A new adaptive filtering subband algorithm for two-channel acoustic noise reduction and speech enhancement

•We propose a new subband implementation of the Forward BSS algorithm.•We propose a new two-channel criterion combination adapted to the proposed FBSS algorithm.•The proposed FBSS algorithm is used to separate speech form noise and therefore enhance the speech signal intelligibility.•The proposed algorithm improves the convergence speed of the cross adaptive filters which are used by the FBSS algorithm.

This paper addresses the problem of acoustic noise reduction and speech enhancement by adaptive filtering algorithms. Most speech enhancement methods and algorithms which use adaptive filtering structure are generally expressed in fullband form. One of these widespread structures is the Forward Blind Source Separation Structure (FBSS). This FBSS structure is often used to separate speech form noise and therefore enhance the speech signal at the processing output. In this paper, we propose a new subband implementation of this FBSS structure. In order to give more robustness to the proposed structure, we adapt then we apply to this subband structure a new combination of criteria based on the system mismatch and the smoothing filtering errors minimizations. The combination between this proposed subband structure with this optimal criteria allows to obtain a new two-channel subband forward (2CSF) algorithm that improves the convergence speed of the cross adaptive filters which are used to separate speech from noise. Objective tests under various environments are presented showing the good behavior of the proposed 2CSF algorithm.

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