Question

这一切都在MATLAB 2010中完成

我的目标是显示以下结果：欠采样，奈奎斯特率/过采样

首先，我需要对.wav文件进行缩减采样，以获得不完整/或公正的数据流，然后我可以重新构建。

下面是我要做的事情的流程图所以流程是模拟信号 - ＆gt;采样模拟滤波器 - ＆gt; ADC - ＆gt;重新取样 - ＆gt;重新取样 - ＆gt; DAC - ＆gt;重建模拟滤波器

需要实现的目标：

F =频率

F（Hz = 1 / s）E.x。 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或采样率   5ms的

这是我第一个使用matlab的信号处理项目。

到目前为止。

% 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)

有谁能告诉我如何让它变得更好，以及如何以奇怪的频率进行采样？

继承人.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);

原件： http://www.4shared.com/audio/11xvNmkd/piano.html

新： http://www.4shared.com/audio/nTRBNSld/PianoUs.html

Answer 1

最简单的方法是通过整数因子更改采样率。 下采样包括通过低通滤波器运行数据，然后丢弃采样，而上采样包括插入采样然后通过低通滤波器运行数据（也称为重建滤波器或内插滤波器）。当跳过过滤步骤或执行不当时会发生别名。因此，为了显示别名的效果，我建议您根据需要简单地丢弃或插入样本，然后以新的采样率创建一个新的WAV文件。要丢弃样本，您可以执行以下操作：

DownSampleRatio = 2;
%# Normally apply a low pass filter here
leftDown = left(1:DownSampleRatio:end); %# extract every N'th sample
fsDown = fs/DownSampleRatio;
wavwrite(leftDown, fsDown, filename);

要创建样本，您可以这样做：

UpSampleRatio = 2;
leftUp = zeros(length(left)*UpSampleRatio, 1);
leftUp(1:UpSampleRatio:end) = left; %# fill in every N'th sample
%# Normally apply a low pass filter here
fsUp = fs*UpSampleRatio;
wavwrite(leftUp, fsUp, filename);

您可以回放写入的WAV文件来听取效果。

顺便说一下，您要求改进代码 - 我更倾向于将t向量初始化为t = (0:(length(left)-1))/fs;。

Answer 2

您需要的DSP技术称为decimation。

我将如何对.wav文件进行下采样，然后使用nyquist重建它？ - 在MATLAB中

2 个答案: