I really should stop by here more often! Anyways, I've just finished work on the Revision 64K Intro Music and Code. Its basically a little bit of hackery which generates / copies / phases / boosts etc samples after P61 has unpacked the samples. It was all a bit of an experiment to be honest, but it worked and we've essentially got a 500k module sat inside our 64k intro! :)
Now that's all finished I've got a bit more of an interest for sound coding. I was wondering where do I start with regards to coding a low or high pass filter? Of course it would need to be in assembly. Bare in mind that I'm not too good at maths and can't read C++ too well.
To start from the simplest angle possible: A low pass filter simply means averaging samples around the current one, like blurring in graphics. You can for example use a box or gaussian function here. The more samples you average, the lower your cut-off frequency. A high pass filter is simply the signal minus the low pass filtered one.
This type of filter is known as FIR filter (finite impulse response). The art is now to calculate the exact filter coefficients, the weights of your blurring function. However, even the simplest approaches get the job done.
