• DocumentCode
    1246612
  • Title

    Adaptive cepstral analysis of speech

  • Author

    Tokuda, Keiichi ; Kobayashi, Takao ; Imai, Satoshi

  • Author_Institution
    Dept. of Electr. & Electron. Eng., Tokyo Inst. of Technol., Japan
  • Volume
    3
  • Issue
    6
  • fYear
    1995
  • fDate
    11/1/1995 12:00:00 AM
  • Firstpage
    481
  • Lastpage
    489
  • Abstract
    This paper proposes an algorithm for adaptive cepstral analysis based on the UELS (unbiased estimation of log spectrum). In the UELS, the model spectrum is represented by cepstral coefficients and the mean square of the inverse filter output is minimized with respect to the cepstral coefficients. By introducing an instantaneous gradient estimate of the criterion in a similar manner of the LMS algorithm, we develop an adaptive cepstral analysis algorithm. In the analysis system, an IIR adaptive filter whose coefficients are given by cepstral coefficients is realized using the log magnitude approximation (LMA) filter. The filter approximates an exponential transfer function and its stability is guaranteed for approximation of speech spectra. To implement the M th 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 LMS algorithm. Several examples of the adaptive cepstral analysis for synthetic signal and natural speech are shown to demonstrate the effectiveness of the algorithm
  • Keywords
    IIR filters; adaptive filters; adaptive signal processing; cepstral analysis; speech processing; transfer functions; IIR adaptive filter; UELS; adaptive cepstral analysis; algorithm; exponential transfer function; fast convergence properties; instantaneous gradient estimate; inverse filter output; log magnitude approximation filter; natural speech; speech analysis; speech spectra; stability; synthetic signal; unbiased estimation of log spectrum; Adaptive filters; Cepstral analysis; Convergence; IIR filters; Least squares approximation; Natural languages; Signal processing algorithms; Speech analysis; Stability; Transfer functions;
  • fLanguage
    English
  • Journal_Title
    Speech and Audio Processing, IEEE Transactions on
  • Publisher
    ieee
  • ISSN
    1063-6676
  • Type

    jour

  • DOI
    10.1109/89.482216
  • Filename
    482216