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
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