DPSK传输系统MATLAB实验

附：DPSK.m
clear all
close all
i=20; %码元的个数
j=4000;
fc=5;
fm=i/10;
B=2*fm;
t=linspace(0,10,j); % 将0-10区间平均分为j份
%%%%%%%%%%%%% 产生基带信号 %%%%%%%%%%%%%%%%
a =round(rand(1,i)); %产生20个随机码，记为a
st1=t;
for n=1:20
if a(n)<1;
for m=j/i*(n-1)+1:j/i*n % j/i为每个码元的 st1(m)=0;
end
else
for m=j/i*(n-1)+1:j/i*n
st1(m)=1;
end
end
end
figure(1);
subplot(411);
plot(t,st1);
title('绝对码');
axis([0,10,-1,2]);
%%%%%%%%%%%%%%%%%差分变换%%%%%%%%%%%%%%%%%%%
b=zeros(1,i);
b(1)=a(1);
for n=2:20
if a(n)>=1;
if b(n-1)>=1
b(n)=0;
else
b(n)=1;
end
else
b(n)=b(n-1);
end
end
st1=t;
for n=1:20
if b(n)<1;
for m=j/i*(n-1)+1:j/i*n
st1(m)=0;
end
else
for m=j/i*(n-1)+1:j/i*n
st1(m)=1;
end
end
end
subplot(412);
plot(t,st1);
title('相对码');
axis([0,10,-1,2]);
st2=t;
for k=1:j;
if st1(k)>=1;
st2(k)=0;
else
st2(k)=1;
end
end;
subplot(413)
plot(t,st2)
title('相对码的反码')
axis([0 10 -1 2])
%%%%%%%%%%%% 载波信号 %%%%%%%%%%%%%%%%%
s1=sin(2*pi*fc*t);
subplot(414);
plot(s1);
title('载波信号'); %%%%%%%%%%%%%% 调制 %%%%%%%%%%%%%%%%%%%%%% d1=st1.*s1;
d2=st2.*(-s1);%相移180
figure(2);
subplot(4,1,1);
plot(t,d1);
title('st1*s1');
subplot(4,1,2);
plot(t,d2);
title('st2*s2');
e_dpsk=d1+d2;
subplot(4,1,3);
plot(t,e_dpsk);
title('调制后波形');
noise=rand(1,j);
dpsk=e_dpsk+0.5*noise;%加入噪声
subplot(4,1,4);
plot(t,dpsk);
title('加噪声信号');
%%%%%%%%%%%%% 与载波相乘 %%%%%%%%%%%%%%%%%%%%% dpsk=dpsk.*sin(2*pi*fc*t);%与载波相乘
figure
subplot(4,1,1);
plot(t,dpsk);
title('与载波相乘后波'); %%%%%%%%%%%%% 低通滤波 %%%%%%%%%%%%%%%%%%%%%%%% [f,af] = T2F(t,dpsk);%通过低通滤波器
[t,dpsk] = lpf(f,af,B);
subplot(4,1,2);
plot(t,dpsk);
title('通过低通滤波器波形'); %%%%%%%%%%%%%%% 抽样判决 %%%%%%%%%%%%%%%%%%%%
st=zeros(1,i);
for m=0:i-1;
if dpsk(1,m*200+100)<0;
st(m+1)=0;
for j=m*200+1:(m+1)*200;
dpsk(1,j)=0;
end
else
for j=m*200+1:(m+1)*200;
st(m+1)=1;
dpsk(1,j)=1;
end
end
end
subplot(4,1,3);
plot(t,dpsk);
axis([0,10,-1,2]);
title('抽样判决后波形') %%%%%%%%%%%%%% 码反变换 %%%%%%%%%%%%%%%%%%
dt=zeros(1,i);
dt(1)=st(1);
for n=2:20;
if (st(n)-st(n-1))<=0&&(st(n)-st(n-1))>-1;
dt(n)=0;
else
dt(n)=1;
end
end
st=t;
for n=1:20
if dt(n)<1;
for m=j/i*(n-1)+1:j/i*n
st(m)=0;
end
else
for m=j/i*(n-1)+1:j/i*n
st(m)=1;
end
end
end
subplot(4,1,4);
plot(t,st);
title('码反变换后波形');
axis([0,10,-1,2]);
子程序1：F2T.m
function [ t,st ] = F2T( f,yf )
%This function calculate the time signal using ifft function for the input %signal's spectrum
df = f(2) - f(1);
Fmx = ( f(end)-f(1)+df );
dt = 1/Fmx;
N = length(yf);
T = dt*N;
sff = fftshift(yf);
t = 0 : dt : T-dt;
st = Fmx * ifft(sff);
st = real(st);
end
子程序2:Ipf.m
function [t,st]=lpf(f,sf,B)
%This function filter an input data using a lowpass filter
%Inputs: f: frequency samples
% sf: input data spectrum samples
% B: lowpass's bandwidth with a rectangle lowpass
%Outputs: t: time samples
% st: output data's time samples
df = f(2)-f(1);
T = 1/df;
hf = zeros(1,length(f));
bf = [-floor( B/df ): floor( B/df )] + floor( length(f)/2 );
hf(bf)=1;
yf=hf.*sf;
[t,st]=F2T(f,yf);
st = real(st);
子程序3：T2F.m
function [f,sf]= T2F(t,st)
%This is a function using the FFT function to calculate a signal's Fourier %Translation
%Input is the time and the signal vectors,the length of time must greater %than 2
%Output is the frequency and the signal spectrum
dt = t(2)-t(1);
T=t(end);
df = 1/T;
N = length(st);
f=-N/2*df:df:N/2*df-df;
sf = fft(st);
sf = T/N*fftshift(sf);
仿真图：。