期刊文献+

基于MATLAB与DSP的自适应滤波器设计与实现 被引量:4

Design and implementation of self-adapting filter based on MATLAB and DSP
下载PDF
导出
摘要 滤波是滤除信号中某些特定频率的波形的技术,在数字信号处理中,主要用于滤除噪声和干扰信号。由于噪声和干扰信号的不确定性,采用固定滤波系数的数字滤波器无法达到最佳的效果。自适应滤波器能够随着环境的改变而改动滤波器自身的参数和结构,从而能够随着噪声和干扰信号的不断变化修正滤波器的参数和结构,最终实现较理想的滤波。本文研究了最小均方差(LMS)算法,并结合自适应滤波器的结构和原理,设计出FIR结构自适应滤波器。最后给出MATLAB仿真结果,并利用DSP验证自适应滤波器的性能。 The filtering is a technology to filter the waveform of some certain frequencies. In digital signal processing, the filtering is mainly used to filter noise and interference signals. Because of the uncertainty of noise and interference signal, using the digital filter of fixed filter coefficients can't achieve the best effect. However, the self-adapting filter can change its parameters and structure with the change of the environments which can accommodate to the changing noise and interference signals so as to realize the ideal filter. After studying the Least Mean Square algorithm (here referred as LMS), as well as analyzing the structure and principle the of the self-adapting filter, the paper designed a self-adapting filter of FIR structure. In the end, it gave the MATLAB simulation results, and used DSP to test the performance of the new design.
出处 《微型机与应用》 2015年第21期16-20,共5页 Microcomputer & Its Applications
基金 福建省高等学校新世纪优秀人才支持计划项目(JA12181) 国家自然科学基金资助项目(51579114)
关键词 自适应滤波器 MATLAB LMS算法 FIR滤波器 DSP adaptive filter MATLAB LMS algorithm FIR filter DSP
  • 相关文献

参考文献6

二级参考文献24

  • 1马伟富,雷勇,滕欢.自适应滤波器(LMS)算法及其在DSP上的实现[J].四川大学学报(自然科学版),2004,41(z1):470-473. 被引量:2
  • 2张靓.通信原理教学实验的MATLAB仿真[J].中国科技信息,2005(6):160-160. 被引量:1
  • 3张丹红,游珍珍.DSP的多领域应用研究[J].计算机技术与发展,2006,16(3):206-207. 被引量:10
  • 4[2]Zayed Ramadan and Alexander Poularikas,A Variable Stepsize Adaptive Noise Canceller Using Signal to Noise Ratio as the Controlling Factor,IEEE,System Theory,Proceeding of the thirtysixth southeastern symposium on,2004,pp.456-461
  • 5[3]S.C.Douglas and T.H.-Y.Meng,Normalized data nonlinearities for LMS adaptation,IEEE Transactions On Signal Processing,vol.42,no.6,June1994,pp.1352-1365
  • 6[4]S.Ikeda and A.Sugiyama,An adaptive noise canceller with low signal distortion for speech codes,IEEE Trans.On Signal,Signal Processing,vol 47,no.3,March 1999,pp665-674.
  • 7Malcolm D M.Performance of the hierarchical LMS algorithm[J].IEEE Signal Processing Letters,2002,12(9):436-437.
  • 8Haykin S.Adaptive Filter Theory[M].[s.l.] :Publishing House of Electronics Industry,2002.
  • 9Han Jianguo.A Model-Free Method Based Kalman Filtering Process for Time-Interval-Variable Sequences with Application to Astronomic Surveying[J].Engineering and Electronics,2003,14(2):29-33.
  • 10王念旭.DSP基础与应用系统设计[M].北京:北京航空航天大学出版社,2002.50-51.

共引文献24

同被引文献25

引证文献4

二级引证文献5

相关作者

内容加载中请稍等...

相关机构

内容加载中请稍等...

相关主题

内容加载中请稍等...

浏览历史

内容加载中请稍等...
;
使用帮助 返回顶部