• 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