Electronic – Mathematically derived waveform vs. captured waveform from an analog synthesizer

analogaudiofouriersumming

I was thinking about how the individual parameters on an analog synthesizer could be controlled digitally. What I mean is, instead of using mechanical potentiometers to adjust some quality of the waveform (for example, the frequency of an LFO), use digital potentiometers. This would allow numerical control of waveform synthesis, similar to how CNC machining produces metal parts.

If the analog output signal was then sampled, couldn't this data be analyzed in comparison to the virtual waveform used by the digital control process? And by this I mean one could see how close the analog signal comes to its intended waveform. There inevitably will be some deviation between the virtual and actual waveforms, in the same way that tolerances describe the deviation of machined parts from their intended dimensions.

Could the analysis then be used to subtract one waveform from the other (within some specified tolerance – if a measured value is close enough to the idealized value, that evaluates as being part of the signal), so that any remaining signal (that didn't cancel out during subtraction) would be the unavoidable noise produced by analog circuits?

If so, could this calculated noise floor then be applied to the original to reduce audio artifacts and so enhance the original analog signal? I assume this would be impossible to do in real time (due to propagation delays), but possible if done in post-production. The intent of which would be to blend the qualities of analog that continue to be in demand with the qualities of digital that make it so popular.

So to summarize, use digital technology to tightly control the operation of analog circuits in a musical synthesizer to produce an output signal that matches very precisely the digital waveform modeled from digital simulations, so that a process could be developed (similar to what's used in noise cancelling headphones) which would produce an enhanced (cleaned up) hybrid of analog and digital?

I'm sure that my electro-fantasy falls flat on its face somewhere, but I'm curious to know where, why, and how badly!


I used the synthesizer as an example, because I didn't know what the term for something was, or to describe something so I could demonstrate (and test) my comprehension. That's why the descriptions are rather long winded, but I really wanted to lay everything out to check accuracy.

The adding & subtracting of signals and the whole process is what I was ultimately interested in. And I'm happy I stumbled upon things I was interested in, but didn't know existed: the "distortion analyzer" and how the difference signal [is] (called the "residual").

I wasn't trying to be misleading with the question. It's just that I didn't know what I was describing was already a thing that existed. Sorry to have created a lot of responses on the subject of the double blind test, and how to go about developing the synthesizer I described. My main interest was about restoring or enhancing old recordings. I've always wondered about such a possibility and how much improvement could be made.

So I'm super happy to have gotten the answer that I accepted. Lucky is getting just what I was after without knowing what to ask for :+D

Best Answer

If the analog output signal was then sampled, couldn't this data be analyzed in comparison to the virtual waveform used by the digital control process? And by this I mean one could see how close the analog signal comes to its intended waveform. There inevitably will be some deviation between the virtual and actual waveforms, in the same way that tolerances describe the deviation of machined parts from their intended dimensions.

Yes, such an instrument has existed for decades (it even predates digital audio) — it's called a "distortion analyzer". Most will provide a real-time output of the difference signal (called the "residual") so that you can analyze further. It consists of a combination of noise and distortion of the original signal.

The problem is to identify which parts of the residual you want to keep and which parts you don't want. This is further complicated by the fact that the nature of the residual will most likely change as the signal changes in level, frequency, etc.

All of the nonlinearities, noise, hum, and other effects associated with analog circuits can and have been replicated with digital processing. You can get plug-ins for all of the popular computer-based audio processing software packages to do exactly that. The output is indistinguishable from analog signals in any proper double-blind test.

You should probably spend time with such software to see whether you can come up with a precise technical definition of what you mean by the "desirable audio qualities of analog signals". If you can, then we can talk about techniques to create them.