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I haven't really thought about that so much. But perhaps implementing an algorithm in C, then playing noise through it (input in a raw audio file format, same with output), then doing a spectrum or waterfall plot of that file - would that be a good approach?
And if there's a peak, can one put a 'snooper' to find the offending path? For example, write the 'output' audio files, but also a few files at selected points? I suppose once the algorithm is written and tested in C, it can be ported into a DSP box either by direct compiling, or hand compiling into assembler.
I just finished up a reverb algorithm (on the AL3201) that is just somewhere around 45 or so allpasses in series. It has a bit of a ring, and a very slow buildup. So I took taps from the first 20 allpasses using sloped tap coefficients with pseudo-random signs, and sum that to the output. That improved the buildup - without it the buildup is rather unnatural. The only thing is that it dies out unnaturally, so some feedback should help, but I've been afraid to try that due to ringing. It sounds kind of like a gated and inverse reverb combined, actually.
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