基于DSP的VoIP语音压缩编解码器的研究与实现

Research and Implement on Low Rate Speech Codec Based on DSP in VoIP

在当前网络状况下，高效的语音压缩编解码器可节省网络传输带宽，解决网络拥挤问题。通过对语音特性的分析，结合共扼结构代数码本激励线性预测（coniugate structure algebraic codeexcited linear prediction，CS—ACELP）算法，提出了一种运算量较小但行之有效的话音激活检测算法，设计出了一种基于TMS320VC5409数字信号处理器的语音压缩编解码器。实验结果表明，利用该语音压缩编解码器可将平均比特率降低到约4kb／s，能很好地满足VoIP中全双工实时语音通信的要求，得到了较好的实际效果。

An efficient low rate speech eodee can get well utilized to solve the problem of network congestion under the circumstance of current network. To the speech characteristic analysed, the voice activity detection （VAD） algorithm that is more few operational quantity and very effective is designed based on conjugate structure algebraic code excited linear predietion（CS-ACELP）, the speech eodee in this paper is implemented on the digital signal proeessor（TMS320VC5409） with low bit rate and good timbre. That will lower down the average bit rate to about 4kb/s. As a result, the demand of real time full-duplex communication in VolP is fulfilled, and the praetical effect is received better.