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IEEE Transactions on Speech and Audio Processing, Volume 9
Volume 9, Number 1, January 2001
- Biing-Hwang Juang:
Why speech synthesis? (in memory of Prof. Jonathan Allen, 1934-2000). 1-2 - Satoshi Takano, Kimihito Tanaka, Hideyuki Mizuno, Masanobu Abe, Shin'ya Nakajima:
A Japanese TTS system based on multiform units and a speech modification algorithm with harmonics reconstruction. 3-10 - Darragh O'Brien, Alex I. C. Monaghan:
Concatenative synthesis based on a harmonic model. 11-20 - Yannis Stylianou:
Applying the harmonic plus noise model in concatenative speech synthesis. 21-29 - Johan Wouters, Michael W. Macon:
Control of spectral dynamics in concatenative speech synthesis. 30-38 - Esther Klabbers, Raymond N. J. Veldhuis:
Reducing audible spectral discontinuities. 39-51 - Jerome R. Bellegarda, Kim E. A. Silverman, Kevin A. Lenzo, Victoria Anderson:
Statistical prosodic modeling: from corpus design to parameter estimation. 52-66
Volume 9, Number 2, February 2001
- Hsi-Wen Nein, Chin-Teng Lin
:
Incorporating error shaping technique into LSF vector quantization. 73-86 - Afshin Rezayee, Saeed Gazor
An adaptive KLT approach for speech enhancement. 87-95 - Alain Biem, Shigeru Katagiri, Erik McDermott, Biing-Hwang Juang:
An application of discriminative feature extraction to filter-bank-based speech recognition. 96-110 - Ajit V. Rao, Kenneth Rose:
Deterministically annealed design of hidden Markov model speech recognizers. 111-126 - Tetsuya Takiguchi, Satoshi Nakamura, Kiyohiro Shikano:
HMM-separation-based speech recognition for a distant moving speaker. 127-140 - Martin Bouchard
Inverse structure for active noise control and combined active noise control/sound reproduction systems. 141-151 - Federico Fontana
Signal-theoretic characterization of waveguide mesh geometries for models of two-dimensional wave propagation in elastic media. 152-161 - Christina Breining:
A robust fuzzy logic-based step-gain control for adaptive filters in acoustic echo cancellation. 162-167 - Trevor N. Yensen, Rafik A. Goubran, Ioannis Lambadaris:
Synthetic stereo acoustic echo cancellation structure for multiple participant VoIP conferences. 168-174 - Michael W. Hoffman, Zhao Li, Devajani Khataniar:
GSC-based spatial voice activity detection for enhanced speech coding in the presence of competing speech. 175-178 - Néstor Becerra Yoma, Fergus R. McInnes, Mervyn A. Jack, Sandra Dotto Stump, Lee Luan Ling:
On including temporal constraints in Viterbi alignment for speech recognition in noise. 179-182
Volume 9, Number 3, March 2001
- André J. W. van der Kouwe, DeLiang Wang, Guy J. Brown
A comparison of auditory and blind separation techniques for speech segregation. 189-195 - Alexandros Potamianos, Petros Maragos:
Time-frequency distributions for automatic speech recognition. 196-200 - Guojun Zhou, John H. L. Hansen, James F. Kaiser:
Nonlinear feature based classification of speech under stress. 201-216 - Elias Nemer, Rafik A. Goubran, Samy Mahmoud:
Robust voice activity detection using higher-order statistics in the LPC residual domain. 217-231 - Yannis Stylianou:
Removing linear phase mismatches in concatenative speech synthesis. 232-239 - Tim Fingscheidt
Softbit speech decoding: a new approach to error concealment. 240-251 - Yuan-Fu Liao
A modular RNN-based method for continuous Mandarin speech recognition. 252-263 - Enrico Bocchieri, Brian Kan-Wing Mak
Subspace distribution clustering hidden Markov model. 264-275 - Koichi Shinoda, Chin-Hui Lee:
A structural Bayes approach to speaker adaptation. 276-287 - Frank Wessel, Ralf Schlüter
Confidence measures for large vocabulary continuous speech recognition. 288-298 - Reinhold Haeb-Umbach
Automatic generation of phonetic regression class trees for MLLR adaptation. 299-302 - Kah-Chye Tan, Sian-Sheng Neo, Benjamin H. C. Yeo:
Ellipsoid representation of reference templates for efficient speaker verification. 302-311
Volume 9, Number 4, May 2001
- Akira Watanabe:
Formant estimation method using inverse-filter control. 317-326 - George Saon
Data-driven approach to designing compound words for continuous speech recognition. 327-332 - Yali Amit, Alejandro Murua:
Speech recognition using randomized relational decision trees. 333-341 - Chafic Mokbel:
Online adaptation of HMMs to real-life conditions: a unified framework. 342-357 - Aravind Ganapathiraju, Jonathan Hamaker, Joseph Picone, Mark Ordowski, George R. Doddington:
Syllable-based large vocabulary continuous speech recognition. 358-366 - Christoph Neukirchen, Jörg Rottland, Daniel Willett, Gerhard Rigoll:
A continuous density interpretation of discrete HMM systems and MMI-neural networks. 367-377 - Brian Kan-Wing Mak
Direct training of subspace distribution clustering hidden Markov model. 378-387 - Qiang Huo, Bin Ma:
Online adaptive learning of continuous-density hidden Markov models based on multiple-stream prior evolution and posterior pooling. 388-398 - Jen-Tzung Chien
Transformation-based Bayesian predictive classification using online prior evolution. 399-410 - Paul M. Baggenstoss:
A modified Baum-Welch algorithm for hidden Markov models with multiple observation spaces. 411-416 - Olivier Siohan, Cristina Chesta, Chin-Hui Lee:
Joint maximum a posteriori adaptation of transformation and HMM parameters. 417-428 - Alberto González
Fast transversal filters for deconvolution in multichannel sound reproduction. 429-440 - Tong Zhang, C.-C. Jay Kuo
Audio content analysis for online audiovisual data segmentation and classification. 441-457 - Ramdas Kumaresan, Yadong Wang:
On the duality between line-spectral frequencies and zero-crossings of signals. 458-461
Volume 9, Number 5, July 2001
- Marine Campedel-Oudot, Olivier Cappé, Eric Moulines:
Estimation of the spectral envelope of voiced sounds using a penalized likelihood approach. 469-481 - Ki-Seung Lee, Richard V. Cox:
A very low bit rate speech coder based on a recognition/synthesis paradigm. 482-491 - Jonas Samuelsson, Per Hedelin:
Recursive coding of spectrum parameters. 492-503 - Rainer Martin
Noise power spectral density estimation based on optimal smoothing and minimum statistics. 504-512 - Peter Eneroth, Steven L. Gay, Tomas Gänsler, Jacob Benesty
A real-time implementation of a stereophonic acoustic echo canceler. 513-523 - Nedelko Grbic, Xiao-Jiao Tao, Sven E. Nordholm
Blind signal separation using overcomplete subband representation. 524-533 - Bor-Shen Lin, Lin-Shan Lee:
Computer-aided analysis and design for spoken dialogue systems based on quantitative simulations. 534-548 - Rathinavelu Chengalvarayan, Li Deng:
A maximum a posteriori approach to speaker adaptation using the trended hidden Markov model. 549-557 - Hong Kook Kim, Richard V. Cox:
A bitstream-based front-end for wireless speech recognition on IS-136 communications system. 558-568 - Qi Li:
A detection approach to search-space reduction for HMM state alignment in speaker verification. 569-578 - Aki Härmä
A comparison of warped and conventional linear predictive coding. 579-588 - Yosef Bendel, David Burshtein, Ofir Shalvi, Ehud Weinstein:
Delayless frequency domain acoustic echo cancellation. 589-597 - Ming Zhang, Hui Lan, Wee Ser:
Cross-updated active noise control system with online secondary path modeling. 598-602
Volume 9, Number 6, September 2001
- Der-Jenq Liu, Chin-Teng Lin
Fundamental frequency estimation based on the joint time-frequency analysis of harmonic spectral structure. 609-621 - Chunyan Li, Peter Lupini, Eyal Shlomot, Vladimir Cuperman:
Coding of variable dimension speech spectral vectors using weighted nonsquare transform vector quantization. 622-631 - Eyal Shlomot, Vladimir Cuperman, Allen Gersho:
Hybrid coding: combined harmonic and waveform coding of speech at 4 kb/s. 632-646 - Montri Karnjanadecha, Stephen A. Zahorian:
Signal modeling for high-performance robust isolated word recognition. 647-654 - Mehmet Bilginer Gülmezoglu, Vakif Dzhafarov
The common vector approach and its relation to principal component analysis. 655-662 - Shaojun Wang, Yunxin Zhao:
Online Bayesian tree-structured transformation of HMMs with optimal model selection for speaker adaptation. 663-677 - Jirí Navrátil:
Spoken language recognition-a step toward multilinguality in speech processing. 678-685 - Dennis R. Morgan, Joseph L. Hall, Jacob Benesty
Investigation of several types of nonlinearities for use in stereo acoustic echo cancellation. 686-696 - Darren B. Ward, Thushara D. Abhayapala
Reproduction of a plane-wave sound field using an array of loudspeakers. 697-707
Volume 9, Number 7, October 2001
- Philip J. B. Jackson
Pitch-scaled estimation of simultaneous voiced and turbulence-noise components in speech. 713-726 - Tetsuya Shimamura, Hajime Kobayashi:
Weighted autocorrelation for pitch extraction of noisy speech. 727-730 - Jesper Jensen, John H. L. Hansen:
Speech enhancement using a constrained iterative sinusoidal model. 731-740 - Ki Yong Lee, Joohun Lee:
Recognition of noisy speech by a nonstationary AR HMM with gain adaptation under unknown noise. 741-746 - Alvin Wen-Yu Su, Sheng-Fu Liang:
A new automatic IIR analysis/synthesis technique for plucked-string instruments. 747-754 - Hisashi Sano, Toshio Inoue, Akira Takahashi, Kenichi Terai, Yoshio Nakamura:
Active control system for low-frequency road noise combined with an audio system. 755-763
Volume 9, Number 8, November 2001
- Aki Härmä
Linear predictive coding with modified filter structures. 769-777 - Emre Gündüzhan, Kathryn Momtahan:
Linear prediction based packet loss concealment algorithm for PCM coded speech. 778-785 - Oded Gottesman, Allen Gersho:
Enhanced waveform interpolative coding at low bit-rate. 786-798 - Harald Gustafsson, Sven E. Nordholm
Spectral subtraction using reduced delay convolution and adaptive averaging. 799-807 - Régine Le Bouquin-Jeannès
Combined noise and echo reduction in hands-free systems: a survey. 808-820 - Myoung-Wan Koo, Chin-Hui Lee, Biing-Hwang Juang:
Speech recognition and utterance verification based on a generalized confidence score. 821-832 - Ahmed M. Abdelatty Ali
Acoustic-phonetic features for the automatic classification of stop consonants. 833-841 - Jeih-Weih Hung, Jia-Lin Shen, Lin-Shan Lee:
New approaches for domain transformation and parameter combination for improved accuracy in parallel model combination (PMC) techniques. 842-855 - Ramalingam Hariharan, Imre Kiss, I. Viikki:
Noise robust speech parameterization using multiresolution feature extraction. 856-865 - Chao-Shih Huang, Hsiao-Chuan Wang, Chin-Hui Lee:
An SNR-incremental stochastic matching algorithm for noisy speech recognition. 866-873 - Hui Jiang, Li Deng:
A Bayesian approach to the verification problem: applications to speaker verification. 874-884 - Dietrich Fränken, Klaus Meerkötter, Joachim Wassmuth:
Passive parametric modeling of dynamic loudspeakers. 885-891 - Seon Joon Park, Jeong Hyeon Yun, Young-Cheol Park, Dae Hee Youn:
A delayless subband active noise control system for wideband noise control. 892-899 - Doh-Suk Kim:
On the perceptually irrelevant phase information in sinusoidal representation of speech. 900-905 - Johan Hellgren, Urban Forssell:
Bias of feedback cancellation algorithms in hearing aids based on direct closed loop identification. 906-913 - Eric D. Scheirer:
Structured audio, Kolmogorov complexity, and generalized audio coding. 914-931 - Jang-Chyuan Jenq, Shih-Fu Hsieh:
Acoustic echo cancellation using iterative-maximal-length correlation and double-talk detection. 932-942 - Yiteng Huang, Jacob Benesty
Real-time passive source localization: a practical linear-correction least-squares approach. 943-956
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