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Semantic ScholarICASSP 2001: Salt Lake City, Utah, USA
IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2001, 7-11 May, 2001, Salt Palace Convention Center, Salt Lake City, Utah, USA, Proceedings. IEEE 2001, ISBN 0-7803-7041-4- Liam Comerford, David Frank, Ponani S. Gopalakrishnan, Ramesh Gopinath, Jan Sedivý:
The IBM Personal Speech Assistant. 1-4 - Olli Viikki, Imre Kiss, Jilei Tian:
Speaker- and language-independent speech recognition in mobile communication systems. 5-8 - Xuedong Huang, Alex Acero, Ciprian Chelba, Li Deng, Jasha Droppo, Doug Duchene, Joshua Goodman, Hsiao-Wuen Hon, Derek Jacoby, Li Jiang, Ricky Loynd, Milind Mahajan, Peter Mau, Scott Meredith, Salman Mughal, Salvado Neto, Mike Plumpe, Kuansan Steury, Gina Venolia, Kuansan Wang, Ye-Yi Wang:
MiPad: a multimodal interaction prototype. 9-12 - Sadaoki Furui, Koji Iwano, Chiori Hori, Takahiro Shinozaki, Yohei Saito, Satoshi Tamura:
Ubiquitous speech processing. 13-16 - Richard C. Rose, Sarangarajan Parthasarathy, Bojana Gajic, Aaron E. Rosenberg, Shrikanth S. Narayanan:
On the implementation of ASR algorithms for hand-held wireless mobile devices. 17-20 - Volker Stahl, Alexander Fischer, Rolf Bippus:
Acoustic synthesis of training data for speech recognition in living room environments. 21-24 - Frank Wessel, Ralf Schlüter, Hermann Ney:
Explicit word error minimization using word hypothesis posterior probabilities. 33-36 - Nicola Bertoldi, Fabio Brugnara, Mauro Cettolo, Marcello Federico, Diego Giuliani:
From broadcast news to spontaneous dialogue transcription: portability issues. 37-40 - Christoph Neukirchen, Dietrich Klakow, Xavier L. Aubert:
Generation and expansion of word graphs using long span context information. 41-44 - Daniel Povey, Philip C. Woodland:
Improved discriminative training techniques for large vocabulary continuous speech recognition. 45-48 - Luís Felipe Uebel, Philip C. Woodland:
Improvements in linear transform based speaker adaptation. 49-52 - Yuqing Gao, Bhuvana Ramabhadran, C. Julian Chen, Hakan Erdogan, Michael Picheny:
Innovative approaches for large vocabulary name recognition. 53-56 - Thomas Hain, Philip C. Woodland, Gunnar Evermann, Daniel Povey:
New features in the CU-HTK system for transcription of conversational telephone speech. 57-60 - C. Julian Chen, Haiping Li, Liqin Shen, Guokang Fu:
Recognize tone languages using pitch information on the main vowel of each syllable. 61-64 - Hagen Soltau, Thomas Schaaf, Florian Metze, Alex Waibel:
The ISL evaluation system for Verbmobil-II. 65-68 - Akinobu Lee, Tatsuya Kawahara, Kiyohiro Shikano:
Gaussian mixture selection using context-independent HMM. 69-72 - Sirko Molau, Michael Pitz, Ralf Schlüter, Hermann Ney:
Computing Mel-frequency cepstral coefficients on the power spectrum. 73-76 - Ahmed M. Abdelatty Ali, Jan Van der Spiegel, Paul Mueller:
Robust classification of stop consonants using auditory-based speech processing. 81-84 - Bojana Gajic, Kuldip K. Paliwal:
Robust feature extraction using subband spectral centroid histograms. 85-88 - Shi-Huang Chen, Jhing-Fa Wang:
Extraction of pitch information in noisy speech using wavelet transform with aliasing compensation. 89-92 - Wen-Hsing Lai, Sin-Horng Chen:
A novel syllable duration modeling approach for Mandarin speech. 93-96 - Karl Schnell, Arild Lacroix:
Pole zero estimation from speech signals by an iterative procedure. 109-112 - Qifeng Zhu, Abeer Alwan:
An efficient and scalable 2D DCT-based feature coding scheme for remote speech recognition. 113-116 - Hiroshi Matsumoto, Masanori Moroto:
Evaluation of mel-LPC cepstrum in a large vocabulary continuous speech recognition. 117-120 - Roberto Gemello, Dario Albesano, Loreta Moisa, Renato De Mori:
Integration of fixed and multiple resolution analysis in a speech recognition system. 121-124 - Liang Gu, Kenneth Rose:
Perceptual harmonic cepstral coefficients for speech recognition in noisy environment. 125-128 - Takashi Fukuda, Masashi Takigawa, Tsuneo Nitta:
Peripheral features for HMM-based speech recognition. 129-132 - Ralf Schlüter, Hermann Ney:
Using phase spectrum information for improved speech recognition performance. 133-136 - Sachin S. Kajarekar, Bayya Yegnanarayana, Hynek Hermansky:
A study of two dimensional linear discriminants for ASR. 137-140 - Ho Young Hur, Hyung Soon Kim:
Formant weighted cepstral feature for LSP-based speech recognition. 141-144 - Rathinavelu Chengalvarayan:
On the use of matrix derivatives in integrated design of dynamic feature parameters for speech recognition. 145-148 - Zekeriya Tufekci, John N. Gowdy:
Subband feature extraction using lapped orthogonal transform for speech recognition. 149-152 - Conrad Sanderson, Kuldip K. Paliwal:
Noise compensation in a multi-modal verification system. 157-160 - Martin Heckmann, Frédéric Berthommier, Kristian Kroschel:
Optimal weighting of posteriors for audio-visual speech recognition. 161-164 - Gerasimos Potamianos, Juergen Luettin, Chalapathy Neti:
Hierarchical discriminant features for audio-visual LVCSR. 165-168 - Juergen Luettin, Gerasimos Potamianos, Chalapathy Neti:
Asynchronous stream modeling for large vocabulary audio-visual speech recognition. 169-172 - Hervé Glotin, D. Vergyr, Chalapathy Neti, Gerasimos Potamianos, Juergen Luettin:
Weighting schemes for audio-visual fusion in speech recognition. 173-176 - Sabri Gurbuz, Zekeriya Tufekci, Eric K. Patterson, John N. Gowdy:
Application of affine-invariant Fourier descriptors to lipreading for audio-visual speech recognition. 177-180 - Adriano Vilela Barbosa, Hani C. Yehia:
Measuring the relation between speech acoustics and 2D facial motion. 181-184 - Takanobu Nishiura, S. Nakanura, Kiyohiro Shikano:
Speech enhancement by multiple beamforming with reflection signal equalization. 189-192 - Panikos Heracleous, Satoshi Nakamura, Kiyohiro Shikano:
A microphone array-based 3-D N-best search algorithm for the simultaneous recognition of multiple sound sources in real environments. 193-196 - Dinei A. F. Florêncio, Henrique S. Malvar:
Multichannel filtering for optimum noise reduction in microphone arrays. 197-200 - Scott M. Griebel, Michael S. Brandstein:
Microphone array speech dereverberation using coarse channel modeling. 201-204 - Firas Jabloun, Benoît Champagne:
A multi-microphone signal subspace approach for speech enhancement. 205-208 - Jasha Droppo, Alex Acero, Li Deng:
Efficient on-line acoustic environment estimation for FCDCN in a continuous speech recognition system. 209-212 - Christophe Cerisara, Luca Rigazio, Robert Boman, Jean-Claude Junqua:
Environmental adaptation based on first order approximation. 213-216 - Hui Jiang, Frank K. Soong, Chin-Hui Lee:
Hierarchical stochastic feature matching for robust speech recognition. 217-220 - Yunxin Zhao, Shaojun Wang, Kuan-Chieh Yen:
Recursive estimation of time-varying environments for robust speech recognition. 225-228 - Mohamed Afify, Olivier Siohan:
Sequential noise estimation with optimal forgetting for robust speech recognition. 229-232 - Qi Li, Jinsong Zheng, Qiru Zhou, Chin-Hui Lee:
Robust, real-time endpoint detector with energy normalization for ASR in adverse environments. 233-236 - Arnaud Martin, Delphine Charlet, Laurent Mauuary:
Robust speech/non-speech detection using LDA applied to MFCC. 237-240 - Hong Kook Kim, Richard V. Cox:
Feature enhancement for a bitstream-based front-end in wireless speech recognition. 241-244 - Osamu Segawa, Kazuya Takeda, Fumitada Itakura:
Continuous speech recognition without end-point detection. 245-248 - Ramalingam Hariharan, Juha Häkkinen, Kari Laurila:
Robust end-of-utterance detection for real-time speech recognition applications. 249-252 - Astrid Hagen, Hervé Bourlard, Andrew C. Morris:
Adaptive ML-weighting in multi-band recombination of Gaussian mixture ASR. 257-260 - Claude Barras, Lori Lamel, Jean-Luc Gauvain:
Automatic transcription of compressed broadcast audio. 265-268 - Alexandros Potamianos, Vijitha Weerackody:
Soft-feature decoding for speech recognition over wireless channels. 269-272 - Rita Singh, Michael L. Seltzer, Bhiksha Raj, Richard M. Stern:
Speech in Noisy Environments: robust automatic segmentation, feature extraction, and hypothesis combination. 273-276 - Konstantinos Koumpis, Søren Kamaric Riis:
Adaptive transition bias for robust low complexity speech recognition. 277-280 - Christian Uhl, Markus Lieb:
Experiments with an extended adaptive SVD enhancement scheme for speech recognition in noise. 281-284 - Volker Stahl, Alexander Fischer, Rolf Bippus:
Acoustic synthesis of training data for speech recognition in living room environments. 285-288 - Robert W. Morris, Mark A. Clements:
Maximum-likelihood compensation of zero-memory nonlinearities in speech signals. 289-292 - Masakiyo Fujimoto, Yasuo Ariki:
Continuous speech recognition under non-stationary musical environments based on speech state transition model. 297-300 - Li Deng, Alex Acero, Li Jiang, Jasha Droppo, Xuedong Huang:
High-performance robust speech recognition using stereo training data. 301-304 - Dusan Macho, Yan Ming Cheng:
SNR-dependent waveform processing for improving the robustness of ASR front-end. 305-308 - Satya Dharanipragada, Bhaskar D. Rao:
MVDR based feature extraction for robust speech recognition. 309-312 - Jon P. Nedel, Richard M. Stern:
Duration normalization for improved recognition of spontaneous and read speech via missing feature methods. 313-316 - Kuan-Ting Chen, Hsin-Min Wang:
Eigenspace-based maximum a posteriori linear regression for rapid speaker adaptation. 317-320 - Eugene Jon, Dong Kook Kim, Nam Soo Kim:
EMAP-based speaker adaptation with robust correlation estimation. 321-324 - George Saon, Geoffrey Zweig, Mukund Padmanabhan:
Linear feature space projections for speaker adaptation. 325-328 - Jen-Tzung Chien, Chih-Hsien Huang:
Online speaker adaptation based on quasi-Bayes linear regression. 329-332 - Hakan Erdogan, Yuqing Gao, Michael Picheny:
Rapid adaptation using penalized-likelihood methods. 333-336 - Trausti T. Kristjansson, Brendan J. Frey, Li Deng, Alex Acero:
Towards non-stationary model-based noise adaptation for large vocabulary speech recognition. 337-340 - Shinichi Yoshizawa, Akira Baba, Kanako Matsunami, Yuichiro Mera, Miichi Yamada, Kiyohiro Shikano:
Unsupervised speaker adaptation based on sufficient HMM statistics of selected speakers. 341-344 - Nick J.-C. Wang, Sammy S.-M. Lee, Frank Seide, Lin-Shan Lee:
Rapid speaker adaptation using a priori knowledge by eigenspace analysis of MLLR parameters. 345-348 - S. Douglas Peters:
Hypothesis-driven adaptation (Hydra): a flexible eigenvoice architecture. 349-352 - Henrik Botterweck:
Anisotropic MAP defined by eigenvoices for large vocabulary continuous speech recognition. 353-356 - Delphine Charlet:
Confidence-measure-driven unsupervised incremental adaptation for HMM-based speech recognition. 357-360 - John W. McDonough, Florian Metze, Hagen Soltau, Alex Waibel:
Speaker compensation with sine-log all-pass transforms. 369-372 - Roland Kuhn, Florent Perronnin, Patrick Nguyen, Jean-Claude Junqua, Luca Rigazio:
Very fast adaptation with a compact context-dependent eigenvoice model. 373-376 - Changxue Ma, Mark A. Randolph, Joe Drish:
A support vector machines-based rejection technique for speech recognition. 381-384 - Denis Jouvet, S. Droguet:
On combining recognizers for improved recognition of spelled names. 385-388 - Benoît Maison, Ramesh A. Gopinath:
Robust confidence annotation and rejection for continuous speech recognition. 389-392 - Rubén San Segundo, Bryan L. Pellom, Kadri Hacioglu, Wayne H. Ward, José M. Pardo:
Confidence measures for spoken dialogue systems. 393-396 - Timothy J. Hazen, Issam Bazzi:
A comparison and combination of methods for OOV word detection and word confidence scoring. 397-400 - Hakan Altinçay, Mübeccel Demirekler:
Comparison of different objective functions for optimal linear combination of classifiers for speaker identification. 401-404 - Adriano Petry, Dante Augusto Couto Barone:
Fractal dimension applied to speaker identification. 405-408 - B. Yegnanarayana, K. Sharat Reddy, S. Prahallad Kishore:
Source and system features for speaker recognition using AANN models. 409-412 - Kazumasa Mori, Seiichi Nakagawa:
Speaker change detection and speaker clustering using VQ distortion for broadcast news speech recognition. 413-416 - Shai Fine, Jirí Navrátil, Ramesh A. Gopinath:
A hybrid GMM/SVM approach to speaker identification. 417-420 - Jereme M. Lovekin, Robert E. Yantorno, Kasturi Rangan Krishnamachari, Daniel S. Benincasa, Stanley J. Wenndt:
Developing usable speech criteria for speaker identification technology. 421-424 - Douglas E. Sturim, Douglas A. Reynolds, Elliot Singer, Joseph P. Campbell:
Speaker indexing in large audio databases using anchor models. 429-432 - Chiyomi Miyajima, Yosuke Hattori, Keiichi Tokuda, Takashi Masuko, Takao Kobayashi, Tadashi Kitamura:
Speaker identification using Gaussian mixture models based on multi-space probability distribution. 433-436 - Gil-Jin Jang, Te-Won Lee, Yung-Hwan Oh:
Learning statistically efficient features for speaker recognition. 437-440 - Roland Auckenthaler, Michael J. Carey, John S. D. Mason:
Language dependency in text-independent speaker verification. 441-444 - Jerome R. Bellegarda, Devang Naik, Matthias Neeracher, Kim E. A. Silverman:
Language-independent, short-enrollment voice verification over a far-field microphone. 445-448 - Luc Gagnon, Peter Stubley, Ghislain Mailhot:
Password-dependent speaker verification using quantized acoustic trajectories. 449-452 - Marcos Faúndez-Zanuy:
A combination between VQ and covariance matrices for speaker recognition. 453-456 - Lit Ping Wong, Martin J. Russell:
Text-dependent speaker verification under noisy conditions using parallel model combination. 457-460 - Upendra V. Chaudhari, Jirí Navrátil, Ganesh N. Ramaswamy, Stéphane H. Maes:
Very large population text-independent speaker identification using transformation enhanced multi-grained models. 461-464 - Roberto Togneri, Li Deng:
An EKF-based algorithm for learning statistical hidden dynamic model parameters for phonetic recognition. 465-468 - Andreas Tuerk, Steve J. Young:
Indicator variable dependent output probability modelling via continuous posterior functions. 473-476 - Lori Lamel, Jean-Luc Gauvain, Gilles Adda:
Investigating lightly supervised acoustic model training. 477-480 - Andrew Aaron, Scott Saobing Chen, Paul S. Cohen, Satya Dharanipragada, Ellen Eide, Martin Franz, Jean-Michel LeRoux, X. Luo, Benoît Maison, Lidia Mangu, T. Mathes, Miroslav Novak, Peder A. Olsen, Michael Picheny, Harry Printz, Bhuvana Ramabhadran, Andrej Sakrajda, George Saon, Borivoj Tydlitát, Karthik Visweswariah, D. Yuk:
Speech recognition for DARPA Communicator. 489-492 - Tsuneo Kato, Shingo Kuroiwa, Tohru Shimizu, Norio Higuchi:
Efficient mixture Gaussian synthesis for decision tree based state tying. 493-496 - Konstantin Markov, Seiichi Nakagawa, Satoshi Nakamura:
Discriminative training of HMM using maximum normalized likelihood algorithm. 497-500