Friday, October 5, 2012

Baby Mammoth is now fully mature

My 5U modular synthesizer system, named the Baby Mammoth, is now up to completion :

The system is dual brand, consisting of 2 rows of 12MU each of Mos-Lab modules (faithful reproductions of the original Moog modules) :

2x CP-10 mixer
1x 992 controls to oscillator
1x 921-A oscillator driver
3x 921-B oscillator
2x 995 attenuators
1x 911 envelope generator
2x 902 voltage controlled amplifier
1x 904-A voltage controlled low pass filter
1x 904-B voltage controlled high pass filter
1x 904-C filter coupler
2x 905 reverberation unit

The bottom two rows are occupied by Original Rob Hordijk Design modules :

1x Dual phaser
1x Triple Input Filter
1x Dual Fader
1x Rungler
1x Triple LF-VCO
2x Dual Env
1x Mini Matrix / Node Processor combo

Last weekend I had Rob Hordijk coming over to my place to deliver the latest additions to the system, the two Dual Env, the third VCO HRM and the Dual Phaser :

The Dual Env, as it's name suggests, consits of two voltage controlable envelope generators.
Both share the same Gate input.
The first one is of the ADBDR type (Attack, Decay I, Break, Decay II, Release).
Each stage has its own control voltage input (unattenuated), except for the Break parameter.
The second envelope is a simple voltage controlled Decay with direct output and polarized output.
In addition, there is a S&H circuit that can be used indepently .
When self-patched or cross-patched with a Triple LF-VCO, the Dual Env proves to be very effective at creating complex, polyrhythmic modulations.
Here's the block diagram and panel layout of the Dual Env:

And here's a more detailed description of the module :

"The Dual Envelope module (DUAL ENV) is a fully voltage controlled envelope generator specifically designed to be used with sequenced music. There are two different types of envelopes available, one is a four stage (attack, decay1, break level, decay2, release) envelope and the second is a one stage envelope (decay only). Both envelope generators share the same gate input, meaning that they can not be triggered separately. Triggering treshold is at roughly 100mV above ground and also accepts e.g. triangle waves. The ADBDR envelope is primarily intended to be used for volume envelopes. When the decay2 knob on the first envelope generator is fully open the decay2 acts like the sustain that you find on most of the traditional envelope generators. In this case the break control will act like the sustain level. There are CV inputs for the attack, decay1, decay2 and release rates. The CV for the attack is inversed, so increasing the CV level will shorten the attack time while increasing the decay times for the decay1, decay2 and release. This means that when e.g. the key velocity voltage is used a higher velocity will shorten the attack and increase the other decay times. Rate settings can be from really snappy to pretty slow. Care was taken that you still have good control over the rates when in the snappy range. The second envelope generator is intended as a modulation envelope generator to e.g. sweep a filter or control the harmonic waveshaping of an OSC HRM module. It has an extra output that is controlled by a bipolar mix knob that can invert the envelope shape and also gives some extra overall ‘sink’ or ‘lift’ when the output level is increased. Rate can be set from a glitch to about a minute. When modulating decay times with control voltages it is good advise to keep the voltage fixed while the envelope is developing. E.g. trying to modulate the decay time with an audio rate signal does in general not produce sensible results. A S&H is integrated into the module to sample the decaytime modulation input signal for the second envelope generator on every new gate trigger. This way the modulation amount will stay fixed until the module is triggered again by a new gate pulse. The sampled signal is also brought out on a connector, so it can be routed to a CV input on the first ADBDR envelope generator. Or be used in any other S&H application. The ADBDR envelope is designed in a way that is hás to finish its attack phase to reach its peak level before it can be retriggered. When used for sequencing this allows for complex envelope shapes that give interesting rhythmic effects, but when used for keyboard play it might feel a bit strange to play the module with long attack times and fast play. Note that there are no attenuator knobs to set the amount of modulation for the ADBDR envelope CV inputs, these inputs are at full sensitivity. They can be connected directly to e.g. the velocity CV or CC# CV outputs of a MIDItoCV converter, but when modulated from other sources one might need an extra CV mixer module to set the modulation levels properly."

The Dual Phaser is more of a complex multiple filter system than your average, bread & butter phaser. The two phase shifters are independent and can be used separately or can be used in parallel or series. The sound is lush, precise, punchy, crispy, deep and organic. It delivers wonderfully at the classic sweep on the pad/noise patches, but also gives more unusual vocal, liquid and percussive sounds when patched in a less conventional way.

Here's a more detailed description, along with the panel layout :

"The internal CV voltage scale is 1V/Oct. Each phaser has a reasonably accurate one volt per octave direct control input that can track the keyboard voltage. Normalization is used, routing the V/Oct input signal of phaser1 into phaser2 when the phaser2 V/Oct input is left unplugged. Total control range is about 18 octaves. The Frq knob goes over the top 9 octaves of this range. Through the V/Oct and Modulation inputs you can go deeper, but you get into the LFO range and audible phasing effects would disappear. It is however possible to use the phasing effect on LFO control signals in the 1Hz to 10Hz range by supplying the V/Oct with e.g. a fixed -5V control signal, which can create quite interesting LFO effects on e.g. drones. All inputs and outputs are DC coupled, so CV signals can pass the module equally well as audio signals. Only the internal resonance is AC coupled, so resonance drops off below roughly 10Hz. Additionally each phaser has a modulation input, also at 1V/Oct when the mode is set to sweep. When the mode is set to spread it behaves like the modulation sensitivity is halved, also when it is in half mode where only half of the poles in each phaser are modulated by this input. These inputs are not normalized, in fact if no plug is connected the modulation level knobs receive a fixed voltage so a manual spread value can be set. Audio input is maximum 12V peak/peak before clipping occurs and there is 6dB attenuation from input to output to enable resonance peaks without clipping. Audio routing is as follows: If a jack is connected to input1, and if input2 is unconnected, then the audio will route into both phasers. In this mode you can use the two phaser outputs as a stereo signal. Connecting a jack to input2 will override this internal input1->input2 connection and separate both phasers. If audio is routed into input1 and if input2 is left unconnected, and if a jack is connected into ónly output2, then the two phasers are automatically set to "inverse parallel" mode. Meaning that if both phasers are set to exactly the same knob settings the phaser outputs would be in exact reverse phase and thus result in almost silence. If audio is routed into input1, and if output1 is connected with a short cable to input2, and if output2 is taken as the overall output, the two phasers are in series and thus result in one 16-pole phaser. To summarize: you can use the phasers fully separated, parallel with two (stereo) outputs on one input signal, parallel with mono output but with one phaser in reversed phase before the mixing of the outputs of the phasers take place on output2, or in series. All this is accomplished by the internal switches in the connectors and only depends on which inputs and outputs have a plug."

Another very interesting module, which I haven't covered before, is the Mini Matrix / Node Processor combo.

This combo can be associated to a Triple LF-VCO, a Dual Env or a Rungler in a triple panel.
The Mini Matrix is a buffered 6 inputs and 4 outputs routing and mixing system using TRS jacks.
Each row of the matrix can be used as a simple 4-way buffered multi when inserting mono jacks in the row's nodes.
Multiple inputs can be routed/mixed to one output using TRS jacks (with shorted tip-ring, or with a resistor between tip and ring for attenuation). The matrix comes with some ready-made shorting jacks with a female banana socket to interface with other systems using banana plugs.

Here's a more detailed description of the matrix system :

"The Active Matrix module is a fully buffered six by four matrix where any one of six input signals (rows) can be added to any one of four outputs (columns). By using ¼-inch tip-ring-sleeve insert jacks for the matrix nodes (equal to stereo jacks) a whole range of applications become possible. First the row input signal is buffered and then routed to the tips of the nodes in that row. The ring signals of the nodes are basically summing inputs and summed to the final output signals at the ends of the columns. By connecting a stereo jack where the tip and the ring are connected directly together, a connection with unity gain is made from a row input to a column output. If the tip-ring connection in the jack goes through a resistor an additional attenuation can be accomplished. E.g. a 30k resistor will attenuate by 6dB and a 91k resistor by 12dB. When a stereo audio cable is soldered to a jack plug and on the other side of the cable a potentiometer is attached, the potentiometer will act like a pot on the node, enabling to set the mix level by the pot. Basically each node is an insert, just like the inserts on a mixing desk. And can thus be used in the same way. So, using a jack with a pot means to ‘insert’ the pot into the signal path. This means that you can insert any other external device in the signal path by using an insert cable with a stereo jack on one side and two mono jacks on the other, provided signal levels match of course (e.g. 5V pp oscillator output signals will severely overload line level inputs on e.g. a digital effects rack or the guitar input of a stompbox). One could also connect a resistive sensor like a light dependent resistor (LDR) to a jack and make the node light sensitive. Each row also acts like a multiple. When a mono jack is connected into a node it will pick up the row input signal from the tip. But the ring input is now short circuited to the ground through the sleeve of the mono jack and will so disable any input from this particular node to the column output. This will not interfere with any other nodes in the same row or column, because of the full buffering of both the row inputs and column outputs. So, any node that is not used to route a signal to a column output can be used as a multiple output. Meaning that the matrix is also six multiples with one buffered input and four buffered outputs on each multiple."

Now for the Node Processor :
It consists of a quantizer with independant range and scale control,  a Gain pot that provides up to 20dB boost, a bi-polar offset generator (-5V -> +5V), two voltage controlled bi-polar VCAs and two audio tapers. In addition to the Node Processor I/O jacks, there's a Transpose input.
Node Processor functions are simply inserted into the nodes of the matrix by using stereo (TRS) patch cables.

The Mini Matrix / Node Processor combo is a crucial addition to any small to medium sized system with little or no dedicated utilities and/or mixers.

Here you can download the manual : Working with the Mini Matrix and Node Processor
All these together add up to form a powerful, flexible and versatile yet compact modular system with bags of character, attitude and loads of creative potential ... Thank you Mr. Hordijk !

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