Some VCVRack modules
New modules in 2.1.0: PAD2, µPad2, PhO, PhS, PHSR, SPL, YAC, Faders, PLC
New modules in 2.1.1: PHSR2, CSOSC, BWF, EVA, Ratio
New modules in 2.2.0: Osc1, Osc2, Osc3, Osc4, Osc5, Osc6, PRB, SPF, SKF, L4P, USVF, CHBY, AP, LWF, DRM, SWF, JWS, CLP, CWS, RndH2, AUX, OFS and OFS3
New modules in 2.3.0: Osc7, Osc8, Osc9, Drum, PPD
New modules in 2.4.0: FS6, Pulsar, OscA1, OscS, OscP
New modules in 2.5.0: X16, X8, X6, X4, CVS, DTrg
See also the demo patches on PatchStorage or on youtube
Table of Contents
- Modular Phase Distortion
- Sound Generators
- Additive Oscillators
- Some CPU friendly polyphonic filters
- Some cpu friendly polyphonic Waveshapers
- Effects
- Random
- Sequencing
- Matrix Mixer
- Others
The following modules are built for a modular phase distortion synthesis system which consists of:
- A start module which delivers a phase. This can be anything outputting an audio wave, but the most useful start phase is a clean saw wave e.g. the output of the VCV LFO Saw. Later also a sine wave or triangle wave can be considered.
- A chain of modules which modify the phase. This also can be anything which has an input and an output.
- A phase driven oscillator. This is the key point of the system. Such an oscillator is capable of taking an external phase to produce its output.
Let's start with the last point:
A 16 harmonics additive phase driven oscillator. The harmonics are passed using the polyphonic input "Pts". When fed with a clean saw wave it outputs the additive wave. With the dmp parameter the harmonics can be attenuated with an exponential falling curve towards the higher harmonics.
Now lets take a look what happens if the phase is modified somehow. Here we take a wave shaper to modify the phase:
pho-a1.mp4
Other interesting possibilities are using all kind of filters (Fundamental VCF, VULT Tangents etc).
This module provides the shaper extracted from Valley's Terrorform module for shaping a phase. There are a lot of different shapes available as shown. Each shape can be adjusted with the Amt parameter and input.
Here an example:
PhS.mp4
This module is a tiny Phasor module which can be used as start phase generator It provides a polyphonic V/Oct input, a reset input and outputs for a clean saw wave, a sinus wave and a clean triangle wave.
New in 2.1.1: Frequency modulation -- By using FM PHSR can be used as FM Operator. When used as carrier the frequency modulation is applied to the target where its phase is connected to. So FM can also be utilised on phase driven oscillators which have no FM implemented by their own.
An advanced Phasor with frequency modulation (including linear FM through zero). The phase can be distorted via moving the points (up to 16). The inner points (2 up to (Len-2)) can be modulated via the modulation inputs:
- The Y values are in the range -5/5V
- The X values are in range 0/10V and influence the actual x coordinate relative to the surrounding points.
There are several modules which can be phase driven:
- FS6 (new in 2.4.0)
- PRB (new in 2.2.0)
- CSOSC (new in 2.1.1)
- SPL (new, see below): generates spline waveforms from up to 16 points
- GeneticTerrain
- Geneticsuperterrain
- SuperLFO (if the frequency is set to the lowest value)
- Valley's Terrorform: connect the phase to the FM input, set the attenuator to 0.5 and set "zero mode" on.
- Osc1 is a polyphonic oscillator making complex waveforms out of up to 16 given points which can be modulated (see PHSR2).
- Aliasing is suppressed via 16 times oversampling.
- Osc1 provides exponential FM and linear FM through zero.
- The PD output delivers an alias suppressed phase distorted sine wave using the given line segment.
- Osc2 provides two sinus oscillators with adjustable phase difference.
- It outputs the maximum wave and the wave of the first oscillator clipped by the second.
- The second oscillator uses its own v/oct input if connected.
- The phase can be realigned using the reset input. (if the v/oct inputs differ the phases of the two oscillators will diverge)
- Osc2 provides exponential FM and linear FM through zero.
- Aliasing is suppressed via 16 times oversampling but is turned off by default (see menu).
- Osc3 is a polyphonic oscillator making complex waveforms for a set of up to 16 given values via the polyphonic Pts input (similar to SPL).
- It outputs the line segments and the step segments.
- Aliasing is suppressed via 16 times oversampling.
- Osc3 provides exponential FM and linear FM through zero.
- Osc4 is classic polyphonic oscillator with a smooth transition from triangle to square wave.
- Wave input behaviour:
- 0 = Triangle
- 0.3 = Saw Wave
- 0.6 = Square Wave
- 0.6-0.98 = Square wave with modulated pulse width
- Osc4 provides exponential FM,linear FM through zero and hard and soft sync
- Aliasing is suppressed via 16 times oversampling.
- Osc5 is an alias suppressed polyphonic pulse oscillator
- It makes 4 pulse per cycle.
- The BP1 parameter controls the position in time of the 3rd pulse
- The BP2 parameter controls the relative positions of the second and forth pulse
- Osc5 provides exponential FM and linear FM through zero.
- Aliasing is suppressed via 16 times oversampling.
- Osc6 is a polyphonic complex oscillator which generates its waves with integer arithmetic and byte logic.
- i.e. it computes its wave via the formula:
where t is an increasing integer depending on the input frequency (via Freq, V/Oct and Oct) (see also bytebeat or ByteBeat from Voxglitch, however this module is more intended to be an oscillator. In many cases a bytebeat can be obtained by going e.g. 6 octaves down).
((x&(t*xc>>xa))|t*xd>>xb)*(yp+((y&(t*yc>>ya))|t*yd>>yb)*ym) - The X, Y parameters are given in bits either set or unset. The X/Y inputs must provide polyphonic gates. The inputs and parameters are combined with a logical OR.
- Note that the bits to be set in the X or Y parameter which are above the bit parameter value are ignored.
- If X and Y = 2^bits-1 (all bits set) and only the parameters xc and yadd are 1 (the default preset) then a saw wave is produced
- If XC>0, YM>0 and YC>0 then it gets very complex as then the term t*t is present in the formula,
- Aliasing is suppressed via 16 times oversampling (can be turned off in the menu, as DC blocking).
- There are some factory presets to check out.
- Osc7 is a complex oscillator producing some harsh phase distorted waves.
- It provides 8 types of phase distortion which can be controlled via an amount parameter and input.
- Aliasing is suppressed via 16 times oversampling.
- Osc8 has two additive oscillators
- The oscillators can TZFM modulate each other (in both directions or cross modulation) controlled via the FM inputs.
- They provide up to 16 partials which can be offset, stretched, decayed and filtered (odd/even)
- There is a ratio and fine control for pitch
Osc9 is a wave shaping oscillator providing two controls for shaping the sound -- try out!
- PRB is a very fast polyphonic oscillator providing sinus like waves via polynomials
- With degree 2 it is a sinus wave with slightly odd harmonics above (picture above)
- With higher degrees (parameter 'deg') it converges in direction to a square wave
- PRB provides exponential FM and linear FM through zero.
- PRB has a phs input for phase distortion synthesis
- FS6 provides 6 very fast sinus wave approximations with far less harmonic content than PRB
- It is useful for FM Synthesis via linear FM through zero and phase distortion synthesis
- CSOSC is a cosine oscillator with adjustable clip (towards pulse) and skew (towards saw).
- It is implemented by distoring the phase. This distorted phase can be retrieved through the phs output and connected to phase driven oscillators.
- Via the phs input CSOSC itself can be phase driven.
- CSOSC can be used as FM Operator.
SPL takes up to 16 points through the polyphonic PTS input and generates a smooth spline wave. Additionally, it outputs the line segments and step sequence of the points through the Line/Step outputs.
If SPL is fed with a phase i.e. the phase input is connected, then the V/Oct input is ignored.
Here an example. The spline wave is generated randomly and smoothly changed over time, and a bit strechted randomly over time with the PhS BEND shaper.
SPL.mp4
Geneticterrain is a polyphonic oscillator using wave terrain synthesis.
The 27 provided base terrains can be combined 4 times so each terrain is a "genetic" sequence with length 4 (sums up to 531441 possible terrains but don't worry many of them are similar). There are 10 different curves available and each of it has a curve parameter. The curves can be moved on terrain via dragging the blue circle or the X/Y Knobs. The view of the terrains can be moved by dragging not on the blue circle.
On the left side you can select the terrains, in v2 simple right-click on the knob and select. The polyphonic GATE input is mainly for the display of the curve for each voice and has no influence on the sound. If WYSIWYG make sure the GATE input reflects the GATE you are using for e.g. the VCA or ADSR.
On the right side there are the modulation parameters for the curve. The modulation inputs are polyphonic, so each voice can be parametrized separately as shown in the image above.
NOTE: There is no band-limiting implemented for different reasons, so you have to handle possible aliasing by yourself. In general: scale down the curves to cut down high frequencies.
NOTE: There are a lot of performance improvements implemented, but it costs still about 2% CPU per Quad Voice so maximal about 8% when using 16 Voices also depending on how many and which Terrains are combined.
If the phs input is connected when driving it with an external phase the v/oct input is ignored.
Geneticsuperterrain is the same as Geneticterrain but with curves produced by the superformula. (https://en.wikipedia.org/wiki/Superformula#Generalization)
Don't use it ;-)
And if you use it beware of the following issues:
-
The curves are not on all combinations of M1,M2 (aka Y,Z) closed and have poles (closed in infinity). For example if N1>0 and there exists an n,m in Z with 2*M1/M2 = 2m+1/n i.e curves with (3,2) (5,2) (7,2) etc and (5,4) (6,4) (7,4) (9,4) etc. have a pole which is noticeable when listening. If N1 < 0 then the curve is reversed and the poles go towards zero in this case (see https://web.archive.org/web/20171208231427/http://ftp.lyx.de/Lectures/SuperformulaU.pdf)
-
The speed of the point on some curves can vary dramatically which results in high frequencies (but also very interesting sounds) and may upset connected VCFs.
If you still want to use it, I'd recommend to start with simple curves e.g. M1=M2 and avoid sharp, pointy curves. (use rather the red curve in the image above and not the yellow one).
The SuperLFO module outputs super formula curves (x and y). It can be used as LFO and as normal polyphonic Oscillator by moving the freq button to maximum (261.63 HZ).
If the frequency is set to the lowest value it is treated as zero. Then it can act as phase driven oscillator by using the phs input.
The current curve can be visualized by placing the plotter module on the right side of SuperLFO.
AddSynth is a monophonic two-dimensional additive synthesizer. The first dimension is defined by the ratios of the partials and the corresponding amps which are modulated by the polyphonic amp inputs. The second dimension is defined by the wave which is applied for each ratio.
The incoming amps can be attenuated by an exponential falling curve towards the higher frequencies. A dmp value of 0 causes the strongest attenuation.
The ratios and waves are defined in res/addsynth_default.json in the dbRackModules plugin directory. You can copy this file to addsynth.json in the same directory and make your own version of it. If it can be read without errors it will be taken instead otherwise it falls back to addsynth_default.json (which should never be changed)
This module makes pads using Paul Nascas PadSynth algorithm. Simply checkout these beautiful sounds by playing around with the for parameters after connecting a polyphonic v/oct source. Hint: the partials amps used for the IFFT are randomly generated (parameter seed) using different methods (parameter Mth).
New in 2.1.0:
- The amp of the first partial can now be controlled with the amp knob - was 1.0 before.
- Presets will be applied directly without clicking on a knob.
- External parameter changes (e.g. from µMap) can be applied by sending a trigger signal to the new trig input (sse example below).
- In the menu there can be set a fade time which causes that on every change the sound changes smoothly to the new.
- There is new method "static" which produces a descending partial sequence. If the seed is 0.5 all partials are part of the sequence (saw like wave). If the seed is zero only odd partials (pulse wave) are used and if the seed is one than only even partials are used. Every seed in between will be interpolated.
pad.mp4
This module makes pads using Paul Nascas PadSynth algorithm. It is maybe the first module implementing the PadSynth algorithm which is capable of smooth handling every parameter change. The wave is computed completely in the background not affecting any CPU usage inside VCV Rack. But it may turn on the fan of your computer if a lot of parameter changes are made in short distances e.g. connecting an LFO to the BW input. The background processing may occupy another CPU with about 20-30%.
Changes to PAD:
- 48 partials can be edited
- There are polyphonic inputs for the controlling the partials.
- The random function updates the faders and the sound can be adjusted afterwards
- There are inputs for BW and SCL including CV attenuation.
- Advanced: The phases for the IFFT are random generated. With the Phs-Seed parameter the random seed can be changed, which does have a slight effect.
- Advanced: The fundamental frequency is set to a low 32.7 HZ by default. This can be changed with the Fund parameter. It matters if only high frequencies are played - then it may be useful to increase the fund parameter. Or when experimenting with vocal frequencies ....
Here an example:
pad2.mp4
Thanks to Paul Piko and Dave Benham for testing and feature requests.
This is a small version of PAD2. It has to be fully controlled by the partial inputs and inputs for BW and Scl.
The Faders module can be used for controlling µPad2. The following example shows how to transfer a sound from PAD2 to µPad2 using the Fader module. The Fader module has a special function for fetching the values of PAD2 by placing it on the right side of PAD2, placing the mouse cursor over the Faders module and pressing 'f'.
mpad2.mp4
This is a port of Supercolliders gendy which is based on the work of Peter Hoffmann who has rebuild GENDYN from Iannis Xenakis.
A tiny sample based drum module with pitch, gain, decay controls.
- Pulsar is a pulsar synthesis oscillator
- It produces trains of short wave forms adjustable with the duty parameter
- The wave form can be smoothly controlled from sine to triangle to square to saw
- The pulses can be clustered and enveloped via the cluster and window parameters
- Aliasing is suppressed via 16 times oversampling.
Some alias-free oscillators which generate the wave form via adding 256-512 partials. The computation is done in the background and not in the main audio thread. Therefore, there is a menu which controls the buffer size used to communicate between the generator and audio output. If the oscillator makes noisy sounds the buffer size should be increased. The communication via the buffer on higher sizes may introduce a small delay of the audio output which can be compensated via a delay of e.g. 10ms on the trigger gates for vca/env module connected behind
- An additive oscillator similar to Osc8 (see there for the parameter descriptions)
- OscA1 provides additionally a spectral comb filter
- Produces an additive generated alias-free saw wave
- Produces an additive generated alias-free pulse wave
- Provides PWM
A fast polyphonic Steiner Parker filter.
A fast polyphonic stereo Sallen Key filter. As the filter is reacting very sensitive on changes of the cutoff frequency, there is a slew limiter on the frequency input. The slew amount can be configured in the menu.
A fast polyphonic stereo linear 4 pole filter
A fast polyphonic unstable state variable filter with overdrive control
A fast polyphonic 4 pole Chebyshev type 1 filter
A polyphonic allpass filter. A building block for phasers and reverbs.
A cpu friendly polyphonic Rack v2 version of the Lockhard Wavefolder released in the JE-plugin for Rack v1.
A cpu friendly polyphonic Rack v2 version of the Diode Rimg Modulator released in the JE-plugin for Rack v1.
A cpu friendly stereo polyphonic Rack v2 version of the Sharp Wavefolder of the Agave Rack v1 plugin.
A cpu friendly stereo polyphonic waveshaper based on the De Jong Attractor algorithm of Nozoïd Sin WS.
A cpu frienfly stereo polyphonic clipper/waveshaper/satrurator/overdrive with 9 different selectable algorithms
- Soft: Softclip via TANH approximation
- Hard: Hardclip
- Sin: Sinus as shaper function
- OD: Overdrive
- Exp:
sgn(x)*(1-e^(-|x|)) - Sqr:
x/sqrt(1-x*x) - Abs:
x/(1-|x|) - AAS: Anti Aliased Softclip
- AAH: Anti Aliased Hardclip
A cpu friendly polyphonic chebyshev waveshaper with up to 16 coefficients (to be passed in the polyphonic input)
This is a stereo polyphonic dc blocker.
Use the default order to dc block and the lower orders for stronger highpass filtering. This is a reimplementation of the Csound dcblock2 algorithm for Rack.
An fft-based pitch shifter from the gamma library.
Port from Csounds reverbsc opcode. This is an 8 delay line reverb.
Port of a 5x5 waveguide mesh reverb from Jon Christopher Nelson and Michael Gogins (https://github.com/gogins/csound-extended/tree/master/Opcodes/MVerb) with the following differences: There is no equalizer on each delay line (would cause 1000 biquad filter operations). The randomization of the mesh frequencies can be done by the frequency modulation inputs.
The standard presets are provided in the factory presets of the module.
New in 2.1.0: The MVerb can now be run in a threaded mode (the default setting in the menu) which puts the heavy computing into a background thread. It should now run below 1-2% CPU. If there are audio glitches the thread buffer size can be increased in the menu.
Yet another chorus, this one can be modulated.
Long time I wondered why chorus, flangers almost only have an internal LFO with a sine wave. So I built this one. If the ModL and ModR inputs are connected they were taken instead of the internal LFO. The rate parameter is ignored in this case and the depth parameter is used as attenuator for the incoming modulation signal.
Here an example using Caudal from VULT as modulator. With intent a just chord is used to avoid 'built in' chorus effects which occur in 12edo (de-)tuning.
YAC.mp4
A brickwall filter. Uses FFT to cut frequencies. It is more an effect as a filter ... The fft size which determines the latency can be configured in the menu.
A bpm controlled mono to stereo ping pong delay with sends and returns.
RndH is a polyphonic, seedable random and hold with 3 Outputs serving different distributions:
- Min outputs the minimum of k uniform random values (k is adjusted with the Strength-Parameter). In bi-polar mode the output is randomly flipped at zero.
- WB outputs the weibull distribution with sigma = 1/k. In bi-polar mode the output is randomly flipped at zero.
- Tri outputs the triangular distribution i.e. outputs the average of k uniform random values. With higher k it approximates the gaussian normal distribution.
If Strength = 1 all outputs deliver a uniform distribution. The image below shows the difference between the distributions for higher strength values.
If the seed input is not connected or 0 the system time is used as seed on every RST trigger. Otherwise, the seed is built from the provided value 0<x<=10. In this case for each seed you will get exactly the same sequence on every RST Trigger.
The number of output channels can be adjusted with the Chn Knob.
When no clock is connected RndH outputs the noise as audio stream.
RndC is similar to RndH but outputs a (very) smooth polyphonic signal using cubic interpolation. Because of this RndC may produce lower/higher voltages as 0V/10V in unipolar mode and -5V/5V in bipolar mode. I have foregone to scale it as it is done better outside (and potentially different for each usecase). By scaling with 0.8 and adding 1V the normal range should be achieved. But this you may not want for the min distribution or weibull.
This image shows from left to right: min, weibull, tri with strength 10 and uniform.
Additions over RndH:
- The clock input is polyphonic, the Chn parameter is only used if the channel count is one.
- Can be seeded with a parameter knob
- Has Range and Offset parameters which can be modulated
- Has and additional Uni output which provides still a uniform distribution regardless of the strength parameter
- Has a slew limiter
The module does the same as Count Modulas Clocked Random Gates but is full polyphonic, seedable and a bit smaller sized. The probabilities can be provided via the polyphonic prop input (-5V/5V, 0V = 50%).
RndHvs3 uses a 3d HVS Cube (similar to Csounds hvs3 opcode) filled with seedable random values of different distributions. The cube values are defined by a 3d lattice of size 7x7x7 where on each node in the lattice there are put 32 random values.
The values in between the nodes and on the vertices are linearly interpolated.
The x,y,z inputs (0-10V) define the (continuous) point in the cube for which the 32 values are outputted through the two polyphonic output ports. It will output a smooth signal on every output channel if the input is a smooth signal.
A polyphonic gaussian random trigger around a given frequency and deviation. The used random values can also be plugged in via the src input. If Chn > 1 then on each internal trigger the output channel is randomly chosen.
HexSeq is a 12x64 step trigger sequencer in a very compact format.
It accepts hex digits 0123456789ABCDEFabcdef* an interprets one hex digit as 4 steps according
to the bit representation of the hex number. The original idea is from Steven Yi in his Csound live coding toolset.
The asterix (*) causes a hex char to be randomly choosen on every hit.
The only thing you must master to remember are the bits of the 16 digits:
0 0000
1 0001
2 0010
3 0011
4 0100
5 0101
6 0110
7 0111
8 1000
9 1001
A 1010
B 1011
C 1100
D 1101
E 1110
F 1111
The sequence is always looped according to length*4 of the input. So the input '8' makes a trigger per beat if clocked with 1/16, '2' makes the typical hihat, '08' makes the typical snare.
If the hex sequence is edited or changed it appers as red. You must press enter to activate it (makes it green).
As usual due to the one sample delay per cable, it is recommended always to feed the clock delayed by 2-10 samples so that the rst is triggered for sure before the clock.
New features in 2.0.4:
-
new key commands for active fields:
- TAB: moves focus a field down without saving
- UP: saves/activates the field and moves focus a field up
- DOWN: saves/activates the field and moves focus a field down
- p: randomizes the current active field, P: additionally activates/save
- r: rotates the current active field one bit right, R: additionally activates/save
- l: rotates the current active field one bit left, L: additionally activates/save
-
if the mouse is over the module the field 1-9 can be focused by pressing key 1-9
-
new Expander HexSeqExp
- The expander for HexSeq provides additional outputs:
- Gte: outputs constant 10V if the last bit was set, 0V otherwise.
- Clk: represents the incoming clock gate if bit is set
- Inv: outputs the inverted trigger signal i.e. fires if the current bit is 0
- NOTE: the expander can only be placed on the right side of HexSeq.
- The expander for HexSeq provides additional outputs:
-
Presets for learning purposes (Thanks to Andras Szabo)
-
Menus for randomization settings - Bit density, Length Range
-
Lights for the current Step - can be turned on/off in the menu
Thanks to Andras Szabo and Dave Benham for feature suggestions/requests.
HexSeqP is a 16x16x64 step trigger sequencer. Similar to HexSeq but with 16 selectable patterns.
The patterns can be selected via the pattern input in two modes (configurable in the menu):
- Poly Mode: The pattern is selected on a trigger on the respective polyphonic channel (left case in image below)
- Otherwise: The pattern is selected by the incoming voltage (0V-10V) (right case in image below)
For the description of the polyphonic outputs see HexSeq and HexSeqExp.
An internal incoming clock delay can be configured in the menu.
Frac is a monophonic function and hold module with a simple function. It devides two integer numbers N and P and outputs step by step their expansion for the given base (ranging from 2 to 16) scaled by the Scl Parameter and added to base volt value. The offset parameter causes the sequence to be started on an RST trigger from the given value.
Hopa is a monophonic function and hold module which outputs the sequence of the hopalong attractor for the given start values x0,y0 and the parameters a,b,c.
x=x0, y=y0
xnew=y-sqrt(abs(b*x-c))*sign(x)
ynew=a-x
If the CLK input is not connected it outputs the hopalong attractor as audio. In order to hear this interesting sound it is best to filter out the 10K band a much as possible and push the lower bands.
Some compact polyphonic matrix mixers with modulation.
- The inputs and outputs are polyphonic.
- The polyphonic channels of the modulation inputs are applied to the corresponding input.
- So there is only monophonic modulation per input and column.
- All inputs and outputs are full polyphonic.
- The polyphonic modulation inputs take 0-10V and scale the value to 0-1V. Then te value is multiplied with the knob value.
- All inputs and outputs are full polyphonic.
- Each modulation input has a cv attenuator and is added to the knob value.
- See X6.
This module is kind of similar to the unless towers module. It provides the following features:
- It has three fader blocks and three knobs
- There are 100 voltage addressable slots controlled by the Pat parameter and input
- There is a glide parameter which causes morphing from one slot to another
- On click on a fader the value according to the mouse position is immediately set.
- Adjustable voltage ranges {-10V,10V}, {-5V,5V}, {-3V,3V}, {-2V,2V}, {-1V,1V}, {0V,10V}, {0V,5V}, {0V,3V}, {0V,2V},{0V,1V} in the menu
- Polyphonic channels configurable in the menu
- Snap function: "None","Semi","0.1","Pat 32","0.5","Pat 16","1","Pat 8". If another value than None is selected in the menu, the output value is quantized according to the given step value. 'Pat X' denotes 10V/X and is for CV addressing.
- Each block can be randomized separately in the menu.
This module does the same as PolyCon from Bogaudio but
- It is thinner
- The channels and ranges can be set in the menu.
- The values stay the same if the range is changed and the values are still in the range.
- CPU friendly polyphonic modules for offset and scale cv values.
An ADSR with built in VCA. Designed for using with FM Operators.
This module applies a ratio to the incoming V/Oct signal. Designed for using with FM Operators.
AUX mixes a stereo polyphonic signal down to a stereo monophonic signal. It can either be used to sum a polyphonic signal and additionally attenuate single channels, or it can be used as 16 channel stereo mixer by merging monophonic stereo signals into a polyphonic signals. The values (0-10V) of the polyphonic Lvl input are divided by 10 and then multiplied with the corresponding parameter value. By using 10V Gates on the Lvl input single channels can be turned on and off. To avoid clicks there is a slew limiter processing the incoming level values.
A polyphonic Schmitt Trigger module
JTScaler tunes a 12edo V/Oct standard input to a just intonation scale.
The scales selectable via the scale knob are defined in res/default_scales.json in the dbRackModules plugin directory. You can copy this file to scales.json in the same directory and make your own version of it. If it can be read without errors it will be taken instead otherwise it falls back to default_scales.json (which should never be changed)
GenScale a polyphonic 16 note scale defined by the toggled buttons.
Interface is a utility to define an input/output interface for a group of modules or saved selections. It does not apply anything to the inputs.
Plots a polyphonic 2D Signal.
- CVS has 16 polyphonic inputs and the polyphonic inputs P1-P4.
- the channel Y of PX is used for input number Y channel X (vertical assignment) if the input Y is not connected.
- The CV input can address continuously the 16 inputs in the range from 0-10V
E.g. 0.625 would output input1,
0.625+0.625/2 = 0.9375would outputinput1*0.5 + input2*0.5. - The address value is always taken by modulo Len. So if e.g. Len = 2 0.125 would output input1
- The trig output is high (10V) whenever the output changed. This seems to be very complicated but the following examples show some easy understandable usecases:
P16 delivers exactly the values for addressing the inputs (no interpolation).
In this example the rndh delivers arbitrary values between 0-10V. So the inputs are interpolated.
Here a lfo is used to scan 4 pulsars (continuously)
In this example rndc with 4 channels scans each note of the chord differently.
- a simple cv change detector
- if the window is 0 the output is high if the input changes (compared to the last sample).
- otherwise the output is high if the current sample is in the interval last-window, last+window.