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ICASSP 2002: Orlando, Florida, USA
- Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2002, May 13-17 2002, Orlando, Florida, USA. IEEE 2002, ISBN 0-7803-7402-9
- Tatsuya Kawahara, Masahiro Hasegawa:
Automatic indexing of lecture speech by extracting topic-independent discourse markers. 1-4 - Jay Billa
, Mohammed Noamany, Amit Srivastava, Daben Liu, Rebecca Stone, Jinxi Xu, John Makhoul, Francis Kubala:
Audio Indexing of Arabic broadcast news. 5-8 - Chiori Hori, Sadaoki Furui, Robert G. Malkin, Hua Yu, Alex Waibel:
Automatic speech summarization applied to English broadcast news speech. 9-12 - Claude Barras, Alexandre Allauzen, Lori Lamel, Jean-Luc Gauvain:
Transcribing audio-video archives. 13-16 - Cosmin Popovici, Marco Andorno, Pietro Laface, Luciano Fissore, Mario Nigra, Claudio Vair:
Learning new user formulations in automatic Directory Assistance. 17-20 - Premkumar Natarajan
, Rohit Prasad, Richard M. Schwartz, John Makhoul:
A scalable architecture for Directory Assistance automation. 21-24 - Imed Zitouni, Hong-Kwang Jeff Kuo, Chin-Hui Lee:
Combination of boosting and discriminative training for natural language call steering systems. 25-28 - Marie Rochery, Robert E. Schapire, Mazin G. Rahim, Narendra K. Gupta, Giuseppe Riccardi, Srinivas Bangalore, Hiyan Alshawi, Shona Douglas:
Combining prior knowledge and boosting for call classification in spoken language dialogue. 29-32 - Owen Kimball, Rukmini Iyer, Herbert Gish, Scott Miller, Fred Richardson:
Extracting descriptive noun phrases from conversational speech. 33-36 - Roger Argiles Solsona, Eric Fosler-Lussier, Hong-Kwang Jeff Kuo, Alexandros Potamianos, Imed Zitouni:
Adaptive language models for spoken dialogue systems. 37-40 - Ye-Yi Wang, Alex Acero:
Evaluation of spoken language grammar learning in the ATIS domain. 41-44 - Olivier Pietquin, Steve Renals
:
ASR system modeling for automatic evaluation and optimization of dialogue systems. 45-48 - Shai Fine, George Saon
, Ramesh A. Gopinath:
Digit recognition in noisy environments via a sequential GMM/SVM system. 49-52 - Brian Kingsbury, George Saon
, Lidia Mangu, Mukund Padmanabhan, Ruhi Sarikaya:
Robust speech recognition in Noisy Environments: The 2001 IBM spine evaluation system. 53-56 - Jasha Droppo, Alex Acero, Li Deng:
Uncertainty decoding with SPLICE for noise robust speech recognition. 57-60 - Trausti T. Kristjansson, Brendan J. Frey:
Accounting for uncertainity in observations: A new paradigm for Robust Automatic Speech Recognition. 61-64 - Kalle J. Palomäki, Guy J. Brown, Jon P. Barker:
Missing data speech recognition in reverberant conditions. 65-68 - Peter Jancovic, Ji Ming:
Combining the union model and missing feature method to improve noise robustness in ASR. 69-72 - Peng Ding, Zhenbiao Chen, Yang Liu, Bo Xu:
Asymmetrical Support Vector Machines and applications in speech processing. 73-76 - Nathan D. Smith, Mark J. F. Gales:
Using SVMS and discriminative models for speech recognition. 77-80 - Mohamed Kamal Omar, Mark Hasegawa-Johnson
:
Maximum mutual information based acoustic-features representation of phonological features for speech recognition. 81-84 - Ricardo de Córdoba
, Philip C. Woodland, Mark J. F. Gales:
Improved cross-task recognition using MMIE training. 85-88 - Anton Lilchododev, Yuqing Gao:
Direct models for phoneme recognition. 89-92 - Geoffrey Zweig, Jeff A. Bilmes, Thomas Richardson, Karim Filali, Karen Livescu
, Peng Xu, Kirk Jackson, Yigal Brandman, Eric D. Sandness, Eva Holtz, Jerry Torres, Bill Byrne:
Structurally discriminative graphical models for automatic speech recognition - results from the 2001 Johns Hopkins Summer Workshop. 93-96 - Qi Li, Biing-Hwang Juang:
A new algorithm for fast discriminative training. 97-100 - Yik-Cheung Tam, Brian Mak:
An alternative approach of finding competing hypotheses for better minimum classification error training. 101-104 - Daniel Povey, Philip C. Woodland:
Minimum Phone Error and I-smoothing for improved discriminative training. 105-108 - Carsten Meyer:
Utterance-level boosting of HMM speech recognizers. 109-112 - Hui Jiang, Olivier Siohan, Frank K. Soong, Chin-Hui Lee:
A dynamic in-search discriminative training approach for large vocabulary speech recognition. 113-116 - Tomoko Matsui, Frank K. Soong, Biing-Hwang Juang:
Classifier design for verification of multi-class recognition decision. 117-120 - Delphine Charlet:
Speaker indexing for retrieval of voicemail messages. 121-124 - Brett Y. Smolenski, Robert E. Yantorno, Daniel S. Benincasa, Stanley J. Wenndt:
Co-channel speaker segment separation. 125-128 - Jack McLaughlin, Douglas A. Reynolds:
Speaker detection and tracking for telephone transactions. 129-132 - Yifan Gong:
Noise-robust open-set speaker recognition using noise-dependent Gaussian mixture classifier. 133-136 - Nobuaki Minematsu, Mariko Sekiguchi, Keikichi Hirose:
Automatic estimation of one's age with his/her speech based upon acoustic modeling techniques of speakers. 137-140 - Frederick Weber, Linda Manganaro, Barbara Peskin, Elizabeth Shriberg:
Using prosodic and lexical information for speaker identification. 141-144 - Qin Jin, Tanja Schultz, Alex Waibel:
Speaker identification using multilingual phone strings. 145-148 - Walter D. Andrews, Mary A. Kohler, Joseph P. Campbell, John J. Godfrey, Jaime Hernandez-Cordero
:
Gender-dependent phonetic refraction for speaker recognition. 149-152 - Scott Axelrod:
Speaker identification using online, frame dependent, and diffusive variance adaptation. 153-156 - Say Wei Foo, Eng Guan Lim:
Speaker recognition using adaptively boosted decision tree classifier. 157-160 - William M. Campbell:
Generalized linear discriminant sequence kernels for speaker recognition. 161-164 - Hervé Taddei, Sean A. Ramprashad, Carl-Erik W. Sundberg, Hui-Ling Lou:
Mode adaptive Unequal Error Protection for transform predictive speech and audio coders. 165-168 - Masahiro Serizawa, Hironori Ito:
A packet loss recovery method using packet arrived behind the playout time for CELP decoding. 169-172 - Jonas Lindblom, Per Hedelin:
Packet loss concealment based on sinusoidal extrapolation. 173-176 - Xin Zhong, Biing-Hwang Juang:
Multiple description speech coding with diversities. 177-180 - Christoph Erdmann, David Bauer, Peter Vary:
Pyramid CELP: Embedded speech coding for packet communications. 181-184 - M. K. Vinay, P. V. Suresh Babu:
Context-based error recovery technique for GSM AMR speech codec. 185-188 - Kaisheng Yao, Kuldip K. Paliwal
, Satoshi Nakamura:
Noise adaptive speech recognition in time-varying noise based on sequential kullback proximal algorithm. 189-192 - Kimmo Pärssinen, Petri Salmela, Mikko Harju, Imre Kiss:
Comparing Jacobian adaptation with cepstral mean normalization and parallel model combination for noise robust speech recognition. 193-196 - Hiroshi Shimodaira, Nobuyoshi Sakai, Mitsuru Nakai, Shigeki Sagayama:
Jacobian joint adaptation to noise, channel and vocal tract length. 197-200 - Christophe Cerisara, Jean-Claude Junqua, Luca Rigazio:
Dynamic estimation of a noise over estimation factor for Jacobian-based adaptation. 201-204 - Robert W. Morris, Michael E. Deisher:
Efficient second-order adaptation for large vocabulary distributed speech recognition. 205-208 - Hong Kook Kim, Richard C. Rose:
Cepstrum-domain model combination based on decomposition of speech and noise for noisy speech recognition. 209-212 - Ananth Sankar, Ashvin Kalman:
Automatic confidence score mapping for adapted speech recognition systems. 213-216 - Takaharu Sato, Muhammad Ghulam, Takashi Fukuda, Tsuneo Nitta:
Confidence scoring for accurate HMM-based word recognition by using SM-based monophone score normalization. 217-220 - Jacques Duchateau, Kris Demuynck, Patrick Wambacq:
Confidence scoring based on backward language models. 221-224 - Kadri Hacioglu, Wayne H. Ward:
A concept graph based confidence measure. 225-228 - Yi-Chung Lin, Huei-Ming Wang:
Probabilistic integration of multiple confidence measures and context information for concept verification. 229-232 - Sameer S. Pradhan, Wayne H. Ward:
Estimating semantic confidence for spoken dialogue systems. 233-236 - Peter Jax, Peter Vary:
An upper bound on the quality of artificial bandwidth extension of narrowband speech signals. 237-240 - Dar Ghulam Raza, Cheung-Fat Chan
:
Enhancing quality of CELP coded speech via wideband extension by using voicing GMM interpolation and HNM re-synthesis. 241-244 - Mitsuhiro Hosoki, Takayuki Nagai, Akira Kurematsu:
Speech signal band width extension and noise removal using subband HMN. 245-248 - Ilyas Potamitis, Nikos Fakotakis, George Kokkinakis:
Gender-dependent and speaker-dependent speech enhancement. 249-252 - Rainer Martin
:
Speech enhancement using MMSE short time spectral estimation with gamma distributed speech priors. 253-256 - Thomas F. Quatieri, Robert B. Dunn:
Speech enhancement based on auditory spectral change. 257-260 - Wesley Pereira, Peter Kabal:
Improved spectral tracking using interpolated linear prediction parameters. 261-264 - Phu Chien Nguyen, Masato Akagi:
Improvement of the restricted temporal decomposition method for line spectral frequency parameters. 265-268 - Sunil Shukla, Ali Erdem Ertan, Thomas P. Barnwell III:
Circular LPC modeling and constant pitch transform for accurate speech analysis and high quality speech synthesis. 269-272 - Parham Aarabi, Albarz Mahdavi:
The relation between speech segment selectivity and source localization accuracy. 273-276 - Khosrow Lashkari, Toshio Miki:
Joint optimization of model and excitation in parametric speech coders. 277-280 - Ian C. Bruce, Neel V. Karkhanis, Eric D. Young, Murray B. Sachs:
Robust formant tracking in noise. 281-284 - Tsuneo Kato, Masaki Naito, Tohru Shimizu:
Noise-robust cellular phone speech recognition using codec-adapted speech and noise models. 285-288 - Marco Matassoni, Maurizio Omologo, Alfiero Santarelli, Piergiorgio Svaizer
:
On the joint use of noise reduction and MLLR adaptation for in-car hands-free speech recognition. 289-292 - Rafid A. Sukkar, Rathi Chengalvarayan, John Jacob:
Unified speech recognition for the landline and wireless environments. 293-296 - Jitendra Ajmera, Iain McCowan, Hervé Bourlard:
Robust HMM-based speech/music segmentation. 297-300 - Mauro Cettolo:
Porting an audio partitioner across domains. 301-304 - Georg F. Meyer
, Jeff Mulligan:
Continuous audio-visual digit recognition using decision fusion. 305-308 - Satoshi Nakamura, Ken'ichi Kumatani, Satoshi Tamura:
Robust bi-modal speech recognition based on state synchronous modeling and stream weight optimization. 309-312 - Hervé Glotin:
Enhanced posteriors bias prediction for robust multi-stream ASR combining voicing and estimate reliabilities. 313-316 - Zoran Cvetkovic, Baltasar Beferull-Lozano, Andreas Buja:
Robust phoneme discrimination using acoustic waveforms. 317-320 - Jianping Zhang, Wayne H. Ward, Bryan L. Pellom:
Phone based voice activity detection using online Bayesian adaptation with conjugate normal distributions. 321-324 - Hong-Kwang Jeff Kuo, Eric Fosler-Lussier, Hui Jiang, Chin-Hui Lee:
Discriminative training of language models for speech recognition. 325-328 - Tomohiro Tanaka, Takao Kobayashi, Dhany Arifianto, Takashi Masuko:
Fundamental frequency estimation based on instantaneous frequency amplitude spectrum. 329-332 - Xuejing Sun:
Pitch determination and voice quality analysis using Subharmonic-to-Harmonic Ratio. 333-336 - Andrei Jefremov, W. Bastiaan Kleijn
:
Spline-based continuous-time pitch estimation. 337-340 - Wenyao Zhang, Gang Xu, Yuguo Wang:
Pitch estimation based on Circular AMDF. 341-344 - Dmitry E. Terez:
Robust pitch determination using nonlinear state-space embedding. 345-348 - Anastasis Kounoudes
, Patrick A. Naylor
, Mike Brookes
:
The DYPSA algorithm for estimation of glottal closure instants in voiced speech. 349-352 - Holger Quast, Olaf Schreiner, Manfred R. Schroeder:
Robust pitch tracking in the car environment. 353-356 - Yih-Ru Wang, I-Je Wong, Teng-Chun Tsao:
A statistical pitch detection algorithm. 357-360 - Kavita Kasi, Stephen A. Zahorian:
Yet Another Algorithm for Pitch Tracking. 361-364 - Douglas J. Nelson:
Recovery of the harmonic fundamental from the mixed partial derivatives of the STFT phase. 365-368 - Mingyang Wu, DeLiang Wang, Guy J. Brown:
A multi-pitch tracking algorithm for noisy speech. 369-372 - Jeih-Weih Hung, Lin-Shan Lee:
Data-driven temporal filters for robust features in speech recognition obtained via Minimum Classification Error (MCE). 373-376 - Dimitrios Dimitriadis, Petros Maragos, Alexandros Potamianos:
Modulation features for speech recognition. 377-380 - Brian Mak, Yik-Cheung Tam, Peter Qi Li:
Discriminative auditory features for robust speech recognition. 381-384 - Chung-Hsien Wu, Yu-Hsien Chiu, Huigan Lim:
Perceptual speech modeling for noisy speech recognition. 385-388 - Taisuke Ito, Kazuya Takeda, Fumitada Itakura:
Acoustic analysis and recognition of whispered speech. 389-392 - Younes Souilmi, Luca Rigazio, Patrick Nguyen, David Kryze, Jean-Claude Junqua:
Blind channel estimation based on speech correlation structure. 393-396 - Kiyoaki Aikawa, Kentaro Ishizuka:
Noise-robust speech recognition using a new spectral estimation method "PHASOR". 397-400 - Ángel de la Torre, José C. Segura, M. Carmen Benítez, Antonio M. Peinado, Antonio J. Rubio:
Non-linear transformations of the feature space for robust Speech Recognition. 401-404 - Chris Pal, Brendan J. Frey, Trausti T. Kristjansson:
Noise robust speech recognition using Gaussian basis functions for non-linear likelihood function approximation. 405-408 - José C. Segura
, M. Carmen Benítez, Ángel de la Torre
, Stéphane Dupont
, Antonio J. Rubio:
VTS residual noise compensation. 409-412 - Qin Yan, Saeed Vaseghi:
A comparative analysis of UK and US English accents in recognition and synthesis. 413-416 - Matthias Eichner, Matthias Wolff, Rüdiger Hoffmann:
Improved duration control for speech synthesis using a multigram language model. 417-420 - Oliver Jokisch, Hongwei Ding, Hans Kruschke:
Towards a multilingual prosody model for text-to-speech. 421-424 - Kazuaki Yoshida, Michiko Kazama, Mikio Tohyama:
Pitch and speech-rate conversion using envelope modulation modeling. 425-428 - Hong-Goo Kang, Hong Kook Kim:
A phase generation method for speech reconstruction from spectral envelope and pitch intervals. 429-432 - Hamid Sheikhzadeh, Etienne Cornu, Robert L. Brennan, Todd Schneider:
Real-time speech synthesis on an ultra low-resource, programmable DSP system. 433-436 - Sadao Hiroya, Masaaki Honda:
Determination of articulatory movements from speech acoustics using an HMM-based speech production model. 437-440 - Jintao Jiang, Abeer Alwan, Lynne E. Bernstein, Edward T. Auer, Patricia A. Keating:
Similarity structure in perceptual and physical measures for visual Consonants across talkers. 441-444 - Jun Huang, Stephen E. Levinson, Donald Davis, Scott Slimon:
Articulatory speech synthesis based upon fluid dynamic principles. 445-448 - Brian Gabelman, Abeer Alwan:
Analysis by synthesis of FM modulation and aspiration noise components in pathological voices. 449-452 - Min Chu, Chun Li, Hu Peng, Eric Chang:
Domain adaptation for TTS systems. 453-456 - Shaw-Hwa Hwang, Cheng-Yu Yei:
The synthesis unit generation algorithm for Mandarin TTS. 457-460 - Ivan Bulyko, Mari Ostendorf, Jeff A. Bilmes:
Robust splicing costs and efficient search with BMM Models for concatenative speech synthesis. 461-464 - Tomoki Toda
, Hisashi Kawai, Minoru Tsuzaki, Kiyohiro Shikano:
Unit selection algorithm for Japanese speech synthesis based on both phoneme unit and diphone unit. 465-468 - Phuay Hui Low, Saeed Vaseghi:
Synthesis of unseen context and spectral and pitch contour smoothing in concatenated text to speech synthesis. 469-472 - Chih-Chung Kuo, Chi-Shiang Kuo:
Speech segment selection for concatenative synthesis based on prosody-aligned distance measure. 473-476 - Çaglayan Erdem, Hans-Georg Zimmermann:
A data-driven method for input feature selection within neural prosody generation. 477-480 - David Escudero Mancebo
, Valentín Cardeñoso-Payo, Antonio Bonafonte
:
Corpus based extraction of quantitative prosodic parameters of stress groups in Spanish. 481-484