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
Link To Document :
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