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 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 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
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