I've been working on the stable feedback between a loudspeaker and microphone for a little while....and I'm still working on it. I recommend first of all to try compression, as a means of reigning in your unstable feedback. Once the volume becomes loud enough, the roll-off will decrease the loop gain to the point where it is no longer unstable (won't blow up and clip persistently). It will however oscillate between low and high volumes this way.
To make your effects stable, you will need to do some analysis. For instance with convolution, you have to have the fourier components of the convolving signal everywhere less than 1. You are guaranteed stability if the integral of the square of the signal is less than 1. For instance with reverb effects, you can truncate your room reverb signal at the point where the integral is less than 1 (or adjust the gain by the integral of the signal)
The method I'm working on is for subband adaptive filtering to equalize frequency response. It's not going so well....I've been trying a frequency domain algorithm for a little while now...and I've tried a lot of different things to try to get it to converge, like averaging and selection, with no good results yet. Since I've already got it to record 64 consecutive measurements, I will try to program an independent components analysis next.
Chuck