I am writing software for processing audio given as 16-bit PCM samples. The first stage of the processing involves calculating the energy (or total variation) in a certain frequency range (above a certain cut-off frequency).
What I am currently doing is subtracting the energy of the low-pass filtered signal from the energy of the original signal. I found out that a lot of the processing is dedicated to converting the integer sampels to a floating point representation.
So my question is, is there a technnique for filtering the integer samples without converting them to floating point?