Karlos wrote:
Right, lets see
My basic assumptions were as follows
1) We analyse a window of sound to get the frequency spectrum for that window.
2) We identify the lowest and highest frequencies for which the intensity (contribution to the overall signal strength) is greater than a threshold set by the user.
3) We use those frequencies as the limits to build an FIR filter and process the original window samples to eliminate as much of those frequencies above and below as possible.
By doing this for each window in the sample we get an adaptive filtering for the sound as it changes over time.
Still not sure what effect you're tring to achive!!!
