Bayesian inference model for applications of time-varying acoustic system identification

A major challenge in acoustic signal processing lies in the uncertainty regarding the current state of the acoustic environment. The relevant applications in the field of speech and audio signal processing include the multichannel sound capture, the signal processing for spatial sound control, and the acoustic echo/interference cancellation. In this paper, a Bayesian impulse response model is proposed for acoustic system identification. It is justified by the stochastic nature of time-varying and noisy environments. In particular, we argue for a state-space dynamical model of the unknown impulse responses as a suitable form to incorporate a priori information of the acoustic environment. For the echo/interference cancellation case, we then describe the Bayesian inference of the acoustic system. It is structurally and experimentally compared to maximum-likelihood and least-squares estimators which are both rooted in deterministic system modeling. Algorithmic structure and performance, both speak for the Bayesian inference.

[1]  Ea-Ee Jan,et al.  Matched-filter processing of microphone array for spatial volume selectivity , 1995, Proceedings of ISCAS'95 - International Symposium on Circuits and Systems.

[2]  Don H. Johnson,et al.  Statistical Signal Processing , 2009, Encyclopedia of Biometrics.

[3]  Peter Vary,et al.  Frequency-domain adaptive Kalman filter for acoustic echo control in hands-free telephones , 2006, Signal Process..

[4]  Edward J. Wegman,et al.  Statistical Signal Processing , 1985 .

[5]  Jacob Benesty,et al.  A new class of doubletalk detectors based on cross-correlation , 2000, IEEE Trans. Speech Audio Process..

[6]  Jacob Benesty,et al.  Adaptive multi-channel least mean square and Newton algorithms for blind channel identification , 2002, Signal Process..

[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]  Masato Miyoshi,et al.  Inverse filtering of room acoustics , 1988, IEEE Trans. Acoust. Speech Signal Process..

[9]  S. Haykin,et al.  Adaptive Filter Theory , 1986 .

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

[11]  A. Berkhout,et al.  Acoustic control by wave field synthesis , 1993 .

[12]  E. Hänsler,et al.  Acoustic Echo and Noise Control: A Practical Approach , 2004 .

[13]  Arun Ross,et al.  Microphone Arrays , 2009, Encyclopedia of Biometrics.

[14]  Mickael Tanter,et al.  Sound focusing in rooms: the time-reversal approach. , 2003, The Journal of the Acoustical Society of America.

[15]  Boaz Rafaely,et al.  Microphone Array Signal Processing , 2008 .

[16]  Karim Abed-Meraim,et al.  Blind system identification , 1997, Proc. IEEE.