DocumentCode :
2311042
Title :
An efficient adaptive LSP method for speech encoding
Author :
Ching, P.C. ; Ho, K.C.
Author_Institution :
Dept. of Electron. Eng., Chinese Univ. of Hong Kong, Shatin, Hong Kong
fYear :
1990
fDate :
24-27 Sep 1990
Firstpage :
324
Abstract :
The authors describe an adaptive split-path filter structure for calculating line spectral pair (LSP) coefficients by using the least-mean-squares (LMS) algorithm. The two split filters are constructed as a cascade of second-order sections and the filter parameters are equivalent to the LSP coefficients. Adaptation is based on minimizing the output errors of the two filters independently, which is computationally simpler and has a much faster convergence rate. A subband predictive coder that divides speech into three subbands, each encoded with fourth-order LSP coefficients representing each of the three predictive filters, has been simulated on an IBM PC. Preliminary results indicate that output speech produced at 12 to 16 kb/s is highly intelligible and has a quality comparable to the 16 kb/s continuously variable slope delta (CVSD) method
Keywords :
encoding; filtering and prediction theory; speech analysis and processing; 12 to 16 kbit/s; LMS algorithm; LSP coefficients; adaptive split-path filter structure; convergence rate; least-mean-squares; line spectral pair; output errors; predictive filters; speech encoding; speech intelligibility; speech quality; subband predictive coder; Adaptive filters; Bit rate; Convergence; Encoding; Signal synthesis; Speech analysis; Speech coding; Speech enhancement; Speech synthesis; Transversal filters;
fLanguage :
English
Publisher :
ieee
Conference_Titel :
Computer and Communication Systems, 1990. IEEE TENCON'90., 1990 IEEE Region 10 Conference on
Print_ISBN :
0-87942-556-3
Type :
conf
DOI :
10.1109/TENCON.1990.152625
Filename :
152625
Link To Document :
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