building spring reverb

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Enzo said:
And why not go to the Accutronics web site and read up on the drive requirements and the output signal levels.


Some thoughts.

Ever notice bass amps don;t have reverbs? It just makes mud. Guitar amps typically feed the reverb driver with a signal with the bottom end rolled off. You might consider that when making up your drive circuits.

Yeah, I usually have some sort of EQ or filtering with most of my time based effects. So I can just EQ the bus that is sending to the reverb unit and loop it back to monitor in real time and blend with other instruments. The idea is like a poor man's plate.

I guess where the biggest source of confusion with me is about the impedance. Bridging vs Matching and what you should do with a passive device. If I was to bridge impedance the numbers would be flip flopped and I would order the wrong thing? And it just seems a little confusing because with a device like this the impedance is increased with the frequency of the signal.
 
Key said:

And it just seems a little confusing because with a device like this the impedance is increased with the frequency of the signal.


that's because the transducers are inductive. i think that's the reason some use constant current drive on them. it's not much of a problem if you roll off the low frequencies. little considered fact: op amps also increase in output impedance with frequency, yet for most purposes it doesn't have a large enough effect to matter much.
 
But are full fidelity time based effects realistic? If I was guessing I would think that the source which excites reverberation and echoes is usuallu degraded a bit in fidelity by the time it hits the boundary in a real situation. How does standing air interact with a sound wave? Is the interaction what you would call linear?
 
i think the real nonlinearities of real room reverberation are more from reflections from objects that aren't flat than from air, although air motion relative to the source and destination, such as forced air from vents, etc... can introduce minute doppler shifts. transmission through certain objects can introduce refractions and phase shifts from objects of varying densities. there are a lot more variables than can be simulated easily. those nonlinearities usually don't turn up in the direct sound, but in the reflections. if you ever listen to a well done and properly set up surround system, you will notice that most of what you hear from the surround and rear speakers is garbled and distorted, yet when you stand near the center of the room, it sounds extremely realistic.

that's why makers of DSP effects actually go to the venues they are trying to simulate and ping the actual room. this gives a very real sounding profile to the DSP version of the room. then the DSP setup routine pings the user's actual room and makes the calculations and adjustments necessary to make receiver and speakers in the user's room fit the profile of the simulated venue. it's far beyond spring reverb, and it sounds a lot more real, and it captures the nonlinearities of the original room extremely well, but it takes a lot more gadgetry to do than a spring reverb, tape echo, BBD line, or simple digital delay (there are even DSP versions of classic spring reverbs, tape echo boxes, and bucket brigade devices).
 
My main field of experimentation has been surround sound and stereo to surround sound decoding for a while now. I don't exactly know what you mean by distorted or garbled rears. Most of the time when I get distortions in the rear speakers it's due to out of phase artifacts in the stereo source recording.

I agree that surround sound with reverbs gives a great illusion of reality. To my ears instruments sound real and 3-D. It's as much a clearer picture compared to stereo as stereo is to mono.

But one thing it has made me appreciate more than anything - a good mixing engineer knows how to use reverb well. I think this is the weakest link more than what types of reverbs are used. You can have the best convolution based reverb in the world and if you crank it up too much to compensate for the limits of stereo it will sound too heavy handed and fake. If you don't use enough it will sound dry like a rough mix. It's a tough balancing act that I think has been done well with all sorts of methods. Some people swear by the lexicons etc.. some swear by the plate and there are a lot of new proponents of the impulse method.

My system is pretty brutal and I just am finding I like the sounds in some of the old 60s recordings. I also am pretty shifty with genres and how I record and am just looking for new colors in the studio to mess with. Really I have more digital stuff than I could ever use so I am looking for something different maybe a little more dirty. Now that I have 192kHz 24bit I would like to experiment with making my own impulse responses of things I can make etc...
 
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