clc;
close all;
fs = 44100;
% x = wavread('b.wav');
t = -5*pi:pi/100:5*pi;
x = sin(t);
x = x(:);
sx = size(x,1);
subplot(2,2,1);
plot(x);axis([0 sx -1 1]);
% 原信号FFT
xf = fft(x,1024);
subplot(2,2,3);
plot(abs(xf));
% 添加高斯噪声
t = 0 : 1/fs : (sx-1)/fs;
noise = 0.2*randn(size(x)); % 均值为0,方差为0.5的标准正态噪声
x1 = x + noise;
subplot(2,2,2);
plot(x1);axis([0 sx -1 1]);
% 信号加噪声后的FFT
xf = fft(x1,1024);
subplot(2,2,4);
plot(abs(xf));
% LMS自适应滤波
param.M = 50;
param.w = ones(param.M, 1) * 0.1;
param.u = 0.1;
param.max_iter = 100;
param.min_err = 0.5;
[yn,err] = zx_lms(x1(:,1), x(:,1), param);
figure,
plot(yn)
ynf = fft(yn(param.M:end), 1024);
figure,
plot(abs(ynf));
function [yn ,err] = zx_lms(xn, dn, param)
% x 输入信号
% dn 期望输出
% param Structure for using LMS, must include at least
% .w - 初始化权值
% .u - 学习率
% .M - 滤波器阶数
% .max_iter - 最大迭代次数
% .min_err - 迭代最小误差
%
% y 经过滤波器后的输出信号
% error 误差输出
W = param.w; % 初始权值
M = param.M; % 滤波器阶数
if length(W) ~= M
error('param.w的长度必须与滤波器阶数相同.\n');
end
if param.max_iter > length(xn) || param.max_iter < M
error('迭代次数太大或太小,M<=max_iter<=length(xn)\n');
end
iter = 0;
for k = M:param.max_iter
x = xn(k:-1:k-M+1); % 滤波器M个抽头的输入
y = W.*x;
err = dn(k) - y;
% 更新滤波器权值系数
W = W + 2*param.u*x;
iter = iter + 1;
if (abs(err) < param.min_err); break; end
end
% 求最优时滤波器的输出序列
yn = inf * ones(size(xn));
for k = M:length(xn)
x = xn(k:-1:k-M+1);
yn(k) = W(:,end).'* x;
end
end
去除高斯噪声滤波器
最新推荐文章于 2022-09-15 09:42:16 发布