title: Simple audio filter-bank
body: <p>I'm new to audio filters so please excuse me if i'm saying something wrong.</p>
  
  <p>I like to write a code which can split up audio stored in PCM samples into two or three frequency bands and do some manipulation (like modifying their audio levels) or analysis on them then reconstruct audio samples from the output.</p>
  
  <p>As far as i read on the internet for this task i could use FFT-IFFT and do manipulation on the complex form or use a time domain based filterbank which for example is used by the MP2 audio encoding format. Maybe a filter-bank is a better choice, at least i read somewhere it can be more CPU usage friendly in real time streaming environments. However i'm having hard times understanding the mathematical stuff behind a filterbank. I'm trying to find some source code (preferably in Java or C/C++) about this topic, so far with no success.</p>
  
  <p>Can somebody provide me tips or links which can get me closer to an example filter bank?</p>