Blind dereverberation based on estimates of signal transmission channels without precise information on channel order [speech processing applications]

This paper addresses the blind dereverberation problem of single-input multiple-output acoustic systems. Most approaches require an exact knowledge of the order of the room transfer functions. In this paper, we propose an equalization algorithm that is less sensitive to the order of the estimated transfer functions. First, the transfer functions are estimated using an overestimated order, and the inverse filter set for this estimated transfer functions is calculated. Since the estimated transfer functions have a common part, the signal processed by the inverse filter set contains distortion. Then, we compensate for this distortion using a common polynomial extraction technique. This algorithm enables a reverberated speech signal to be dereverberated as long as the channel is overestimated. Simulation results show that the proposed method is robust even when the order is highly overestimated.

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

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

[3]  K. Furuya,et al.  Two-channel blind deconvolution of nonminimum phase FIR systems , 1997 .

[4]  Tomohiro Nakatani,et al.  One Microphone Blind Dereverberation Based on Quasi-periodicity of Speech Signals , 2003, NIPS.

[5]  Tomohiro Nakatani,et al.  Blind dereverberation of single channel speech signal based on harmonic structure , 2003, 2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03)..

[6]  Masato Miyoshi,et al.  Blind algorithm for calculating common poles based on linear prediction , 2004, 2004 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[7]  Chrysostomos L. Nikias,et al.  EVAM: an eigenvector-based algorithm for multichannel blind deconvolution of input colored signals , 1995, IEEE Trans. Signal Process..

[8]  S.C. Douglas,et al.  Multichannel blind deconvolution and equalization using the natural gradient , 1997, First IEEE Signal Processing Workshop on Signal Processing Advances in Wireless Communications.

[9]  Xiaoan Sun,et al.  A NATURAL GRADIENT CONVOLUTIVE BLIND SOURCE SEPARATION ALGORITHM FOR SPEECH MIXTURES , 2001 .

[10]  Georgios B. Giannakis,et al.  Signal Processing Advances in Wireless and Mobile Communications, Volume 2: Trends in Single- and Multi-User Systems , 2000 .

[11]  Mohamed Najim,et al.  Cancelling convolutive and additive coloured noises for speech enhancement , 2004, 2004 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[12]  César Caballero-Gaudes,et al.  Robust blind identification of SIMO channels: a support vector regression approach , 2004, 2004 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[13]  D. Harville Matrix Algebra From a Statistician's Perspective , 1998 .

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