• DocumentCode
    2174653
  • Title

    Gammatone sub-band magnitude-domain dereverberation for ASR

  • Author

    Kumar, Kshitiz ; Singh, Rita ; Raj, Bhiksha ; Stern, Richard

  • Author_Institution
    Carnegie Mellon Univ., Pittsburgh, PA, USA
  • fYear
    2011
  • fDate
    22-27 May 2011
  • Firstpage
    4604
  • Lastpage
    4607
  • Abstract
    We present an algorithm for dereverberation of speech signals for automatic speech recognition (ASR) applications. Often ASR systems are presented with speech that has been recorded in environments that include noise and reverberation. The performance of ASR systems degrades with increasing levels of noise and reverberation. While many algorithms have been proposed for robust ASR in noisy environments, reverberation is still a challenging problem. In this paper, we present ´ an approach for dereverberation that models reverberation as a convolution operation in the speech spectral domain. Using a least-squares error criterion we decompose reverberated spectra into clean spectra convolved with a filter. We incorporate non-negativity and sparsity of the speech spectra as constraints within a non-negative matrix factorization (NMF) frame work to achieve the decomposition. In ASR experiments where the system is trained with unreverberated and reverberated speech, we show that the proposed approach can provide upto 40% and 19% relative reduction respectively in performance.
  • Keywords
    error analysis; least squares approximations; matrix decomposition; speech recognition; ASR systems; NMF framework; automatic speech recognition systems; gammatone subband magnitude-domain dereverberation; least-squares error criterion; nonnegative matrix factorization framework; speech signals; speech spectra; speech spectral domain; Frequency domain analysis; Noise; Reverberation; Robustness; Spectral analysis; Speech; Speech recognition; Dereverberation; NMF; Spectral decomposition; Spectral modeling; Speech recognition;
  • fLanguage
    English
  • Publisher
    ieee
  • Conference_Titel
    Acoustics, Speech and Signal Processing (ICASSP), 2011 IEEE International Conference on
  • Conference_Location
    Prague
  • ISSN
    1520-6149
  • Print_ISBN
    978-1-4577-0538-0
  • Electronic_ISBN
    1520-6149
  • Type

    conf

  • DOI
    10.1109/ICASSP.2011.5947380
  • Filename
    5947380