IP电话系统中语音芯片的DSP实现  

Realization of a Speech Coder Being Used in IP Telephone System on DSP

作　　者：刘丹[1] 王芙蓉[1] 黄本雄[1] 

机构地区：[1]华中科技大学电信系,武汉430074

出　　处：《计算机工程》2002年第8期206-208,共3页Computer Engineering

摘　　要：介绍了一种具有语音编解码、自适应回声抵消和编解码功能的语音芯片,以满足电话系统中低速率语音通信的需G.723.1DTMFIP要。根据内核的特点,对算法进行了优化,降低了算法复杂度;回声抵消部分使用归一化最小二乘自适应滤波器实现;DSPG.723.1(NL MS)采用算法实现的检测。使用较少的资源,在一块芯片上集成了以上全部功能。A speech CODEC chip is introduced in this article,which has the facilities of G.723.1 speech CODEC, adaptive echo canceling and DTMF CODEC to meet the requirement of low bit rate communication in IP telephone systems. The G.723.1 algorithm is optimized to reduce the computational complexity according to the architecture of the DSP core; the echo cancellation is realized with the normalized least mean square(NLMS) adaptive filter; uses Goertzel algorithm to achieve DTMF detecting.All the modules are integrated in one TMS320C5402 DSP chip using less memory sources. ;;;

关 键 词：IP电话系统 语音芯片 DSP INTERNET 

分 类 号：TN916.1[电子电信—通信与信息系统] TP393.4[电子电信—信息与通信工程]