IMPORTANT "CAG Asy Releases do not work on the new platforms : Paca or Pacarana!" -Christiaan Gelauff
http://www.christiaangelauff.nl/
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RayYoung - 13 Mar 2011
This sound emerged in the course of time while experimenting and playing around with modules from Christiaan Gelauff's
Microsounds and my new Continuum, extending parameter modulations and trying VCS structuring. You have to load the appropriate CAG asy release (e.g.
CAGAsyRelease4) in order to play the sound. You also need sin48pnt.aif, which is contained in the
CAG asy release in the folder LUTs. As always with me: this is (never-ending) work in progress.
Because developed for the Continuum, it is most expressively played with a Continuum.
I apologize for being too lazy to document all switches and Hot Parameters used; just play around, roll the dice and enjoy.
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EckardVossas - 22 Oct 2006
Really cool to see my own example being used! Mmm...you like writing long lines in the eventvalues! If feel you are the opposite of me, I like to keep it as short as possible. It is amazing what some extreem FM with just sinewaves can sound like (Not bad that simple Twiddle!). The
HarmonicResonator? gives it the string flavor. I do not own a continuum, but it still gave me a good hint what it can sound like. I was suppriced by the variaty of sounds you produced by listening to the different presets in the VCS.
If I make some suggestions: use a logarithmic scale for setting filter cutoff, it feels more natural (to my opinion). You used an LFO (normal oscillator) that is pasted into the scale field of an envelope (btw LFOWellen(9,4096).aif I do not have!) and wrote something like: 1 - !switch*LFO L (last being the sound). This is a wast of processing power and also it will introduce zipper noise (could be it was intented). I do not know what should in the LFOWellen, but if it is something like a triangle or sine wave only the positive part of that waveform is used! To use the full resolution of the LFO you should multiply the LFO with Evenlope and add the envelope again. In the Scale of the LFO you should write: !switch negated. But it could be done on purpose.
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ChristiaanGelauff - 06 Dec 2006
For Christiaan and others who would like to try out the Continuum examples but do not yet have a fingerboard, you can play a monophonic approximation of these examples using a Wacom tablet.
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CarlaScaletti - 06 Dec 2006
Due to lack of time only a short answer at the moment. I also was rather surprised about the delightful possibilities of this "simple" sound construct. Your suggestions I will try, when I have some time. At this moment I can't well remember my intention with this envelope/LFO construct: but I guess it was: LFO and envelope should be switchable, the LFO should be multi-waveform-capable and (this is the point I can't remember) either the envelope should modulate the LFO or the LFO should yield an anvelope varying in time or something like this ... leading to a "living" and time varying (somehow unpredictable) modulation source (I probed several combinations of envelope and LFO in scale fields and came up with this solution by examination in the oscillator ... it looked nice at last ;-)). Using a Continuum gives more control over the "shape" of the sound (in terms of timbre and amplitude envelope), thus also resulting in a more "living" sound while playing. Sorry about having forgotten the wave forms for the LFO: here they are (the original name
LFOWellen(9,4096).aif got lost when attaching the file here)
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EckardVossas - 07 Dec 2006
In the meantime I had the opportunity to look into this sound and to remember my intentions. The LFO modulates an ADSR in order to make it more unsteady and vibrating. The ADSR is intentionally of type linear: therefore the Scale parameter may have negative values, the negative parts of the LFO waves are used in this case. This ADSR modulates in turn the Cutoff frequency of the VCF, leading to something like a "timbre vibrato" (oscillating unsteadily around the selected Cutoff frequency also using negative values of the ADSR resulting from the modulation with the LFO).
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EckardVossas - 20 Dec 2006
Sorry to have mislead you: your scale field of that ADSR said "1-!LFOmod asLogicValue+[LFO(multi-wave)]L". That is a bit different from what I wrote up. And Yes.. when !LFOmod = 1 you will get the positive and negative part of the sound into the scale field. But if !LFOmod = 0, do you expect modulation to be gone? If your answer is yes, then you make a mistake! Simple explaination: your [LFO(multi-wave)]L will have a range -1,...,+1 that is added to 1 so the outcome will have a range 0,..,2 but remember signals can not grow above +1 so the actual range = 0,...,1 and the upper half of your [LFO(multi-wave)]L waveform is clipped. Are you aware of this?
Next to this once more I like to indicate you use the
CutoffModulator? input to the VCF which is an audio input for controlling the cutoff of the filter. You used an ADSR for envelope modulation which output an audio signal. To modulate this ADSR you use an Oscillator(=LFO) which output an audio signal. But instead of multiplying the ADSR with the LFO you past the LFO into the scale field of the ADSR. By doing this you reduce the sample rate of your LFO from let say 44100 Hz to control rate 1kHz. If this was the intention OK. Again my question are you aware of this?
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ChristiaanGelauff - 20 Dec 2006
I've to admit, that my thoughts (and resulting formulas) are sometimes complicated and weird. But as far as I remember (I don't have access to my Kyma system at this moment) the Hot Value !LFOmod is also used inside the LFO and (!LFOmod = 0) means at the same time, that the LFO outputs 0, i.e. the amplitude of the LFO wave is set to zero. Then this construct works as intended (1 - 0 + 0 ==> 1, i.e. the unmodulated ADSR envelope).
To the second question: Because I only wanted to have a rough (and drastic) modulation of the ADSR envelope == of the Cutoff frequency, I am not concerned about reducing the sample rate of the LFO to control rate. If I had multiplied the LFO with the ADSR, then setting the Hot Value !LFOmod to 0 (==> setting the amplitude of the LFO to 0) would have resulted in erasing also the ADSR ([0 .. 1] * 0 ==> 0), which was certainly not what I intended. Thus: simply multiplying doesn't work in my sense.
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EckardVossas - 21 Dec 2006
Oke, good to see you understand your own stuff. When multiplying the LFO with an ADSR you have to insert a
ScaleAndOffset? prototype after your LFO (Scale and Offset = 0.5) to get the thing you wanted. If you wanted to have a !LFOAmp, you put this in the scale field of your LFO oscillator and put (2-!LFOAmp)*0.5 in the offset field of the
ScaleAndOffset? sound.
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ChristiaanGelauff - 23 Dec 2006
