DocumentCode :
3142713
Title :
An expectation-maximization algorithm for multichannel adaptive speech dereverberation in the frequency-domain
Author :
Schmid, Dominic ; Malik, Sarmad ; Enzner, Gerald
Author_Institution :
Inst. of Commun. Acoust., Ruhr-Univ. Bochum, Bochum, Germany
fYear :
2012
fDate :
25-30 March 2012
Firstpage :
17
Lastpage :
20
Abstract :
This paper presents an online dereverberation algorithm that is derived within the maximum-likelihood expectation-maximization (ML-EM) framework. We formulate an overlap-save observation model for the multichannel blind problem in the DFT-domain. The modeling of acoustic channel impulse responses as random variables with a first-order Markov property facilitates the ensuing algorithm to cope with time-varying conditions. We then show that the ML-EM learning rules for the multichannel state-space model at hand take the form of a recursive posterior estimator for the channels, followed by an equalization stage for recovering the speech signal subject to an expectation with respect to the estimated channel posterior. Our derivation thus results in an iterative ML algorithm for blind equalization and channel identification (ML-BENCH) which comprises two distinct and coupled subsystems. The dereverberation performance of the proposed system is evaluated by considering spectrograms and instrumental quality measures.
Keywords :
Markov processes; blind equalisers; discrete Fourier transforms; expectation-maximisation algorithm; maximum likelihood estimation; speech processing; transient response; DFT-domain; ML-BENCH; ML-EM framework; acoustic channel impulse response; blind equalization; channel identification; coupled subsystems; expectation-maximization algorithm; first-order Markov property; frequency-domain analysis; instrumental quality measures; iterative ML algorithm; maximum-likelihood expectation-maximization framework; multichannel adaptive speech dereverberation; multichannel blind problem; multichannel state-space model; online dereverberation algorithm; overlap-save observation model; random variables; recursive posterior estimator; spectrograms measures; speech signal; time-varying conditions; Frequency modulation; Microphones; Noise; Reverberation; Speech; Vectors; Expectation-maximization; frequency-domain adaptive filtering; multichannel dereverberation; state-space model;
fLanguage :
English
Publisher :
ieee
Conference_Titel :
Acoustics, Speech and Signal Processing (ICASSP), 2012 IEEE International Conference on
Conference_Location :
Kyoto
ISSN :
1520-6149
Print_ISBN :
978-1-4673-0045-2
Electronic_ISBN :
1520-6149
Type :
conf
DOI :
10.1109/ICASSP.2012.6287806
Filename :
6287806
Link To Document :
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