bit phase synth

[astroid power-up!]

i didn't know what else to call this one. kept it simple so the idea would come through-no filters, one osc.

the red block has the kernel of the idea-an osc goes into an envelope, the post env signal is converted via the a/d convertor, while the envelope itsself is converted by another convertor. they are cross-faded together.

now, the blue group is doing the same thing, but exactly out of phase with the red group.

they are crossfaded together using the same technique in the purple group.

i was shocked at the sounds from this. usually my weird synthesis experiements end in banality, but this sounds like a fluid cross between pwm and bit reduction, maybe with a bit of hard sync thrown in. the delay effect-as in variation number 3, is especially wierd. i like it, but don't quite know what's going on. yay! variation 4 is my fave.

if you do attempt variation, remember to mess with the level convertors-that's how the blue env is getting thrown out of phase.

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bit phase synth.pch2
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[Blue Hell]

Nice one to toy with ! Has some very nice possibilities in it especially when retriggering the envelopes from the AD convertors and applying some step filtering at the output.

[astroid power-up!]

retriggering from the adc's. hmm sounds cool.

[astroid power-up!]

doing logical operations on the bits is interesting, too.

[Blue Hell]

[astroid power-up!]

'cept there's no way to sequence with those truth tables because they're those dastardly drop-down menus.

should be able to make the same kind of logical gates out of switches, though.

[Blue Hell]

Or throw in some more fixed gates in parallel and then select from that with a multiplexer or something.

[astroid power-up!]

tried that one.

all those logic sounds are fairly similar-smooth bit reduction.

what i really want to do is to take the bit info and hack it out to audio-"granular" style, to do a kind of resynthesis.

i've tried lots of stuff for resynthesis using the general idea of the split bits.

here's the quandry (forgive my lack of understanding):

you basically end up with 8 fixed amplitudes per sample, an 8-bit word. so, you could see the data as a set of 8 squarewaves of those fixed amplitudes, that are changing constantly in frequency. so, you should be able to reduce the info to a set of frequencies at any sample. rather a derivative of the frequency change.

but how to do that is kind of racking my brain. how would you do a derivative on the g2? is that even the right answer?

PS: this is how my brain works-throw random words together and try to make sense of it later.

[astroid power-up!]

whelp, i guess i'm trying to reinvent the discrete fourier transform.

and the math for the dft breaks my head apart