Blind Estimation and Suppression of Late Reverberation Utilising Auditory Masking

A new method for blind estimation and suppression of late reverberation of speech signals is presented. The proposed algorithm consists of two steps. In a first step, the reverberation time is blindly determined from the reverberant signal. Then, an approximation of the power spectrum of late reverberation is subtracted from the power spectrum of the reverberant signal. Hence, a preliminary estimation of the anechoic speech spectrum is derived. In a second step, the auditory masking threshold of the clean spectrum estimation is calculated and used to define the coefficients for a nonlinear filter for the reverberant signal, which produces the final enhanced speech signal. The performance of the algorithm is demonstrated on artificially generated signals. Subjective tests are conducted and their results indicate that the quality of the speech signals obtained by the proposed method is superior when compared to previous methods.