多路语音传输前软件压缩处理技术  

Software Compression Technology of Multiplexing Audio before Transport

作　　者：韩希明[1] 李治柱[1] 

机构地区：[1]上海交通大学计算机系,上海200030

出　　处：《计算机仿真》2005年第3期151-152,174,共3页Computer Simulation

摘　　要：语音传输技术(VOIP)广泛地应用于IP电话等领域,它将语音通讯与数据通讯广泛地融合起来,促进了Internet应用技术的发展。在VOIP技术应用里,语音数据发送入网路之前,通过静音检测、语音压缩等技术,可有效减少数据发送量,降低网络占用率,提高传输效率。应用静音检测和MPEG语音编码压缩技术,采用RTP实时传输协议,开发了一个多路语音传输系统,能够处理并行多路实时语音传输。系统具有较低的CPU占用率,为小规模多路并行实时语音传输的实现提供借鉴。Technology of Voice over Internet Protocol (VOIP) is widely used in fields like IP phone. It combines audio communication and data communication, promotes the development of Internet application. In VOIP Tech, before sound data is put into net, it is processed by silence detection and audio compression. This can help to reduce data amount, down net use, and improve transport efficiency. With silence detection, MPEG1 LAYER3 and RTP protocol, the author has developed a multiplexing voice transport system, which can deal with multiplexing real time voice transport. The system is low CPU rate. It makes an example for small scale multiplexing real time voice transport.

关 键 词：静音检测 语音压缩 实时传输协议 

分 类 号：TP391.9[自动化与计算机技术—计算机应用技术]