DSP - Filtering frequencies using DFT

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Published on 2010-05-20T16:58:45Z Indexed on 2010/05/21 11:50 UTC
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I'm trying to implement a DFT-based 8-band equalizer for the sole purpose of learning. To prove that my DFT implementation works I fed an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good.

I'm using the so-called 'standard way of calculating the DFT' which is by correlation. This method calculates the real and imaginary parts both N/2 + 1 samples in length. To attenuate a frequency I'm just doing:

float atnFactor = 0.6;
Re[k] *= atnFactor;
Im[k] *= atnFactor;

where 'k' is an index in the range 0 to N/2, but what I get after resynthesis is a slighty distorted signal, especially at low frequencies.

The input signal sample rate is 44.1 khz and since I just want a 8-band equalizer I'm feeding the DFT 16 samples at a time so I have 8 frequency bins to play with.

Can someone show me what I'm doing wrong? I tried to find info on this subject on the internet but couldn't find any.

Thanks in advance.

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