摘要
在数字信号处理过程中,常常要处理一些无法预知的信号、噪声或时变信号,如果采用具有固定滤波系数的数字滤波器无法实现最优滤波。实现较好的滤波,必须设计自适应滤波器,以跟踪信号和噪声的变化。主要根据自适应滤波的结构及原理并对最小均方误差(LMS)算法进行了研究,同时在DSP的集成开发环境下利用C语言编程设计一个16阶LMS自适应滤波器,并在软件模拟器上实现了仿真。通过不断调整滤波器的自适应步长,对实验结果进行对比分析,最后给出了结论。
In the process of Digital Signal Processing, some random signal, noise or the time -varying signal must be dealt with. The adoption of digital filters with fixed coefficient can not realize the optimized filter. Therefore it is necessary to design an Adaptive Filter to track the changes of the signal and noise. This paper mainly introduces the structure and principle of Adaptive Filter and the research of Least Mean Squareal (LMS) algorithm, and presents the implementation of simulation in the software simulator of a sixteenth orders Adaptive Filter which is designed in C language in DSP' s integrated development environment. Through adjusting the adaptive step of the Filter constantly, and analyzing the experimental results comparatively, a conclusion is given in the end.
出处
《计算机仿真》
CSCD
北大核心
2009年第9期281-284,共4页
Computer Simulation
关键词
自适应滤波器
最小均方误差算法
数字信号处理
Adaptive filter
Least mean squareal algorithm
Digital signal processing(DSP)