SirSickSik Contributions

#551

NEW MODULES

OSC

"waveformGenerator"

Generates waveforms to be used with table-oscillators/LFO's.
Waveforms are generated at start-up, this takes some time, but when finished, this doesn't ask any CPU anymore! Of course, you could also save the table to a file and later just re-call that file for future wavetable use.

The size of the waveforms (sample-length) can be set with an attribute, allowing you to make HQ waveforms for the 2Dtablemorph oscillator.

First 9 waveforms are 'standardised" waveforms
sine,triangle,saw,square,peak etc.
-harmonics, sets the number of harmonics that will be summed together for the standard waves. So setting this to 10 will only create the first 10 harmonics of eg. the saw.

Depending on the "waveforms"-setting, it fills the rest of the presets with random generated waveforms, though these are limited to the settings below the "waveforms" setting.

-minharm sets the minimum amount of harmonics that can be present in the random waveforms, though it might perfectly happen that it will sum the same harmonic multiple times (because of the random harmonic-number-selection).
-maxharm sets the maximum amount of harmonics that can be present in the random waveforms.
-The harmonic-numbers that are chosen to add are randomly selected.

-maxLvl sets the maximum output level for the waveform. 64 will fill the whole scope-display.
As my axoloti already starts clipping around 40, I often put this to 32 to save me some headroom.

ps. when using this module for my older table-oscillators, set the size to 1024.
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"2Dtablemorph"
a cousin of the 1D and 3D table morphers..but not completely the same.. as it uses the above module to serve it's waveforms. (the other two only use 1024-sample waveforms)

though the mix-control is the same as the 1D and 3D morphers.
Using a LFO as input, you can set to how many waveforms it will morph, where it will start the morph and what the stepsize is between selected waveforms (skipping waveforms in-between).

#552

NEW MODULE

FILTER

"dualMorphSVF"

a continous morphing variation on the complexDualSVF

Dual Mode-morphing state-variable-filter.
-Using a saw-wave LFO as input for the ModeX(1/2/A) creates a quadrature reading through:
LP->BP->HP->Notch->LP
ModeX1 and ModeX2 will control the modes of filters 1 and 2 respectively.
ModeXA controls both modes at the same time.
-depending in which input you send your audio, it will respond differently:
-in1->normal response
-in2->inversed filter response at filter 1, normal for filter 2
-in3->only send to filter 2 as inverse filter response
-modeY1 controls the offset from the center, this control has an inversed response, meaning, a high input will force the mix to a centered mix of all filter modes. When above 64, it will inverse the filter responses, turning LP to HP, HP to LP, BP to Notch and Notch to BP.
-SerialSel controls the amount of each audio input to be send to filter2 input when serial mode is mixed in.
-SerPar mixes between serial and parallel mode for filter 1 and 2.
-drive controls the drive over the resonance (BP filter part that influences the other filters)

#553

NEW MODULE

LFO

"rubiLFO"
tempo-synced LFO version of the rubisik oscillator (eurorack rubicon-oscillator based waveforms)

#554

NEW MODULE

LFO

"2DmorphLFO"
tempo-synced LFO version of the 2Dtablemorph oscillator
combine it with the "waveformGenerator" module to generate LFO waveforms (save waveform-table to a file for future use)

#555

NEW MODULE

MATH

"baseRatios"
to be used with my upcoming glitchBeast

Module to calculate only ratio's of the input signal based on the selected base-values.
You can either just include a single base value, like 3, to only have "power of two" multiplications/divisions of 3 as output values:
1,3,6,12,etc divided by 1,3,6,12etc
Or use 3 bases, like 2,3 and 5 and their "power of two" multiplications, for more complex ratios.
The whole list is created at start-up and sorted out from lowest to highest value automatically.

The "total" control sets how many values will be created for the M and D selecting inputs.
The M input selects a value from the created list to use as a multiplier for the time input.
The D input selects a value from the created list to use as a divider for the time input.
Both inputs expect a fractioal control value between 0 and 64. (like a normal control module)

#556

NEW MODULE

FX

"glitchBeast"

very heavy multi glitch effect with 32 effect parameters to control (at least one per fx)
all effects can be used simultanuously!
this module is almost a patch-filling module on it's own (with automation recorders added). So you could use this for a stand-alone glitch-fx module for your hardware.
each effect first starts with a gate input (eg. Glag) and then has one or more CV-inputs for the parameter (eg lag) (refer to fx-patch in my community folder how to use it with midi-note&touch-modules and automation recorders)
Delay times can be controlled with my host-tempo-timing modules

the effects:
-lag: introduces a (tempo-synced) delay to the audio output->100% wet. Actually, the module constantly reads out the buffer, even when everything is off (no delaytime used, just direct readout of latest recorded input). Lot of the delay-effects do something to the readout-position of the delay.
-retrigger: feeds back the output audio back into the start of the delaybuffer and bypasses the input. This is not your regular retrigger! ALL the other effects are WITHIN the retrigger effect for instant granular mahem!
-gap: introduces a volume gap to the audio at the rate of the current retrigger time. "Gap" controls the gap size while "offset" changes the phase of the gap in respect to the delay, allowing a different part to come through.
-reverse: reverses the play direction of the buffer reader, reversing the audio. The CV controls the rate the audio goes into reverse play (hard press is instant reverse)
-tapestop: stops the buffer reader, slowing down the audiorate to zero. CV controls the rate at which it stops, the higher the CV, the quicker the stop.
-mod: modulates the delaytime using a sine LFO with CV's for rate and modulation width
-samplerate: reduces the sample rate with higher CV's
-spatial: offsets the left or right channel up to 64 samples delay. (32=no delay to either channel)
-phaser: some kind of weird phaser/chorus/modulated delay effect. Prerate controls the modulation rate at which the incoming audio is written to the buffer (interpolated and summed with buffer which is set to zero just before the writer starts adding audio), feedrate controls the modulation rate of the feedback time modulation (readout) and feedback controls the volume/amount of thefeedback.
-repeater: repeater delay repeats the incoming audio several times (up to 16) at the same volume and adds this to the original (or retriggering) signal. RepeatSize sets the time of the delay and Repeats sets the amount of delays.
-ringmodulator: ringmodulates the audio and adds it to original. Ring controls the pitch of the modulator wave, ringgain adds gain to the triangle wave, turning it into a squared trapezoid.
-panning: panning pans the stereo signal between left and right, panRate controls the rate of the panning
-multi mode filter: SVF filter with controls for cutoff, resonance and filter-mode. Morphs continously through 4 filter modes: LP->BP->HP->Notch
-grainuliser: read out a position somewhere within the recorded buffer and use it as a waveshape for an oscillator. GrainSize sets the size of the buffer being read. Grain position controls where in the buffer the samples are taken and GrainRate sets the play-rate of the grain
-pitchshifter: pitchshifts the signal up or down. Pitchshift sets the shifting size while Window sets the buffer size for the pitchshifter.
-scratch: uses the CV directly to offset the readout position. Kind of like "lag", but lag is meant to use tempo-synced delay times while this uses continous filtered signals.
-random delaytime jam: randomises several delay-buffer-readouts throughout the effect, causing a lot of hard-glitching. CV controls the rate at which the randomisation is updating.
-Octaver: simple rectifier&highpass based octaving effect, more like a distortion. CV controls feedback amount, causing even higher frequencies.
-suboctaver: generates suboctaves by using a counter. subrate controls the rate at which the sub-generator responds to an up/down command of the counter->slow rate causes a triangle shape of low volume.

here's a demo patch, to be used with a polyphonic-aftertouch capable midi controller like beatstep pro. The settings will set the first effect on C in the middle of the keyboard (no octaves up/down).
the patch also shows how to use midiNote modules for triggering certain effects and recording aftertouch for seperate notes.

glitch fx2.axp (67.1 KB)

Glitch Beast help?
#557

NEW MODULE

MIDI

"rndMidiNotes"

This module generates random internal midi-notes.
-touch controls the polyphonic midi aftertouch that's being send with the random generated notes
-generate controls the amount of random notes that are being generated (max=16)
-offset controls the aftertouch-offset of each new extra generated random note.

this module is a complement to the glitchBeast module, so effects can be triggered randomly while the timing of the random fx can be recorded using the "touchrec" module.

#558

Hey @SirSickSik

Did you sync library? These are missing here when loading the patch you supplied:

sss/midi/rndMidiNotes
sss/math/baseRatios
sss/fx/glitchBeast

Thanks

#559

yeah, I hit sync somewhere in the middle, but not at the end anymore, thnks for reminding

#560

Thanks works now :slight_smile: .............

#561

tonight at the bar with a beer I suddenly figured... I haven't made any live recording&play of samples modules.. if anyone has more of these "new" ideas, which I haven't covered yet... let me know..

#562

NEW MODULE

MATH

"divscale"

Multiply (attenuate) with a constant value, then divide by an integer.
To be used with the divide-outputs of the table-oscillators/LFO's to scale the input back to maximal quantification 4 steps, whatever the amount of quantification of the quantizer of the oscillator/LFO.

#563

NEW MODULE

DELAY

"tappedDelay"
multi-tap delay with lowpass filter on each tap.
You enter the tap-velocity/timing/cutoff for each tap by external tapping.
eg. when using a midi note module:
connect gate-output of midi module to the tap-input of this module
connect note-output of midi module to the cutoft-input of this module
connect velocity-output of midi module to the vel-input of this module

first tap you play will reset the timing-counter to the "zero" position->incoming note
taps after this will set the timing/velocity/cutoff of the delays.
Up to 16 taps can be recorded.

when you want to set new timings for the delay, just hit "rst" and play a new rhythm.

on-module cutoff-knob is added to the recorded cutoff.

there is an onboard control for feedback, but be careful with this one. Full feedback will be 100% feedback for ALL taps, thus making the signal louder and louder as taps will be fed back into each other and multiply the volume..

#564

NEW MODULE

ENV

"hahdEnv"
delay/Attack/hold/decay envelope, linear attack, exponential decay.
triggered, not gated

#565

NEW MODULE

OSC

"2DHQtable"
updated version of the "2DtableMorph" oscillator.
Added interpolation for the readout of the tables and an internal sine generator to be able to fade out the table-harmonics using an external envelope.

#566

NEW MODULE

DELAY

"loop4"

16bit 4 channel audio looper with channel "summing2one"
Allows you to record 4 seperate audio loops.
Syncs to external clocks.
Recorded loops can be saved/loaded to/from SDcard.
Channels can be combined using the combine-function.

-rstCount: send a trigger to this input to restart the audio loop. This resets the counter for recording/reading the audio buffer to zero.
-playChnl1 to 4: mute/unmute the audio for this channel.
-EditChnl: input between 0 and 3, selects the internal channel-buffer to save the incoming audio to.
-rec: overwrites the current buffer-position of the selected channel with the incoming audio.
-dub: sums to the current buffer-position of the selected channel with the incoming audio.
-delete: deletes the current buffer-position of the selected channel.
-clear: clears the whole channel, though at a low rate to prevent hick-ups. This can cause the channel to still sound until it re-loops again.
-combine2channel: selects to which channel the current selected edit-channel will be summed to. (0, 1, 2 or 3)
-combine: combines the selected channels into the combine2Chnl-channel.
-save: saves the current audio buffers to your SDcard.
-load: loads a saved buffer from your SDcard into the audio-buffer
-filename: use this to set the filename of your audioloop-file

the next two are for adjusting the play-position or to shuffle the loop by an external CV-pattern
-offset: this enables you to quick-offset the reading position.
-size: this sets the division/quantification of the offset. Higher numbers will divide the total loop-length (of the rstCount-timing) into smaller parts, allowing you to offset the audio in smaller steps. So, setting this to 4, while "offset" is 1, will offset the read-out of the audio-buffer a quarter of it's looping length.

-size attribute: sets the maximum size of a single channel. Total table-size will be 4 times this big!

ps. I experienced some "bugs" having audio recorded into other channels. If the sync/reset time is longer then the buffer, it starts recording the extra time into the next channel. I let this be for the moment, as it might be useful somehow.. if you don't want this to happen, just make sure the reset is within the table-size.

#567

NEW MODULE

CTRL

"cfgKnobF" and "cfgKnobI"
knob with a text-config for configuring the width of the knob with a minimum/maximum value.
Expects integer numbers between -128 and +128 and scales it to the right 32-bit range for the cfgKnobF module.
how to config?
eg. for a knob output range between -32 and 64. (don't forget the ; at the end of each line!)

press the "config" attribute and type:

max=64;
min=-32;

the "cfgKnobI" module outputs integers and expects just any integer min/max values.
so with this one you'ld just add like:

max=56252315;
min=234;

#568

NEW MODULE

DIST

"softComp"
direct-audio-compression with soft-treshold feature. No internal envelope follower, attack or decay, so it's more like a distortion, but based on the native dynamics module

#569

NEW MODULE

DYN

"compressor2"
compressor with soft-threshold function, attack/release, ratio and gain.
very nice distortion with short attack/decay rates
has readouts for input/output volume, compression amount and treshold reach

#570

14 more modules to reach 400 modules and 5 days left! :wink: