Abstract

This paper addresses the problem of acoustic noise reduction and speech enhancement in new telecommunication systems by adaptive filtering algorithms. We propose a new two-channel forward BSS algorithm based on signal prediction to give an automatic algorithm with a very nice behavior at the output. This algorithm is called the two-channel fast normalized least mean square (TCFNLMS) algorithm. This new algorithm shows superiority in comparison with classical two-channel NlMS in terms of the following objective criteria: the segmental signal to noise ratio (SegSNR), the cepstral distance (CD), the system mismatch (SM) and the segmental mean square error (SegMSE).

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