GUI design: adventures in Blender.
Comments
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Bowlle, use an volume loudness sensor behind the filter and let it keep the relative volume the same when you change the filter. I'm sure there a million compressors in the users section. I prefer the type that uses a humans sense of hearing in the different frequency areas to determine the amount of compression or in the case, volume compensation.
Later..
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Playing around with recreating an existing design by Company Who Shall Not Be Named, just for fun.
Plan is to at some point use it for a VA synth build and name it Minibaum.
This is for personal use and will not be distributed via the Reaktor User Library or anywhere else.
Greets,
Bolle
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Trying to recreate a square wave and i need help.
The left is the one i want to recreate. The right (pixeled scope) is where i'm at with recreating it.
The peaks i can round off with a simple filter. But i then want to add that little sharp bump at the end of the ramps. And, if i eventually get there, i need it to work equally across all note pitches.
Greets,
Bolle
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This would probably be close enough. If i could reverse it. 😂
I'll get there eventually.
Greets,
Bolle
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I have to admit, I sure do like the 3D views you make with Blender. That's cool man!
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i need it to work equally across all note pitches
It looks like at least in part to be a high pass filter at low cutoff - like 20Hz, but that would change somewhat across all note pitches. What is the source? and are you sure the waveform doesn't change at different frequencies.
Is the wave from an analogue source, or digital?
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Creating an alternative GUI for the Beezlebaum.
First image is a carbon copy of a Moog Minitaur, with dual function Decay/Release knobs for the envelopes and Frequency Modulation of the oscillators.
Second image is a Beezlebaum version with full ADSR envelopes, no FM, and an oscilloscope. The knobs turn NI standard +/- 150° instead of Moog's 140°
Oscillator buttons will cycle through 4 wave shapes by lmb clicks, with the selected wave being displayed on the button.
Greets,
Bolle
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I ditched this plan. Might go back to it later.
The image i have from a Minitaur square wave would indeed vary across different notes.
Greets,
Bolle
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Minibaum available for download.
Get it here: https://www.native-instruments.com/en/reaktor-community/reaktor-user-library/entry/show/14983/
Greets,
Bolle
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Greets,
Bolle
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Question:
What is the proper mathematics setup for a Filter ( Pro-52 in this case) with P and F input, making use of a Cutoff control, an LFO and an ADSR envelope?
How do i stitch these together? Or which Reaktor synth would be a good source of inspiration for this basic, subtractive setup?
I didn't do it properly building the Beezlebaum. Not a problem, it sounds great. My current project however, requires a more proper and accurate setup.
Greets,
Bolle
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Also:
How does one keep the output volume of the oscillators of a polyphonic oscillator at bay when many keys are played? Compression?
Greets,
Bolle
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Reskin of Alt Fidelity's classic studio compressor, Inga Naïve Leveller, reuploaded.
Get it here: https://www.native-instruments.com/en/reaktor-community/reaktor-user-library/entry/show/15003/
Greets,
Bolle
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Should it not be louder when there are many keys being played?
If not, then what if you are playing one key, then while still holding that key, you hit 5 others simultaneously? should the first suddenly get quieter?
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The P input is the Midi note number, where each +1 or -1 is like moving one note along a piano keyboard. Every +1 factors the frequency by the 12th root of 2, about 1.059. Doing this 12 times means the frequency is doubled, which is an octave.
The F input changes the actual frequency of the filter and is in Hz.
So, P is logarithmic, and F is linear.
Internally, the filter uses both inputs. It first converts the P input into a frequency and adds the result to the F input. That sum is the actual cutoff of the filter.
As a builder, the big difference to me is that the P accepts events, and the F accepts audio. If you want all modulations to occur at the audio rate, tie the P input to a constant, a really low number, like -200. This is because an unconnected P input is actually Midi note 0, which is about 8 Hz. Not intuitive, perhaps, but mathematically correct.
LFOs used for vibrato, and keyboard tracking, are most easily done in the land of P. So, if the event rate is sufficient, use it. If not, tie the filter's P input to -200 and put a P>F converter between the LFO or keyboard and the filter's F input.
Envelopes pretty much have to run at audio rate to sound decent. Tying an envelope directly to an F input means that there is no keyboard scaling at all. If that's not what you want, one solution is to run the envelope through a P>F converter before sending it to the F input. Put your modulation amount control between the envelope and the converter.
Core filters do everything using F, so use P>F converters for things like vibrato and pitchbend, where the depth should be the same regardless of the note being played.
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