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IEEE Transactions on Speech and Audio Processing, Volume 10
Volume 10, Number 1, January 2002
- Rongshan Yu, Chi Chung Ko:
A warped linear-prediction-based subband audio coding algorithm. 1-8 - Hui Jiang, Li Deng:
A robust compensation strategy for extraneous acoustic variations in spontaneous speech recognition. 9-17 - Eric A. Durant, Gregory H. Wakefield:
Efficient model fitting using a genetic algorithm: pole-zero approximations of HRTFs. 18-27 - Chao-Shih Huang, Hsiao-Chuan Wang, Chin-Hui Lee:
Correction to "An SNR-incremental stochastic matching algorithm for noisy speech recognition". 28
Volume 10, Number 2, February 2002
- Mark J. F. Gales:
Maximum likelihood multiple subspace projections for hidden Markov models. 37-47 - Takeshi Yamada, Satoshi Nakamura, Kiyohiro Shikano:
Distant-talking speech recognition based on a 3-D Viterbi search using a microphone array. 48-56 - Yifan Gong:
Noise-dependent Gaussian mixture classifiers for robust rejection decision. 57-64 - Shrikanth S. Narayanan, Alexandros Potamianos:
Creating conversational interfaces for children. 65-78 - Mohamed Afify, Olivier Siohan, Chin-Hui Lee:
Upper and lower bounds on the mean of noisy speech: application to minimax classification. 79-88 - Rita Singh, Bhiksha Raj, Richard M. Stern:
Automatic generation of subword units for speech recognition systems. 89-99 - Geert Rombouts, Marc Moonen:
A sparse block exact affine projection algorithm. 100-108 - M. Marzinzik, Birger Kollmeier:
Speech pause detection for noise spectrum estimation by tracking power envelope dynamics. 109-118 - Johan Hellgren:
Analysis of feedback cancellation in hearing aids with Filtered-x LMS and the direct method of closed loop identification. 119-131
Volume 10, Number 3, March 2002
- Liang Gu, Kenneth Rose:
Substate tying with combined parameter training and reduction in tied-mixture HMM design. 137-145 - Qi Li, Jinsong Zheng, Augustine Tsai, Qiru Zhou:
Robust endpoint detection and energy normalization for real-time speech and speaker recognition. 146-157 - Néstor Becerra Yoma, Miguel Villar Fernandez:
Speaker verification in noise using a stochastic version of the weighted Viterbi algorithm. 158-166 - Zihou Meng, Kimihiro Sakagami, Masayuki Morimoto, Guoan Bi, Alex ChiChung Kot:
Extending the sound impulse response of room using extrapolation. 167-172 - Jaco Vermaak, Christophe Andrieu, Arnaud Doucet, Simon J. Godsill:
Particle methods for Bayesian modeling and enhancement of speech signals. 173-185 - Tom Bäckström, Paavo Alku, Erkki Vilkman:
Time-domain parameterization of the closing phase of glottal airflow waveform from voices over a large intensity range. 186-192 - Imed Zitouni:
A hierarchical language model based on variable-length class sequences: the MCnnu approach. 193-198
Volume 10, Number 4, May 2002
- William M. Campbell, Khaled T. Assaleh, Charles C. Broun:
Speaker recognition with polynomial classifiers. 205-212 - Sven Johansson, Sven Nordebo, Ingvar Claesson:
Convergence analysis of a twin-reference complex least-mean-squares algorithm. 213-221 - Nam Phamdo, Udar Mittal:
A joint source-channel speech coder using hybrid digital-analog (HDA) modulation. 222-231 - Darryl W. Purnell, Elizabeth C. Botha:
Improved generalization of MCE parameter estimation with application to speech recognition. 232-239
Volume 10, Number 5, July 2002
- Stefan Gustafsson, Rainer Martin, Peter Jax, Peter Vary:
A psychoacoustic approach to combined acoustic echo cancellation and noise reduction. 245-256 - Tomas Gänsler, Jacob Benesty:
New insights into the stereophonic acoustic echo cancellation problem and an adaptive nonlinearity solution. 257-267 - Jen-Tzung Chien:
Quasi-Bayes linear regression for sequential learning of hidden Markov models. 268-278 - Ahmed M. Abdelatty Ali, Jan Van der Spiegel, Paul Mueller:
Robust auditory-based speech processing using the average localized synchrony detection. 279-292 - George Tzanetakis, Perry R. Cook:
Musical genre classification of audio signals. 293-302 - Berlin Chen, Hsin-Min Wang, Lin-Shan Lee:
Discriminating capabilities of syllable-based features and approaches of utilizing them for voice retrieval of speech information in Mandarin Chinese. 303-314 - Marie A. Roch, Richard R. Hurtig:
The integral decode: a smoothing technique for robust HMM-based speaker recognition. 315-324 - Yuan-Hao Huang, Tzi-Dar Chiueh:
A new audio coding scheme using a forward masking model and perceptually weighted vector quantization. 325-335
Volume 10, Number 6, September 2002
- David Burshtein, Sharon Gannot:
Speech enhancement using a mixture-maximum model. 341-351 - Lucas C. Parra, Christopher V. Alvino:
Geometric source separation: merging convolutive source separation with geometric beamforming. 352-362 - Ran D. Zilca:
Text-independent speaker verification using utterance level scoring and covariance modeling. 363-370 - Ivan Magrin-Chagnolleau, Geoffrey Durou, Frédéric Bimbot:
Application of time-frequency principal component analysis to text-independent speaker identification. 371-378 - Gerald Schuller, Bin Yu, Dawei Huang, Bernd Edler:
Perceptual audio coding using adaptive pre- and post-filters and lossless compression. 379-390 - Ronald M. Aarts, Roy Irwan, Augustus J. E. M. Janssen:
Efficient tracking of the cross-correlation coefficient. 391-402 - Ji Ming, Peter Jancovic, Francis Jack Smith:
Robust speech recognition using probabilistic union models. 403-414 - Lutz Welling, Hermann Ney, Stephan Kanthak:
Speaker adaptive modeling by vocal tract normalization. 415-426
Volume 10, Number 7, October 2002
- Mukund Padmanabhan, George Saon, Jing Huang, Brian Kingsbury, Lidia Mangu:
Automatic speech recognition performance on a voicemail transcription task. 433-442 - Néstor Becerra Yoma, Jorge F. Silva:
MAP speaker adaptation of state duration distributions for speech recognition. 443-450 - Juan Manuel Huerta:
Alignment-based codeword-dependent cepstral normalization. 451-459 - Stephen Cox, Srinandan Dasmahapatra:
High-level approaches to confidence estimation in speech recognition. 460-471 - C. Chandra Sekhar, B. Yegnanarayana:
A constraint satisfaction model for recognition of stop consonant-vowel (SCV) utterances. 472-480 - Fu-Chiang Chou, Chiu-yu Tseng, Lin-Shan Lee:
A set of corpus-based text-to-speech synthesis technologies for Mandarin Chinese. 481-494 - Frank Baumgarte:
Improved audio coding using a psychoacoustic model based on a cochlear filter bank. 495-503 - Lie Lu, Hong-Jiang Zhang, Hao Jiang:
Content analysis for audio classification and segmentation. 504-516 - Khaled A. Mayyas:
Stereophonic acoustic echo cancellation using lattice orthogonalization. 517-525
Volume 10, Number 8, November 2002
- Harry Printz, Isabel Trancoso:
Editorial. 529-530 - Eric Chang, Frank Seide, Helen M. Meng, Zhuoran Chen, Yu Shi, Yuk-Chi Li:
A system for spoken query information retrieval on mobile devices. 531-541 - Satya Dharanipragada, Salim Roukos:
A multistage algorithm for spotting new words in speech. 542-550 - Sabine Deligne, Satya Dharanipragada, Ramesh A. Gopinath, Benoît Maison, Peder A. Olsen, Harry Printz:
A robust high accuracy speech recognition system for mobile applications. 551-561 - Imre Varga, Stefanie Aalburg, Bernt Andrassy, Sergey Astrov, Josef G. Bauer, Christophe Beaugeant, Christian Geißler, Harald Höge:
ASR in mobile phones - an industrial approach. 562-569 - Alexis Bernard, Abeer Alwan:
Low-bitrate distributed speech recognition for packet-based and wireless communication. 570-579 - Constantinos Boulis, Mari Ostendorf, Eve A. Riskin, Scott Otterson:
Graceful degradation of speech recognition performance over packet-erasure networks. 580-590 - Hong Kook Kim, Richard V. Cox, Richard C. Rose:
Performance improvement of a bitstream-based front-end for wireless speech recognition in adverse environments. 591-604 - Li Deng, Kuansan Wang, Alex Acero, Hsiao-Wuen Hon, Jasha Droppo, Constantinos Boulis, Ye-Yi Wang, Derek Jacoby, Milind Mahajan, Ciprian Chelba, Xuedong Huang:
Distributed speech processing in miPad's multimodal user interface. 605-619 - Bruno Bessette, Redwan Salami, Roch Lefebvre, Milan Jelinek, J. Rotola-Pukkila, Janne Vainio, Hannu Mikkola, Kari Järvinen:
The adaptive multirate wideband speech codec (AMR-WB). 620-636 - Antonio Servetti, Juan Carlos De Martin:
Perception-based partial encryption of compressed speech. 637-643 - Jhing-Fa Wang, Jia-Ching Wang, Han-Chiang Chen, Tai-Lung Chen, Chin-Chan Chang, Ming-Chi Shih:
Chip design of portable speech memopad suitable for persons with visual disabilities. 644-658
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