Blind-Matched Filtering for Speech Enhancement with Distributed Microphones

A multichannel noise reduction and equalization approach for distributed microphones is presented. The speech enhancement is based on a blind-matched filtering algorithm that combines the microphone signals such that the output SNR is maximized. The algorithm is developed for spatially uncorrelated but nonuniform noise fields, that is, the noise signals at the different microphones are uncorrelated, but the noise power spectral densities can vary. However, no assumptions on the array geometry are made. The proposed method will be compared to the speech distortion-weighted multichannel Wiener filter (SDW-MWF). Similar to the SDW-MWF, the new algorithm requires only estimates of the input signal to noise ratios and the input cross-correlations. Hence, no explicit channel knowledge is necessary. A new version of the SDW-MWF for spatially uncorrelated noise is developed which has a reduced computational complexity, because matrix inversions can be omitted. The presented blind-matched filtering approach is similar to this SDW-MWF for spatially uncorrelated noise but additionally achieves some improvements in the speech quality due to a partial equalization of the acoustic system.

[1]  Peter Vary,et al.  Low Delay Noise Reduction and Dereverberation for Hearing Aids , 2009, EURASIP J. Adv. Signal Process..

[2]  Marc Delcroix,et al.  Dereverberation and Denoising Using Multichannel Linear Prediction , 2007, IEEE Transactions on Audio, Speech, and Language Processing.

[3]  Régine Le Bouquin-Jeannès,et al.  A Two-Sensor Noise Reduction System: Applications for Hands-Free Car Kit , 2003, EURASIP J. Adv. Signal Process..

[4]  Reinhold Häb-Umbach,et al.  Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition , 2007, IEEE Transactions on Audio, Speech, and Language Processing.

[5]  B. Holter,et al.  The optimal weights of a maximum ratio combiner using an eigenfilter approach , 2002 .

[6]  Israel Cohen,et al.  On speech enhancement under signal presence uncertainty , 2001, 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221).

[7]  Dominic Schmid,et al.  Robust subsystems for iterative multichannel blind system identification and equalization , 2009, 2009 IEEE Pacific Rim Conference on Communications, Computers and Signal Processing.

[8]  G. Carter,et al.  The generalized correlation method for estimation of time delay , 1976 .

[9]  Neviano Dal Degan,et al.  Acoustic noise analysis and speech enhancement techniques for mobile radio applications , 1988 .

[10]  Marc Moonen,et al.  Speech Distortion Weighted Multichannel Wiener Filtering Techniques for Noise Reduction , 2005 .

[11]  O. L. Frost,et al.  An algorithm for linearly constrained adaptive array processing , 1972 .

[12]  Rainer Martin,et al.  Soft decision combining for dual channel noise reduction , 2006, INTERSPEECH.

[13]  K. Miller On the Inverse of the Sum of Matrices , 1981 .

[14]  L. J. Griffiths,et al.  An alternative approach to linearly constrained adaptive beamforming , 1982 .

[15]  S. Applebaum,et al.  Adaptive arrays , 1976 .

[16]  Rainer Martin,et al.  Combined acoustic echo cancellation, dereverberation and noise reduction: a two microphone approach , 1994 .

[17]  Richard C. Hendriks,et al.  Noise Correlation Matrix Estimation for Multi-Microphone Speech Enhancement , 2012, IEEE Transactions on Audio, Speech, and Language Processing.

[18]  Jont B. Allen,et al.  Image method for efficiently simulating small‐room acoustics , 1976 .

[19]  S. Gannot,et al.  Speech enhancement based on the general transfer function GSC and postfiltering , 2004, IEEE Trans. Speech Audio Process..

[20]  Emanuel A. P. Habets,et al.  Multi-microphone speech dereverberation using LIME and least squares filtering , 2008, 2008 16th European Signal Processing Conference.

[21]  Henry Cox,et al.  Robust adaptive beamforming , 2005, IEEE Trans. Acoust. Speech Signal Process..

[22]  Marc Moonen,et al.  Robust time-delay estimation in highly adverse acoustic environments , 2001, Proceedings of the 2001 IEEE Workshop on the Applications of Signal Processing to Audio and Acoustics (Cat. No.01TH8575).

[23]  Jürgen Freudenberger,et al.  Microphone Diversity Combining for In-Car Applications , 2010, EURASIP J. Adv. Signal Process..

[24]  Gerhard Schmidt,et al.  Speech and Audio Processing in Adverse Environments , 2008 .

[25]  Richard M. Schwartz,et al.  Enhancement of speech corrupted by acoustic noise , 1979, ICASSP.

[26]  Gerhard Doblinger,et al.  Computationally efficient speech enhancement by spectral minima tracking in subbands , 1995, EUROSPEECH.

[27]  Marc Moonen,et al.  Acoustic Beamforming for Hearing Aid Applications , 2010 .

[28]  Sofiène Affes,et al.  A signal subspace tracking algorithm for microphone array processing of speech , 1997, IEEE Trans. Speech Audio Process..

[29]  Marc Moonen,et al.  Subspace Methods for Multimicrophone Speech Dereverberation , 2003, EURASIP J. Adv. Signal Process..

[30]  Mohsen Rahmani,et al.  An iterative noise cross-PSD estimation for two-microphone speech enhancement , 2009 .

[31]  E.A.P. Habets,et al.  Single-Channel Speech Dereverberation based on Spectral Subtraction , 2004 .

[32]  Marc Moonen,et al.  Frequency-domain criterion for the speech distortion weighted multichannel Wiener filter for robust noise reduction , 2007, Speech Commun..

[33]  Ehud Weinstein,et al.  Signal enhancement using beamforming and nonstationarity with applications to speech , 2001, IEEE Trans. Signal Process..

[34]  Jürgen Freudenberger,et al.  Spectral combining for microphone diversity systems , 2009, 2009 17th European Signal Processing Conference.