-
1
-
-
0023985457
-
Beamforming:A versatile approach to spatial filtering
-
Apr.
-
B. D. V. Veen and K. M. Buckley, "Beamforming:A versatile approach to spatial filtering," IEEE ASSP Mag., vol. 5, no. 2, pp. 4-24, Apr. 1988.
-
(1988)
IEEE ASSP Mag.
, vol.5
, Issue.2
, pp. 4-24
-
-
Veen, B.D.V.1
Buckley, K.M.2
-
2
-
-
0141603901
-
Superdirective microphone arrays
-
M. Brandstein and D. Ward, Eds., 1st ed. Berlin, Germany: Springer, ch. 2
-
J. Bitzer and K. U. Simmer, "Superdirective microphone arrays," in Microphone Arrays: Signal Processing Techniques and Applications, M. Brandstein and D. Ward, Eds., 1st ed. Berlin, Germany: Springer, 2001, ch. 2, pp. 19-38.
-
(2001)
Microphone Arrays: Signal Processing Techniques and Applications
, pp. 19-38
-
-
Bitzer, J.1
Simmer, K.U.2
-
3
-
-
84948594425
-
An algorithm for linearly constrained adaptive array processing
-
Aug.
-
O. L. Frost, III, "An algorithm for linearly constrained adaptive array processing," Proc. IEEE, vol. 60, no. 8, pp. 926-935, Aug. 1972.
-
(1972)
Proc. IEEE
, vol.60
, Issue.8
, pp. 926-935
-
-
Frost, O.L.1
-
4
-
-
0019928857
-
An alternative approach to linearly constrained adaptive beamforming
-
Jan.
-
L. J. Griffiths and C. W. Jim, "An alternative approach to linearly constrained adaptive beamforming," IEEE Trans. Antennas Propag., vol. AP-30, no. 1, pp. 27-34, Jan. 1982.
-
(1982)
IEEE Trans. Antennas Propag.
, vol.AP-30
, Issue.1
, pp. 27-34
-
-
Griffiths, L.J.1
Jim, C.W.2
-
5
-
-
0000424380
-
Spatially selective sound capture for speech and audio processing
-
Oct.
-
J. L. Flanagan, A. C. Surendran, and E. E. Jan, "Spatially selective sound capture for speech and audio processing," Speech Commun., vol. 13, no. 1-2, pp. 207-222, Oct. 1993.
-
(1993)
Speech Commun.
, vol.13
, Issue.1-2
, pp. 207-222
-
-
Flanagan, J.L.1
Surendran, A.C.2
Jan, E.E.3
-
6
-
-
0031234613
-
A signal subspace tracking algorithm for microphone array processing of speech
-
Sep.
-
S. Affes and Y. Grenier, "A signal subspace tracking algorithm for microphone array processing of speech," IEEE Trans. Speech Audio Process., vol. 5, no. 5, pp. 425-437, Sep. 1997.
-
(1997)
IEEE Trans. Speech Audio Process.
, vol.5
, Issue.5
, pp. 425-437
-
-
Affes, S.1
Grenier, Y.2
-
7
-
-
0035424281
-
Signal enhancement using beamforming and nonstationarity with applications to speech
-
Aug.
-
S. Gannot, D. Burshtein, and E. Weinstein, "Signal enhancement using beamforming and nonstationarity with applications to speech," IEEE Trans. Signal Process., vol. 49, no. 8, pp. 1614-1626, Aug. 2001.
-
(2001)
IEEE Trans. Signal Process.
, vol.49
, Issue.8
, pp. 1614-1626
-
-
Gannot, S.1
Burshtein, D.2
Weinstein, E.3
-
8
-
-
0033321732
-
A robust adaptive beamformer for microphone arrays with a blocking matrix using constrained adaptive filters
-
Oct.
-
O. Hoshuyama, A. Sugiyama, and A. Hirano, "A robust adaptive beamformer for microphone arrays with a blocking matrix using constrained adaptive filters," IEEE Trans. Signal Process., vol. 47, no. 10, pp. 2677-2684, Oct. 1999.
-
(1999)
IEEE Trans. Signal Process.
, vol.47
, Issue.10
, pp. 2677-2684
-
-
Hoshuyama, O.1
Sugiyama, A.2
Hirano, A.3
-
9
-
-
22544485893
-
An acoustic human-machine front-end for multimedia applications
-
W. Herbordt, H. Buchner, and W. Kellermann, "An acoustic human-machine front-end for multimedia applications," EURASIP J. Appl. Signal Process., vol. 2003, no. 1, pp. 21-31, 2003.
-
(2003)
EURASIP J. Appl. Signal Process.
, vol.2003
, Issue.1
, pp. 21-31
-
-
Herbordt, W.1
Buchner, H.2
Kellermann, W.3
-
10
-
-
5744231382
-
Spatially pre-processed speech distortion weighted multi-channel Wiener filtering for noise reduction
-
Dec.
-
A. Spriet, M. Moonen, and J. Wouters, "Spatially pre-processed speech distortion weighted multi-channel Wiener filtering for noise reduction," Signal Process., vol. 84, no. 12, pp. 2367-2387, Dec. 2004.
-
(2004)
Signal Process.
, vol.84
, Issue.12
, pp. 2367-2387
-
-
Spriet, A.1
Moonen, M.2
Wouters, J.3
-
11
-
-
34447286933
-
Frequency-domain criterion for the speech distortion weighted multichannel Wiener filter for robust noise reduction
-
-Aug.
-
S. Doclo, A. Spriet, J. Wouters, and M. Moonen, "Frequency-domain criterion for the speech distortion weighted multichannel Wiener filter for robust noise reduction," Speech Commun., vol. 49, no. 7-8, pp. 636-656, Jul.-Aug. 2007.
-
(2007)
Speech Commun.
, vol.49
, Issue.7-8
, pp. 636-656+Jul
-
-
Doclo, S.1
Spriet, A.2
Wouters, J.3
Moonen, M.4
-
12
-
-
67651154520
-
Beamforming with a maximum negentropy criterion
-
Jul.
-
K. Kumatani, J. McDonough, B. Rauch, D. Klakow, P. N. Garner, and W. Li, "Beamforming with a maximum negentropy criterion," IEEE Trans. Audio, Speech, Lang. Process., vol. 17, no. 5, pp. 994-1008, Jul. 2009.
-
(2009)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.17
, Issue.5
, pp. 994-1008
-
-
Kumatani, K.1
McDonough, J.2
Rauch, B.3
Klakow, D.4
Garner, P.N.5
Li, W.6
-
13
-
-
51449123556
-
Blind acoustic beamforming based on generalized eigenvalue decomposition
-
Jul.
-
E. Warsitz and R. Haeb-Umbach, "Blind acoustic beamforming based on generalized eigenvalue decomposition," IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 5, pp. 1529-1539, Jul. 2007.
-
(2007)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.15
, Issue.5
, pp. 1529-1539
-
-
Warsitz, E.1
Haeb-Umbach, R.2
-
14
-
-
14344274593
-
A new method based on spectral subtraction for speech dereverberation
-
May
-
K. Lebart, J. M. Boucher, and P. N. Denbigh, "A new method based on spectral subtraction for speech dereverberation," Acta Acust. united with Acust., vol. 87, no. 3, pp. 359-366, May 2001.
-
(2001)
Acta Acust. United with Acust.
, vol.87
, Issue.3
, pp. 359-366
-
-
Lebart, K.1
Boucher, J.M.2
Denbigh, P.N.3
-
15
-
-
77955697587
-
Late reverberant spectral variance estimation based on a statistical model
-
Sep.
-
E. A. P. Habets, S. Gannot, and I. Cohen, "Late reverberant spectral variance estimation based on a statistical model," IEEE Signal Process. Lett., vol. 16, no. 9, pp. 770-773, Sep. 2009.
-
(2009)
IEEE Signal Process. Lett.
, vol.16
, Issue.9
, pp. 770-773
-
-
Habets, E.A.P.1
Gannot, S.2
Cohen, I.3
-
16
-
-
65249167097
-
Suppression of late reverberation effect on speech signal using long-term multiple-step linear prediction
-
May
-
K. Kinoshita, M. Delcroix, T. Nakatani, and M. Miyoshi, "Suppression of late reverberation effect on speech signal using long-term multiple-step linear prediction," IEEE Trans. Audio, Speech, Lang. Process., vol. 17, no. 4, pp. 534-545, May 2009.
-
(2009)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.17
, Issue.4
, pp. 534-545
-
-
Kinoshita, K.1
Delcroix, M.2
Nakatani, T.3
Miyoshi, M.4
-
17
-
-
77955680097
-
Correlation-based and model-based blind single-channel late-reverberation suppression in noisy time-varying acoustical environments
-
Sep.
-
J. S. Erkelens and R. Heusdens, "Correlation-based and model-based blind single-channel late-reverberation suppression in noisy time-varying acoustical environments," IEEE Trans. Audio, Speech, Lang. Process., vol. 18, no. 7, pp. 1746-1765, Sep. 2010.
-
(2010)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.18
, Issue.7
, pp. 1746-1765
-
-
Erkelens, J.S.1
Heusdens, R.2
-
18
-
-
77955676316
-
Model-based dereverberation preserving binaural cues
-
Sep.
-
M. Jeub, M. Schäfer, T. Esch, and P. Vary, "Model-based dereverberation preserving binaural cues," IEEE Trans. Audio, Speech, Lang. Process., vol. 18, no. 7, pp. 1732-1745, Sep. 2010.
-
(2010)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.18
, Issue.7
, pp. 1732-1745
-
-
Jeub, M.1
Schäfer, M.2
Esch, T.3
Vary, P.4
-
19
-
-
0023961145
-
Inverse filtering of room acoustics
-
Feb.
-
M. Miyoshi and Y. Kaneda, "Inverse filtering of room acoustics," IEEE Trans. Acoust., Speech, Signal Process., vol. 36, no. 2, pp. 145-152, Feb. 1988.
-
(1988)
IEEE Trans. Acoust., Speech, Signal Process.
, vol.36
, Issue.2
, pp. 145-152
-
-
Miyoshi, M.1
Kaneda, Y.2
-
20
-
-
0037235030
-
A class of frequency-domain adaptive approaches to blind multichannel identification
-
Jan.
-
Y. Huang and J. Benesty, "A class of frequency-domain adaptive approaches to blind multichannel identification," IEEE Trans. Signal Process., vol. 51, no. 1, pp. 11-24, Jan. 2003.
-
(2003)
IEEE Trans. Signal Process.
, vol.51
, Issue.1
, pp. 11-24
-
-
Huang, Y.1
Benesty, J.2
-
21
-
-
0242271432
-
Subspace methods for multimicrophone speech dereverberation
-
S. Gannot and M. Moonen, "Subspace methods for multimicrophone speech dereverberation," EURASIP J. Appl. Signal Process., vol. 2003, no. 11, pp. 1074-1090, 2003.
-
(2003)
EURASIP J. Appl. Signal Process.
, vol.2003
, Issue.11
, pp. 1074-1090
-
-
Gannot, S.1
Moonen, M.2
-
22
-
-
84555197048
-
Cross-relation-based blind SIMO identifiability in the presence of near-common zeros and noise
-
Jan.
-
D. Schmid and G. Enzner, "Cross-relation-based blind SIMO identifiability in the presence of near-common zeros and noise," IEEE Trans. Signal Process., vol. 60, no. 1, pp. 60-72, Jan. 2012.
-
(2012)
IEEE Trans. Signal Process.
, vol.60
, Issue.1
, pp. 60-72
-
-
Schmid, D.1
Enzner, G.2
-
23
-
-
79951653603
-
Robust speech dereverberation based on blind adaptive estimation of acoustic channels
-
May
-
M. A. Haque, T. Islam, and M. K. Hasan, "Robust speech dereverberation based on blind adaptive estimation of acoustic channels," IEEE Trans. Audio, Speech, Lang. Process., vol. 19, no. 4, pp. 775-787, May 2011.
-
(2011)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.19
, Issue.4
, pp. 775-787
-
-
Haque, M.A.1
Islam, T.2
Hasan, M.K.3
-
24
-
-
84856278998
-
A forced spectral diversity algorithm for speech dereverberation in the presence of near-common zeros
-
Mar.
-
X. Lin, A. W. H. Khong, and P. A. Naylor, "A forced spectral diversity algorithm for speech dereverberation in the presence of near-common zeros," IEEE Trans. Audio, Speech, Lang. Process., vol. 20, no. 3, pp. 888-899, Mar. 2012.
-
(2012)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.20
, Issue.3
, pp. 888-899
-
-
Lin, X.1
Khong, A.W.H.2
Naylor, P.A.3
-
25
-
-
50449100228
-
Robust speech dereverberation using multichannel blind deconvolution with spectral subtraction
-
Jul.
-
K. Furuya and A. Kataoka, "Robust speech dereverberation using multichannel blind deconvolution with spectral subtraction," IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 5, pp. 1579-1591, Jul. 2007.
-
(2007)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.15
, Issue.5
, pp. 1579-1591
-
-
Furuya, K.1
Kataoka, A.2
-
26
-
-
85091450770
-
TRINICON for dereverberation of speech and audio signals
-
P. A. Naylor and N. D. Gaubitch, Eds., 1st ed. London, U.K.: Springer, ch. 10
-
H. Buchner and W. Kellermann, "TRINICON for dereverberation of speech and audio signals," in Speech Dereverberation, P. A. Naylor and N. D. Gaubitch, Eds., 1st ed. London, U.K.: Springer, 2010, ch. 10, pp. 311-385.
-
(2010)
Speech Dereverberation
, pp. 311-385
-
-
Buchner, H.1
Kellermann, W.2
-
27
-
-
34548569780
-
Precise dereverberation using multichannel linear prediction
-
Feb.
-
M. Delcroix, T. Hikichi, and M. Miyoshi, "Precise dereverberation using multichannel linear prediction," IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 2, pp. 430-440, Feb. 2007.
-
(2007)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.15
, Issue.2
, pp. 430-440
-
-
Delcroix, M.1
Hikichi, T.2
Miyoshi, M.3
-
28
-
-
34548555989
-
Dereverberation by using time-variant nature of speech production system
-
T. Yoshioka, T. Hikichi, and M. Miyoshi, "Dereverberation by using time-variant nature of speech production system," EURASIP J. Adv. Signal Process., vol. 2007, no. 1, pp. 1-15, 2007.
-
(2007)
EURASIP J. Adv. Signal Process.
, vol.2007
, Issue.1
, pp. 1-15
-
-
Yoshioka, T.1
Hikichi, T.2
Miyoshi, M.3
-
29
-
-
70350458846
-
Speech dereverberation based on maximum-likelihood estimation with time-varying Gaussian source model
-
Nov.
-
T. Nakatani, B.-H. Juang, T. Yoshioka, K. Kinoshita, M. Delcroix, and M. Miyoshi, "Speech dereverberation based on maximum-likelihood estimation with time-varying Gaussian source model," IEEE Trans. Audio, Speech, Lang. Process., vol. 16, no. 8, pp. 1512-1527, Nov. 2008.
-
(2008)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.16
, Issue.8
, pp. 1512-1527
-
-
Nakatani, T.1
Juang, B.-H.2
Yoshioka, T.3
Kinoshita, K.4
Delcroix, M.5
Miyoshi, M.6
-
30
-
-
77955698459
-
Speech dereverberation based on variance-normalized delayed linear prediction
-
Sep.
-
T. Nakatani, T. Yoshioka, K. Kinoshita, M. Miyoshi, and B.-H. Juang, "Speech dereverberation based on variance-normalized delayed linear prediction," IEEE Trans. Audio, Speech, Lang. Process., vol. 18, no. 7, pp. 1717-1731, Sep. 2010.
-
(2010)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.18
, Issue.7
, pp. 1717-1731
-
-
Nakatani, T.1
Yoshioka, T.2
Kinoshita, K.3
Miyoshi, M.4
Juang, B.-H.5
-
31
-
-
84898688036
-
Speech denoising and dereverberation using probabilistic models
-
H. Attias, J. C. Platt, A. Acero, and L. Deng, "Speech denoising and dereverberation using probabilistic models," Adv. Neural Inf. Process. Syst., vol. 13, pp. 758-764, 2001.
-
(2001)
Adv. Neural Inf. Process. Syst.
, vol.13
, pp. 758-764
-
-
Attias, H.1
Platt, J.C.2
Acero, A.3
Deng, L.4
-
32
-
-
70350435249
-
Integrated speech enhancement method using noise suppression and dereverberation
-
Feb.
-
T. Yoshioka, T. Nakatani, and M. Miyoshi, "Integrated speech enhancement method using noise suppression and dereverberation," IEEE Trans. Audio, Speech, Lang. Process., vol. 17, no. 2, pp. 231-246, Feb. 2009.
-
(2009)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.17
, Issue.2
, pp. 231-246
-
-
Yoshioka, T.1
Nakatani, T.2
Miyoshi, M.3
-
33
-
-
0042362199
-
Blind single channel deconvolution using nonstationary signal processing
-
Sep.
-
J. R. Hopgood and P. J. W. Rayner, "Blind single channel deconvolution using nonstationary signal processing," IEEE Trans. Speech Audio Process., vol. 11, no. 5, pp. 476-488, Sep. 2003.
-
(2003)
IEEE Trans. Speech Audio Process.
, vol.11
, Issue.5
, pp. 476-488
-
-
Hopgood, J.R.1
Rayner, P.J.W.2
-
34
-
-
44649112061
-
Parametric modelling for single-channel blind dereverberation of speech from a moving speaker
-
Jun.
-
C. Evers and J. R. Hopgood, "Parametric modelling for single-channel blind dereverberation of speech from a moving speaker," IET Signal Process., vol. 2, no. 2, pp. 59-74, Jun. 2008.
-
(2008)
IET Signal Process.
, vol.2
, Issue.2
, pp. 59-74
-
-
Evers, C.1
Hopgood, J.R.2
-
35
-
-
79956341145
-
Multichannel online blind speech dereverberation with marginalization of static observation parameters in a Rao-Blackwellized particle filter
-
C. Evers and J. R. Hopgood, "Multichannel online blind speech dereverberation with marginalization of static observation parameters in a Rao-Blackwellized particle filter," J. Signal Process. Syst., vol. 63, no. 3, pp. 315-332, 2011.
-
(2011)
J. Signal Process. Syst.
, vol.63
, Issue.3
, pp. 315-332
-
-
Evers, C.1
Hopgood, J.R.2
-
36
-
-
3543081155
-
-
Ph.D. dissertation, Univ. College London, London, U.K.
-
M. J. Beal, "Variational algorithms for approximate Bayesian inference," Ph.D. dissertation, Univ. College London, London, U.K., 2003.
-
(2003)
Variational Algorithms for Approximate Bayesian Inference
-
-
Beal, M.J.1
-
38
-
-
84867589358
-
An expectation-maximization algorithm for multichannel adaptive speech dereverberation in the frequency-domain
-
D. Schmid, S. Malik, and G. Enzner, "An expectation-maximization algorithm for multichannel adaptive speech dereverberation in the frequency-domain," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., Kyoto, Japan, Mar. 2012, pp. 17-20.
-
Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., Kyoto, Japan, Mar. 2012
, pp. 17-20
-
-
Schmid, D.1
Malik, S.2
Enzner, G.3
-
39
-
-
84957613950
-
A maximum a posteriori approach to multichannel speech dereverberation and denoising
-
D. Schmid, S. Malik, and G. Enzner, "A maximum a posteriori approach to multichannel speech dereverberation and denoising," in Proc. Int. Workshop Acoust. Signal Enhanc., Aachen, Germany, Sep. 2012.
-
Proc. Int. Workshop Acoust. Signal Enhanc., Aachen, Germany, Sep. 2012
-
-
Schmid, D.1
Malik, S.2
Enzner, G.3
-
40
-
-
33645935392
-
Frequency-domain adaptive Kalman filter for acoustic echo control in hands-free telephones
-
Jun.
-
G. Enzner and P. Vary, "Frequency-domain adaptive Kalman filter for acoustic echo control in hands-free telephones," Signal Process., vol. 86, no. 6, pp. 1140-1156, Jun. 2006.
-
(2006)
Signal Process.
, vol.86
, Issue.6
, pp. 1140-1156
-
-
Enzner, G.1
Vary, P.2
-
41
-
-
84887981929
-
-
Ph.D. dissertation, Inst. of Commun. Acoust., Ruhr-Univ. Bochum, Bochum, Germany
-
S. Malik, "Bayesian learning of linear and nonlinear acoustic system models in hands-free communication," Ph.D. dissertation, Inst. of Commun. Acoust., Ruhr-Univ. Bochum, Bochum, Germany, 2012.
-
(2012)
Bayesian Learning of Linear and Nonlinear Acoustic System Models in Hands-free Communication
-
-
Malik, S.1
-
42
-
-
84887925478
-
A variational Bayesian learning approach for nonlinear acoustic echo control
-
Dec.
-
S. Malik and G. Enzner, "A variational Bayesian learning approach for nonlinear acoustic echo control," IEEE Trans. Signal Process., vol. 61, no. 23, pp. 5853-5867, Dec. 2013.
-
(2013)
IEEE Trans. Signal Process.
, vol.61
, Issue.23
, pp. 5853-5867
-
-
Malik, S.1
Enzner, G.2
-
43
-
-
84911479930
-
-
Ph.D. dissertation, Inst. of Commun. Acoust., Ruhr-Univ. Bochum, Bochum, Germany
-
D. Schmid, "Multichannel dereverberation and noise reduction for hands-free speech communication systems," Ph.D. dissertation, Inst. of Commun. Acoust., Ruhr-Univ. Bochum, Bochum, Germany, 2014.
-
(2014)
Multichannel Dereverberation and Noise Reduction for Hands-free Speech Communication Systems
-
-
Schmid, D.1
-
44
-
-
78049379406
-
Variational Bayesian blind estimation of SIMO channels
-
K. Harada and H. Sakai, "Variational Bayesian blind estimation of SIMO channels," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., Dallas, TX, USA, Mar. 2010, pp. 3218-3221.
-
Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., Dallas, TX, USA, Mar. 2010
, pp. 3218-3221
-
-
Harada, K.1
Sakai, H.2
-
45
-
-
0026753533
-
Frequency-domain andmultirate adaptive filtering
-
Jan.
-
J. J. Shynk, "Frequency-domain andmultirate adaptive filtering," IEEE Signal Process. Mag., vol. 9, no. 1, pp. 14-37, Jan. 1992.
-
(1992)
IEEE Signal Process. Mag.
, vol.9
, Issue.1
, pp. 14-37
-
-
Shynk, J.J.1
-
46
-
-
2742574944
-
-
Ph.D. dissertation, Univ. of Sussex, Brighton and Hove, U.K.
-
D. J. Salmond, "Tracking in uncertain environments," Ph.D. dissertation, Univ. of Sussex, Brighton and Hove, U.K., 1989.
-
(1989)
Tracking in Uncertain Environments
-
-
Salmond, D.J.1
-
47
-
-
56749171334
-
Generating nonstationary multisensor signals under a spatial coherence constraint
-
Nov.
-
E. A. P. Habets, I. Cohen, and S. Gannot, "Generating nonstationary multisensor signals under a spatial coherence constraint," J. Acoust. Soc. Amer., vol. 124, no. 5, pp. 2911-2917, Nov. 2008.
-
(2008)
J. Acoust. Soc. Amer.
, vol.124
, Issue.5
, pp. 2911-2917
-
-
Habets, E.A.P.1
Cohen, I.2
Gannot, S.3
-
48
-
-
78049402824
-
Online maximum-likelihood learning of time-varying dynamical models in block-frequency-domain
-
S. Malik and G. Enzner, "Online maximum-likelihood learning of time-varying dynamical models in block-frequency-domain," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., Dallas, TX, USA, Mar. 2010, pp. 3822-3825.
-
Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., Dallas, TX, USA, Mar. 2010
, pp. 3822-3825
-
-
Malik, S.1
Enzner, G.2
-
49
-
-
47649120090
-
Prediction of energy decay in room impulse responses simulated with an image-source model
-
Jul.
-
E. A. Lehmann and A. M. Johansson, "Prediction of energy decay in room impulse responses simulated with an image-source model," J. Acoust. Soc. Amer., vol. 124, no. 1, pp. 269-277, Jul. 2008.
-
(2008)
J. Acoust. Soc. Amer.
, vol.124
, Issue.1
, pp. 269-277
-
-
Lehmann, E.A.1
Johansson, A.M.2
-
50
-
-
0018455820
-
Image method for efficiently simulating small-room acoustics
-
Apr.
-
J. B. Allen and D. A. Berkley, "Image method for efficiently simulating small-room acoustics," J. Acoust. Soc. Amer., vol. 65, no. 4, pp. 943-950, Apr. 1979.
-
(1979)
J. Acoust. Soc. Amer.
, vol.65
, Issue.4
, pp. 943-950
-
-
Allen, J.B.1
Berkley, D.A.2
-
51
-
-
85132863054
-
Models, measurement and evaluation
-
P. A. Naylor and N. D. Gaubitch, Eds., 1st ed. London, U.K.: Springer, ch. 2
-
P. A. Naylor, E. A. P. Habets, J. Y.-C. Wen, and N. D. Gaubitch, "Models, measurement and evaluation," in Speech Dereverberation, P. A. Naylor and N. D. Gaubitch, Eds., 1st ed. London, U.K.: Springer, 2010, ch. 2, pp. 21-56.
-
(2010)
Speech Dereverberation
, pp. 21-56
-
-
Naylor, P.A.1
Habets, E.A.P.2
Wen, J.Y.-C.3
Gaubitch, N.D.4
-
52
-
-
0003548585
-
-
Linguistic Data Consortium, Philadelphia, PA, USA
-
J. S. Garofolo, L. F. Lamel, W. M. Fisher, J. G. Fiscus, D. S. Pallett, N. L. Dahlgren, and V. Zue, TIMIT Acoustic-Phonetic Continuous Speech Corpus, Linguistic Data Consortium, Philadelphia, PA, USA, 1993.
-
(1993)
TIMIT Acoustic-Phonetic Continuous Speech Corpus
-
-
Garofolo, J.S.1
Lamel, L.F.2
Fisher, W.M.3
Fiscus, J.G.4
Pallett, D.S.5
Dahlgren, N.L.6
Zue, V.7
-
54
-
-
44149106061
-
Evaluation of objective quality measures for speech enhancement
-
Jan.
-
Y. Hu and P. C. Loizou, "Evaluation of objective quality measures for speech enhancement," IEEE Trans. Audio, Speech, Lang. Process., vol. 16, no. 1, pp. 229-238, Jan. 2008.
-
(2008)
IEEE Trans. Audio, Speech, Lang. Process.
, vol.16
, Issue.1
, pp. 229-238
-
-
Hu, Y.1
Loizou, P.C.2
-
55
-
-
85032751613
-
Making machines understand us in reverberant rooms: Robustness against reverberation for automatic speech recognition
-
Nov.
-
T. Yoshioka, A. Sehr, M. Delcroix, K. Kinoshita, R. Maas, T. Nakatani, and W. Kellermann, "Making machines understand us in reverberant rooms: Robustness against reverberation for automatic speech recognition," IEEE Signal Process. Mag., vol. 29, no. 6, pp. 114-126, Nov. 2012.
-
(2012)
IEEE Signal Process. Mag.
, vol.29
, Issue.6
, pp. 114-126
-
-
Yoshioka, T.1
Sehr, A.2
Delcroix, M.3
Kinoshita, K.4
Maas, R.5
Nakatani, T.6
Kellermann, W.7
-
56
-
-
70450143974
-
-
Linguistic Data Consortium, Philadelphia, PA, USA
-
J. Garofalo, D. Graff, D. Paul, and D. Pallett, CSR-I (WSJ0) Sennheiser, Linguistic Data Consortium, Philadelphia, PA, USA, 1993.
-
(1993)
CSR-I (WSJ0) Sennheiser
-
-
Garofalo, J.1
Graff, D.2
Paul, D.3
Pallett, D.4
-
57
-
-
51449115975
-
-
Univ. of Cambridge, Cambridge, U.K., Tech. Rep., [Online]. Available
-
K. Vertanen, "Baseline WSJ acoustic models for HTK and Sphinx: Training recipes and recognition experiments," Univ. of Cambridge, Cambridge, U.K., Tech. Rep., 2006. [Online]. Available: http://www.keithv.com/pub/baselinewsj
-
(2006)
Baseline WSJ Acoustic Models for HTK and Sphinx: Training Recipes and Recognition Experiments
-
-
Vertanen, K.1
-
59
-
-
84947967217
-
Dereverberation preprocessing and training data adjustments for robust speech recognition in reverberant environments
-
D. Schmid, P. Thüne, D. Kolossa, and G. Enzner, "Dereverberation preprocessing and training data adjustments for robust speech recognition in reverberant environments," in Proc. ITG Conf. Speech Commun., Braunschweig, Germany, Sep. 2012, pp. 235-238.
-
Proc. ITG Conf. Speech Commun., Braunschweig, Germany, Sep. 2012
, pp. 235-238
-
-
Schmid, D.1
Thüne, P.2
Kolossa, D.3
Enzner, G.4
-
60
-
-
0036475447
-
A tutorial on particle filters for online nonlinear/non-Gaussian Bayesian tracking
-
Feb.
-
M. S. Arulampalam, S. Maskell, N. Gordon, and T. Clapp, "A tutorial on particle filters for online nonlinear/non-Gaussian Bayesian tracking," IEEE Trans. Signal Process., vol. 50, no. 2, pp. 174-188, Feb. 2002.
-
(2002)
IEEE Trans. Signal Process.
, vol.50
, Issue.2
, pp. 174-188
-
-
Arulampalam, M.S.1
Maskell, S.2
Gordon, N.3
Clapp, T.4
-
61
-
-
0003732219
-
-
Berlin, Germany: Springer
-
J. Benesty, T. Gänsler, D. R. Morgan, M. M. Sondhi, and S. L. Gay, Advances in Network and Acoustic Echo Cancellation, 1st ed. Berlin, Germany: Springer, 2001.
-
(2001)
Advances in Network and Acoustic Echo Cancellation, 1st Ed.
-
-
Benesty, J.1
Gänsler, T.2
Morgan, D.R.3
Sondhi, M.M.4
Gay, S.L.5
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