语音编码算法在局域网通信系统中的优化实现  

Optimal implementation of speech coding algorithm in LAN communication system

作　　者：吴游[1] 冯金金[1] 

机构地区：[1]江苏科技大学电子信息学院,江苏镇江212003

出　　处：《现代电子技术》2016年第9期70-74,共5页Modern Electronics Technique

摘　　要：针对多媒体通信技术在网络带宽有限和传输音频数据量大的情况下,带宽利用率低,传输效率低的问题,在局域网通信系统中选择并优化实现了G.723.1编解码算法。在VC++6.0平台上,采用TCP/IP协议,在两台计算机之间分别采用客户端/服务器模式完成了系统设计。实验结果表明,优化后的算法提高了性能,有较高的压缩比,编码后音频数据占用存储空间小,可以节省传输带宽,提高传输效率。Since the bandwidth utilization ratio and transmission efficiency are low when the multimedia communication technology is used in the situation of the limited network bandwidth and mass audio data transmission,the G.723.1 coding and decoding algorithm was selected and optimally implemented in the local area network（LAN）communication system. The client/server modes are respectively adopted between two computers to accomplish the system design on VC＋＋ 6.0 platform by using TCP/IP protocol. The experimental results show that the performance of the optimized algorithm is improved,its compression ratio is high,and the coded audio data occupies less storage space,which can save the transmission bandwidth and improve the transmission efficiency.

关 键 词：局域网通信系统 语音编解码 G.723.1 TCP/IP协议 

分 类 号：TN912.32[电子电信—通信与信息系统] TP391[电子电信—信息与通信工程]