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Originally Posted by zmix  Well obviously the EMT250 does a lot of both #1 and #3, it's remained highly useful despite (but probably *because of*) it's rather low mode density. The Ursa Major Space station uses a form of #1 and something like #3 though it hasn't really got a similar structure to other reverberators. The 224 uses these methods as well, but it's quite easy remove all modulation in that machine and still have a very useful reverb. |
I would describe all 3 of these algorithms as relying on #1 only. [EDIT] I presume that the EMT250 has linear interpolated delay lines in its feedback loops, but I may be mistaken. [/EDIT] Method #3 implies changing the gains of feedback operators, presumably in a unitary fashion. A few examples:
- Parallel delay lines coupled by a feedback matrix, where the matrix is changing values in a unitary method. Blesser has this in his recent patent, but it looks like he removed the claims that cover this particular technique. I had worked with similar ideas around the same time as Blesser, but wasn't too happy with the results. It sounds like cycling through different fixed resonances, instead of smoothly varying the resonant frequencies.
- Changing the coefficients of allpasses embedded within a delay loop. This can result in some nasty sounds, as well as very obvious changes in reverb echo density. When I have tried it, it sounds either like water sploshing in a metal pan (for slow variation speeds), or like tortured souls screaming from the pit of Hell (for faster variation speeds). I used one of my failed experiments as a sound to scare trick or treaters one year.
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When I listen to the "Random Hall" algo (which seems to use #2 above) or the later Lex algos with the "Spin" and "Wander' parameters I hear a lack of 'distance' in the tail. The sound seems to float like a cloud rather than travel away from the listener. I think the Griesinger paper provides a compelling argument against this sort of modulation for realistic spaces.
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It seems like you prefer the sidebands to build up over time, instead of staying at a fixed level. Method #2, or any other feedforward method (which could be any of the 3 methods, if you change #3 to include feedforward matrices) will probably be better for preserving the pitch of sustained signals. Having said that, I love the sound of time variation in the feedback loops.
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How would you characterize the differences in depth that you have observed when experimenting with these three methods?
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I need to revisit some of the other methods. Most of my experimentation with #3 happened in the early 2000's, when I wasn't as good at setting the delay lengths of a reverb. I have barely tried #2. Method #1 has so many variations, that I have spent much of my time exploring different variations - and I still feel like I have barely scratched the surface.