这一切都在 MATLAB 2010 中完成
我的目标是展示以下结果：欠采样、奈奎斯特率/过采样
首先，我需要对 .wav 文件进行下采样以获得不完整/或公正的数据流，然后我可以重建该数据流。
这是我要做什么的流程图所以流程是模拟信号->采样模拟滤波器->ADC->重采样->重采样->DAC->重建模拟滤波器
需要达到的目标：
F= 频率
F(Hz=1/s) Ex 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f)
示例问题：1000 hz = 最高频率 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc 或 5ms 的采样率
这是我第一个使用 matlab 进行信号处理的项目。
到目前为止我有什么。
% Fs = frequency sampled (44100hz or the sampling frequency of a cd)
[test,fs]=wavread('test.wav'); % loads the .wav file
left=test(:,1);
% Plot of the .wav signal time vs. strength
time=(1/44100)*length(left);
t=linspace(0,time,length(left));
plot(t,left)
xlabel('time (sec)');
ylabel('relative signal strength')
**%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.***
soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however)
谁能告诉我如何让它变得更好，以及如何以不同的频率进行采样？
这是 .wav 文件http://www.4shared.com/audio/11xvNmkd/piano.html
编辑：
%Play decimated file ( soundsc(y,fs) )
%Play Original file ( soundsc(play,fs ) )
%Play reconstucted File ( soundsc(final,fs) )
[piano,fs]=wavread('piano.wav'); % loads piano
play=piano(:,1); % Renames the file as "play"
t = linspace(0,time,length(play)); % Time vector
x = play;
y = decimate(x,25);
stem(x(1:30)), axis([0 30 -2 2]) % Original signal
title('Original Signal')
figure
stem(y(1:30)) % Decimated signal
title('Decimated Signal')
%changes the sampling rate
fs1 = fs/2;
fs2 = fs/3;
fs3 = fs/4;
fs4 = fs*2;
fs5 = fs*3;
fs6 = fs*4;
wavwrite(y,fs/25,'PianoDecimation');
%------------------------------------------------------------------
%Downsampled version of piano is now upsampled to the original
[PianoDecimation,fs]=wavread('PianoDecimation.wav'); % loads piano
play2=PianoDecimation(:,1); % Renames the file as "play
%upsampling
UpSampleRatio = 2; % 2*fs = nyquist rate sampling
play2Up=zeros(length(PianoDecimation)*UpSampleRatio, 1);
play2Up(1:UpSampleRatio:end) = play2; % fill in every N'th sample
%low pass filter
ResampFilt = firpm(44, [0 0.39625 0.60938 1], [1 1 0 0]);
fsUp = (fs*UpSampleRatio)*1;
wavwrite(play2Up,fsUp,'PianoUpsampled');
%Plot2
%data vs time plot
time=(1/44100)*length(play2);
t=linspace(0,time,length(play2));
stem(t,play2)
title('Upsampled graph of piano')
xlabel('time(sec)');
ylabel('relative signal strength')
[PianoUpsampled,fs]=wavread('PianoUpsampled.wav'); % loads piano
final=PianoUpsampled(:,1); % Renames the file as "play"
%-------------------------------------------------------------
%resampleing
[piano,fs]=wavread('piano.wav'); % loads piano
x=piano(:,1); % Renames the file as "play"
m = resample(x,3,2);