audio snippets triggering modular synth

Anything about the Scope modular synths

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at0m
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Post by at0m »

Hi,

I made audio snippets to trigger samples or synths inside modular. This works good for one trigger or sound, but if I send more triggers (each to it's own mod input port), the triggers seem to trigger other samples too, not only the one connected. Hm... there's invisible wires. ):

Here's an example of simple mod patch:

Image

Does anyone know how to make sure the audio is hard panned or at least doesn't trigger other samples? For example I make another drum in the mod above, connected to another input, and it triggers the first sample too.

I hope someone shares some experience.
Thanks.

/at0mic.
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at0m
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Post by at0m »

Problem will be solved in a while with audio-to-gate module. They're working on it.

In the meanwhile, enjoy the experiments :wink:
marcuspocus
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Post by marcuspocus »

You mean that creamware is working on an audio2gate new module for your mod?
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at0m
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Post by at0m »

Why CW?

I asked around and it's coming.

Obsidian had made an audio2midi device, but unfortunately that is not compatible with my setup (XP or SFP, I don't know). IMO this will be something similar, but outputting gate instead of MIDI note.

Imagine what one can do with it!

:cool:

<font size=-1>[ This Message was edited by: at0mic on 2002-12-17 14:37 ]</font>
marcuspocus
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Post by marcuspocus »

Yes! You bet mister! I'll have fun with this!
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Post by Spirit »

Atomic, Glad to see you're keeping your Levitating Enigmatic Fire Breathing Sphinx-like Overlord reputation intact. :smile:
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at0m
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Post by at0m »

Back to the future: analog sequencing on DSP.

Thanks Spirit and marcus :smile:

I hope this is not so enigmatic but rather usefull and concrete. :razz:

Since I cannot wait -hey life goes on- I still like to know why connections that are not made, trigger wrong drum synths.

I realise it takes very low sample values to trigger, any sample value that is not 'zero' will trigger the drum. Hard panning doesn't do the job.

One could wait for the audio2gate, but IMO I'll need hard (read: exclusive) panning for other stuff sooner or later.


I'm looking for ways how to make ASIO bandwidth more efficient. For now, I can trigger 2 samples w one channel: one triggered by max. value, other triggered by min. value. Don't need 'waves' for it, it can be done w snippets of a certain value without wave form.


The wave form I'd like to use for other stuff, like modulations, velocity or filter assignments etc.

Solution might be j9k's LSB/MSB devices, but my problem is: how the heck do I code the wave forms so I know what their LSB/MSB and intermediate values will be? I realise these are simple volume changes that make up MSB/LSB values, still I cannot reverse engineer it yet and in advance determine LSB/MSB values.

Do you realise how much bandwidth ASIO channels have as opposed to MIDI!? Let's use it!!!

C'mon peeps try this at home and let us know. Experiment. Think of the weirdest stuff first, apply Oblique Strategies here. Your imagination is the limit.

I'm just home from work, got some fresh idea's and gonna experiment w it now.

Laterz,

/at0m.
hubird

Post by hubird »

I hardly know what you're talking about, but it are the freaks on this planets who brought us to the future, so... Image

_________________
Let There Be Music!

<font size=-1>[ This Message was edited by: hubird on 2002-12-18 01:14 ]</font>
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at0m
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Post by at0m »

LOL

A bit more general technical talks:

Over a stereo ASIO channel you can send 2x44.100 16bit values per second. Why only send it to speakers? I know this sounds crazy, but we have the tools to use ASIO for totally different purposes. We can analyse the data, dissolve it and use it to control most different things. ASIO is a very stable and correct medium.

The other available medium is MIDI. It's easy and compatible with most of our gear. MIDI has small bandwidth, bad timing and is not stable.

One of my loves is very minimal electronic music. I think this could be a big step forward fine controlling the delicate changes in minimal.
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valis
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Post by valis »

Very nice....digital audio as digital control data...with the degree of control that implies I'm paying attention.

One question though, as you change bitrates & sampling rates on the asio streams will the devices detect & compensate or will the same absolute values still apply, only with a greater range of control?
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at0m
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Post by at0m »

Logically, as you change the card's frequency, it depends on how the wavs are converted. Timestretched waves are sync for modulation, while snippets can keep sample length.

I have no idea what different bit depths will do to some of it. Once on DSP, all is 32bit.

Fail-safe is to be consequent w samplerates and bitdepths.
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