How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab [closed]
- by martin
This is all done in MatLab 2010
My objective is to show the results of: undersampling, nyquist rate/ oversampling
First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct.
Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter
what needs to be achieved:
F= Frequency
F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec)
F(s)= 1/(2f)
Example problem: 1000 hz = Highest
frequency 1/2(1000hz) = 1/2000 =
5x10(-3) sec/cyc or a sampling rate of
5ms
This is my first signal processing project using matlab.
what i have so far.
% 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)
Can anyone tell me how to make it better, and how to do the various sampling at different frequencies?
