Y. Avargel and I. Cohen, On multiplicative transfer function approximation in the short-time Fourier transform domain, IEEE Signal Processing Letters, vol.14, issue.5, pp.337-340, 2007.

S. Gannot, D. Burshtein, and E. Weinstein, Signal enhancement using beamforming and nonstationarity with applications to speech, IEEE Transactions on Signal Processing, vol.49, issue.8, pp.1614-1626, 2001.

X. Li, L. Girin, R. Horaud, and S. Gannot, Estimation of relative transfer function in the presence of stationary noise based on segmental power spectral density matrix subtraction, IEEE International Conference on Acoustics, Speech and Signal Processing, pp.320-324, 2015.
URL : https://hal.archives-ouvertes.fr/hal-01119186

H. L. , Van Trees, Detection, estimation, and modulation theory, 2004.

O. Yilmaz and S. Rickard, Blind separation of speech mixtures via time-frequency masking, IEEE Transactions on Signal Processing, vol.52, issue.7, pp.1830-1847, 2004.

M. I. Mandel, R. J. Weiss, and D. P. Ellis, Model-based expectationmaximization source separation and localization, IEEE Transactions on Audio, Speech, and Language Processing, vol.18, issue.2, pp.382-394, 2010.

S. Winter, W. Kellermann, H. Sawada, and S. Makino, MAP-based underdetermined blind source separation of convolutive mixtures by hierarchical clustering and 1-norm minimization, EURASIP Journal on Applied Signal Processing, issue.1, pp.81-81, 2007.

S. Gannot, E. Vincent, S. Markovich-golan, and A. Ozerov, A consolidated perspective on multimicrophone speech enhancement and source separation, IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol.25, issue.4, pp.692-730, 2017.
URL : https://hal.archives-ouvertes.fr/hal-01414179

M. Kowalski, E. Vincent, and R. Gribonval, Beyond the narrowband approximation: Wideband convex methods for under-determined reverberant audio source separation, IEEE Transactions on Audio, Speech, and Language Processing, vol.18, issue.7, pp.1818-1829, 2010.
URL : https://hal.archives-ouvertes.fr/hal-00435897

S. Arberet, P. Vandergheynst, J. Carrillo, R. E. Thiran, and Y. Wiaux, Sparse reverberant audio source separation via reweighted analysis, IEEE Transactions on Audio, Speech, and Language Processing, vol.21, issue.7, pp.1391-1402, 2013.

M. Miyoshi and Y. Kaneda, Inverse filtering of room acoustics, IEEE Transactions on Acoustics, Speech, and Signal Processing, vol.36, issue.2, pp.145-152, 1988.

M. Kallinger and A. Mertins, Multi-channel room impulse response shaping-a study, IEEE International Conference on Acoustics Speech and Signal Processing Proceedings (ICASSP), vol.5, pp.101-104, 2006.

A. Mertins, T. Mei, and M. Kallinger, Room impulse response shortening/reshaping with infinity-and p-norm optimization, IEEE Transactions on Audio, Speech, and Language Processing, vol.18, issue.2, pp.249-259, 2010.

I. Kodrasi, S. Goetze, and S. Doclo, Regularization for partial multichannel equalization for speech dereverberation, IEEE Transactions on Audio, Speech, and Language Processing, vol.21, issue.9, pp.1879-1890, 2013.

I. Kodrasi and S. Doclo, Joint dereverberation and noise reduction based on acoustic multi-channel equalization, IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol.24, issue.4, pp.680-693, 2016.

T. Hikichi, M. Delcroix, and M. Miyoshi, Inverse filtering for speech dereverberation less sensitive to noise and room transfer function fluctuations, EURASIP Journal on Advances in Signal Processing, issue.1, pp.1-12, 2007.
DOI : 10.1155/2007/34013
URL : https://asp-eurasipjournals.springeropen.com/track/pdf/10.1155/2007/34013

Y. Huang, J. Benesty, and J. Chen, A blind channel identification-based two-stage approach to separation and dereverberation of speech signals in a reverberant environment, IEEE Transactions on Speech and Audio Processing, vol.13, issue.5, pp.882-895, 2005.

H. Yamada, H. Wang, and F. Itakura, Recovering of broadband reverberant speech signal by sub-band MINT method, IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), pp.969-972, 1991.

H. Wang and F. Itakura, Realization of acoustic inverse filtering through multi-microphone sub-band processing, IEICE Transactions on Fundamentals of Electronics, vol.75, issue.11, pp.1474-1483, 1992.

S. Weiss, G. W. Rice, and R. W. Stewart, Multichannel equalization in subbands, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, pp.203-206, 1999.
DOI : 10.1109/aspaa.1999.810885
URL : https://eprints.soton.ac.uk/251923/1/weiss99a.pdf

N. D. Gaubitch and P. A. Naylor, Equalization of multichannel acoustic systems in oversampled subbands, IEEE Transactions on Audio, Speech, and Language Processing, vol.17, issue.6, pp.1061-1070, 2009.

F. Lim and P. A. Naylor, Robust speech dereverberation using subband multichannel least squares with variable relaxation, European Signal Processing Conference (EUSIPCO), 2013.
DOI : 10.1109/taslp.2014.2329632

Y. Avargel and I. Cohen, System identification in the short-time Fourier transform domain with crossband filtering, IEEE Transactions on Audio, Speech, and Language Processing, vol.15, issue.4, pp.1305-1319, 2007.

R. Talmon, I. Cohen, and S. Gannot, Relative transfer function identification using convolutive transfer function approximation, IEEE Transactions on Audio, Speech, and Language Processing, vol.17, issue.4, pp.546-555, 2009.
DOI : 10.1109/tasl.2008.2009576

R. Talmon, I. Cohen, and S. Gannot, Convolutive transfer function generalized sidelobe canceler, IEEE transactions on audio, speech, and language processing, vol.17, pp.1420-1434, 2009.
DOI : 10.1109/tasl.2009.2020891

X. Li, L. Girin, R. Horaud, and S. Gannot, Estimation of the directpath relative transfer function for supervised sound-source localization, IEEE/ACM Transactions on Audio, Speech and Language Processing, vol.24, issue.11, pp.2171-2186, 2016.
URL : https://hal.archives-ouvertes.fr/hal-01349691

X. Li, L. Girin, R. Horaud, and S. Gannot, Multiple-speaker localization based on direct-path features and likelihood maximization with spatial sparsity regularization, IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol.25, issue.10, 1997.
DOI : 10.1109/taslp.2017.2740001
URL : https://hal.archives-ouvertes.fr/hal-01413417

X. Li, L. Girin, and R. Horaud, Audio source separation based on convolutive transfer function and frequency-domain lasso optimization, IEEE International Conference on Acoustics, Speech, and Signal Processing, 2017.
DOI : 10.1109/icassp.2017.7952214
URL : https://hal.archives-ouvertes.fr/hal-01430754

S. Leglaive, R. Badeau, and G. Richard, Multichannel audio source separation: variational inference of time-frequency sources from timedomain observations, IEEE International Conference on Acoustics, Speech and Signal Processing, 2017.
DOI : 10.1109/icassp.2017.7951791
URL : https://hal.archives-ouvertes.fr/hal-01416347

S. Leglaive, R. Badeau, and G. Richard, Separating time-frequency sources from time-domain convolutive mixtures using non-negative matrix factorization, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), 2017.
URL : https://hal.archives-ouvertes.fr/hal-01548469

R. Badeau and M. D. Plumbley, Multichannel high-resolution NMF for modeling convolutive mixtures of non-stationary signals in the timefrequency domain, IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol.22, issue.11, pp.1670-1680, 2014.

B. Schwartz, S. Gannot, and E. A. Habets, Online speech dereverberation using kalman filter and EM algorithm, IEEE/ACM Transactions on Audio, Speech and Language Processing, vol.23, issue.2, pp.394-406, 2015.
DOI : 10.1109/taslp.2014.2372342

X. Li, L. Girin, and R. Horaud, An EM algorithm for audio source separation based on the convolutive transfer function, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), 2017.
URL : https://hal.archives-ouvertes.fr/hal-01568818

P. L. Combettes and J. Pesquet, A douglas-rachford splitting approach to nonsmooth convex variational signal recovery, IEEE Journal of Selected Topics in Signal Processing, vol.1, issue.4, pp.564-574, 2007.
URL : https://hal.archives-ouvertes.fr/hal-00621820

P. L. Combettes and V. R. Wajs, Signal recovery by proximal forwardbackward splitting, Multiscale Modeling & Simulation, vol.4, issue.4, pp.1168-1200, 2005.
DOI : 10.1137/050626090
URL : https://hal.archives-ouvertes.fr/hal-00017649

S. Doclo, A. Spriet, J. Wouters, and M. Moonen, Speech distortion weighted multichannel wiener filtering techniques for noise reduction, pp.199-228, 2005.

X. Li, R. Horaud, and S. Gannot, Blind multichannel identification and equalization for dereverberation and noise reduction based on convolutive transfer function, CoRR, 2017.
URL : https://hal.archives-ouvertes.fr/hal-01568835

X. Li, L. Girin, S. Gannot, and R. Horaud, Non-stationary noise power spectral density estimation based on regional statistics, IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pp.181-185, 2016.
URL : https://hal.archives-ouvertes.fr/hal-01250892

C. Forbes, M. Evans, N. Hastings, and B. Peacock, Erlang distribution, Statistical Distributions, pp.84-85, 2010.

R. T. Rockafellar, Convex analysis, 2015.

M. J. Fadili and J. Starck, Monotone operator splitting for optimization problems in sparse recovery, IEEE International Conference on Image Processing, pp.1461-1464, 2009.
URL : https://hal.archives-ouvertes.fr/hal-00813889

Y. Nesterov, Gradient methods for minimizing composite objective function," tech. rep., International Association for Research and Teaching, 2007.

A. Beck and M. Teboulle, A fast iterative shrinkage-thresholding algorithm for linear inverse problems, SIAM journal on imaging sciences, vol.2, issue.1, pp.183-202, 2009.

E. Hadad, F. Heese, P. Vary, and S. Gannot, Multichannel audio database in various acoustic environments, International Workshop on Acoustic Signal Enhancement, pp.313-317, 2014.

J. S. Garofolo, L. F. Lamel, W. M. Fisher, J. G. Fiscus, D. S. Pallett et al., Getting started with the DARPA TIMIT CD-ROM: An acoustic phonetic continuous speech database, vol.107, 1988.

D. R. Morgan, J. Benesty, and M. M. Sondhi, On the evaluation of estimated impulse responses, IEEE Signal processing letters, vol.5, issue.7, pp.174-176, 1998.

E. Vincent, R. Gribonval, and C. Févotte, Performance measurement in blind audio source separation, IEEE transactions on audio, speech, and language processing, vol.14, pp.1462-1469, 2006.
URL : https://hal.archives-ouvertes.fr/inria-00544230

A. W. Rix, J. G. Beerends, M. P. Hollier, and A. P. Hekstra, Perceptual evaluation of speech quality (PESQ)-a new method for speech quality assessment of telephone networks and codecs, IEEE International Conference on Acoustics, Speech, and Signal Processing, vol.2, pp.749-752, 2001.

I. Cohen, S. Gannot, and B. Berdugo, An integrated real-time beamforming and postfiltering system for nonstationary noise environments, EURASIP Journal on Applied Signal Processing, pp.1064-1073, 2003.

S. Gannot and I. Cohen, Speech enhancement based on the general transfer function GSC and postfiltering, IEEE Transactions on Speech and Audio Processing, vol.12, issue.6, pp.561-571, 2004.