele, Juergen Schimanowski. 1937-1940 [doi]
- Evaluating discourse understanding in spoken dialogue systemsRyuichiro Higashinaka, Noboru Miyazaki, Mikio Nakano, Kiyoaki Aikawa. 1941-1944 [doi]
- Assessment of spoken dialogue system usability - what are we really measuring?Lars Bo Larsen. 1945-1948 [doi]
- Evaluation of a speech-driven telephone information service using the PARADISE framework: a closer look at subjective measuresPaula M. T. Smeele, Juliette A. J. S. Waals. 1949-1952 [doi]
- Quantifying the impact of system characteristics on perceived quality dimensions of a spoken dialogue serviceSebastian Möller, Janto Skowronek. 1953-1956 [doi]
- A programmable policy manager for conversational biometricsGanesh N. Ramaswamy, Ran D. Zilca, Oleg Alecksandrovich. 1957-1960 [doi]
- Integration of speaker recognition into conversational spoken dialogue systemsTimothy J. Hazen, Douglas A. Jones, Alex Park, Linda C. Kukolich, Douglas A. Reynolds. 1961-1964 [doi]
- Discriminative optimization of large vocabulary Mandarin conversational speech recognition systemPeng Ding, Zhenbiao Chen, Sheng Hu, Shuwu Zhang, Bo Xu. 1965-1968 [doi]
- Speech recognition with dynamic grammars using finite-state transducersJohan Schalkwyk, I. Lee Hetherington, Ezra Story. 1969-1972 [doi]
- FLavor: a flexible architecture for LVCSRKris Demuynck, Tom Laureys, Dirk Van Compernolle, Hugo Van Hamme. 1973-1976 [doi]
- An architecture for rapid decoding of large vocabulary conversational speechGeorge Saon, Geoffrey Zweig, Brian Kingsbury, Lidia Mangu, Upendra V. Chaudhari. 1977-1980 [doi]
- MMI-MAP and MPE-MAP for acoustic model adaptationDaniel Povey, M. J. F. Gales, Do Yeong Kim, Philip C. Woodland. 1981-1984 [doi]
- Lattice segmentation and minimum Bayes risk discriminative trainingVlasios Doumpiotis, Stavros Tsakalidis, William J. Byrne. 1985-1988 [doi]
- Spoken language condensation in the 21st centuryKlaus Zechner. 1989-1992 [doi]
- Robust methods in automatic speech recognition and understandingSadaoki Furui. 1993-1998 [doi]
- Parsing spontaneous speechRodolfo Delmonte. 1999-2004 [doi]
- Model compression for GMM based speaker recognition systemsDouglas A. Reynolds. 2005-2008 [doi]
- The awe and mystery of t-normJiri Navratil, Ganesh N. Ramaswamy. 2009-2012 [doi]
- Gaussian dynamic warping (GDW) method applied to text-dependent speaker detection and verificationJean-François Bonastre, Philippe Morin, Jean-Claude Junqua. 2013-2016 [doi]
- Modeling duration patterns for speaker recognitionLuciana Ferrer, Harry Bratt, Venkata Ramana Rao Gadde, Sachin S. Kajarekar, Elizabeth Shriberg, M. Kemal Sönmez, Andreas Stolcke, Anand Venkataraman. 2017-2020 [doi]
- Improved speaker verification through probabilistic subspace adaptationSimon Lucey, Tsuhan Chen. 2021-2024 [doi]
- An improved model-based speaker segmentation systemPeng Yu, Frank Seide, Chengyuan Ma, Eric Chang. 2025-2028 [doi]
- A latent analogy framework for grapheme-to-phoneme conversionJerome R. Bellegarda. 2029-2032 [doi]
- Conditional and joint models for grapheme-to-phoneme conversionStanley F. Chen. 2033-2036 [doi]
- Mixed-lingual text analysis for polyglot TTS synthesisBeat Pfister, Harald Romsdorfer. 2037-2040 [doi]
- Identifying speakers in children s stories for speech synthesisJason Y. Zhang, Alan W. Black, Richard Sproat. 2041-2044 [doi]
- Experimental tools to evaluate intelligibility of text-to-speech (TTS) synthesis: effects of voice gender and signal qualityCatherine Stevens, Nicole Lees, Julie Vonwiller. 2045-2048 [doi]
- Arabic in my hand: small-footprint synthesis of egyptian arabicLaura Mayfield Tomokiyo, Alan W. Black, Kevin A. Lenzo. 2049-2052 [doi]
- Schema-based modeling of phonemic restorationSoundararajan Srinivasan, DeLiang Wang. 2053-2056 [doi]
- Perception of voice-individuality for distortions of resonance/source characteristics and waveformsHisao Kuwabara. 2057-2060 [doi]
- The perceptual cues of a high level pitch-accent pattern in Japanese: pitch-accent patterns and durationTsutomu Sato. 2061-2064 [doi]
- Illusory continuity of intermittent pure tone in binaural listening and its dependency on interaural time differenceMamoru Iwaki, Norio Nakamura. 2065-2068 [doi]
- CART-based factor analysis of intelligibility reduction in Japanese EnglishNobuaki Minematsu, Changchen Guo, Keikichi Hirose. 2069-2072 [doi]
- Harmonic alternatives to sine-wave speechLászló Tóth, András Kocsor. 2073-2076 [doi]
- Non-intrusive assessment of perceptual speech quality using a self-organising mapDorel Picovici, Abdulhussain E. Mahdi. 2077-2080 [doi]
- Inhibitory priming effect in auditory word recognition: the role of the phonological mismatch length between primes and targetsSophie Dufour, Ronald Peereman. 2081-2084 [doi]
- Recognising real-life speech with spem: a speech-based computational model of human speech recognitionOdette Scharenborg, Louis ten Bosch, Lou Boves. 2085-2088 [doi]
- The effect of speech rate and noise on bilinguals speech perception: the case of native speakers of arabic in israelJudith Rosenhouse, Liat Kishon-Rabin. 2089-2092 [doi]
- Subjective evaluations for perception of speaker identity through acoustic feature transplantationsOytun Türk, Levent M. Arslan. 2093-2096 [doi]
- Modelling human speech recognition using automatic speech recognition paradigms in speMOdette Scharenborg, James M. McQueen, Louis ten Bosch, Dennis Norris. 2097-2100 [doi]
- The effect of amplitude compression on wide band telephone speech for hearing-impaired elderly peopleMutsumi Saito, Kimio Shiraishi, Kimitoshi Fukudome. 2101-2104 [doi]
- Word activation model by Japanese school children without knowledge of roman alphabetTakashi Otake, Miki Komatsu. 2105-2108 [doi]
- Multi-resolution auditory scene analysis: robust speech recognition using pattern-matching from a noisy signalSue Harding, Georg Meyer. 2109-2112 [doi]
- Investigation of emotionally morphed speech perception and its structure using a high quality speech manipulation systemHisami Matsui, Hideki Kawahara. 2113-2116 [doi]
- Usefulness of phase spectrum in human speech perceptionKuldip K. Paliwal, Leigh D. Alsteris. 2117-2120 [doi]
- Perception of English lexical stress by English and Japanese speakers: effect of duration and realistic intensity changeShinichi Tokuma. 2121-2124 [doi]
- French intonational rises and their role in speech seg mentation [sic]Pauline Welby. 2125-2128 [doi]
- Physical and perceptual configurations of Japanese fricatives from multidimensional scaling analysesWon Tokuma. 2129-2132 [doi]
- An acquisition model of speech perception with considerations of temporal informationChing-Pong Au. 2133-2136 [doi]
- Dynamic channel compensation based on maximum a posteriori estimationHuayun Zhang, Zhaobing Han, Bo Xu. 2137-2140 [doi]
- Far-field ASR on inexpensive microphonesLaura Docío Fernández, David Gelbart, Nelson Morgan. 2141-2144 [doi]
- Evaluation of ETSI advanced DSR front-end and bias removal method on the Japanese newspaper article sentences speech corpusSatoru Tsuge, Shingo Kuroiwa, Kenji Kita. 2145-2148 [doi]
- Environment adaptive control of noise reduction parameters for improved robustness of ASRChng Chin Soon, Bernt Andrassy, Josef G. Bauer, Günther Ruske. 2149-2152 [doi]
- Speech enhancement with microphone array and fourier / wavelet spectral subtraction in real noisy environmentsYuki Denda, Takanobu Nishiura, Hideki Kawahara. 2153-2156 [doi]
- Environmental sound source identification based on hidden Markov model for robust speech recognitionTakanobu Nishiura, Satoshi Nakamura, Kazuhiro Miki, Kiyohiro Shikano. 2157-2160 [doi]
- High-likelihood model based on reliability statistics for robust combination of features: application to noisy speech recognitionPeter Jancovic, Münevver Köküer, Fionn Murtagh. 2161-2164 [doi]
- Noise robust digit recognition with missing framesCenk Demiroglu, David V. Anderson. 2165-2168 [doi]
- A noise-robust ASR back-end technique based on weighted viterbi recognitionXiaodong Cui, Alexis Bernard, Abeer Alwan. 2169-2172 [doi]
- Voice quality normalization in an utterance for robust ASRMuhammad Ghulam, Takashi Fukuda, Tsuneo Nitta. 2173-2176 [doi]
- Environmental sniffing: robust digit recognition for an in-vehicle environmentMurat Akbacak, John H. L. Hansen. 2177-2180 [doi]
- Energy contour extraction for in-car speech recognitionTai-Hwei Hwang. 2181-2184 [doi]
- Noise-robust ASR by using distinctive phonetic features approximated with logarithmic normal distribution of HMMTakashi Fukuda, Tsuneo Nitta. 2185-2188 [doi]
- Noise-robust automatic speech recognition using orthogonalized distinctive phonetic feature vectorsTakashi Fukuda, Tsuneo Nitta. 2189-2192 [doi]
- Language model accuracy and uncertainty in noise cancelling in the stochastic weighted viterbi algorithmNéstor Becerra Yoma, Ivan Brito, Jorge Silva. 2193-2196 [doi]
- An integrated system for smart-home control of appliances based on remote speech interactionIlyas Potamitis, Kallirroi Georgila, Nikos Fakotakis, George K. Kokkinakis. 2197-2200 [doi]
- A spoken language interface to an electronic programme guideJianhong Jin, Martin J. Russell, Michael J. Carey 0002, James Chapman, Harvey Lloyd-Thomas, Graham Tattersall. 2201-2204 [doi]
- Towards a personal robot with language interfaceLuís Seabra Lopes, António J. S. Teixeira, Mário Rodrigues, D. Gomes, C. Teixeira, L. Ferreira, P. Soares, J. Girao, N. Senica. 2205-2208 [doi]
- Preference, perception, and task completion of open, menu-based, and directed prompts for call routing: a case studyJason D. Williams, Andrew T. Shaw, Lawrence Piano, Michael Abt. 2209-2212 [doi]
- An integrated toolkit deploying speech technology for computer based speech training with application to dysarthric speakersAthanassios Hatzis, Phil Green, James Carmichael, Stuart Cunningham, Rebecca Palmer, Mark Parker, Peter O Neill. 2213-2216 [doi]
- Towards best practices for speech user interface designBernhard Suhm. 2217-2220 [doi]
- Design and evaluation of a limited two-way speech translatorDavid Stallard, John Makhoul, Fred Choi, Ehry MacRostie, Premkumar Natarajan, Richard M. Schwartz, Bushra Zawaydeh. 2221-2224 [doi]
- Multimodal interaction on PDA s integrating speech and pen inputsSorin Dusan, Gregory J. Gadbois, James L. Flanagan. 2225-2228 [doi]
- Towards multimodal interaction with an intelligent roomPetra Gieselmann, Matthias Denecke. 2229-2232 [doi]
- A multimodal conversational interface for a concept vehicleRoberto Pieraccini, Krishna Dayanidhi, Jonathan Bloom, Jean-Gui Dahan, Michael Phillips, Bryan R. Goodman, K. Venkatesh Prasad. 2233-2236 [doi]
- Context awareness using environmental noise classificationL. Ma, D. J. Smith, Ben P. Milner. 2237-2240 [doi]
- Simple designing methods of corpus-based visual speech synthesisTatsuya Shiraishi, Tomoki Toda, Hiromichi Kawanami, Hiroshi Saruwatari, Kiyohiro Shikano. 2241-2244 [doi]
- Comparing the usability of a user driven and a mixed initiative multimodal dialogue system for train timetable informationJanienke Sturm, Ilse Bakx, Bert Cranen, Jacques M. B. Terken. 2245-2248 [doi]
- Read my tongue movements: bimodal learning to perceive and produce non-native speech /r/ and /l/Dominic W. Massaro, Joanna Light. 2249-2252 [doi]
- Low resource lip finding and tracking algorithm for embedded devicesJesus F. Guitarte Perez, Klaus Lukas, Alejandro F. Frangi. 2253-2256 [doi]
- Detection and separation of speech segment using audio and video information fusionFutoshi Asano, Yoichi Motomura, Hideki Asoh, Takashi Yoshimura, Naoyuki Ichimura, Kiyoshi Yamamoto, Nobuhiko Kitawaki, Satoshi Nakamura. 2257-2260 [doi]
- Resynthesis of 3d tongue movements from facial dataOlov Engwall, Jonas Beskow. 2261-2264 [doi]
- Acquiring lexical information from multilevel temporal annotationsThorsten Trippel, Felix Sasaki, Benjamin Hell, Dafydd Gibbon. 2265-2268 [doi]
- LUCIA a new italian talking-head based on a modified cohen-massaro s labial coarticulation modelPiero Cosi, Andrea Fusaro, Graziano Tisato. 2269-2272 [doi]
- A visual context-aware multimodal system for spoken language processingNiloy Mukherjee, Deb Roy. 2273-2276 [doi]
- Maximum entropy good-turing estimator for language modelingJuan P. Piantanida, Claudio Estienne. 2277-2280 [doi]
- Exploiting order-preserving perfect hashing to speedup n-gram language model lookaheadXiaolong Li, Yunxin Zhao. 2281-2284 [doi]
- Stem-based maximum entropy language models for inflectional languagesDimitris Oikonomidis, Vassilios Digalakis. 2285-2288 [doi]
- Combination of a hidden tag model and a traditional n-gram model: a case study in czech speech recognitionPavel Krbec, Petr Podveský, Jan Hajic. 2289-2292 [doi]
- Unlimited vocabulary speech recognition based on morphs discovered in an unsupervised mannerVesa Siivola, Teemu Hirsimäki, Mathias Creutz, Mikko Kurimo. 2293-2296 [doi]
- Evaluation of the stochastic morphosyntactic language model on a one million word hungarian dictation taskMate Szarvas, Sadaoki Furui. 2297-2300 [doi]
- Locus equations determination using the speechdat(II)Bojan Petek. 2301-2304 [doi]
- A memory-based approach to Cantonese tone recognitionMichael Emonts, Deryle Lonsdale. 2305-2308 [doi]
- Experimental evaluation of the relevance of prosodic features in Spanish using machine learning techniquesDavid Escudero Mancebo, Valentín Cardeñoso-Payo, Antonio Bonafonte. 2309-2312 [doi]
- Dominance spectrum based v/UV classification and f_0 estimationTomohiro Nakatani, Toshio Irino, Parham Zolfaghari. 2313-2316 [doi]
- Analysis and modeling of f_0 contours of portuguese utterances based on the command-response modelHiroya Fujisaki, Shuichi Narusawa, Sumio Ohno, Diamantino Freitas. 2317-2320 [doi]
- Covariation and weighting of harmonically decomposed streams for ASRPhilip J. B. Jackson, David M. Moreno, Martin J. Russell, Javier Hernando. 2321-2324 [doi]
- Automatic title generation for Chinese spoken documents considering the special structure of the languageLin-Shan Lee, Shun-Chuan Chen. 2325-2328 [doi]
- Statistical speech-to-speech translation with multilingual speech recognition and bilingual-chunk parsingBo Xu, Shuwu Zhang, Chengqing Zong. 2329-2332 [doi]
- Automatic extraction of bilingual chunk lexicon for spoken language translationLimin Du, Boxing Chen. 2333-2336 [doi]
- Multi-scale document expansion in English-Mandarin cross-language spoken document retrievalWai Kit Lo, Yuk-Chi Li, Gina-Anne Levow, Hsin-Min Wang, Helen M. Meng. 2337-2340 [doi]
- Mandarin speech prosody: issues, pitfalls and directionsChiu-yu Tseng. 2341-2344 [doi]
- A contrastive investigation of standard Mandarin and accented MandarinAijun Li, Xia Wang. 2345-2348 [doi]
- Emotion control of Chinese speech synthesis in natural environmentJianhua Tao. 2349-2352 [doi]
- Optimality criteria in inverse problems for tongue-jaw interactionAlexander S. Leonov, Victor N. Sorokin. 2353-2356 [doi]
- FEM analysis based on 3-d time-varying vocal tract shapeKoji Sasaki, Nobuhiro Miki, Yoshikazu Miyanaga. 2357-2360 [doi]
- Consideration of muscle co-contraction in a physiological articulatory modelJianwu Dang, Kiyoshi Honda. 2361-2364 [doi]
- Robust techniques for pre- and post-surgical voice analysisClaudia Manfredi, Giorgio Peretti. 2365-2368 [doi]
- Analysis of lossy vocal tract models for speech productionKarl Schnell, Arild Lacroix. 2369-2372 [doi]
- Estimation of vocal noise in running speech by means of bi-directional double linear predictionFrédéric Bettens, Francis Grenez, Jean Schoentgen. 2377-2380 [doi]
- Visualisation of the vocal tract based on estimation of vocal area functions and formant frequenciesAbdulhussain E. Mahdi. 2381-2384 [doi]
- Reproducing laryngeal mechanisms with a two-mass modelDenisse Sciamarella, Christophe d Alessandro. 2385-2388 [doi]
- Methods for estimation of glottal pulses waveforms exciting voiced speechMilan Bostik, Milan Sigmund. 2389-2392 [doi]
- Acoustic modeling of american English lateral approximantsZhaoyan Zhang, Carol Y. Espy-Wilson, Mark Tiede. 2393-2396 [doi]
- Translation and rotation of the cricothyroid joint revealed by phonation-synchronized high-resolution MRISayoko Takano, Kiyoshi Honda, Shinobu Masaki, Yasuhiro Shimada, Ichiro Fujimoto. 2397-2400 [doi]
- GMM-based voice conversion applied to emotional speech synthesisHiromichi Kawanami, Yohei Iwami, Tomoki Toda, Hiroshi Saruwatari, Kiyohiro Shikano. 2401-2404 [doi]
- Probability models of formant parameters for voice conversionDimitrios Rentzos, Saeed Vaseghi, Qin Yan, Ching-Hsiang Ho, Emir Turajlic. 2405-2408 [doi]
- Perceptually weighted linear transformations for voice conversionHui Ye, Steve Young. 2409-2412 [doi]
- Voice conversion with smoothed GMM and MAP adaptationYining Chen, Min Chu, Eric Chang, Jia Liu, Runsheng Liu. 2413-2416 [doi]
- A system for voice conversion based on adaptive filtering and line spectral frequency distance optimization for text-to-speech synthesisÖzgül Salor, Mübeccel Demirekler, Bryan L. Pellom. 2417-2420 [doi]
- Speaker conversion in ARX-based source-formant type speech synthesisHiroki Mori, Hideki Kasuya. 2421-2424 [doi]
- Implementing an SSML compliant concatenative TTS systemAndrew P. Breen, Steve Minnis, Barry Eggleton. 2425-2428 [doi]
- Acoustic variations of focused disyllabic words in Mandarin Chinese: analysis, synthesis and perceptionZhenglai Gu, Hiroki Mori, Hideki Kasuya. 2429-2432 [doi]
- An approach to common acoustical pole and zero modeling of consecutive periods of voiced speechPedro J. Quintana-Morales, Juan L. Navarro-Mesa. 2433-2436 [doi]
- Estimating the vocal-tract area function and the derivative of the glottal wave from a speech signalHuiqun Deng, Michael P. Beddoes, Rabab Kreidieh Ward, Murray Hodgson. 2437-2440 [doi]
- Glottal closure instant synchronous sinusoidal model for high quality speech analysis/synthesisParham Zolfaghari, Tomohiro Nakatani, Toshio Irino, Hideki Kawahara, Fumitada Itakura. 2441-2444 [doi]
- Mixed physical modeling techniques applied to speech productionMatti Karjalainen. 2445-2448 [doi]
- An expandable web-based audiovisual text-to-speech synthesis systemSascha Fagel, Walter F. Sendlmeier. 2449-2452 [doi]
- A reconstruction of farkas kempelen s speaking machineP. Nikleczy, Gábor Olaszy. 2453-2456 [doi]
- Acoustic model selection and voice quality assessment for HMM-based Mandarin speech synthesisWentao Gu, Keikichi Hirose. 2457-2460 [doi]
- Modeling of various speaking styles and emotions for HMM-based speech synthesisJunichi Yamagishi, Koji Onishi, Takashi Masuko, Takao Kobayashi. 2461-2464 [doi]
- Towards the development of a brazilian portuguese text-to-speech system based on HMMR. da S. Maia, Heiga Zen, Keiichi Tokuda, Tadashi Kitamura, Fernando Gil Vianna Resende Jr.. 2465-2468 [doi]
- Grapheme to phoneme conversion and dictionary verification using graphonemesPaul Vozila, Jeff Adams, Yuliya Lobacheva, Ryan Thomas. 2469-2472 [doi]
- Improving the accuracy of pronunciation prediction for unit selection TTSJustin Fackrell, Wojciech Skut, Kathrine Hammervold. 2473-2476 [doi]
- Detection of list-type sentencesTaniya Mishra, Esther Klabbers, Jan P. H. van Santen. 2477-2480 [doi]
- A new pitch synchronous time domain phoneme recognizer using component analysis and pitch clusteringRamon Prieto, Jing Jiang, Chi-Ho Choi. 2481-2484 [doi]
- Mixed-lingual spoken word recognition by using VQ codebook sequences of variable length segmentsHiroaki Kojima, Kazuyo Tanaka. 2485-2488 [doi]
- Low memory acoustic models for HMM based speech recognitionTommi Lahti, Olli Viikki, Marcel Vasilache. 2489-2492 [doi]
- Nearest-neighbor search algorithms based on subcodebook selection and its application to speech recognitionJosé A. R. Fonollosa. 2493-2496 [doi]
- Non-linear maximum likelihood feature transformation for speech recognitionMohamed Kamal Omar, Mark Hasegawa-Johnson. 2497-2500 [doi]
- Automatic generation of context-independent variable parameter models using successive state and mixture splittingSoo-Young Suk, Ho-Youl Jung, Hyun-Yeol Chung. 2501-2504 [doi]
- Data driven generation of broad classes for decision tree construction in acoustic modelingAndrej Zgank, Zdravko Kacic, Bogomir Horvat. 2505-2508 [doi]
- An efficient integrated gender detection scheme and time mediated averaging of gender dependent acoustic modelsPeder A. Olsen, Satya Dharanipragada. 2509-2512 [doi]
- Syllable-based acoustic modeling for Japanese spontaneous speech recognitionJun Ogata, Yasuo Ariki. 2513-2516 [doi]
- Cross-stream observation dependencies for multi-stream speech recognitionÖzgür Çetin, Mari Ostendorf. 2517-2520 [doi]
- Pruning transitions in a hidden Markov model with optimal brain surgeonBrian Kan-Wing Mak, Kin-Wah Chan. 2521-2524 [doi]
- Using pitch frequency information in speech recognitionMathew Magimai-Doss, Todd A. Stephenson, Hervé Bourlard. 2525-2528 [doi]
- Hidden feature models for speech recognition using dynamic Bayesian networksKaren Livescu, James R. Glass, Jeff Bilmes. 2529-2532 [doi]
- An efficient viterbi algorithm on DBNsWei Hu, Yimin Zhang, Qian Diao, Shan Huang. 2533-2536 [doi]
- Speech recognition based on syllable recoveryLi Zhang, William H. Edmondson. 2537-2540 [doi]
- HARTFEX: a multi-dimensional system of HMM based recognisers for articulatory features extractionTarek Abu-Amer, Julie Carson-Berndsen. 2541-2544 [doi]
- Automatic baseform generation from acoustic dataBenoît Maison. 2545-2548 [doi]
- Data-driven pronunciation modeling for ASR using acoustic subword unitsThurid Spiess, Britta Wrede, Gernot A. Fink, Franz Kummert. 2549-2552 [doi]
- Time is of the essence - dynamic approaches to spoken languageSteven Greenberg. 2553-2556 [doi]
- Spectro-temporal interactions in auditory and auditory-visual speech processingKen W. Grant, Steven Greenberg. 2557-2560 [doi]
- Brain imaging correlates of temporal quantization in spoken languageDavid Poeppel. 2561-2564 [doi]
- Temporal aspects of articulatory controlElliot Saltzman. 2565-2568 [doi]
- The temporal organisation of speech as gauged by speech synthesisBrigitte Zellner Keller. 2569-2572 [doi]
- Localized spectro-temporal features for automatic speech recognitionMichael Kleinschmidt. 2573-2576 [doi]
- Modulation spectral filtering of speechLes E. Atlas. 2577-2580 [doi]
- A comparison of the data requirements of automatic speech recognition systems and human listenersRoger K. Moore. 2581-2584 [doi]
- Modeling linguistic features in speech recognitionMin Tang, Stephanie Seneff, Victor W. Zue. 2585-2588 [doi]
- Impact of audio segmentation and segment clustering on automated transcription accuracy of large spoken archivesBhuvana Ramabhadran, Jing Huang, Upendra V. Chaudhari, Giridharan Iyengar, Harriet J. Nock. 2589-2592 [doi]
- Learning linguistically valid pronunciations from acoustic dataFrançoise Beaufays, Ananth Sankar, Shaun Williams, Mitch Weintraub. 2593-2596 [doi]
- Improvement of non-native speech recognition by effectively modeling frequently observed pronunciation habitsNobuaki Minematsu, Koichi Osaki, Keikichi Hirose. 2597-2600 [doi]
- Non-audible murmur recognitionYoshitaka Nakajima, Hideki Kashioka, Kiyohiro Shikano, Nick Campbell. 2601-2604 [doi]
- Variable length mixtures of inverse covariancesVincent Vanhoucke, Ananth Sankar. 2605-2608 [doi]
- Semi-tied full deviation matrices for laplacian density modelsChristoph Neukirchen. 2609-2612 [doi]
- Acoustic modeling with mixtures of subspace constrained exponential modelsKarthik Visweswariah, Scott Axelrod, Ramesh A. Gopinath. 2613-2616 [doi]
- Discriminative estimation of subspace precision and mean (SPAM) modelsVaibhava Goel, Scott Axelrod, Ramesh A. Gopinath, Peder A. Olsen, Karthik Visweswariah. 2617-2620 [doi]
- Model-integration rapid training based on maximum likelihood for speech recognitionShinichi Yoshizawa, Kiyohiro Shikano. 2621-2624 [doi]
- On the use of kernel PCA for feature extraction in speech recognitionAmaro Lima, Heiga Zen, Yoshihiko Nankaku, Chiyomi Miyajima, Keiichi Tokuda, Tadashi Kitamura. 2625-2628 [doi]
- Speaker modeling from selected neighbors applied to speaker recognitionYassine Mami, Delphine Charlet. 2629-2632 [doi]
- Who knows carl bildt? - and what if you don t?Elisabeth Zetterholm, Kirk P. H. Sullivan, James Green, Erik J. Eriksson, Jan van Doorn, Peter E. Czigler. 2633-2636 [doi]
- Improving the competitiveness of discriminant neural networks in speaker verificationCarlos Vivaracho-Pascual, Javier Ortega-Garcia, Luis Alonso Romero, Q. Isaac Moro-Sancho. 2637-2640 [doi]
- On the fusion of dissimilarity-based classifiers for speaker identificationTomi Kinnunen, Ville Hautamäki, Pasi Fränti. 2641-2644 [doi]
- Robust speaker identification using posterior union modelsJi Ming,