-
2
-
-
84906248049
-
-
Barker, J., Christensen, H., Ma, N., Green, P., Vincent, E., 2011. The PASCAL CHiME speech separation and recognition challenge [Online]. Available: http://www.dcs.shef.ac.uk/spandh/chime/challenge.html.
-
(2011)
The PASCAL CHiME Speech Separation and Recognition Challenge [Online]
-
-
Barker, J.1
Christensen, H.2
Ma, N.3
Green, P.4
Vincent, E.5
-
4
-
-
79954589819
-
A perspective on single-channel frequency-domain speech enhancement
-
Morgan & Claypool Publishers
-
J. Benesty, and Y. Huang A perspective on single-channel frequency-domain speech enhancement Synthesis Lectures on Speech and Audio Processing 2011 Morgan & Claypool Publishers
-
(2011)
Synthesis Lectures on Speech and Audio Processing
-
-
Benesty, J.1
Huang, Y.2
-
5
-
-
4544356482
-
TRINICON: A versatile framework for multichannel blind signal processing
-
May Montreal, Canada
-
H. Buchner, R. Aichner, and W. Kellermann TRINICON: a versatile framework for multichannel blind signal processing IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), May, vol. 3 Montreal, Canada 2004 889 892
-
(2004)
IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP)
, vol.3
, pp. 889-892
-
-
Buchner, H.1
Aichner, R.2
Kellermann, W.3
-
6
-
-
4544262407
-
Blind Source Separation for Convolutive Mixtures: A Unified Treatment
-
Y. Huang, J. Benesty, Kluwer Academic Publishers Boston
-
H. Buchner, R. Aichner, and W. Kellermann Blind Source Separation for Convolutive Mixtures: A Unified Treatment Y. Huang, J. Benesty, Audio Signal Processing for Next-generation Multimedia Communication Systems 2004 Kluwer Academic Publishers Boston 255 293
-
(2004)
Audio Signal Processing for Next-generation Multimedia Communication Systems
, pp. 255-293
-
-
Buchner, H.1
Aichner, R.2
Kellermann, W.3
-
7
-
-
11144223199
-
A generalization of blind source separation algorithms for convolutive mixtures based on second order statistics
-
H. Buchner, R. Aichner, and W. Kellermann A generalization of blind source separation algorithms for convolutive mixtures based on second order statistics IEEE Transactions on Speech Audio Processing 13 January (1) 2005 120 134
-
(2005)
IEEE Transactions on Speech Audio Processing
, vol.13
, Issue.1 January
, pp. 120-134
-
-
Buchner, H.1
Aichner, R.2
Kellermann, W.3
-
8
-
-
33750368310
-
An audio-visual corpus for speech perception and automatic speech recognition
-
M. Cooke, J. Barker, S. Cunningham, and X. Shao An audio-visual corpus for speech perception and automatic speech recognition Journal of the Acoustical Society of America 120 5 2006 2421 2424
-
(2006)
Journal of the Acoustical Society of America
, vol.120
, Issue.5
, pp. 2421-2424
-
-
Cooke, M.1
Barker, J.2
Cunningham, S.3
Shao, X.4
-
9
-
-
85009070292
-
Large-vocabulary speech recognition under adverse acoustic environments
-
L. Deng, A. Acero, M. Plumpe, and X. Huang Large-vocabulary speech recognition under adverse acoustic environments Int. Conf. on Spoken Language Processing (ICSLP), vol. 3 2000 806 809
-
(2000)
Int. Conf. on Spoken Language Processing (ICSLP)
, vol.3
, pp. 806-809
-
-
Deng, L.1
Acero, A.2
Plumpe, M.3
Huang, X.4
-
11
-
-
0021645331
-
Speech enhancement using a minimum mean-square error short-time spectral amplitude estimator
-
Y. Ephraim, and D. Malah Speech enhancement using a minimum mean-square error short-time spectral amplitude estimator IEEE Transactions on Acoustics, Speech and Signal Processing 32 December (6) 1984 1109 1121
-
(1984)
IEEE Transactions on Acoustics, Speech and Signal Processing
, vol.32
, Issue.6 December
, pp. 1109-1121
-
-
Ephraim, Y.1
Malah, D.2
-
12
-
-
0021892216
-
Speech enhancement using a minimum mean-square error log-spectral amplitude estimator
-
Y. Ephraim, and D. Malah Speech enhancement using a minimum mean-square error log-spectral amplitude estimator IEEE Transactions on Acoustics, Speech and Signal Processing 33 April (2) 1985 443 445
-
(1985)
IEEE Transactions on Acoustics, Speech and Signal Processing
, vol.33
, Issue.2 April
, pp. 443-445
-
-
Ephraim, Y.1
Malah, D.2
-
14
-
-
84948594425
-
An algorithm for linearly constrained adaptive array processing
-
O.L. Frost An algorithm for linearly constrained adaptive array processing Proceedings of the IEEE 60 August (8) 1972 926 935
-
(1972)
Proceedings of the IEEE
, vol.60
, Issue.8 August
, pp. 926-935
-
-
Frost, O.L.1
-
15
-
-
70349227947
-
The application of hidden Markov models in speech recognition
-
M. Gales, and S. Young The application of hidden Markov models in speech recognition Foundations and Trends in Signal Processing 1 3 2007 195 304
-
(2007)
Foundations and Trends in Signal Processing
, vol.1
, Issue.3
, pp. 195-304
-
-
Gales, M.1
Young, S.2
-
17
-
-
0035424281
-
Signal enhancement using beamforming and nonstationarity with applications to speech
-
S. Gannot, D. Burshtein, and E. Weinstein Signal enhancement using beamforming and nonstationarity with applications to speech IEEE Transactions on Signal Processing 49 August (8) 2001 1614 1626
-
(2001)
IEEE Transactions on Signal Processing
, vol.49
, Issue.8 August
, pp. 1614-1626
-
-
Gannot, S.1
Burshtein, D.2
Weinstein, E.3
-
18
-
-
0019928857
-
An alternative approach to linearly constrained adaptive beamforming
-
L. Griffiths, and C. Jim An alternative approach to linearly constrained adaptive beamforming IEEE Transactions on Antennas and Propagation 30 January (1) 1982 27 34
-
(1982)
IEEE Transactions on Antennas and Propagation
, vol.30
, Issue.1 January
, pp. 27-34
-
-
Griffiths, L.1
Jim, C.2
-
19
-
-
84856168858
-
The MVDR beamformer for speech enhancement
-
I. Cohen, J. Benesty, S. Gannot, Springer Berlin/Heidelberg
-
E.A.P. Habets, J. Benesty, S. Gannot, and I. Cohen The MVDR beamformer for speech enhancement I. Cohen, J. Benesty, S. Gannot, Speech Processing in Modern Communication 3 2010 Springer Berlin/Heidelberg 225 254
-
(2010)
Speech Processing in Modern Communication
, vol.3
, pp. 225-254
-
-
Habets, E.A.P.1
Benesty, J.2
Gannot, S.3
Cohen, I.4
-
21
-
-
33846210450
-
Application of a double-talk resilient DFT-domain adaptive filter for bin-wise stepsize controls to adaptive beamforming
-
May Sapporo, Japan
-
W. Herbordt, H. Buchner, S. Nakamura, and W. Kellermann Application of a double-talk resilient DFT-domain adaptive filter for bin-wise stepsize controls to adaptive beamforming Intl. Workshop on Nonlinear Signal and Image Processing (NSIP), May Sapporo, Japan 2005
-
(2005)
Intl. Workshop on Nonlinear Signal and Image Processing (NSIP)
-
-
Herbordt, W.1
Buchner, H.2
Nakamura, S.3
Kellermann, W.4
-
22
-
-
0033321732
-
A robust adaptive beamformer for microphone arrays with a blocking matrix using constrained adaptive filters
-
O. Hoshuyama, A. Sugiyama, and A. Hirano A robust adaptive beamformer for microphone arrays with a blocking matrix using constrained adaptive filters IEEE Transactions on Signal Processing 47 October (10) 1999 2677 2684
-
(1999)
IEEE Transactions on Signal Processing
, vol.47
, Issue.10 October
, pp. 2677-2684
-
-
Hoshuyama, O.1
Sugiyama, A.2
Hirano, A.3
-
23
-
-
33947691700
-
Pocketsphinx: A free real-time continuous speech recognition system for hand-held devices
-
May Toulouse, France
-
D. Huggins-Daines, M. Kumar, A. Chan, A.W. Black, M. Ravishankar, and A.I. Rudnicky Pocketsphinx: a free real-time continuous speech recognition system for hand-held devices IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), May Toulouse, France 2006
-
(2006)
IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP)
-
-
Huggins-Daines, D.1
Kumar, M.2
Chan, A.3
Black, A.W.4
Ravishankar, M.5
Rudnicky, A.I.6
-
26
-
-
0016990291
-
The generalized correlation method for estimation of time delay
-
August
-
C.H. Knapp, and G.C. Carter The generalized correlation method for estimation of time delay IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), August, vol. 24, no. 4 1976 320 327
-
(1976)
IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP)
, vol.24
, Issue.4
, pp. 320-327
-
-
Knapp, C.H.1
Carter, G.C.2
-
28
-
-
0029288633
-
Maximum likelihood linear regression for speaker adaptation of continuous density hidden Markov models
-
C.J. Leggetter, and P.C. Woodland Maximum likelihood linear regression for speaker adaptation of continuous density hidden Markov models Computer Speech and Language 9 1995 171 185
-
(1995)
Computer Speech and Language
, vol.9
, pp. 171-185
-
-
Leggetter, C.J.1
Woodland, P.C.2
-
29
-
-
0018642851
-
Enhancement and bandwidth compression of noisy speech
-
J.S. Lim, and A.V. Oppenheim Enhancement and bandwidth compression of noisy speech Proc. of the IEEE 67 December (12) 1979 1586 1604
-
(1979)
Proc. of the IEEE
, vol.67
, Issue.12 December
, pp. 1586-1604
-
-
Lim, J.S.1
Oppenheim, A.V.2
-
30
-
-
79957740368
-
TDOA estimation for multiple sound sources in noisy and reverberant environments using broadband independent component analysis
-
A. Lombard, Y. Zheng, H. Buchner, and W. Kellermann TDOA estimation for multiple sound sources in noisy and reverberant environments using broadband independent component analysis IEEE Transactions on Audio, Speech, and Language Processing 19 August (6) 2011 1490 1503
-
(2011)
IEEE Transactions on Audio, Speech, and Language Processing
, vol.19
, Issue.6 August
, pp. 1490-1503
-
-
Lombard, A.1
Zheng, Y.2
Buchner, H.3
Kellermann, W.4
-
31
-
-
84869432703
-
A two-channel acoustic front-end for robust automatic speech recognition in noisy and reverberant environments
-
September Florence, Italy
-
R. Maas, A. Schwarz, Y. Zheng, K. Reindl, S. Meier, A. Sehr, and W. Kellermann A two-channel acoustic front-end for robust automatic speech recognition in noisy and reverberant environments International Workshop on Machine Listening in Multisource Environments (Satellite Event of Interspeech 2011), September Florence, Italy 2011 41 46
-
(2011)
International Workshop on Machine Listening in Multisource Environments (Satellite Event of Interspeech 2011)
, pp. 41-46
-
-
Maas, R.1
Schwarz, A.2
Zheng, Y.3
Reindl, K.4
Meier, S.5
Sehr, A.6
Kellermann, W.7
-
34
-
-
0346707504
-
Microphone array post-filter based on noise field coherence
-
I.A. McCowan, and H. Bourlard Microphone array post-filter based on noise field coherence IEEE Transactions on Speech Audio Processing 11 November (6) 2003 709 716
-
(2003)
IEEE Transactions on Speech Audio Processing
, vol.11
, Issue.6 November
, pp. 709-716
-
-
McCowan, I.A.1
Bourlard, H.2
-
35
-
-
70349203006
-
Robust two-channel TDOA estimation for multiple speaker localization by using recursive ICA and a state coherence transform
-
April
-
F. Nesta, P. Svaizer, and M. Omologo Robust two-channel TDOA estimation for multiple speaker localization by using recursive ICA and a state coherence transform IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), April 2009 4597 4600
-
(2009)
IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP)
, pp. 4597-4600
-
-
Nesta, F.1
Svaizer, P.2
Omologo, M.3
-
36
-
-
0036753896
-
Geometric source separation: Merging convolutive source separation with geometric beamforming
-
L.C. Parra, and C.V. Alvino Geometric source separation: merging convolutive source separation with geometric beamforming IEEE Transactions on Speech Audio Processing 10 September 2002 352 362
-
(2002)
IEEE Transactions on Speech Audio Processing
, vol.10
, Issue.September
, pp. 352-362
-
-
Parra, L.C.1
Alvino, C.V.2
-
37
-
-
77953867956
-
Speech enhancement for binaural hearing aids based on blind source separation
-
March Limassol, Cyprus
-
K. Reindl, Y. Zheng, and W. Kellermann Speech enhancement for binaural hearing aids based on blind source separation Proc. 4th IEEE Intl. Symposium on Communications, Control, and Signal Processing (ISCCSP), March Limassol, Cyprus 2010
-
(2010)
Proc. 4th IEEE Intl. Symposium on Communications, Control, and Signal Processing (ISCCSP)
-
-
Reindl, K.1
Zheng, Y.2
Kellermann, W.3
-
40
-
-
0009653561
-
Post-filtering techniques
-
M. Brandstein, D. Ward, Springer
-
K.U. Simmer, J. Bitzer, and C. Marro Post-filtering techniques M. Brandstein, D. Ward, Microphone Arrays: Signal Processing Techniques and Applications 2001 Springer 39 60
-
(2001)
Microphone Arrays: Signal Processing Techniques and Applications
, pp. 39-60
-
-
Simmer, K.U.1
Bitzer, J.2
Marro, C.3
-
41
-
-
65249179051
-
Blind spatial subtraction array for speech enhancement in noisy environment
-
Y. Takahashi, T. Takatani, K. Osako, H. Saruwatari, and K. Shikano Blind spatial subtraction array for speech enhancement in noisy environment IEEE Transactions on Audio, Speech, and Language Processing 17 May (4) 2009 650 664
-
(2009)
IEEE Transactions on Audio, Speech, and Language Processing
, vol.17
, Issue.4 May
, pp. 650-664
-
-
Takahashi, Y.1
Takatani, T.2
Osako, K.3
Saruwatari, H.4
Shikano, K.5
-
43
-
-
84940453976
-
Speech enhancement by conditional estimation: Noise reduction, error concealment & bandwidth extension, what makes the difference?
-
September Seattle, WA, USA
-
P. Vary Speech enhancement by conditional estimation: Noise reduction, error concealment & bandwidth extension, what makes the difference? Int. Workshop Acoustic Echo Noise Control (IWAENC), September Seattle, WA, USA 2008
-
(2008)
Int. Workshop Acoustic Echo Noise Control (IWAENC)
-
-
Vary, P.1
-
45
-
-
33744975847
-
Performance measurement in blind audio source separation
-
E. Vincent, R. Gribonval, and C. Févotte Performance measurement in blind audio source separation IEEE Transactions on Audio, Speech, and Language Processing 14 July (4) 2006 1462 1469
-
(2006)
IEEE Transactions on Audio, Speech, and Language Processing
, vol.14
, Issue.4 July
, pp. 1462-1469
-
-
Vincent, E.1
Gribonval, R.2
Févotte, C.3
-
46
-
-
33645209480
-
Sphinx-4: A flexible open source framework for speech recognition
-
Walker, W., Lamere, P., Kwok, P., Raj, B., Singh, R., Gouvea, E., Wolf, P., Woelfel, J., 2004. Sphinx-4: a flexible open source framework for speech recognition. Sun Microsystems Technical Report, No. TR-2004-139.
-
(2004)
Sun Microsystems Technical Report, No. TR-2004-139
-
-
Walker, W.1
Lamere, P.2
Kwok, P.3
Raj, B.4
Singh, R.5
Gouvea, E.6
Wolf, P.7
Woelfel, J.8
-
49
-
-
0023773764
-
A microphone array with adaptive post-filtering for noise reduction in reverberant rooms
-
April
-
R. Zelinski A microphone array with adaptive post-filtering for noise reduction in reverberant rooms IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), April 1988 2578 2581
-
(1988)
IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP)
, pp. 2578-2581
-
-
Zelinski, R.1
|