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《Journal of Data Acquisition & Processing》 2005-01
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Wideband Speech Coding Algorithm Based on Adaptive Interpolation of Weighted Spectrum

LING Zhen-hua, DAI Li-rong, WANG Ren-hua, SHUANG Zhi-wei, ZHOU Bin (Department of Electronic Engineering and Information Science, University of Science and Technology of China, Hefei, 230027,China)  
Based on speech transformation and representation using adaptive interpolation of weighted spectrum (STRAIGHT), a wideband speech coding algorithm is presented. The input speech signals are firstly decomposed into pitch parameters and spectral parameters by STRAIGHT,and then compressed effectively by sampling in temporal domain and modeling in frequency domain. Because of the introduction of adaptive sampling with variable frame lengths, the bitrates can be more reasonably allocated accoding to the actural movement of speech signals. Subjective listening test demonstrates that the decoded quality of proposed algorithm at 6 kbps for 16 kHz sampled speech signal corresponds to that of AMR-WB(G.722.2) at 8.85 kbps. Besides, the method has flexible modification ability on duration, pitch and spectrum of decoded speech. So it can be widely applied in the fields, such as speech synthesis with parametric modification, voice conversion and so on.
【CateGory Index】: TN912.3
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Chinese Journal Full-text Database 10 Hits
1 XIA Juan, SHEN Min (Department of Communication Engineering, Chongqing University of Posts and Telecommunications,Chongqing 400065,P.R.China);TD-SCDMA based adaptive multi-rate codec[J];Journal of Chongqing University of Posts and Telecommunications;2004-01
2 DAI Qin-yun, HU Han-ying, TAN Wei (Information Engineering University of PLA, Zhengzhou 450002,China);Performance Characterization of AMR Speech Codec in 3G Partnership Project[J];Telecommunication Engineering;2002-02
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5 LI Zhi-hong1, Zhang Xue-ying2, Wang An-hong1, QI Xiang-dong1 ( 1. College of Electronics and Information, Taiyuan University of Science and Technology, Taiyuan 030024, China; 2. Information Engineering College of Taiyuan University of Technology, Taiyuan 030024, China );Nonlinear predictive speech coding system based on DWNN[J];Journal of Circuits and Systems;2005-05
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7 DAI Qin-yun,HU Han-ying;Mechanism of Source Controlled Rate[J];Audio Engineering;2002-06
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9 HUANG Hong-kai, LIU Jia-kang, XIAO Guo-jun, ZHANG Yu-bing(Department of Electronic Engineering, School of Information Science and Technology, Beijing Institute of Technology, Beijing 100081, China);Channel Coding Techniques for AMR and Its DSP Implementation[J];Audio Engineering;2005-02
10 XIE Jun1, YI Qing-ming1, ZHOU De-hua2 (1. Department of Electronic Engineering, Ji'nan University,Guangzhou 510632, China; 2. Department of Computer Science, Ji'nan University,Guangzhou 510632, China);Design and Realization of Voice Real-Time Transmission Based on Adaptive Multi-Rate Coding[J];Audio Engineering;2005-05
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