• DocumentCode
    788399
  • Title

    Subband Likelihood-Maximizing Beamforming for Speech Recognition in Reverberant Environments

  • Author

    Seltzer, Michael L. ; Stern, Richard M.

  • Author_Institution
    Speech Technol. Group, Microsoft Res., Redmond, WA
  • Volume
    14
  • Issue
    6
  • fYear
    2006
  • Firstpage
    2109
  • Lastpage
    2121
  • Abstract
    In this paper, we introduce subband likelihood-maximizing beamforming (S-LIMABEAM), a new microphone-array processing algorithm specifically designed for speech recognition applications. The proposed algorithm is an extension of the previously developed LIMABEAM array processing algorithm. Unlike most array processing algorithms which operate according to some waveform-level objective function, the goal of LIMABEAM is to find the set of array parameters that maximizes the likelihood of the correct recognition hypothesis. Optimizing the array parameters in this manner results in significant improvements in recognition accuracy over conventional array processing methods when speech is corrupted by additive noise and moderate levels of reverberation. Despite the success of the LIMABEAM algorithm in such environments, little improvement was achieved in highly reverberant environments. In such situations where the noise is highly correlated to the speech signal and the number of filter parameters to estimate is large, subband processing has been used to improve the performance of LMS-type adaptive filtering algorithms. We use subband processing principles to design a novel array processing architecture in which select groups of subbands are processed jointly to maximize the likelihood of the resulting speech recognition features, as measured by the recognizer itself. By creating a subband filtering architecture that explicitly accounts for the manner in which recognition features are computed, we can effectively apply the LIMABEAM framework to highly reverberant environments. By doing so, we are able to achieve improvements in word error rate of over 20% compared to conventional methods in highly reverberant environments
  • Keywords
    adaptive filters; filtering theory; least mean squares methods; microphone arrays; reverberation; speech processing; speech recognition; LMS-type adaptive filtering algorithms; additive noise; microphone-array processing; reverberant environments; speech recognition; speech signal; subband likelihood-maximizing beamforming; subband processing principles; waveform-level objective function; Adaptive filters; Additive noise; Algorithm design and analysis; Array signal processing; Computer architecture; Optimization methods; Process design; Speech enhancement; Speech processing; Speech recognition; Adaptive beamforming; microphone array processing; speech recognition;
  • fLanguage
    English
  • Journal_Title
    Audio, Speech, and Language Processing, IEEE Transactions on
  • Publisher
    ieee
  • ISSN
    1558-7916
  • Type

    jour

  • DOI
    10.1109/TASL.2006.872614
  • Filename
    1709899