Overlap and windowing a signal for signal processing

Question

I’d like to digitally signal process an audio data. Therefore my idea was to:

• Split the audio data into individual blocks of 16 samples each.
• Each block has an overlap of 50% (8 samples are shared across each consecutive block).
• Each block of data is multiplied with a window (hanning),
• Do the FFT,
• apply an Equalizer to the FFT data (Which I will implement later),
• iFFT the data and recover the original signal back.

Due to the Equalizer, the output might differ a little from the input, but shouldn’t be distorted .

To this end, I implemented a MATLAB code,

% I am using an audio file sampled at 16kHz 
[Input_sig_16k,Fs] = audioread(‘filename);

window_size = 16;        % individual block size 
overlap_win = 8;         % 50% overlap
data_full = zeros(1,length(Input_sig));
w = hanning(16);         % hanning windowing   
for ii = 1:(length(Input_sig)/win_size)-1

  data_range = ((ii-1)*overlap_win)+1:((ii-1)*overlap_win)+16;
%     [data_range(1) data_range(end)]
  data_block = Input_sig_16k(data_range);

  % windowing function
  data_block = data_block.*w;
%     plot(data_block,'-rx');
%     ii 

 fft_sig = fft(data_block);                 %FFT of the signal

 % I will multiply the frequency data with an Equalizer

 ifft_sig = ifft(fft_sig);                  %iFFT of the signal

 data_full(((ii-1)*window_size)+1:((ii-1)*window_size)+16) = ifft_sig;

 end

subplot(2,1,1), plot(Input_sig_16k,'-og'), grid on;   % plot and compare both the signals
subplot(2,1,2), plot(data_full,'-xr'), grid on;

However the output data is quite distorted. I’m not sure what I need to do to recover the original signal. Could someone let me know the mistake behind what I have done.

So how do I obtain the correct output?

Answer 1

Don't overlap. Don't use a Von Hann window. Zero-pad by at least the length of the impulse response of your desired equalization filter. Then use an overlap add or overlap save fast convolution method or algorithm to combine your zero-padded and filtered FFT/IFFT blocks.

Otherwise circular convolution artifacts will corrupt (severely distort) your attempts at filtering your signal.

If you want to analyze or visualize the audio, use separate overlapped and windowed FFT blocks, not the ones used for overlap add/save fast convolution filtering.