VoIP G.723.1语音算法的优化与实现  

Optimization and Realization of G.723.1 algorithm in VoIP

作　　者：廖日坤[1] 纪越峰[1] 李慧[1] 

机构地区：[1]北京邮电大学电信工程学院

出　　处：《西安外事学院学报》2007年第3期101-104,共4页

基　　金：国家自然科学基金(No60572021);国家杰出青年科学基金(No60325104)

摘　　要：提出一种VoIP G.723.1双速率语音算法汇编优化及其DSP实现方法。通过阐述G.723.1信源编码的实现过程,讨论纯C代码在DSP平台上运行速度慢的原因。对关键代码全汇编优化以及优化过程的同步调试,采用该方法的优化结果和预期优化结果基本符合。FIFO环形缓冲区可以实现实时语音播放和IP-Phone,PESQ测试结果表明系统满足VoIP语音的质量要求。An assembly optimized method of VoIP-G.723.1 Dual Rate Speech algorithm and DSP implementation was proposed.Through describing the realization of G.723.1 codec,the reason why causes the pure C code running slowly was discussed.With the method of complete assembly optimization for the key code and synchronous debugging in optimized process,the optimization result basically coincides with that predicted by this method.Circular FIFO buffer can realize of real-time speech playing and IP-Phone on which satisfies the VoIP speech quality with the PESQ(Perceptual Evaluation of Speech Quality)results.

关 键 词：G.723.1 DSP 优化技术 语音编码 

分 类 号：TN912.3[电子电信—通信与信息系统]