DSP reverb, parallel functions help

Hi,
This is my first post on AAC's forums, been reading here for a while and find many of you here are of great knowledge.

I'm learning to work on a DSP, TMS3220C6713 if my memory is correct. (DSK6713). I implemented a reverb effect using a buffer method for the length of the effect and a g factor for the gain control. Really simple.
I use a 16kHz sampling freq., C language, and an interrupt method for the sampling. Now what I'd like to use is multiple parallel filter and be able to modify the frequence density (Df) and the echo density (Dt) so that sqrt(Df*Dt) = Number of parallel filtre needed. Each have a gain g1, g2, g3... and each have a buffer lenght... and the sum of all this should be my new output.

Can anyone point me in the right direction on how I can implement this?

 

In other words, I'd like to program something that looks like the figure 13 of this webpage: http://www.music.mcgill.ca/~gary/307/week3/reverb.html