Online Speech Dereverberation Using Kalman Filter and EM Algorithm

被引:63
|
作者
Schwartz, Boaz [1 ]
Gannot, Sharon [1 ]
Habets, Emanuel A. P. [2 ]
机构
[1] Bar Ilan Univ, Fac Engn, IL-5290002 Ramat Gan, Israel
[2] Joint Inst Univ Erlangen Nuremberg & Fraunhofer I, Int Audio Labs Erlangen, D-91058 Erlangen, Germany
关键词
Dereverberation; recursive parameter estimation; recursive expectation-maximization; convolution in STFT; EXPECTATION-MAXIMIZATION ALGORITHM; RECURSIVE EM; ENHANCEMENT; DOMAIN; IDENTIFICATION; CONVERGENCE; REVERBERANT; SUPPRESSION; QUALITY; NOISE;
D O I
10.1109/TASLP.2014.2372342
中图分类号
O42 [声学];
学科分类号
070206 ; 082403 ;
摘要
Speech signals recorded in a room are commonly degraded by reverberation. In most cases, both the speech signal and the acoustic system of the room are unknown and time-varying. In this paper, a scenario with a single desired sound source and slowly time-varying and spatially-white noise is considered, and a multi-microphone algorithm that simultaneously estimates the clean speech signal and the time-varying acoustic system is proposed. The recursive expectation-maximization scheme is employed to obtain both the clean speech signal and the acoustic system in an online manner. In the expectation step, the Kalman filter is applied to extract a new sample of the clean signal, and in the maximization step, the system estimate is updated according to the output of the Kalman filter. Experimental results show that the proposed method is able to significantly reduce reverberation and increase the speech quality. Moreover, the tracking ability of the algorithm was validated in practical scenarios using human speakers moving in a natural manner.
引用
收藏
页码:394 / 406
页数:13
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