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Extending the soundscape of a hexaphonic guitar pickup
sorveltaja:
Today i've been mostly testing the vocoder, as a separate blocks, and as a whole. Everything should work, but there is no actual vocoder effect. I don't expect to get the sound of full blown 24-channel one, as this has only 8 channels.
Still, there should be clearly audible effect, like on this video of the same Paia vocoder. Demo starts at ~1.10:
Companders work, although they give rather odd waveforms. Not sure why that is. Maybe that's the part of them, that I can't get my head around.
Band pass filters work. To be sure, I did also some aural testing on each of them, using a signal generator's 'random noise'(not really random, more like a loop, which can be heard at a low frequency like 1Hz). They all have different, distinct audio ranges.
One thing that came to mind, was to replace the companders with something simpler, like Led-Ldr(Light dependent resistor) pairs. After a bit of search on the net, conclusion was, that they aren't generally used for a vocoder.
All the diy -designs, or versions that I'm aware of, use IC-based solutions for that.
The Led-Ldr pairs are used on some diy compressors, phase shifters and so on. Of course the linearity isn't at the same level with the IC-based 'level adjustment' -devices.
Despite of all that, I think it's worth of testing, how they could possibly work for this purpose. The form will probably be like on an envelope follower from Craig Anderton's "Electronic projects for musicians":
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Basically, the above Led-Ldr envelope follower works as follows: an audio signal is fed to the input, and goes through the op-amp(IC1), which drives the Led portions of the optoisolators(OI1 and OI2), so that the leds light up, roughly following the input signal's amplitude. Amount of that light tells the Ldr, how much it can 'open', by lowering its resistance.
So, plan is to start by cropping the above schematic's parts count to bare minimum.
sorveltaja:
To see, if the compander bus board could be used for led-ldr pairs, I made adapters for the pin headers, so that it's easier to connect the breadboard to the rest of the circuitry:
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I did roll some temporary optoisolators to test the concept. They aren't lightproof, and need to be covered with something like a cardboard box, when testing. I had to order more ldr's, as it is probably better to have the same model for all the eight pairs:
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First I connected some leds to compander bus boards' header pins(which are connected to band pass filter's(bpf) outputs), and it was like a christmas tree. Frequency sweep from 0 to 6000Hz showed, how each channel responded to certain frequencies. Adjustable gain, first stages(for mic and instrument) had expected results; when turning the gain down, leds dimmed correspondingly.
So far, it looks like the bpf's outputs itself might have enough drive to make the leds light up at the usable levels.
One part of me tells that the signals should be buffered, before they hit the leds, but not yet; only if there are several, noticeable impacts to the rest of the circuitry, then it should be done.
Version 0.1 layout of the possible compander-replacing led-ldr portion:
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Mic input signals drive the leds, and should ideally be in a dc form. Diodes do some rectifying by allowing only positive peaks of the ac-form audio waves to pass through, and then the 47uF caps smooth out rest of the signals.
Instrument input signals go through the ldr's, that act like variable potentiometers. Again, microphone's signal controls the intrument's signal output level.
As can be seen in the above layout, 'outputs' of the ldr's are fed to the final stage, being an op-amp. They are simply tied together, which shouldn't work, as all of them will fight against each other, to get a place in the sun. Buffering seems to be the key, and is added later on, if definitely needed.
sorveltaja:
First aural test, using previously mentioned circuit. Currently 5 of 8 channels are connected:
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Resulting audio, rather muted and distorted, with little of an eq added to clarify:
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Audio files used to drive the vocoder: a sample from an audiobook "Gods of Mars"(freely available from archive.org) for mic input. For instrument input, an audio file, using the hex pickup, recorded back when I made final adjustments for it.
It's very tricky to get the adjustments to the point, where the effect is most clearly audible. Changes in input signal levels affect the result also.
I'm thinking of testing the inputs by using compressed audio signals, to get more stable action. First it could be done by compressing the audio file's signals, and if it helps the overall performance, then build a compressor for each input. But we'll see.
After all, I am very much surprised, of how that rudely simple concept, where I just threw some components to breadboard, gives signs of life. Absolutely worth further testings.
vtsteam:
Amazing. And fascinating. Beautifully presented.
:proj:
sorveltaja:
Vtsteam, thanks.
The ordered ldr's arrived, and now all the 8 channels are connected. First sample was made using the same hex pickup track, to feed the instrument input. Control input (Mic) was fed with 2Hz(120bpm) square wave. Raw output, so there was no post audio processing(who needs a drum machine, sequencer, or a keyboard anymore):
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This one was made by feeding the same clip from previously mentioned audio book(speech) for mic input, and 50Hz square wave for the instrument input(or was it vice versa). But anyways, that required some heavy eq'ing afterwards, but still lacks the definition:
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So far, as this vocoder is a bare-bone version, it has already exceeded my expectations, by opening a whole new audio landscape for experimenting.
Some aspects, that need to be worked on:
Compressing the signals in audio files by using an audio editing software(like Audacity), especially speech, is more complicated, than it seems.
I'm not an audio engineer, and just fiddled with some settings, but it's way too easy to add unwanted distortion to the original signal, by 'ballparking'.
Another way to approach the problem, could be to use a soft-clipping circuit, to squeeze the speech signal between certain values, and adding needed harmonic content to it at the same time.
To realize that, something like 4049 cmos ic(which is known for its properties to clip in tube-like manner).
I should have some of those ic's in the shelf to boot, so it's going to be one of the subjects of forthcoming testings.
Then there is one thing, that I noticed earlier: the band pass filters don't have equal output levels, even at their fundamental, or 'peak' frequency ranges. I have no idea, if that is what they are supposed to be, but I'm going to find out, if they could be 'leveled' in a simple way, like using trimpots, that I have.
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