G.723.1语音压缩算法的分析及DSP实现  

The analysis of G.723.1 Speech Codec Algorithm and its Implement Based on DSP

作　　者：王伟[1] 李剑雄[1] 周广禄[1] 郭恒业[1] 

机构地区：[1]哈尔滨工业大学(威海),山东省威海市264209

出　　处：《微计算机信息》2008年第12期-,共3页Control & Automation

基　　金：国防科工委基础科研项目基金(编号不公开)

摘　　要：本文首先阐述了G.723.1语音编解码标准的流程。根据低功耗、高性能的16位定点的数字信号处理芯片TMS320DM642的特点对源代码中运算量较大的模块进行了代码优化。最后,在DSP硬件平台上对优化后的编解码器的复杂度和性能进行了测试。从实验结果看,经改进优化后的程序在内存占用和运算复杂度方面都达到了预期目标,语音信号经编码器编码解码之后失真很小。This article first specifies the process of G.723.1 standard. Then,it optimizes the larger operational quantity modules of the source code,according to the characteristics of low power loss and high-performance 16-bit fixed-point digital signal processing chip TMS320DM642. Finally,the performance is tested and evaluated and the optimized programs have met with the expectation in data stor-age and operational complexity. The distortion of the decoded speech is rather small.

关 键 词：G.723.1 语音编码 DSP 

分 类 号：TN912.3[电子电信—通信与信息系统]