Triple LF-VCO

This module has three independent low frequency voltage controlled oscillators or LF-VCO's that can also be synced and cross-modulate. Three LEDs give visual feedback of the activity of the three LFOs. The modulation inputs of all VC-LFO's are normalized at the input connectors in such a way that everything can crossmodulate and sync.

LF-VCO A provides a Sine wave wich has no corners, giving very smooth modulation. It has a wide frequency range going from quite slow to well into the audio range, so it can be used as a sound source as well as a modulator. This LFO is tied internally to LF-VCO C. The latter will modulate the former in FM and in combined FM-AM called "fluctuation". This "fluctuation" modulation soft-syncs to the harmonics of the modulating signal. Therefore the modulation depth 'feels' more natural even when the frequency of the modulated waveform is increased. This results in an extremely natural sounding vibrato, much gentler than through a normal LFO. Pitch is controlled manually. By mixing LF-VCO A & LF-VCO A Fluct, at a certain point, one can only hear the amplitude modulation which is interesting when modulating the pitch of an oscillator.

LF-VCO B is useful for clocking a sequencer or envelopes with the pulse. It has a slower range than LFO A. There is a triangle output and a pulse output. These outputs are unipolar. The wave shaping control is set manually. The triangle output can be modulated from downsloping saw to a triangle to an upsloping saw. The triangle output can then be used as envelope. The pulsewidth also varies with the waveshaping control.

The frequency of LF-VCO B can be modulated internally by LF-VCO C or by an external signal.

LF-VCO B has three operating modes, selectable with a switch:
- MODE : LF-VCO B is free running
- SYNC : LF-VCO B is synced to LF-VCO C or to an external signal
- HALT : LF-VCO B oscillates at a frequency set by its Rate control during the Rise of the Triangle wave of LF-VCO C and will "freeze" during the Fall of the Triangle wave of LF-VCO C. LF-VCO B will restart with the next cycle of LF-VCO C. When "frozen", LF-VCO B will output a fixed, momentary voltage. Halt mode works best on pitch.

LF-VCO C has a triangle and an pulse output. Rate goes from several minutes to around 100Hz. The LF-VCO C Mod slows down the rate of the downgoing slope of the triangle. The rate of the upgoing slope remains the same. So, when e.g. speeding up the LFO and opening the C MOD knob the waveform will take on the form of a fast attack and slow decay envelope like a sawtooth wave. The pulse output is high when the slope is going up and low when it is going down, so the pulse length will remain the same but the time between the pulses will
increase.

Regrettably this selfmod feature is obsolete as Rob developed a S&H instead.
Here is Rob's description about it :

The modulation input of LFO C has a S&H right after the input connector, but just before the input level knob. LFO C triggers the S&H circuit. 
On every positive peak and on every negative peak of the triangle waveform the S&H samples the LFO C modulation input signal and holds the sampled value during the slope that follows. This means that the S&H causes every upslope and every downslope of the triangle to have a different duration, defined by the momentary value the S&H happened to sample. However, the slopes will remain perfectly linear, only their steepness is affected. The effect is that there is a more or less random spread in time. On the pulse output this causes pulses of different duration, an effect also named clustering in time.

The output of the S&H is only available internally, but also normalizes to the LFO A modulation input. This can be overridden by inserting a jack with another signal into the LFO A modulation input. When no jack is connected to this LFO A input it will follow the S&H on the LFO C modulation input.

When no signal is inserted in the LFO C modulation input it is normalized to the output of LFO A. This means that when no jacks are applied to the LFO A and LFO C inputs, but their input level knobs and the LFO A fluctuation knob are opened, the result is constantly varying pitches and rates on both LFO A and LFO C, caused by the crossmodulation between LFO C and LFO A. On LFO A it will always sound like a stepped pattern unless you also open the fluctuation knob, as that will modulate the pitch slightly with the slopes of LFO C or with an external signal when the fluctuation input has a jack with a signal plugged in.

Note that for pitch or frequency modulation there is a behaviour that a fast pitch can easily be deeply modulated by a slow pitch or rate, but a slow pitch or rate is much harder to modulate deeply with a much faster pitch. A S&H changes this FM equation into a pure statistic function that becomes pitch independent, resulting in slow rates reacting much deeper to faster rate modulation signals as without the S&H. E.g. when a 1Hz LFO is modulated straightforward with a 1kHz audio signal there may only be a tiny bit of 1kHz zippery noise on the LFO signal, but the 1Hz will not seem to change much. But through the S&H the LFO will instead go all over the place, defined only by the average amplitude of the 1kHz signal.

There are maximum possible pitches for the three available LFO’s, they basically can not go faster as the maximum setting on their rate knobs. To use the S&H modulation on LFO C it is often a good idea to not set the LFO C rate knob to its maximum. There is not such a limit for the minimum rate times, though a very large negative modulation signal can cause a LFO to actually stop oscillating. If this happens when the LFO C rate knob is at its minumum, the modulation level knob is fully open, and a sampled negative signal is stopping the LFO, and thus also not sampling a new value that can start it again. In this case LFO C may appear to be frozen. In that case you can just open the rate knob a bit until it starts oscillating again. Or wait for a very long time for the S&H capacitor to eventually loose its charge.

A tip is to e.g. apply the LFO C Pulse output to the S&H input of one of the EnvGens while clocking that EnvGen with a stable clock. Then feed the output of that S&H to the Gate input of the second EnvGen. While the first EnvGen is triggered at a stable rate the second will trigger only now and then in an irregular rate but synced to the clock of the first EnvGen. If LFO C is deeply modulated the second EnvGen gates seem to appear in clusters in time.

And there are many other uses if you want to introduce a smaller or larger amount of variation in dynamics to an otherwise static pattern, or to spread events unevenly in time.

A video of the new feature is at the end of the page.

Here is the flowchart of the Triple LFO with LFO C selfmod.

Here is the flowchart of the newer Triple LFO with S&H.

Here is a video of the Triple LF-VCO. As I couldn't upload the one at Novars please follow this link : https://vimeo.com/239786705

Here is the explanation of the new S&H feature.

Thanks to Todd Barton for providing the picture of the module.

The Dual Fader

A Swiss Army knife for modular.

The Dual Fader is a voltage controlled mixer module with several options useful for different applications for dynamic signal. It can be used in various ways and at different points in your patches. It contains two RMS 'equal loudness' voltage controlled faders with both two inputs and two outputs. Depending on how jacks are connected the basic functions are crossfader, panner or alternating VCA.

The inputs are DC coupled which means it will process CV signals as well as audio signals. You could have one channel set up as a fader for CV signals and the other as a panner for audio signals.

In a two inputs/one output configuration, the channels will act as individually voltage controlled crossfaders usingAX and/or BX outputs and the signal goes from one signal to another with the fade control. Fade parameter is voltage controlled.

In a one input/two outputs configuration, the channels will act as individually voltage controlled panners with "energy loss compensation" set at -3dB with the control centered (equal loudness curve).

It is good to note that when a signal is connected to only input A1 (or B1) and the output is taken only from AX/L out (or BX/L out) the input signal will go to both VCAs and the output will be taken from the mix of the VCAs.

Each channel has a 20dB boost available to, for example, adapt external line level signals to the signal levels inside the modular. This is also very useful to crack up the level of stompboxes to synth signal level. This way it is easy to use a fader to e.g. crossfade between the input and output of an external guitar stompbox effect or line level effects processor. With some clever routing using the matrix it is also possible to create external feedback paths for external effects and have the feedback under voltage control. You can watch examples in the videos at the bottom of the page. This 20dB boost is applied to the A2 & B2 inputs. A1 & B1 inputs are normalized to the A2 & B2 inputs respectively, so that when no jack is inserted in the inputs A2 & B2, the signals present at A1 & B1 are duplicated and can benefit of the 20dB boost. It now has an interesting feature, as when it is opened it will start to clip the input signal when a high level signals like from an Osc or the output of a filter is used. Now you can use the module as a distortion by letting the signal clip and basically set a mix between the clipped and the unclipped signal by the fader knob.

One application for this boost would be to feed a signal into e.g the A1 input and the input of a stompbox effect. Then route the output of the stompbox into the A2 input and crank up the gain from the stompbox output (that is often at line level) until it matches the A1 input level and now crossfade between the dry sound and the wet stompbox effect, e.g controlled by an envelope or a sequencer. Alternatively it is also possible to use a fader to set e.g. echo time on a stompbox, in this case one sends the output of a fader to the input of an echo-box, and reconnect the output of the echo-box to the A2 input, with the audio signal to be processed on the A1 input, while listening to the fader output. When the fader is turned left the dry input signal will be present on the output. By turning the fader to the right the output signal that is also routed into the echo-box will now route back a delayed signal into input A2 and the delayed signal will also be rerouted into the echo-box and thus start to create echo repeats. When turned more and more to the right (and the gain level set correctly) this patch will eventually create an infinite repeat. And this is now under full voltage control.

The AX TO B LEVEL knob receives a signal that is the crossfade mix of input A1 and A2. This signal is generated internally and is fed through the knob to the second fader and internally added to the B1 and B2 inputs.

The switch is in the line to the B2 input and it has three possible functions:

1) The AX MIX position.

The signal from fader A is passed through the AX LEVEL knob in an equal amount to both B inputs. In this mode you can for instance make a mix of two signals with fader A and then pan this mix to outputs BX/L and BR. So, you would e.g. connect two oscs to inputs A1 and A2, not use any of the A outputs, open the AX LEVEL knob and connect the BX/L and BR outputs to the left and right channels of a mixing desk or to two different audio processors. The FADE A knob sets the amount of mix of the two inputs and the FADE B knob pans the stereo position of the mix from left to right. The AX TO B LEVEL knob can be used as an overall volumeknob. Modulating the FADE B MOD at low rates yields tremolo / leslie effect-like results.

2) The AX>B position.

In this mode the internal mix signal from the A1 and A2 inputs is fed only to the B1 input. In this mode you can e.g. mix three signals to the BX/L output. Connect two signals to the A1 and A2 inputs and a third signal to the B2 input. The FADE A knob sets the mix between the A1 and A2 signals andthe FADE B knob sets the mix between the A1/A2 mix and the signal on the B2 input. This way you can mix three signals in any ratio with only two knobs and always have an output signal, as long as the AX TO B LEVEL knob is opened. You can even add a fourth signal by plugging it into the B1 input and set its input level with the B1 IN LEVEL knob.

3) The RM position.

In this case the internal mix signal of the A1 and A2 input signals is fed to both the B1 and the B2 inputs through the AX TO B LEVEL knob, but the signal that goes to B2 is inverted before mixed with the B2 input signal. This means that when the FADE B knob is in its middle position the signal that goes to the B1 input is cancelled by the inverted signal that goes to the B2 input. If you start to modulate the FADE B knob with an audio signal on the MOD B IN input and opening the FADE B MOD knob, while the AX TO B LEVEL knob is open the output signal on BX/L will alternate between its normal signal and the inverted copy of the normal signal and ringmodulation will happen.

To do this just connect an osc to the A1 input (your carrier signal can also be a composite signal of A1 and A2), set the FADE A knob to its middle position, set the switch to RM mode, open the AX TO B LEVEL knob, plug a second oscillator into the MOD B IN connector, listen to the BX/L output, close the FADE B MOD knob and adjust the FADE B knob to the middle until you hear silence. Now slowly open the FADE B MOD knob and you will hear a ringmodulator effect. Tune the oscs so that they produce a nice sound and play both oscs with the same 1V/Oct signal from e.g. a keyboard. Changing the tuning ratio between the two oscs, setting their basic timbre with the ODD and EVEN HRM knobs and playing with the FADE B and FADE B MOD knobs will take you through a whole range of ringmodulator timbres. Due to the nonlinear crossfade curve there will be some distortion in the ringmodulation, and the modulation depth will define the amount of distortion. But as the curve now has sort of a compressing effect on the modulation waveform a triangle wave will be sort of compressed to a sinewave at a certain modulation depth and fine tonal control can be achieved by setting the modulation depth at certain levels for certain modulating waveforms. E.g. you can create the typical vintage 'diode-transformer' ring modulator sound from the fifties and sixties as used by e.g. Karl-Heinz Stockhausen or those SF-timbres from many sixties and seventies SF movies. The A2 IN GAIN control may be used to slightly overdrive carrier input into the modulator, for example leading to more 'vintage' sounding results.

The 'ducking' mode happens when two different signals are connected to e.g. the A1 and A2 inputs and the AX/L and AR outputs are both connected to two inputs of a mixing desk. In this mode the A1 signal goes straight to the AX/L output and the A2 signal to the AR output. When playing the FADE A knob one signal will increase while the other automatically decreases in volume. If the FADE A MOD input would now receive e.g. a side chain CV signal from a compressor that signal would automatically drop the volume on the A1 > AX/L line and open the A2 > AR line. This is often used when a narrative is added to a music track for a film or documentary, so that when the speech comes the music is automatically lowered in volume. It would however need an extra compressor with a side chain CV output or an envelope follower module. Of course you could also modulate this 'ducking' effect with an LFO or an envelope signal.

Here is the flowchart of the DualFader :

Check out these videos, you will find tons of good examples of patching this great module.

Thanks to Todd Barton for providing the picture of the module.