VoIP中语音压缩编解码器的研究与实现

Research and Implementation on Low Rate Speech Codec in VoIP

在当前网络状况下,高效的语音压缩编解码器可以节省网络传输带宽,解决网络拥挤问题。本文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算法提出了一种运算量较小但有效的话音激活检测算法,进一步将平均比特率降低到约4kb/s。经过实验分析,以上设计和实现能够很好地满足VoIP中全双工实时语音通信的要求。

An efficient low rate Speech Codec can get well utilized to solve the problem of network congestion under the circumstance of current network. The Speech Codec accomplished in this thesis is implemented on the Digital Signal Processor with low bit rate （8kb/s） and good timbre. And the Voice Activity Detection （VAD） algorithm is designed on the basis of it. That will lower down the average bit rate to about 4kb/s. As a result, the demand of real time full-duolex communication in VolP will be fulfilled.