• DocumentCode
    336221
  • Title

    A new technique to filter reduction for speech signal processing systems

  • Author

    Li, Luowen ; Xie, Lihua ; Li, Gang ; Soh, Yeng Chai

  • Author_Institution
    Sch. of Electr. & Electron. Eng., Nanyang Technol. Univ., Singapore
  • Volume
    3
  • fYear
    1999
  • fDate
    15-19 Mar 1999
  • Firstpage
    1513
  • Abstract
    In many applications, one needs to approximate a filter of very high order with that of lower order. To reduce the order of the filter, some techniques such as the balanced model reduction approach are often applied. In this paper, we introduce a new technique which is based on minimizing the H2-norm between the filter of very high order and the reduced one. This technique shows much better performance than other existing model reduction methods and is applied to estimating the vocal tract filter for speech processing systems. A speech processing example is presented to demonstrate the design procedure and the performance of the proposed algorithm
  • Keywords
    digital filters; filtering theory; parameter estimation; poles and zeros; speech processing; H2-norm; algorithm performance; balanced model reduction; filter order reduction; ple-zero filter; speech signal processing systems; time-invariable linear digital filter; very high order filter; vocal tract filter estimation; Algorithm design and analysis; Digital filters; Finite impulse response filter; IIR filters; Reduced order systems; Signal processing; Signal processing algorithms; Speech analysis; Speech processing; Transfer functions;
  • fLanguage
    English
  • Publisher
    ieee
  • Conference_Titel
    Acoustics, Speech, and Signal Processing, 1999. Proceedings., 1999 IEEE International Conference on
  • Conference_Location
    Phoenix, AZ
  • ISSN
    1520-6149
  • Print_ISBN
    0-7803-5041-3
  • Type

    conf

  • DOI
    10.1109/ICASSP.1999.756271
  • Filename
    756271