function part2(wavefile,newfile,percent)

[y,fs,nbits]=wavread(wavefile); % Reads in the stereo wavfile
ym=y(:,1); % Takes channel 1 audio
N=length(ym)+1; % Number of samples (assumed odd).
ymf=fft(ym,N); % We take a fast Fourier Transform of the song
clear ym;
zmf=ymf; % Make a copy of the FFT

%Here we zero out anything that is less than percent*(max in zmf)
idx = find(zmf<percent*max(zmf));
zmf(idx)=0;

%Plot the results
figure(1)
plot(abs(ymf),'b'); % Spectrum of the original signal
hold on;
plot(abs(zmf),'r'); % Spectrum of signal after setting a portion to zero
hold off;
title('red: retained, blue: filtered out');
szmf=sparse(zmf); % The sparsity step removes memory allocation for zeros
whos szmf zmf; % Gives you memory usage times 2 since fft is symmetric
clear szmf;

%Calculate the Error
errorf=(1/N)*sqrt(sum((ymf(1:N)-zmf(1:N)).^2));
error=real(ifft(errorf))


zm=real(ifft(zmf)); % Inverse FFT
wavwrite(zm,fs,nbits,newfile); % Compressed song is written