WREN module documentation

Audio input module, connects physical audio inputs to the patch.

This is a MIDI note input module.

The ch selector sets the MIDI channel to listen to. OMNI means that it listens to all MIDI channels, and ch 1 - 16 are the usual MIDI channels.

When a note on is seen the gate output will go on and the note number will be put on the 'note' output. The velocity will go to the 'velocity on' output.

When a subsequent note off is seen for the currently active note the gate will turn off, and the note off velocity will be set on the 'velocity off' output.

Note on messages with a velocity of zero are treated as note offs (with a note off velocity of zero).

When the channel is changed the current note will be turned off (with zero note off velocity), and the same will happen when a MIDI all notes off command (CC 123 with value 0) is seen or when the 'Panic' function is used (Action menu / Panic or ctrl+P).

This is a MIDI four note input module.

This module is superceded by the Midi8NoteIn module which uses the same internal coded with the additionl flexibility of setting a limit to the actual amount of notes to handle. For this module the channel count is four and that is fixed.

The ch selector sets the MIDI channel to listen to, OMNI means that it listens to all MIDI channels, and ch 1 - 16 are the usual MIDI channels.

When a note on is seen the gate output will go on and the note number will be put on the 'note' output. The velocity will go to the 'velocity on' output. When a subsequent note in arrives it will be routed to the 2nd output set, etc.

When a subsequent note off is seen for the currently active note the gate will turn off, and the note off velocity will be set on the 'velocity off' output.

Note on messages with a velocity of zero are treated as note offs (with a note offf velocity of zero).

When the channel is changed the current notes will be turned off (with zero note off velocity), and the same will happen when a MIDI all notes off command (CC 123 with value 0) is seen or when the Panic function is used (menu Action / Panic, or ctrl + P).

The 'note stealing' mode comes in effect when there are four active notes and a new one comes in, the different modes act like:

• none : no note stealing - find first free, if none free, skip the new note.

• cyclic : round robin, just use the next note slot regardless of how long it was on.

• softest : replace the softest note. This needs the level input, connect this to an envelope for instance. When no such note was found this mode acts like the cyclic mode.

• oldest : replace oldest note, this will replace the note that was active for the longest time, when no such note was found this mode acts like the cyclic mode.

• lowest : replace lowest note, the note with the lowest note number is replaced, when no such note was found this mode acts like the cyclic mode.

• highest : replace highest note, the note with the highest note number is replaced, when no such note was found this mode acts like the cyclic mode.

• newest : replace newest note, the note that was turned on last will be replaced, when no such note was found this mode acts like the cyclic mode.

This module handles incoming pitch bend messages and after touch.

This is a MIDI CC input module.

The ch selector and the cc selector set the CC event to listen for. When ch = OMNI it will match all MIDI channels, values 1 - 16 to the corresponding MIDI channels.

When a new CC event is detected the gate input will go on for a short time, and the value will be put on the value output. The value did not necessarily change when the gate input becomes active, it reacts to a matching incoming CC message.

The incoming value range [0,127] is mapped to an output range [0,1].

This is a MIDI note output module.

The ch selector sets the MIDI channel to send to, where ch 1 - 16 are the usual MIDI channels and ic 1 to 16 are internal MIDI channels (MIDI receivers in a patch can just use ch 1 - 16 (or OMNI) to connect to these).

The note input determines what note value will be sent, the velocity inputs determine the on or off velocity to use.

Each time when the gate input changes from inactive to active a note on event is sent with the currently set note on velocity.

Each time when the gate goes from active to inactive a note off event is sent for the note that that was previously turned on, and this event will be sent to the same MIDI channel that the note on had been sent out (even when the set MIDI channels changed in the mean time). For this event the current note off velocity will be used. When no note on event had been sent there will not be a note off event either.

When the note value changes no note event will be sent, only changes on the gate input can cause a note event to be generated (*).

When the channel is changed no note event will be sent out, only changes on the gate input can cause a note event to be generated (*).

(*) This will not result in hanging notes as when the gate goes inactive the channel and note number for the active note will be used to base the note off on.

This is a MIDI Program Change output module.

The ch selector and the prg selector define the program change event to be transmitted, ch 1 - ch 16 to the correspond to to the usual MIDI channels, ic 1 - ic 16 to the internal MIDI channels (MIDI receivers in a patch can just use ch 1 - 16 (or OMNI) to connect to these).

When the gate input goes from inactive to active the event will be sent out using the SUM of the current values as set by the prg control and read from the progr. input.

When the ch or the progr input change there will not be a new MIDI event generated - MIDI events occur only when the gate goes from inactive to active, or when the prg control is changed manually.

The sent output will go active after the MIDI message was sent, this can be used to trigger a chain of events to happen in a specified order.

A MIDI SysEx output module.

NOTE: this module is not working yet.

A sequence of numbers and parameters can be filled in from which a MIDI SysEx message will be computed.

The parameters are the eight inputs which can be used as %1 .. %8.

On the rising edge of the gate input the parameters will be read in and will be filled into the message which will then be sent out over MIDI.

NOTE: this module is not working yet.

Send or receive OSC messages.

This module can both send and receive OSC messages, depending on the role chosen for this Wren instance.

For an OSC transmitter it will send the value present on the blue (send) input when the green input goes from low to high to the address present in the address box. In this case the value will also be copied to the output (so this doubles as a slow rate sample and hold).

For an OSC receiver it will listen to OSC type f messages on the address specified in the address box and it will send the value found in the message to the output. Note that when you you clock in input values as well the output value may toggle between the OSC received value and the clocked in value.

To set an address control click the address field to pop up a text editor, and when done use ctrl+enter to save the edit. In the address the leading slash (/) character must not be specified, Wrenn will add it by itself.

Each address can be used only once, when it occurs multiple times an arbitrary instance of the module will handle it and the other instances will think it to be handled already after that. There can be multiple senders on the same address for a Wren instance that is set to be an OSC transmitter though.

Messages may be sent to Wren with more than one parameter, however only the first parameter will be used and only when it's type is 'f' (float).

Note: OSC address strings are case sensitive, so 'module' does not match 'Module'.

Note: All Wren modules can receive OSC messages when their name starts with an @ character.

A controller to trigger auto-randomize, auto-mutate, auto-mate or auto-morph.

Recorder

A 1 to 8 channel audo recorder that can write it's recording to a wave file on disk.

The tape length can be set from 1 to 30 minutes and when the tape is full recording will continue writing over old material.

The record control starts or stops recording, the write control starts a disk write (for which a file has to be selected before) and the clear control will erase the tape.

Write, clear and record can be controlled by the buttons as well as by logic input signals.

The rec on ouput is active when recording is on.

The rec of output is active when recording is OFF, such that when it is connected to the clear and write inputs the wave gets saved and cleared when recording is stopped.

The full output goes active when the tape loop is fully recorded, such that when connected to the clear and write inputs the wave gets saved when the loop is full.

In the file name %patchname% can be used, this will be replaced by the name of the patch. When you patch is C:\wren\patches\silly-putty.wren the patch name will be silly-putty, and the wave data will be written to C:\wren\recordings\cc-silly-putty.wav.

When 'time stamped' is being selected the file name will be prefixed by a time stamp.

When 'normalized' is selected the wave data will be normalized before it is being written out to the file.

When the file name is clear, the box will show 'Ctrl click to select a file' in that case data will not be written to disk at all, there will be no prompt for a file to save to.

Longest recording times possible are:

An audio rate square wave oscillator.

The freq input is an unattenuated FM input with exponential control.

The FM input provides exponential frequency control which can be attenuated.

The PM input provides linear phase control.

The sync input resets the oscillator to phase zero when it's value changes from zero or less than zero to a value above zero.

The PWM input provides linear Pulse Width Modulation.

The mute button sets the output to zero when active

With the B-Limit control the amount of aliasing can be controlled.

The speed input provides linear trough zero FM. When the input level is 1 the oscillator will run twice as fast as it normally would, when zero the oscillator will stop, and on negative speeds it will run backwards.

An audio rate multi waveform wave oscillator.

This module simultaneously makes a sine-, a trianglar-, a saw- and a square-waveform.

The freq input is an unattenuated FM input with exponential control.

The FM input provides exponential frequency control which can be attenuated.

The PM input provides linear phase control.

The sync input resets the oscillator to phase zero when it's value changes from zero or less than zero to a value above zero.

The PWM input provides linear Pulse Width Modulation (on the square output only).

The mute button sets the output to zero when active

With the B-Limit control the amount of aliasing can be controlled.

The speed input provides linear trough zero FM. When the input level is 1 the oscillator will run twice as fast as it normally would, when zero the oscillator will stop, and on negative speeds it will run backwards.

An audio rate phase distorted sine oscillator.

Phase distortion modulation is very similar to VOSIM.

The freq input is an unattenuated FM input with exponential control.

The FM input provides exponential frequency control, the amount can be controlled by an attenuator.

The LFM input provides linear frequency control, the amount can be controlled by an attenuator.

The sync input resets the oscillator to phase zero when it's value goes from zero or below zero to above zero.

The mute button sets the output to zero when active.

A modulatable frequency fine control is available which can change the tuning by 50 cents up or down.

The speed input provides linear trough zero FM. When the input level is 1 the oscillator will run twice as fast as it normally would, when zero the oscillator will stop, and on negative speeds it will run backwards.

A Karplus-Strong type of oscillator.

The freq input is an unattenuated FM input with exponential control.

The FM input provides exponential frequency control, the amount can be controlled by an attenuator.

A frequency fine control is available which can change the tuning by 50 cents up or down. This setting can be modulated.

The Pluck control controls the amount of input signal (excitation).

The brightness control changes the frequency response of the feedback filter. Brightness can be modulated trough the bmod input and the bmodamt control.

The feedback control controls the amount of feedback. The modulation (fbmod) is added to the main setting with an amount controlled by a depth control (fbmodamt).

The mute button sets the output to zero when active

The 'interp.' selector allows selection between linear, all-pass, algrange and hermite interpolation modes, all with a slightly different sound quality and some responding better on frequency changes.

The module expects impulse like excitation input signals. Other signals work as well, but things may get very loud - use the pluck level controls to compensate.

An audio rate triggered sine, tri, saw and square wave oscillator

The FM input provides exponential frequency control, the amount is under control of an attenuator.

When the trig input changes from zero or less than zero to a value above zero a preset amount of half waves will be sent to to the output. This can be used for VOSIM like synthesis or as an envelope control.

The mute button sets the output to zero when active

A frequency fine control is available which can change the tuning by 50 cents up or down.

With the B-Limit control the amount of aliasing can be controlled (for saw, try and square waves).

The speed input provides linear trough zero FM. When the input level is 1 the oscillator will run twice as fast as it normally would, when zero the oscillator will stop, and on negative speeds it will run backwards.

An audio rate noise generator.

The color control determines the rate of change of the output signal. Set to zero the output freezes and set to max the output will change at it's fastest rate.

Color can be modulated trough the color mod input.

The mute button sets the output to zero when active.

An audio rate Perlin noise generator.

The new input, when active, causes a new noise sample to be set after the current noise sample was played.

The roughness set ... the roughness of the noise, this can be modulated at audio rate.

The layers knob sets the number of layers, each successive layer is a smoothed version of the one above, so the more layers thee are the smoother the noise can be, at the price of some computation overhead.

The length knob sets the length of the current noise pattern, the higher it is set the less tone there there will be in the generated output.

An audio rate Perlin noise generator.

The new input, when active, causes a new noise sample to be set after the current noise sample was played.

The roughness set ... the roughness of the noise, this can be modulated at audio rate.

The layers knob sets the number of layers, each successive layer is a smoothed version of the one above, so the more layers thee are the smoother the noise can be, at the price of some computation overhead.

The length knob sets the length of the current noise pattern, the higher it is set the less tone there there will be in the generated output.

An audio rate strange attractor module.

The freq input is an unattenuated FM input with exponential control.

The FM input provides exponential frequency control, the amount can be controlled by an attenuator.

PadSynth

A PADSynth like module bused on ideas from Nasca Octavian Paul, see: http://zynaddsubfx.sourceforge.net/doc/PADsynth/PADsynth.htm for more on this.

This module use IFFT to generate sound from a spectrum as drawn by the faders and influenced further by the other controls.

The bw / bw-scale controls manipulate the width of the spectral peaks.

The spread and spread type controls manipulate the distance between the peaks.

The module works by generating an in-memory waveform from the spectrum. This wave can then be played at different rates by using the position, speed and frequency controls.

Some of the controls will be clicky as as the IFFT must be recalculated when their positions change.

The module provides its stereo output by using two different tap points on the generated wave.

An 11 channel oscillator module.

The oscillator frequenciies are computed from Chladni's law f = C * ( m + 2n) ^ p where p can be controlled trough an input and n + 2m are distributed over the 11 channels. C is the base frequency.

A triggered wave player

This module can read a wav file from disk into memory. The wav format needs to be 16 bit, 44.1 kHz - stereo and mono are both fine. The maximum file size is not limited by the module, but as the wave has to be read into memory for very large files you may run out of memory resulting in things to get very slow. Recordings with a length of several minutes should not be a problem.

The speed input controls how fast the wave will be played, and the wave will be played backwards for negative values (or stand still when the speed is zero). When the FM input is connected too the speed will be multiplied by the FM amount (and speed must not be zero).

The speed input can be thought of as a linear FM control while the position input can be seen a sa linear PM control.

When a stereo wave is loaded the mono stereo switch will convert the selected wave to mono when set as mono, for a mono wave the left and right outputs will always be the same.

Control click the filename field to pop up a file open dialog.

The FM input provides exponential speed control, for it to work the speed input must be set to something unequal to zero - which can be done by having the on-off control as on, or by connecting something unequal to zero to the speed input.

Wave playing is started from the trigger input, as determined by a window set from trigger level to trigger level plus trigger range. Whenever the trigger input moves into that window a trigger condition is seen and whenever it moves out of that window the trigger condition is removed.

Depending on the trigger mode selected:

• normal: Audio playing will start when a trigger condition occurs, and the full wave will be played then. When the loop mode is set to once the trigger input will then have to first move out of the trigger window before a new trigger can occur. When the loop mode is set to loop wave can start anew without first having to go outside the trigger window.
• retrig: Wave playing can be restarted before it was fully played, which for fast retriggers will lead to granular like effects (except that no fade out is being performed).

A Nord Modular Classic like master oscillator.

This ooscillator does not produce any sound, but it makes a control signal to drive slave oscillators from. In Wren any oscillator having a speed input can be used as a slave oscillator. The idea is to connect the speed output of the master oscillator to the speed input of other oscillators. For the Nord Modular Classic these would be the 'grey signals'.

The freq input is an unattenuated FM input with exponential control.

The FM input provides exponential frequency control, the amount can be controlled by an attenuator.

The Speed input provides linear FM, the lfm input linear FM modulation. For normal speed the control should be set to 0.5.

A frequency fine control is available which can change the tuning by 50 cents up or down. This input can be modulated.

This is an audo rate lookup module, basically. The input value moves a cursor over the horizontal axis and the looked up graph height is then sent to the output. The input wants a positive only signal, the output is bipolar.

Points can be added by double clicking (or with ctrl+click) in an empty area. The same action on an existing point will allow that point to be moved or removed - removing being a move to somewhere out of the graph area.

The leftmost and rightmost points can not be removed, but can only be moved up or down.

This module is better suiatable for audio rates than the DataGraph module, it uses Differential Parabolic Waves internally to suppress the aliasing somewhat.

A right click on the graph area will popup a context menu allowing for some standard wave forms to be created, basic symmetry operations and save to and load from file.

The Phasor module was made to control this module.

the graph on this module uses auto-looping (first and last point will be set to the same Y value) and can optionally use auto-scaling (the full Y range will always be used).

The rnd input, when going from false to true, will set new random Y values for all the points in the graph. The amount of randome being applied is the momentary analog value of the rnd input.

A wave player module

This module can read a wav file from disk into memory. The wav format needs to be 16 bit, 44.1 kHz - stereo and mono are both fine. The maximum file size is not limited by the module, but as the wave has to be read into memory for very large files you may run out of memory resulting in things to get very slow. Recordings with a length of several minutes should not be a problem.

The speed input controls how fast the wave will be played, and the wave will be played backwards for negative values (or stand still when the speed is zero). When the FM input is connected too the speed will be multiplied by the FM amount (and speed must not be zero).

The position input determines which sample will be sent to the output. This can be used with a sawtooth LFO to play the wave.

The speed input can be thought of as a linear FM control while the position input can be seen a sa linear PM control.

When a stereo wave is loaded the mono stereo switch will convert the selected wave to mono when set as mono, for a mono wave the left and right outputs will always be the same.

Control click the filename field to pop up a file open dialog.

The FM input provides exponential speed control, for it to work the speed input must be set to something unequal to zero - which can be done by having the on-off control as on, or by connecting something unequal to zero to the speed input.

A control rate square wave oscillator (LFO).

The FM input provides exponential frequency control.

The PM input provides linear phase control.

The res input resets the oscillator to phase zero when it's value changes from zero or less than zero to a value above zero.

The PWM input provides linear Pulse Width Modulation.

The mute button sets the output to zero when active.

The sync output is active around the start point of the wave.

The output type selector sets the output to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

The speed input provides linear trough zero FM. When the input level is 1 the LFO will run twice as fast as it normally would, when zero the LFO will stop, and on negative speeds it will run backwards.

A control rate noise generator (LFO).

The color control determines the rate of change of the output signal, set to zero the output freezes and set to max the output will change at it's fastest rate. Color can be modulated trough the color mod input.

The mute button sets the output to zero when active.

The output type selector sets the output to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

A set of smooth control rate random generators, they linearly interpolate from the current to the next output value (LFO).

The random mode can be set to to lin or exp, when set to exp the lambda control sets the jumpiness, i.e. the amount of variation between successive target values.

The final random value is smoothed by the distribution control before it is used, distribution can be modulated.

The output type selector sets the output to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

Internal feedback is provided for FM and DISTribution, the amount for each can be set sepately and the amounts can be modulated.

The R outputs provide a none cross-faded signal while the X outputs can be set to cross fade between the current and next output value.

The level value determined from the lev. knob multiplied by it's modulation input will be multiplied by the internal signals, the signals will then be clipped to be in a range [-1.0,1.0] before the output type scaling is being prformed. For high levels the outputs will spend more time at the extreme values.

When the calculated level is below the 'dead level' the output will be zero.

The s&h input supports the special NOT_CONNECTED status.

The correlation (cor) sets the similarity for the output sets. The topmost output will always be the inversion of the bottom one though - correlation is between the 'channels' out(inv)1, out(inv)2, etc. The higher the correlation the the more similar the channels will be. This value can be modulated with the signal input left of it (corrmod). When the correlation is zero all channels will be totaly independant, when it is one all channels will be the same.

A control rate brownian noise, or random walk, generator.

The FM input provides exponential frequency control.

The mute button sets all outputs to zero when active.

The Color control determines the rate of change of the outputs.

This module works by moving a virtual dot in a a virtual cube [(-1,-1,-1),(1,1,1)]. At every clock beat the dot will be moved over a random distance smaller than set by the sum of the distance parameter and the distance modualation input. When the dot hits a wall it will bounce off. The dot position is updated at the rate set by the frequency control.

The random mode can be set to B (linear/brownian), E (Exponential) or L (Levy flight). When set to E or L the Lamba control determines the jumpiness, turn right for larger jumps.

The output type selector sets the outputs to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

The sync output goes active when the ouput values change and it will go inactive again halfway the LFO period.

An LFO rate strange attractor module.

The output type selector sets the output to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

A control rate triggered sine, tri, saw and square wave oscillator (LFO).

The FM input provides exponential frequency control, the amount being controlled by an attenuator.

When the trig input changes from zero or less than zero to a value above zero a preset amount of half waves will be sent to to the output. This can be used for VOSIM like synthesis or as an envelope control.

The mute button sets the output to zero when active.

The rdy output becomes active when the module is ready for a new trigger signal.

The output type selector sets the output to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

The mode selector can set the the module into normal or retrig mode. Together with a graph module and when set to a saw shaped wave this will allow for arbitrary envelope shapes.

The speed input provides linear trough zero FM. When the input level is 1 the LFO will run twice as fast as it normally would, when zero the LFO will stop, and on negative speeds it will run backwards.

A control rate triggered noise generator (LFO).

When the tr input changes from zero or less than zero to a value above zero a new random value will be sent to to the output.

The color control determines the rate of change of the output. When it is set to zero the output will be frozen, when it is set to max the output values will change most from sample to sample.

The mute button sets the output to zero when active.

The output type selector sets the output to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

ClockGen2, a Clock generator.

This module was especially made for the DataGraph module, the two can make a versatile sequencer together.

The bars control sets the number of bars to be used, that is the number of time divisions to be made in the DataGraph module (which got an input to control its horizontal divisions).

Bars/m or bars per minute sets the play rate for the bars - or the number of divisions per minute to be played in the DataGrap. This sets the module's cycle rate and it is output through the sin, tri, saw and sq(uare) outputs.

The saw output is always affected by the phase controls, the sin, tri and sq ouputs can be switched on or off for this effect (with the samll button below the outputs). The signal type to be used is set for all four with the same control.

Quantization is a rate multiplied version of of the cycle, it is meant to be connected to the DataGraph's quant input and it determines the quantization times to be used - or the note granularity to be used. This is a square wave output and it can have swing (as set by the swing controls) applied to it.

Accents is a rate divided down version of the quantize signal, it can be used to apply accents - it is not involved with the DataGraph module.

There is an overall speed control input for dynamically changing the speed.

The res input provides for a means to sync the module to other signals.

And the active input switches the module between active and inactive, it is xor-ed with the ON / off button. When the module goes from inactive to active it will be reset as well.

There are modulation inputs for Phase, Swing, Bars/m, quantization and accents - all are multiplicative and are 1.0 when not connected.

An audio rate sine, tri, saw, square, random and random-square wave oscillator (LFO).

This is an LFO module running at audio rates, it allows for smoother control of delay line based effects.

The FM input provides exponential frequency control, the amount is under control of an attenuator.

The PM input provides linear phase control, the amount being controlled by an attenuator.

The res input resets the oscillator to phase zero when it's value changes from zero or less than zero to a value above zero.

The mute button sets the output to zero when active.

The sync output is active around the start point of the wave.

The output type selector sets the output to either:

• Full range
• Full range - inverted signal
• Positive only
• Positive only - inverted signal
• Negative only
• Negative only - inverted signal

The speed input provides linear trough zero FM. When the input level is 1 the LFO will run twice as fast as it normally would, when zero the LFO will stop, and on negative speeds it will run backwards.

A retriggerable Attack Release type Envelope.

The envelope will start when the trig input goes from zero, or below zero, to above zero. When the envelope period was not finished when a new trigger comes in the Attack phase will start from the value where the output was at that moment.

The Envelope amplitude is set by the gain input, when this input is not connected the gain will be 1.

The leftmost output will output the envelope signal as is, while the rightmost output will apply the envelope to the signal present at the input above it.

The Attack and Release shapes can be set to Log, Lin, Exp or S type.

The mute button will set all outputs to zero when active.

A and R time modulation is present, the modulation values are added and / or subtracted to / from the preset times.

With the Inh(ibit) input the envelope can be disabled (when it's value is >= 0.5).

The A, R and H outputs go active when the A, R or H state is active, the EOC output goes active when none of the A, R, or H states is active.

The A/D mode knob selects between the usual digital trigger mode and a couple of analog mode. In analog mode the trig input is level controlled, where the level is being quantised to levels set by the 2, 3, 4, 5, 6 and A modes and whenever that quantised value changes the envelope is triggered. The modes 2, 3, 4, 5 and 6 indicate the number of levels used, while the A mode has an 'infinite' number of levels.

An Attack Decay Sustain Release type envelope generator.

The attack, decay and release types can be set to Linear, Logarithmic or exponential.

The timing ranges can be set as fast, medium or slow.

The trigger type can be set to gate or trig mode. In gate mode the envelope period needs to be finished before it can restart, in trig mode it can restart at any time.

The envelope will start when the trig input goes from zero, or below zero, to above zero.

The Envelope amplitude is set by the gain input, when this input is not connected the gain will be 1.

The leftmost output will output the envelope signal as is, while the rightmost output will apply the envelope to the signal present at the input above it.

The mute button will set all outputs to zero when active.

For the Attack, Decay and Release phases the mode can be set to exponential, linear, logarithmic or s-curved.

All the times and the sustain level can be modulated.

The EOC (End Of Cycle) output goes active when the full envelope period is finished, or when a period nver occured. Fed back into the trigger input sustained self oscillation will take place.

The swell controlls can modify the behaviour of the sustain period.

The A/D mode knob selects between the usual digital trigger mode and a couple of analog mode. In analog mode the trig input is level controlled, where the level is being quantised to levels set by the 2, 3, 4, 5, 6 and A modes and whenever that quantised value changes the envelope is triggered. The modes 2, 3, 4, 5 and 6 indicate the number of levels used, while the A mode has an 'infinite' number of levels.

A retriggerable envelope controller.

The envelope will start when the trig input goes from zero, or below zero, to above zero. The output will then in the set Attack time go to the level set with the level control. When the trig input falls off the output will start getting larger again for the set Release time, till a value of 1 is reached.

At every new trig start the ouput value is reset to 0, and the above cycle will restart.

Both the Attack and Decay phase are linear.

This module is meant to control the GraphControl module, the actual envelope shape is then drawn in that module. The set level will then correspond to the vertical distance in the graph where the hold value will be.

The EOC (End Of Cycle) output goes active when the full envelope period is finished, or when a period never occured. Fed back into the trigger input sustained self oscillation will take place.

A 17 channel envelope follower / multiplier module.

Meant to be used with the splitter and combiner filters etc. to do vocoder like effects.

Unused multiplier (blue) inputs default to a value of 1.

A non-retriggerable Attack Release type Envelope.

The envelope will start when the trig input goes from zero, or below zero, to above zero. The envelope can thereafter only be restarted when it's cycle is completely finished.

The Envelope amplitude is set by the gain input, when this input is not connected the gain will be 1.

The leftmost output will output the envelope signal as is, while the rightmost output will apply the envelope to the signal present at the input above it.

The attack phase is linear only and the release phase is exponential only.

The mute button will set all outputs to zero when active.

A and R time modulation is present, the modulation values are added to the preset times.

With the Inh(ibit) input the envelope can be disabled (when it's value is >= 0.5).

A DC blocking filter.

For audio rates (red mode) the -3dB point can be set from approximately 10/20 Hz to 400/800 Hz (depending on sample rate). In control rate (blue) mode this ranges from approximately 1.25/2.5 Hz to 50/100 Hz (depending on sample rate).

A moving average type of filter.

The filter's alpha parameter is fixed.

The LP output is calculated as:

  Delta           := Inp  - Outp;
  Outp            := Outp + Alpha \* Delta;
  FOutputs[ o_lp] := Normalize( Outp );

This filter is meant to go after a sequencer and before the FM input of an oscillator to filter out fast changes in frequency a bit.

A moving average type of filter.

The knob controls the kernel size, or the number of samples the averaging is computed over.

This filter is most useful for making a lag or portamento type of effect, it can smooth out jumps in the input value.

A 6 dB / Octave band pass filter.

The high pass and the low pass filter are cascaded and can be set independently.

This filter can work at both control and audio rates, but at control rate the frequencies are eight times lower than what is being shown.

A 6 dB / Octave stereo band pass filter.

The high pass and the low pass filter are cascaded and can be set independently. The high cut frequency can be modulated.

This filter can work at both control and audio rates, but at control rate the frequencies are eight times lower than what is being shown.

A 24 dB / Oct moog like low pass filter..

The fm input controls the cutoff frequency, it is exponentially scaled and the FM amount is controlled by an attenuator.

The Q control sets the amount of resonance. The Q value can be modulated trough both a + and a - input.

This filter has some built-in non-linearities, including soft clipping on the output. To avoid soft clipping on the output one can reduce the input level.

There are several filter types implemented, the original (and still default) one is / was Huovilainen, the others are Improved, Krajeski, MicroTracker, MusicDSP, OberheimVar, RKSimulation and Stilson. They are all simulations of a ladder filter, for more information see ddiakopoulos / MoogLadders.

The copyright messages are:

• Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
• Copyright (c) 2015, Miller Puckette. All rights reserved. (for the RKSimulation model).

Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met:

• Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer.

• Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution.

THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

Selfoscillating State Variable Filter with additional 24dB stage, combined 12dB/24dB outputs + individual 12dB LP/BP/HP/BR outputs

Freq ... Cutoff frequency

Q ... Resonance

Input + Gain ... Filter input

Drive ... Input drive amount (0 = no drive, 1 = total clipping)

FM Input + Amount ... Cutoff frequency modulation input (exponential)

QM Input + Amount ... Resonance modulation input

LP ... LowPass filter amount

BP ... BandPass filter amount

HP ... HighPass filter amount

12dB ... Combined 12dB LowPass/BandPass/HighPass output level

24dB ... Combined 24dB LowPass/BandPass/HighPass output level

Outputs:

LP/BP/HP/BR ... Individual 12dB LowPass/BandPass/HighPass/Notch outputs

12dB ... Combined 12dB LowPass/BandPass/HighPass output

24dB ... Combined 24dB LowPass/BandPass/HighPass output

Notes:

Individual outputs are always available and they don't depend on LP/BP/HP amounts or 12dB/24dB levels.

To get notch on 12dB/24dB outputs set LP=1, BP=0 and HP=1.

For selfoscillation some signal must be present at input. For example, set input gain to 0.01 and send some noise to the input.

Be careful with very high resonance values (over 0.9) when using 24dB output.

A differentiator.

This module outputs the difference between the previous input sample and the current one.

There is a mode selector for how the differentiator output behaves. When set to transparent it will output both positive and negative differences. When set to half wave it will output negative differences as zero, while letting thru positive differences. When set to full wave mode the absolute value of the difference will be output. This can be used for example to trigger something when a signal goes up, or when it changes.

The sel input is scaled up and added to the mode selector and then the result taken modulo 3 to determine the actual mode to be used.

A resonating filter

The filter expects impulse like input signals and then will create a damped oscillation from that.

Tilt filter

This uses a modulatable center frequency (F0) and then boosts one of the ranges above or below F0, while doing the opposite with the other range, with a modlatable amount.

http://www.musicdsp.org/showone.php?id=267

A fixed half-octave (tritone) filter bank module.

The filter steps are set to be fixed, 6 semi tones apart (tritone). The labels show rounded values for the frequencies the 40 Hz label is exact.

The low filter can be set into LowPass or BandPass mode.

Thehigh filter can be set into HighPass or BandPass mode

All other filters are bandpass filters.

There is an overall input level control.

The filter Q can be set.

The filter steepness can be set from 1 to 4, where 1 corresponds to a 12 dB filter in low or highpass modes, or a 6 dB filter in band pass mode. Each step up adds an extra filter layer in series - so a steepness of 4 adds up to 68 filters being used.

A fixed half-octave (tritone) frequency splitter module.

The filter steps are set to be fixed, 6 semi tones apart (tritone). The labels show rounded values for the frequencies the 40 Hz label is exact.

The low filter can be set into LowPass or BandPass mode.

The high filter can be set into HighPass or BandPass mode

All other filters are bandpass filters.

There is an overall input level control.

The filter Q can be set.

The filter steepness can be set from 1 to 4, where 1 corresponds to a 12 dB filter in low or highpass modes, or a 6 dB filter in band pass mode. Each step up adds an extra filter layer in series - so a steepness of 4 adds up to 68 filters being used.

Each filter chain has a separate output.

A clipping indicator is provided on each output.

A fixed third-octave (major third) frequency splitter module.

The filter steps are set to be fixed, 6 semi tones apart (tritone). The labels show rounded values for the frequencies the 160 Hz label is exact.

The low filter can be set into LowPass or BandPass mode.

The high filter can be set into HighPass or BandPass mode

All other filters are bandpass filters.

There is an overall input level control.

The filter Q can be set.

The filter steepness can be set from 1 to 4, where 1 corresponds to a 12 dB filter in low or highpass modes, or a 6 dB filter in band pass mode. Each step up adds an extra filter layer in series - so a steepness of 4 adds up to 68 filters being used.

Each filter chain has a separate output.

A clipping indicator is provided on each output.

A fixed third-octave (major third) frequency combiner module.

The filter steps are set to be fixed, 6 semi tones apart (tritone). The labels show rounded values for the frequencies the 160 Hz label is exact.

The low filter can be set into LowPass or BandPass mode.

The high filter can be set into HighPass or BandPass mode

All other filters are bandpass filters.

There is an overall input level control.

The filter Q can be set.

The filter steepness can be set from 1 to 4, where 1 corresponds to a 12 dB filter in low or highpass modes, or a 6 dB filter in band pass mode. Each step up adds an extra filter layer in series - so a steepness of 4 adds up to 68 filters being used.

Each filter chain has a separate input.

A clipping indicator is provided on each input.

An 11 channel modal filter.

The filter settings are loaded fome a text file, wren.modal, and this file has to be located in the Wren application directory.

The default file as distributed with Wren has some instructions in it on how to use it, it can be user modified.

The Q settings and the level settings as determined fom the definition file can be altered with the a and q knobs of the module.

The frequency settings can be modulated with the freq, fm and lfm parameters, the latter being linear FM.

Also there is a modulatable overall Q setting changing the relative channel Q's simultaneously.

The filter preset can be programmatically changed as well.

A fixed filter bank

The lowest frequency is 35Hz the highest one is just under 12kHz

The stereo mode sends the odd channels left and the even ones right, with the knob a mix can be set between full mono and full stereo.

The Q can add or remove resonance.

The lev control sets the input level.

The on/byp switch selects between the effect being on and bypass mode.

The slopes can be either 6dB or 12dB (12dB or 24dB bandpass filters).

A Chainable stereo volume controller.

When a mute is active the module will zero the associated mix outputs.

The attenuation can be set with the level knob and can be modulated trough the mod(ulation) input by an amount set by the mod(ulation) amount control.

A cross fase module.

When the ctrl input is -1 in1 will be sent to the output without mixing any of in2 in. When the ctrl value is 1 this is reversed. For inbetween values of the ctrl signal the output will have a mix of in1 and in2.

The input type selector determines what input swing will be used for a full left to right control based on the direct position control being in the mid position.

A panning module.

When the ctrl input is -1 the in signal will be fully panned tou out1. When the ctrl value is 1 it pans the input to out2. For inbetween values a part of the ipnut signal goes to oit1 and a part 1 - part goes to out2.

The input type selector determines what input swing will be used for a full left to right control based on the direct position control being in the mid position.

A Chainable 4 to 1 mono mixer.

When a mute is active the module will zero the associated mix input

A Chainable 5 to 1 stereo mixer.

When a mute is active the module will zero the associated mix input

Making solo active on the left or right pair will block the audio chain inputs and and it will also block the other audio input pair. Also the solo chain output will then be active which is signalled by a yellow light. When the solo output is connected to a downstream MixS2to1 module that module will block it's local inputs and it will pass on only it's chain inputs. This will be signalled on that module by having an orange solo light.

A Chainable 1 to 1 four channel mixer

When a mute is active the module will zero the associated mix input

A 16 channel mono mixer with modulation control on each fader and output mute.

A fully chainable 6 to 6 mono mixer with gain control on each cross point.

When a mute input is active the module will zero that input for all output columns.

Quad stereo 2 to 1 mixer.

There are four channels each having two stereo inputs with level control. Inputs and outputs can be muted independently.

Quad panner

Four panners in one module. The panning works with a square root panning law.

A Track and Hold module.

The output will follow the input when the trig input is zero or below zero, and it will freeze the output when it goes above zero.

When the trig input is not connected the module is transparent.

The trig input supports the NOT_CONNECTED signal.

A Multi Sample And Hold module.

The outputs will hold the last input values seen when the trig input goes from zero, or below zero, to above zero.

When the trig input is not connected the module is transparent.

The trig input supports the NOT_CONNECTED signal.

A crossfading or holding de-multiplexer

The control input selects to what output the input signal will be sent. With the x-fade control the amount of overlap can be set and in hold mode there is no cross fading but outputs will hold their values when they get deselected.

A digitally controlled de-multiplexer

In Norm(al) mode a deselected output will go to zero, in hold mode a deselected output will keep the last value it had.

A Value Sync module

The module gets triggered by an incoming positive edge on the pulse input, which it will then hold internally along with the value that was present at that time on the value input. When a subsequent positive edge appears on the sync input the Q output will be set active and the captured input value will be passed to the output. Once the pulse got processed the ouput and the internal state are both being cleared when a negative edge is seen on the sync input. The incoming pulse is guaranteed to never get lost.

This modue can be used to sync up a pulse event appearing at a random time to a syncing clock signal while capturing a value that was present at that time and passing it on on a sync only.

This module acts a bit like a sample and hold in that it captures a value on a trigger, but it will delay passing on that value until a sync signal is seen.

A scaler module.

Each input will be multiplied by the set gain and then the set offset will be added to it to obtain the output value.

There is a knob tracking control which can be set into off, lock or mirror mode. In off mode both knobs move independently, in lock mode they will move in tandem and in mirror mode when one goes up the other will go down.

The mute button will set all outputs to zero when active.

A range conversion module

Input values in a range input range ( [0..1], [-1..1], [0..4], [-4..4]) will map to an output range of [lowvalue,highvalue] - low can be larger than high to get a signal inversion.

The inverted outputs give the same range as the regular outputs, but going from high to low where the regular output would go from low to high.

A modulation cotroller

The modulation signal goes to the in input, and the amount of modulation is to be controlled with the mod input.

The internal signals in this module are always assumed to be unipolar, but the in and outgoing signals can be rescaled if needed to different ranges.

The internal computation used is: out = mod * ( in - 1) + 1 , so that when mod = 0 the output will 1 and when mod = 1the ouput will be a copy of the in signal.

Dynamic level type switcher

Select two different output types and switch between the types with the select input. A low vlue selects the left type on the left output and the right type on the right one. A high select input toggles that.

The rate of type change can be controlled with the slew contol.

A change detector

This module detects it's input being changed. When a change occurs a pulse is being output on the trig output and the amount of change is being sent to the out output.

The norm / abs selector determines whether the value (norm mode) or the absolute value (abs mode) of the change is put onto the output.

A rate conversion module.

This module is controlled by it's slave input, which is suppoesed to be connected to a MidiClockIn or a Tapper module. The slave output will just pass this signal on.

The rate multiplier controls the scaling applied to the control output, it will make controlled LFOs go faster or slower by some factor.

The control output can be used some different kind of controls. It can either control an LFO which is set to fast 0.254, medium 2:05 or to BPM mode; a Delay set to 3s or 5 min, or it can control the ClockGen module's BPM value.

The slave in and output can be used to chain up several MidiClockIn, RateConverter or Tapper modules, the first one in the chain will be the master controlling all other Tappers. Other tappers can be set to different control modes to get LFOs and delays synced up properly at different rates.

With the blue input next to the rate multiplier knob the rate can be modulated, it's value will be multiplied by the knob set value. Note that the internal values for the knob range from -23 for the / 24 setting trough 0 for the X 1 setting to 24 for the X 24 setting - so for effective modulation set the rate to X 24. Only whole valued rates will ever be created, or a reciprocal of while values. The modulation input is internally clipped to a range [-1,1]. A values module would be suitable to control it. When unconnected the assumed modulation value is 1.

This module rotates it's input vector (in1,in2) over an angle 2 * PI * angle to calculate an output vector (ou1,out2).

This can be used to make a rotating stereo image for instance, or with an input vector of constants, say (0,1) and the angle driven by a saw signal it will make a sine and a cosine on the outputs (quadrature signal).

This module rectifies it's input signal, either half wave or full wave. In half wave mode only the positive part of the input signal goes through, for the negative part the output will be zero. In full wave mode the output signal will be the absolute value of the input signal.

The inv. output has the inverse signal from the regular output, i.e. it will be negative only.

There also is a transparent mode, in this case the output will follow the input and the inv. output will be the inverse of the input.

Audio to control rate converter. This can be used to prevent portions of a patch to go 'red' (blue, or control rate, modules use less CPU than red, or audio rate, modules).

A slew rate limiter, or control rate low pass filter.

This module can be used to smooth jumps in a control signal.

In single mode the up control controls both the upwards and the downwards slew, in dual mode the down control comes in to make the down slope differ from the up slope.

Modulation is multiplied by the knob value, unconnected modulation inputs are set to a value if 1.0

Square wave rate multiplier.

The rate of the incoming square is being doubled, the pulse / pause information is lost as each edge is translated into a pulse.

This module can be cascaded to get repeated rate doubling.

When the pulses do not 'fit' after rate multiplication eventually a counter will overflow. This will result in an endless stream of pulses with a glitch once every 4 billion pulses or so.

An AM / RM modulator

The in signal is multiplied by a scaled version of the mod input and then sent to the output.

Scaling is applied first on the level shifted mod signal, then the offset is being added.

So the modulation depth is controlled by the mod knob, and an offset can be added so the modulation does not have to go down all the way to zero.

The AM / RM control fades between Amplitude and Ring modulation (or balanced modulation) types.

AM is multiplication by a positive only signal, and RM is multiplication by a bipolar signal.

The mod input is first converted into a bipolar signal for the RM bit and then that signal is internally converted to an unipolar signal for the AM part.

The FX mod controls can be used to fade between AM and RM.

A wave wiper module

This module was constructed after an idea by PHOBoS, you can find more about the original circuit trough the Wren web site.

This module takes a top bit of in1 and a bottom bit of in2, glues these together and outputs that to out1. The split level is set by the break input. Out2 does the inverse from out1.

A signal shaper

This module can transform a linear input signal into an envelope type output signal.

This is a lookup module, basically. The input value moves a cursor over the horizontal axis and the looked up graph height is then sent to the output.

To use the module as a sequencer the ClockGen2 module can be used to control it's in (from the ClockGen2 saw output) it's quant (from the ClockGen2 quantisation output) and it's time divisions (from the bars output).

Points can be added by double clicking (or with ctrl+click) in an empty area. The same action on an existing point will allow that point to be moved or removed - removal being a move to somewhere out of the graph area.

The leftmost and rightmost points can not be removed, but can only be moved up or down.

Although this module can be used at audio rates it is not being anti aliased, so drawn wave forms played as audio will give some serious aliasing.

A right click on the graph area will popup a context menu allowing for some standard wave forms to be created, basic symmetry operations and save to and load from file.

The view and lookup mode can be set as linear, splines, steps or dots.

There is a chg (changed) output which will pulse whenever the output value changed. This is usefull mainly in the steps and dots mode, although when using the sync input all modes are usable.

Sync when not connected leaves the module under full control of the in input. when sync is being connected and pulsed the output value and the chg (changed) output will be synced to the sync signal. The sync input is normal level logic (that is smaller than or equal to zero is low and greater than zero is high (that way it can be synced from, say, a sine LFO output (in bipolar mode))).

The inv and rev inputs can inverse (play upside down) and / or reverse (play right to left) the playing mode.

The rnd button changes all values, not their positions. Likwise the three rnd inputs change vertical positions only, the leftmost alters one at random, the moddle one alters 25% of the values amd the rightmost one does the same as the rnd button - it alters all.

The amount of grid lines can be set with the up down buttons, the top one for vertical lines (x - horizontal divisions( time)) and the bottom one for horizontal lines (y - vertical divisions (value)). The horizontal divisions can also be controlled from the ClockGen2 module.

The sync input supports the special NOT_CONNECTED status.

A Xenakis like number sieve.

I may be more like a generator, nomenclature.

Wren's notation for basic modules is:

* [n,m]       : a sieve module with modulus n and residue m [3,1] -> {1,4,7...}.
* random(f)   : a sequence of random numbers, f selects the chance for any number to be present in the sequence. 0 <= f <= 1, f.i random(0.33)
* fibonacci   : the fibonacci sequence {0,1,1,2,3,5,8,13...}.
* prime       : the prime numbers {2,3,5,7,11,13...}.
* {numbers|n} : a litteral number sequence, f.i. {1,5,13,14,21|24} n is the modulus, 24 in the example.

These can be combined using the following operators:

* \+ : the union of two sieves: [3,0]+[4,0] -> {0,3,4,6,8,9,12...}, the numbers combined (or function).
* \* : the intersection of two sieves: [3,0]*[4.0] -> {0,12,24...}, the numbers present in both (and function).
* ^ : the symmetric difference of two sieves: [3,0]^[4,0] -> {3,4,6,8,9,15...}, the numbers in one but not in the other (xor function).
* ! : the complement of a sieve: ![3,0] -> {1,2,4,5,7...}, the numbers not in the sequence (not function).

Parenthesis can be used to enforce an evaluation order, f.i. as in (([3,0]+4,0)*fibonacci)+prime to first evaluate the left +, then the * and the right + last.

A simplified formula will be calculated when the the formula changes, or when the external period changes. The simplification does not use prime decomposition, it will compute a sequence and from that it will make a sequence of modules joined through union.

The evaluate input will force reevaluation of the formula, this can be useful when a random module is present to obtain a new random sieve.

Otherwise this module functions like the Values module, except that the start value will always be 0 (zero) and the end value is set by the external period.

The difference (top) output will output differences between sieve sequence values, in case the last sequence value was selected it will be the value of the last minus the first element (this cyclic behaviour was not present in the original concept).

Negative numbers can not be used in sieves, the random sieve's probability control is the only floating poin number being used.

The simplified formula is displayed above the user supplied formula, or an error will be displayed when the user supplied formula could not be parsed.

The currently selected sequence value is displayed right of the changed output.

The changed output will go active when the ouput value was changed, the difference output may not have changed then.

The regualar (bottom) output will be scaled such that the external period results in an internal output of 1, this can be changed then with the output type selector.

The difference (top) output will be scaled such that the largest difference value present will result in an output value of 1, again this can be rescaled using the output type selector.

For more information on the subject try googling for Xenakis sieve, or Xenakis sieve algorithm. It may find a 221 page paper by Dimitrios Exarchos (Iannis Xenakis and Sieve Theory), I've founde that to be a very useful text.

The formal-ish parser definition is:

Sieve   = Term                      // Single term
        | Term '+' Term             // Union of two terms
        | Term '*' Term             // Intersection of two terms
        | Term '^' Term             // Symmetric aifference of two terms
        ;

Term    = '[' Module  ']'           // Modulo residue sieve
        | '{' Literal '}'           // Integer sequence
        | '(' Sieve   ')'           // parenthesis for grouping
        | '!' Term                  // Inversion of a Term
                                    // all below of form identifier = [a-zA-Z][a-zA-Z0-9]*
        | 'prime'                   // 2,3,5,7,11..
        | 'fibonacci'               // 1,1,2,3,5,8..
        | 'hanoi'                   // 0,1,3,4,7,8.. with the diff output being the ruler function
        | 'hanoipegs1'              // {0,1}, {0,1,3,5}, {0,1,3,5,6,6,7,8}  non alternating
        | 'hanoipegs2'              // {0,2}, {0,1,3,5}, {0,2,3,4,6,6,8,10} alternating
        | 'hofstadterq'             // chaotic
        | 'metafib' '(' num   ')'   // genralization of hanoi, num being the order
        | 'random'  '(' float ')'   // random numbers, float being the chance param
        ;

Module  = num ',' num               // modulo, residue
        ;

Literal = Nums '|' num              // { ... | residue}
        ;

Nums    =  num                      // Comma separated integer sequence
        |  num ',' Nums
        ;

float   = num                       // floating point number, allows both comma and dot
        | '.' num
        | ',' num
        | num '.' num
        | num ',' num
        ;

num     = [ '0' .. '9'] *           // Integer number
        ;

A Chladni controller.

This module is meant to control the frequency or FM input od an oscillator or a filter. It computes the function f = C * ( m + 2n) ^ p where C is the base frequency as shown in the frequency display.

Globals

Reflects the value of global settings,

For now this is just the globally set LFO relative rate (r.rate).

Easing curves.

The module implements the various Esing cirves as defined at https://github.com/warrenm/AHEasing .

The pos input will make the cursor walk over the graph and the looked up curve value will be scaled by a connected gain (or by one when gain is unconnected) put on the out output.

The inv output will hold 10 minus the out signal. The on the right hand side there is a sigin and a sigout connector, the sigin signal is passed to sigout multiplied by the out signal.

Input signal types for gain and pos can be set with the triangle buttons.

This module will also work as a wave shaper for audio signals - it is rate smart.

A Scala quantizer

Connect the input of this module to a note generator (like a sequencer). Load a Scala keyboard mapping file and a Scala scale file. On the output there will then be the Scala mapping applied to the note value present on the input.

For this to work as intended oscillator frequency should be set to 440 Hz, for filters likewise.

A Transpose / Inversion module

Transposition is calculated as : ( amt + in) mod modulo and Inversion as ( pivot - in ) mod modulo.

The chain inputs are added to the calculated value.

This module implements Euclidean Scales

The steps selector sets the pattern length and the pattern selector selects one of the possible patterns for that amount of steps. The patterns can be altered dynamically with the pattern modulation input.

A quantizer module

The mute control will set the output to zero when active.

The input value will be quantized to the fractional value set by the notes control. Like when this is set to 1/1 half tone steps will be output.

A quantizer module

There are 22 quantized outputs, each quantizing the input to fixed levels.

The quantization levels are 1, 1/2, 1/3, 2/3 .. 7/8.

Suggested usage is to couple some ot the outputs to a multiplexer which can then be used to select the currently wanted one.

A note quantizer module

the input value is quantized to the nearest note selected by the the buttons, when no notes are selected the input is passed unchanged, and when a lower and a higher note are equidistant from the the input value the higher value prevails.

This assumes a 12TET scaling with middle A having note number 69 (or middle E having 64, or middle C having 60).

There is no maximum or minimum on the input signal.

A scaling of 128 notes / unit is used.

The range control attenuates (or inverts) the input signal.

Scales can be selected from a preset and the presets will be dynamically loaded from a file wren.scale that must be in the same folder as the Wren executable file. When this file does not exist Wren will create it. After that you can edit it with a text editor (like Notepad for instance) and Wren will not touch it anymore.

A scale can be altered manually with the note buttons, but that may get you locked, as once when a custom scale is in effect the preset will not work anymore. You can use the enforce button to fix that, it will set the currently selected preset. The mudule show a scale name in case it happens to know the note combination, or Custom when it does not, or Transparent when no notes at all are selected - in this case the input passes unquantized to the output.

A transposition can be applied to the selected scale with the transpose control. Note C4 is the 'neutral' base note (not leading to any changes). Transpositions are not reflected in the shown note pattern.

Note: The inv2 output will follow the set scale, the inv2 output will not. Both are an invesion of the regular output and both play notes from the same octave as the regular output is in.

When the transparent input is active the range scaled in summed with the chain input is passed on to out and the inversion outputs then both hold 1.0 - out. This will also happen when the scale input is set negative (but unequal to NOT_CONNECTED).

The scale input supports the special NOT_CONNECTED status.

This module implements a scale randomizer.

Each time the rnd input goes from inactive to active, or when the rnd button is clicked, a new random scale mapping will be computed.

A scale will always have 12 notes per octave, and all octave mappings will be identical. Within an octave all the 12 notes will be randomly mapped to 12 (probably, but not guaranteed so, different) notes.

Convert note into note and octave

Find nearest value in history

This module will learn the values to find from it's learn input. It will remember the amount of values as set by the history parameter. The results will not be quantized.

Input values will then be converted to the nearest remembered value. The range to be used can be controlled with the range parameter (an attenuator on the in connector)

History length and range can be modulated.

An adder module.

The output will have the sum of all the input signals. An unconnected input will have a value of zero.

The mute button will set the output to zero when active.

A multiplier module.

The output value is the product of all the input values. An unconnected input will have a value of 1.

The mute button will set the output to zero when active.

You can use this module as a ring modulator too by multiplying two signals. A ring modulator is just that, a multiplier.

An power 2 module.

The output value is two to the power of the input value divided by two, so for an input range of -1 to 1 it outputs 1/4 to 1, with 1/2 inbetween as 2^0.

This module is the inverse of the Log2 module.

The mute button will set the output to zero when active.

Eight multipliers in one module.

Each output is the product of the two associated inputs. Unconnected inputs are set to 1 internally.

This module can be seen as a fourfold ring modulator as well.

This module determines the momentary minimum and maximum of it's two inputs.

When in1 > in2 the max output will be in1 and the min output will be in2, and this will reverse when in2 < in1.

Analog AND and OR

The AND (A) output (leftmost) will hold the minimum value of all inputs, or 1.0 when no inputs are connected.

The OR (O) output (rightmost) will hold the maximum value of all inputs, or -1.0 when no inputs are connected.

When more than one input is connected the module behaves as a multi input MINimum and MAXimum finction.

All inputs support the special NOT_CONNECTED status.

Calculates 1 / X

When X is zero the ouput is set to zero, otherwise the output is set as 1 / X. The result however is clipped to a range -1000 to +1000.

Ratio

This module multiplies it's input signals by a selectable ratio.

The multiplier control sets the numerator, the divider control sets the denominator. The numerator can be zero or negative, the denominator must be greater than zero.

The mute control will set the output to zero when active.

Arbitrary value

Use CTRL+RIGHT-CLICK to enter edit mode, type a value there (which may use a modifying suffix), the entered value will be multiplied by the mult input and added to the add input and the result of that will be on the output connector.

An unconnected mult input has a value of one.

Calculates a selectable function of X

Invalid X values will lead to an unchanged output value, otherwise the function is being calculated with the result being clipped to the interval -1000 to +1000

RMS Calculator.

This module calculates the RMS value of it's input value. The averaging time can be set with the time control.

This module converts a point (A,D) in polar coordinates to a point (X,Y) in ectangular coordinates, where A is the angle and D is the distance, or radius.

8 settable constant values with mute and multiplier, the sum output gets the sum of the individual outputs (without mute being applied).

The only difference between this and a mixer is that unused inputs default to 1.0 instead of 0.0.

A base 2 logarithm module.

First the absolute value of the input is taken, when it is >= 1/8 the base 2 logarithm is being calculated and then 1.0 is being added, when the absolute input value <= 1/8 the output value is -2.0.

Thus an input range [1/4 .. 1] results in an output range [-1 .. 1], it is the inverse of the Power2 functionality (with some constraints).

The mute button will set the output to zero when active.

A high-logic NOT module

Logic input values are said to be false when the value is 0.5 or lower. When the value is greater than 0.5 it is said to be true.

Logic output values are -1 for a false result and 1 for a true result.

For each input with a false value the corresponding output value will be true, and inputs with a true value result in the output to be set false.

A high-logic AND. NAND, OR, NOR, XOR, XNOR, ONE or NONE gate module.

A high-logic signal is considered on when it's analog value is over 0.5, otherwise it is off.

The AND. NAND, OR, NOR, XOR and XNOR gates have been generalized to operate on multiple bits, this works by quantizing the input values into a selectable number of bits (1 bit gives the old behaviour) and then performing the logical operation on the resulting bit vector.

The XOR and XNOR ports operate like cascaded ports, which makes them behave like and odd or even parity generator respectivly, the output being active when an odd number (or an even number) of inputs are active.

The ONE port activates it's output when precisely one input is active, the NONE port is the same with an inverter added at it's output. These two port types do not generalize to bit vectors.

When the bit size is set different from one the input signals are AD converted to the set number of bits, after that the logic function is applied and then the signal is converted back to an analog signal. When for instance only one input is being used and the module is set to 'OR' mode and the input value is in the range [ 0, 1] this will result in a quantizer function being performed (with four possible output values in two bit mode, eight in three bit mode, etc., and of course in one bit mode it will have two possible output values).

For the 1 bit mode the following function table applies:

 A B C D   AND   NAND   OR   NOR   XOR   XNOR   ONE   NONE

 0 0 0 0    0      1     0    1     0      1     0      1
 0 0 0 1    0      1     1    0     1      0     1      0
 0 0 1 0    0      1     1    0     1      0     1      0
 0 0 1 1    0      1     1    0     0      1     0      1

 0 1 0 0    0      1     1    0     1      0     1      0
 0 1 0 1    0      1     1    0     0      1     0      1
 0 1 1 0    0      1     1    0     0      1     0      1
 0 1 1 1    0      1     1    0     1      0     0      1

 1 0 0 0    0      1     1    0     1      0     1      0
 1 0 0 1    0      1     1    0     0      1     0      1
 1 0 1 0    0      1     1    0     0      1     0      1
 1 0 1 1    0      1     1    0     1      0     0      1

 1 1 0 0    0      1     1    0     0      1     0      1
 1 1 0 1    0      1     1    0     1      0     0      1
 1 1 1 0    0      1     1    0     1      0     0      1
 1 1 1 1    1      0     1    0     0      1     0      1

A logic divider module.

Logic input values are said to be FALSE when the value is zero or lower. When the value is greater than zero it is said to be TRUE.

Logic output values are -1 for a FALSE result and 1 for a TRUE result.

This module divides down the rate of it's input signal. The number of FALSE / TRUE and TRUE / FALSE input changes will be divided down by the amount set by the knob. This means that when the input frequency is 440 Hz and the knob is set to 2 the output frequency will be 220 Hz, and no matter what shape the input wave had, the output will be a square wave.

A fixed divider module.

The module has a series of outputs with fixed divison rates. After reset all the outputs will be off, to be toggled each time the number of input clock periods matches the division count for that output. The ouput will stay high for one clock period then.

A random logic divider module.

Logic input values are said to be FALSE when the value is zero or lower. When the value is greater than zero it is said to be TRUE.

Logic output values are -1 for a FALSE result and 1 for a TRUE result.

This module divides down the rate of it's input signal. The number of FALSE / TRUE input changes will randomly be divided down by the amount set by the knob.

An RS type flip flop

A rising input edge on the S input (signal goes from zero or negative to positive) will set the flip flop.

A rising edge on the R input will clear the flip flop

When the flip flop is set it's Q output is 1 and it's not Q output is zero, the output state is reversed when the flip flop is cleared.

A logic counter module.

The module will either count up or down depending on the dir input. For an unconnected dir input it will count up.

There are two modes, binary and sequentially. In sequential mode it wil simply count 1, 2, 3, etc. In binary multiple outputs can be active at the same time [], [1], [2], [1,2], [3], etc.

The reset input will reset the counter to 1 or 16 or [] or[all] - depending on mode and direction.

A mode switch will reset the counter.

Digital to Analog Converter.

The highest numbered bit is the most significant bit, and it will add 0.5 to the output when active.

This module compares it's two input values, the < output going active when in1 < in2, etc. The range for equality can be be set with the eq. range control (in dB).

A programmable clock divider.

A low and a high division rate can be set as well as a step value.

When the select input is zero the divider will operate using the low count value, getting the select input up will add the set step value repeatedly to the used count until (for a modulation input of 1) the high count is reached.

A logic pulse delay module.

This module can operate in three modes:

Pulse stretch mode : an incoming pulse will be stretched to the set delay time, or when it is longer than tat it will be truncated. That is the ouput pulse start when the input pulse starts, but it ends when the set time is over, no matter what the input did. This makes changes to the positive part of the incoming signal only.

Pulse delay mode: an incoming pulse will be delayed by the the set time. This will move the entire pulse in time, so it's length will not be changed. This will shift both the positive and negative parts of the incoming signal.

osc mode : when the input is active pulses will be generated continuesly at the set pulse duration rate and whe nthe input is inactive thou output will be off.

The delay time can be modulated from the delay time modulation input, the amount of modulation can be set with the knob.

This module has the same ranges as are used for envelopes, except that the module can be set to run at audio rate. When running at audio rate the set time will be eight times shorter as compared to running at control rate.

In Delay mode only one pulse can be handled properly, meaning the module cycle has to be finished before a new one gets handled properly. When you need to delay a pulse train an audio delay module should be used.

In Stretch mode, when a next pulse comes in before the current one is finished it will be ignored.

The output indicator shows the internal logic output state, this can be modified with the output type control to determine thaa actual values being sent out.

The other indicator shows module activity, e.g. when it is actively stretching, delaying or oscillating.

A rungler module.

This module was modeled after an idea by Rob Hordijk, he used this circuit in his Benjolin device. The length of the shift register can be set from 3 to 64 (the original uses 8) and the number of taps can be set from 1 to 8 (the original uses 3).

The idea is to clock the module from an oscillator and to provide data from another oscillator, the output of the rungler is then fed back to the frequency control input of the data oscillator. This way short varying patterns will be generated. The shift register length matters not much, but it should be set larger than the number of taps used. The number of taps determines the number of distinct levels present in the output signal.

For more information about this go to http://hordijk-synths.info

A 32 bit linear-feedback shift register.

Input from the in input will be XORed with the XOR of the active taps before being added at either the left or the right side of the internal shift register.

A shift is being made on each upgoing edge of the trig input. Shifting can be to the right or to the left depending on the rev (reverse) input. The unreversed direction is from left to right (that is 0 to 1 to 3 etc.).

The indicator bar shows the lowest bit on the left side (same direction as the outputs).

When the free / clocked selector is in the free position reset is immediate, otherwise it will be synced to the trig input.

With the T/G mode selector the output can be switched between triggered and gate mode, where gate mode makes overlapping pulses and trigger mode ands the output with the trig signal.

Branches, a quadruple Bernoulli gate

After a module made by Mutable Instruments

When the input changes it's value is routed to either the top or down output. When p is set to zero it will always be the top one and when p is one it will be the bottom one.

A random notes sequencer.

Will play a sequence of random notes with a set length, a set lowest note and a set highest note. The notes will be picked from a scale set by scale and base note controls.

A low to high on the trig input will play the next note and a low to high on the random input will select a new random sequence.

Scales can be user defied, at program startup a file wren.scale will bee loaded when present in the application directory. When such a file is not present the program will make a default one.

The change inputs are sampled on a rising step input, so a change will occur when such an inpujt is active at the momenent the step input goes from low to high.

A Morse code generator.

Input text is translated into an off pattern on the output connector at a speed determined by the incoming clock signal.

The Reset input will make the sequence restart from the beginning.

When the sequence ends it will repeat from the beginning.

The blue output genrates note values 0 to 7 based on a primitive implementation of SolReSol.

This module can use input from a file by typing :

file:fillename

or

file:fillename regular-expression

The last non-empty line of the file will be read (or an error message will be returned) and that line will be filtered with the optional regular expression.

When the regular expression contains a group named 'result' that will be the matched bit from that last line.

Example:

X:\filedir\filename.log (?:msg\s\[)(?<result>.*?)(?:\]\shost)

pointing to some file on drive X: with a specified filter which has a group named
'result' (and two other unnmaed groups of stuff we are not interested in).

the last non empty line of the specified file could be (this is from a log of an IRC
chat session, a priviat message from jan to group #electro):

  '2022-02-09 01:30:23.784 | priv : [jan -> #electro] msg [pwew .. filter is hard o_o] host [~jan@xxx-xx-xxx-xxx.yyy.zzz.nl]'

The regular expression   matches:  'msg [pwew .. filter is hard o_o] host'.
The named group 'result' matches:  'pwew .. filter is hard o_o'
and that in this case is the bit we want.

A gate sequencer.

Each step has three modes, it can be off (blank) it can be a trigger step (o) active for half a clock period or it can be in gate mode (O) staying active a full clock period.

The rnd button will set a random pattern.

The clr button will clear all steps.

The res input is a synced reset, it will only reset when the signal is active on a low to high transition of the trigger input.

The a-res input is an unsynced reset, it will always reset the module when the signal goes from low to high.

A 16 step sequencer for values and for notes.

It can be set to various signal types.

A cross fade utility is implemented to smooth the signal from one step to the next, cross fading works on all four X outputs, the R outputs are not cross faded (Raw).

Smoothing slope type can be set independently for up going and down going signals to eiher exponential, linear, logarithmic or s-shaped.

This module can be used as a complex envelope generator when properly clocked.

The four outputs are all offset by four steps each, so when output 1 is at step 5, output 2 is at step 9, output 3 at 13 and output 4 will be at step 1.

The chain input is being used on all four outputs when the module is not active (as controlled by the act input).

The sequnecer can be set to run reversed with the dir input, and it can be set to inverse mode with the inv input. Inverse mode being that the slider is read out upside down.

There are three random inputs which will set the sliders to random positions when going active, 1 for one value, some for some values and all for all value.

There are three clear inputs which will set the sliders to zero positions when going active, 1 for one value, some for some values and all for all value.

When the hold input is active the sequencer will still step but will keep the ouptut values frozen, this differs from what the act input does, that one freezes everything.

Skip inputs when active will skip the corresponding step to be used, making the sequence a step shorter. When all steps are skipped the sequencer will stop moving.

The clock skip input, when seen active will cause a clock (trigger) action to not be executed. This differs from the inactive mode in that in inactive mode the sequencer passes on the chain inputs and when the clock is stopped the output will no longer change at all.

The shf (Shuffle) input will shuffle the values for the steps on gong from inactive to active.

The steps output for the current step will go active when the sequencer is active and either the trig input is high or the T(rigger) / (G)ate mode is set to G.

The TimeWarp (tw) output, which can be used to control the speed of a driving clock module, it is meant to drive the speed input of an LFO. The idea is that when steps are being skipped the sequence length can be kept the same`when using the TimeWarp output in this manner.

The changed outputs (in the C column) will pulse briefly when the associated sequencer output changes.

The outputs below the res, dir and act inputs just copy tose inputs.

The res input is a synced reset, it will only reset when the signal is active on a low to high transition of the trigger input.

The a-res input is an unsynced reset, it will always reset the module when the signal goes from low to high.

A lookup table module.

The select input selects one of the faders (or a mix of two when cross fading is in effect) to be sent to the output.

This can also be used as a sequencer when controlled with a saw or triangle shaped signal on the select input.

The four outputs are offset by four steps each, when output 1 outputs step 5, output 2 will output step 9, output 3 step 13 and output4 step 1. Cross fading works on all four outputs.

The steps selector selects the number of steps to use for an input going from zero to one.

The blue inputs below the faders modulate the fader values. When not connected they will have a value of 1. When connected and the xfade control is set to one these inputs will make for a 16 step piecewise linear interpolation processor (processor as opposed to shaper).

A probability based sequencer.

For each note the probablility with which it will be played can be set with a slider (and this value can be multiplied by the vlue on the associated input). Notes have to be be enabled before they can sound, and note enables can be selected from a scale preset.

The presets will be dynamically loaded from a file wren.scale that must be in the same folder as the Wren executable file. When this file does not exist Wren will create it. After that you can edit it with a text editor (like Notepad for instance) and Wren will not touch it anymore.

A transposition can be applied to the selected scale with the transpose control. Note C4 is the 'neutral' base note (not leading to any changes). Transpositions are not reflected in the shown note pattern.

A selected scale can be altered with the note buttons, and such a change will be saved with the patch (making the shown scale preset name meaningless, of course).

The rnd button will set a random probability distribution, the clr button will set a flat distribution. The distribution can be altered with the tilt control. A zero value means that there is no tilt, a positive value maked higher notes more likely to be selected and negative values will produce more low notes.

A transposition can be applied to the selected scale with the transpose control. Note C4 is the 'neutral' base note (not leading to any changes). Transpositions are not reflected in the shown note pattern. The low and high note controls limit the range of possible notes.

The sm(oo)th controls determine the likelihood that the next random value is about the same as the current one.