Stop the war!Остановите войну!
for scientists:
default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCIDICASSP 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
- Ciprian Chelba:
Portability of syntactic structure for language modeling. 1 - 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 - Michiel Bacchiani:
Automatic transcription of voicemail at AT&T. 25-28 - Lidia Mangu, Mukund Padmanabhan:
Error corrective mechanisms for speech recognition. 29-32 - 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 - Wira Gunawan, Mark Hasegawa-Johnson:
PLP coefficients can be quantized at 400 bps. 77-80 - 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 - Peter Kabal, W. Bastiaan Kleijn:
All-pole modelling of mixed excitation signals. 97-100 - Kaliappan Gopalan:
On the effect of stress on certain modulation parameters of speech. 101-104 - Te-Won Lee, Gil-Jin Jang:
The statistical structures of male and female speech signals. 105-108 - 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 - Xue-Wen Chen, Jie Yang:
Visual speech synthesis using quadtree splines. 153-156 - 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 - Iain McCowan, Sridha Sridharan:
Microphone array sub-band speech recognition. 185-188 - 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 - Martin Westphal, Alex Waibel:
Model-combination-based acoustic mapping. 221-224 - 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 - Michael L. Shire:
Multi-stream ASR trained with heterogeneous reverberant environments. 253-256 - Astrid Hagen, Hervé Bourlard, Andrew C. Morris:
Adaptive ML-weighting in multi-band recombination of Gaussian mixture ASR. 257-260 - Ben Milner:
Robust speech recognition in burst-like packet loss. 261-264 - 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 - Carsten Meyer, Georg Rose:
Improved noise robustness by corrective and rival training. 293-296 - 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 - Mark J. F. Gales:
Multiple-cluster adaptive training schemes. 361-364 - Sadaoki Furui, Daisuke Itoh:
Neural-network-based HMM adaptation for noisy speech. 365-368 - 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 - Chak Shun Lai, Bertram E. Shi:
A one-pass strategy for keyword spotting and verification. 377-380 - 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 - Samy Bengio, Johnny Mariéthoz:
Learning the decision function for speaker verification. 425-428 - 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