• DocumentCode
    2885201
  • Title

    Adaptive filtering based on cepstral representation-adaptive cepstral analysis of speech

  • Author

    Tokuda, Keiichi ; Kobayashi, Takehiko ; Shiomoto, Shoji ; Imai, Suguru

  • Author_Institution
    Dept. of Electr. & Electron. Eng., Tokyo Inst. of Technol., Japan
  • fYear
    1990
  • fDate
    3-6 Apr 1990
  • Firstpage
    377
  • Abstract
    An adaptive cepstral analysis method based on an unbiased estimation of the log spectrum is proposed. In the method, an infinite impulse response adaptive filter whose coefficients are given by cepstral coefficients is realized using the log magnitude approximation (LMA) filter. To implement the Mth-order cepstral analysis, the algorithm requires O(M) operations per sample. It is shown that the algorithm has fast convergence properties in comparison with the least-mean-square algorithm. A real-time analysis system is implemented with a general-purpose digital signal processor, and an example of natural speech analysis is shown to demonstrate the convergence
  • Keywords
    adaptive filters; convergence; digital filters; filtering and prediction theory; speech analysis and processing; transfer functions; IIR adaptive filter; adaptive cepstral analysis; cepstral representation; convergence; digital signal processor; exponential transfer function; infinite impulse response adaptive filter; log magnitude approximation filter; log spectrum; natural speech analysis; operations per sample; real-time analysis; unbiased estimation; Adaptive filters; Cepstral analysis; Convergence; Digital signal processors; IIR filters; Natural languages; Real time systems; Signal analysis; Signal processing algorithms; Speech analysis;
  • fLanguage
    English
  • Publisher
    ieee
  • Conference_Titel
    Acoustics, Speech, and Signal Processing, 1990. ICASSP-90., 1990 International Conference on
  • Conference_Location
    Albuquerque, NM
  • ISSN
    1520-6149
  • Type

    conf

  • DOI
    10.1109/ICASSP.1990.115697
  • Filename
    115697