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, A. Tsai, Qiru Zhou:
  Robust endpoint detection and energy normalization for real-time speech and speaker recognition. 146-157
  
• Néstor Becerra Yoma, Miguel Villar:
  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
  
• Tomas 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 MC_n^nu approach. 193-198
  

Volume 10, Number 4, May 2002

• W. 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
  
• D. 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
  
• A. 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
  
• R. 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, F. 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 Silva Sánchez:
  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
  
• K. Mayyas:
  Stereophonic acoustic echo cancellation using lattice orthogonalization. 517-525
  

Volume 10, Number 8, November 2002

• Harry Printz, Isabel Trancoso:
  Editorial. 529-530
  
• E. Chang, F. 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
  
• I. Varga, S. Aalburg, B. Andrassy, S. Astrov, J. G. Bauer, C. Beaugeant, C. Geissler, H. Hoge:
  ASR in mobile phones - an industrial approach. 562-569
  
• A. Bernard, A. 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, R. 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, James Droppo, C. Boulis, Ye-Yi Wang, D. Jacoby, Milind Mahajan, Ciprian Chelba, X. D. Huang:
  Distributed speech processing in miPad's multimodal user interface. 605-619
  
• B. Bessette, Redwan Salami, R. Lefebvre, M. Jelinek, J. Rotola-Pukkila, J. Vainio, H. Mikkola, K. Jarvinen:
  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