Search dance music

Notes: An 'otherworldly' take on the classic delay pedal from Death By Audio, with the addition of a fuzz circuit allowing you to dial in grit and texture as well as ethereal atmospheres with shimmering feedback.

Supplier notes:
Lush Modulating Echo

A powerful, ethereal, and nuanced delay pedal with otherworldly controls. Push the feedback for vast worlds of colourful sound. Introduce a bit of warble with the modulator for ghastly chorus tones, or a ton of it to absolutely warp the delayed signal. Once your delay is thoroughly contorted, turn up the fuzz and pulverize it into repeating, shattered pieces. This is the preferred echo for recording engineers and touring pros around the world because, simply put, it just sounds that good. From delicate, beautiful sounds to yawning voids of fuzz and feedback, the Echo Dream II is ready to transform anything you throw at it.

Controls

Master: Sets the overall volume.
Fuzz: Sets the amount of gain, from clean to fuzzed-out bliss.
Delay: Sets the volume of the delayed signal.
D Time: Controls the delay time (numbers shown in milliseconds) between 20 ms and 1 second. Long delay times can create flutter, noise, and distortion.
FBack: Controls the feedback of the delays, from slapback to wild runaway oscillation.
Left Toggle: Up for clean blend, down to kill the dry signal and hear only the repeats.
Speed: Controls how fast the modulator runs.
Depth: Sets how far the modulator will swing the delay time. At high intervals, the pitch of the repeats will shift several semitones!
Right Toggle: Up for a smoother modulator waveform, down for a sharp, square-shaped wave.

Specifications

Dimensions: 5.87" x 4.72" x 2.35" (including hardware).
Weight: 21 oz.
Power: 9V (runs on standard 2.1mm negative center 9V adapter or included 9V battery).
Current Draw: 40 mA.

Notes: Blistering Reverb

The original synthetic atmosphere creator. The Reverberation Machine offers up drippy, saturated spring reverbs and cavernous expanses in one small package. Push the Altitude and hear the reverb blasted with gloom and distorted shimmer for the ultimate wall of sound. An instant classic for a reason, the Reverberation Machine's unique and experimental sounds are equally at home twisting your sound into a galactic nebula or adding a sinister and yearning beauty to anything you plug in.

Controls

Vol: Controls the master output volume.
Altitude Control: This is similar to gain. Turn it up to bring in a more wide open, distorted sound.
Verb: Blends in the amount of reverb.
Dark/Light Symbol: Dark cavernous reverberation to bright shimmering reverberation.

**When the Altitude and Verb controls are turned past 12 o'clock, the pedal can produce potentially harmful volumes, white noise, and other artifacts**

Specifications

Dimensions: 4.86" x 3.67" x 2.06" (including hardware).
Weight: 14 oz.
Power: 9V (runs ons a positive ring, negative tip 9V DC adapter or included 9V battery).
Current Draw: 66 mA

Notes: Loosely inspired by the look of the classic NS10 studio monitors, Yamaha's HS range offers much more modern sound. Punchy low end and clear highs from this 6.5-inch model.

Supplier's notes:
This second generation of 'HS' nearfield reference monitors, like their predecessors, share the same design philosophy as the legendary NS-10M studio monitors - Providing accurate signal reproduction and superior sonic performance.

The Yamaha HS7 employ a new, highly efficient tweeter providing an extended high frequency range up to 30KHz. The matching HS8S 150W powered subwoofer provides an extend bass when a well defined bottom end is required.

A low resonance enclosure built using a three way mitre joint technique ensures that the corners of the speaker are firmly anchored leading to dramatically improved durability and the elimination of unwanted resonances that can colour the original sound. Yamaha's 'Vortex Sound Control' technology is deployed to decrease unwanted airflows around the speaker ports. This in turn reduces audible noise by up to 6dB.

Product video: http://www.youtube.com/watch?feature=player_embedded&v=Dvj3StP4fVE

Notes: Any signal can be expressed in terms of a few parameters: frequency, phase, amplitude and timbre. It is through modulation of these aspects that synthesists create electronic sound. With Generate 3, it is now possible to create a signal and control all its parameters by through-zero modulation within a single 12 HP wide module.

Everything starts with the exceptionally precise triangle-core voltage controlled oscillator (VCO) at the module’s heart. Through-zero linear frequency modulation (FM) enables rich yet pitch-stable FM sounds, by allowing the oscillator to run backwards into ‘negative frequencies’.

The resultant triangle wave is then processed by the through-zero phase modulator, providing a total phase shift range of 900°. Phase modulation offers an interesting alternative for FM, as it enables deep FM-type tones without actually changing the oscillation frequency.

The timbre of a signal is determined by its harmonic content. This full set of harmonics can be grouped into three main parts: the fundamental, even and odd harmonics. Generate 3’s harmonics generator separates out these three timbral ‘channels’. By adjusting their relative amplitudes, a vast array of different waveforms can be created, including but not limited to the classic sine, saw and square waves.

While the amplitudes can be adjusted manually, each timbral channel supports through-zero amplitude modulation, also known as balanced or ring modulation. When audio-frequency modulation signals are applied, the harmonics become ‘split’, enriching the spectrum.

In addition to all the usual analogue synthesis techniques, the extensive set of modulation options makes Generate 3 especially suited for exploring multiphonic synthesis: a monophonic source creating a complex timbre, much like throat singing, yet by electronic means.

Notes: The Triple Waveshaper (TWS) is a non-linear modifier which can transform a sawtooth wave into a sine wave. This classic module incorporates three independent waveshapers for modifying synthesizer waveforms or for processing signals from preamplified instruments. Although originally designed as a waveshaper for early Serge oscillators, this module has been found to be a excellent modifier of electronic and acoustic sounds, and is highly recommended for subtle timbral modifications beyond the range of simple oscillator/filter patches.

The R*S version extends the classic TWS concept:

+1: An additional (4th) waveshaper has been added (right after the first / bottom waveshaper) that can be dialed in smoothly using the +1 knob. Turned down (CCW) the bottom section acts as in the original (as if that additional waveshaper would not exist). Turned up (CW) the bottom section is actually 2 waveshapers in series.

Switches to link the 3 sections - bottom to top. Using these switches all sections can be set to act in series, starting at the bottom and ending at the top, providing the combined effect of 3 to 4 waveshapers (or anything in between) using the +1 knob.

Linking only the middle and top section and turning the +1 knob up (to have 2 sections in the bottom) allows you to use the module for isntance as two independent 2-stage waveshapers (using bottom in and out for one and middle in and top out for the other) - e.g. for stereo processing.

Notes: Updated version of a truly pioneering Eurorack module that introduced DSP effects to the format. Still sounds great, still versatile, still excellent value.

Supplier's notes:
Back in 2008, the original Z5000 was not only the world's first Eurorack DSP effects module but also the first Tiptop Audio module released. It brought voltage controlled digital effects to Eurorack for the first time and the Z5000 was eventually replaced by the Z-DSP open source platform. Now, years later with over a decade of experience developing audio effects for Eurorack signals we are introducing a totally new version of the Z5000: a compact stereo 8HP multi effect with voltage controlled parameters, VC analogue clock generator driving the DSP chip and 24 of our creme de la creme reverbs, delays, modulation, harmonization and pitch modulated effects. A number of programs in the Z5000 combine more than one effect, in series or parallel, that represent studio techniques from the early days till modern times often leaning toward the more experimental side.

Notes: ZVERB is a collection of all new reverb algorithms developed by Tiptop ranging from typical reverbs to more complex effects that combine delay and pitch shifting with reverb. The effects are grouped by decades based on either the era an algorithm was introduced or the time when the effect was most popular. This variety of reverbs covers small to epic size spaces with the addition of shimmering, feedback tunnelled pitch shifting, tape delay drivers, gated reverbs and a lot more. Analog clocking of the DSP chip and CV over the parameters enables even more experimentation. This is a collection inspired by the best from the past right at your fingertips.

Notes: Instant IDM! An unusual genre-specific module from Mystic Circuits, automatically generating complex trigger sequences. Create loops or apply real-time effects to a trigger or gate input.

Supplier's Notes:
IDUM is a gate processing effect that turns its input effortlessly into IDM. Making complex electronic music with the modular usually requires meticulous sequencing done in advance or a large ecosystem of modules. Unfortunately, the level of expertise needed to accomplish this is not accessible to the average modular user. IDUM aims to make this process easier and put Intelligent Dance Music, also known as IDM, into the hands of people with average intelligence. IDUM does not require much knowledge to get immediate and compelling results; but with some extra familiarity, this module can enable innovative music making techniques. Instead of generating musical patterns on its own, IDUM acts like an effects processor for the sequences that run through it. For example, a filter or a delay allows the user to alter their sound by twisting the knobs or modulating the parameters. Similarly, IDUM manipulates the actual gates and sequences of connected modules, but with a level of playability normally inaccessible to a sequencer. IDUM can vary carefully crafted sequences or turn your music into an onslaught of controlled chaos. Its open source code and hardware based on the beginner friendly Arduino Nano Every encourage users to tweak their modules to taste or to add new features all while ensuring future improvements as IDUM makes its way into the wild.

IDUM has eight effect modes:

HOLD - Alter the length of incoming triggers.

BURST - Generates a ratcheting effect with timing determined by the clock.

MULTIPLY/DIVIDE - Generates a ratcheting effect with timing determined by incoming triggers.

BALL - Generates a bouncing ball effect.

ROTATE - scrambles the connections of inputs and outputs.

DELAY - postpones incoming gates by a dynamic amount of time based on incoming triggers.

BREAK - preset rhythms that turn incoming triggers into breakbeats.

SKIP - manipulates the clock output to multiply the clock speed or skip forward multiple steps.

In addition to these modes, IDUM features an eight step looper that loops incoming gates as well as any effects that were active on those steps.

To control when IDUM's effects are active, the CHANCE slider sets the likelihood that on a given step an effect will activate and the LENGTH slider sets how many steps the effect will last for. The PARAM knob controls a parameter unique to each effect while the MODE knob selects the mode. All of these controls as well as the LOOP status have their own CV modulation inputs.

Unique to IDUM is its ability to manipulate external sequencers using the clock output. This includes functions like: pausing the clock when an effect is active for dynamic clock division, skipping forwards or multiplying the speed of the clock in SKIP mode, and looping a sub-section of sequencer steps in the looper. To keep everything tamed while manipulating the clock, the CYCLE switch will send out bursts of clocks to catch your sequencer up with its original sequencer position after an effect has completed. Keep in mind that not all sequencers like to be clocked fast enough to work with these features, so for best results pair your IDUM with a TREE/ LEAVES combo from Mystic Circuits.

IDUM might be the stupidest way to make IDM and Mystic Circuits are proud of that. Whether you want to add complexity to your dance floor bangers or make the most obtuse and unlistenable music imaginable, IDUM is here to help. See just how dumb your music can get with the Mystic Circuits IDUM.

Notes: Most musicians use their feet as an extension of their playing, either pushing piano pedals, sustain pedals, guitar pedals.

However, feet have been underrated while using modulars. In the past ADDAC released a series of modules using sustain and expression pedals for triggering or controlling signals to help breaking the ice. While these are small and still useful for simple uses, ADDAC wanted to create something more specific this time.

This new solution features 5 footswitches and 5 inputs for external expression pedals with independent controls for each.

The aim was to save HP rack space and so created an external floor pedal with all the footswitches and controls, keeping the module HP to the minimum possible featuring only the 10 outputs and the connector to the external floor pedal.

There are many expression pedals in the market and although the most common one is the TRS pedal there are 2 other standards: RTS and TS.

All five ADDAC311 inputs expect TRS pedals, if you're looking into buying one choose a TRS pedal.

If you already have other pedals channel 4 features jumpers that can be switched in order to be used with a RTS pedal.

Channel 5 features a jumper that can be switched in order to be used with a TS pedal.

External expression pedals are required for their outputs to work.

External expression pedals are not included

Notes: Create without boundaries.

Record, produce, compose, mix, master, and perform-all from a single, intuitive application. Only Studio Oner 5 provides integrated tools that take you from initial inspiration to mastered production, from full-length album to setlist on stage. The standard for end-to-end production, Studio One is the DAW that does it all.

Drag-and-drop everything.

Studio One pioneered the often-imitated drag-and-drop workflow that continues to be a guiding principle in Presonus' design philosophy. Audioloops, virtual instruments, plug-in effects, and even presets can all be dragged into your session from Studio One's innovative Browser. Copy FX Chains from one channel to another simply by dragging-and-dropping. Convert MIDI to audio or audio to MIDI using drag-and-drop. Upload and download files to and from your PreSonus Sphere Workspaces by drag-and-drop. Ease of use is at the heart of Studio One, so your mind stays focused on your inspiration.

Perfect harmony.

The only DAW with Harmonic Editing for both audio and MIDI, Studio Oner 5 offers Chord Track, a songwriter's dream come true. Create, audition, and change chord progressions and even have older parts follow a newer, better chord structure, thanks to the inspiration Harmonic Editing can provide. With version 5.3, you can now drag chords from the Chord Track to an Instrument Track to begin composing right away. Chord notes can even be extracted directly from audio by detecting chords and then dragging the audio event to an Instrument Track.

Inspiring plug-ins.

With a complete suite of effects plug-ins, Studio Oner 5 supercharges your mixing. These plug-ins are so good we've made a few available for use in any DAW by popular demand. From State Space Modeled guitar amps to lush reverbs and polyrhythmic delays, Studio One has all the plug-ins you need to mix and produce. Studio One Professional's unique Pipeline XT plug-in lets you use boutique and vintage analogue signal processors as easily as software plug-ins. Studio One's powerful Splitter plug-in lets you run multiple plug-ins in parallel on a single channel.

Analogue mixing with digital ease.

The proprietary Mix Engine FX plug-ins provide State Space Models of classic analogue consoles that can process your mix across channels on any bus, collectively. This isn't a stereo bus plug-in, this is multi-source processing that re-creates true analogue circuit behaviour like console input-gain saturation, analogue noise, and crosstalk-all painstakingly modelled component by component from real-world hardware.

Produce without limits.

Forget laborious transposition, cutting, pasting, and moving when arranging. Audition new arrangement ideas on the fly with a simple click of your mouse. Use Scratch Pads to test arrangements. The flexible Arranger Track makes moving sections of Songs as easy as moving Parts. Powerful comping tools make assembling a perfect take simple, efficient, and fun. A robust suite of tools ranging from transient detection to multiple slicing and time-stretching features makes editing multitrack drums blazingly fast. And with groundbreaking Harmonic Editing-the most flexible implementation ever-no other program can approach the ease of arranging and songwriting that Studio One 5 delivers.

Edit audio and MIDI- fast.

Clip Gain Envelopes let you polish performances or remove glitches with adding a plug-in to your effects chain. Convert audio to MIDI and MIDI to audio with a simple drag-and-drop for blazing fast sample replacement. Replace drum sounds with samples simply by dragging the source track to an Impact XT track. The combined Arrow and Range Tools boost your audio editing efficiency by merging the most frequently used tools into one. Select, move, split, or duplicate Events; trim start and end points of Events; adjust fades; and change the overall Event level in one seamless action.

A sound as unique as you.

Inspiring virtual instruments and unique Multi Instruments let you combine instruments to create your own unique library of sounds. You'll create beats faster than ever with Impact XT, our comprehensive rhythm-production environment. Build your own sampled instrument with Sample One XT. From polyphonic analogue to monophonic subtractive synthesis, Studio One has the sounds that make you want to create. Presence XT can be expanded with Add-on packs like Deep Flight One and PreSonus Studio Grand. Aux Channels let you easily use your hardware synths, drum machines, and groove boxes as part of your productions, so dust off that TX81Z or Portastudio and put them back to work! However you create, Studio One has a tool that makes the job easy.

Notes: Field recorder meets interface in this clever offering from IK's iRig range. Essentially a 4-in/2-out audio and MIDI interface with built-in mics, the Quattro is portable, versatile and high quality.

Supplier's Notes:
Pro recording and performing, in the palm of your hand

iRig Pro Quattro I/O is a professional-quality 4 in / 2 out portable audio and MIDI interface packed with everything needed for field recording and content creation, for serious sound and flexibility - anywhere.

Small enough to fit in the palm of your hand, the iRig Pro Quattro I/O still packs in four high-quality, low-noise mic preamps; instrument, RCA and TRS inputs; plus XLR balanced, 3.5mm stereo and headphone outputs - and full MIDI I/O!

iRig Pro Quattro I/O works right out of the box with iPhone, iPad, Android, Mac and PC, but can also serve as a stand-alone mic preamp or line mixer, with a built-in limiter and direct monitoring so you're always guaranteed the perfect take.

Packed with a swiss-army-knife's worth of tools, iRig Pro Quattro I/O ensures you're ready for anything, anywhere.

As a mobile interface

Connected to a computer or mobile device, the iRig Pro Quattro I/O will work as a high-quality, multi-channel audio and MIDI interface with a full array of inputs and outputs so you'll always be prepared for any recording or need. Fully MFi-certified, no adaptors are required for iPhone or iPad for reliable field performance on any device.

As a broadcast / podcast interface

Enable STEREO mode and iRig Pro Quattro I/O mixes all sources down to stereo for streaming or recording. Monitoring is easy thanks to a dedicated headphone output with independent volume control and direct monitoring. Or use MONO mode for streaming to social media; Loopback lets you duet your favourite videos easily.

As a field recording interface

With an onboard microphone and optional stereo mic capsules, iRig Pro Quattro I/O turns your iPhone, iPad, Android device into a full-featured, professional field recorder.

As a stand-alone mixer

iRig Pro Quattro I/O also works as a preamp and mixer: just power it via battery, USB or DC IN and use the full set of inputs/outputs to capture and mix multiple sources to your DSLR camera with professional quality.

Always perfect recordings with the safety backup channels

In STEREO and MONO mode, inputs 3 and 4 can be used as safety channels: recording the same signal as input 1 and 2 but reduced by 12dB as a backup in case the sound source unexpectedly overloads the main channels.

Audio quality without compromise... wherever you are

Up to 24-bit, 96 kHz sample rate Ultra-low-noise mic preamps with full +48v phantom power Professional-quality, transparent A/D and D/A conversion Built-in limiter, for input 1-2, to protect against clipping Multitrack, stereo or mono internal mixdown.

Specifications

Microphone Input
Microphone Input: balanced female 3-pin, XLR. Pin 2: hot / Pin 3: cold / Pin 1: ground
Microphone Input Level: from 6 mVpp to 1.4 Vpp
Microphone Input Impedance: 3.2 kOhms
Gain Range: 40 dB
Phantom Power: +48V i 4V
Frequency Response: 20 Hz - 20 kHz +/-1.5dB
Input Noise: -103 dB(A), -97 dB RMS

Instrument Input
Instrument Input: unbalanced, Hi-Z, TS 1/4" Jack, Tip: signal / Shield: ground
Instrument Input Level: from 30 mVpp to 6.9 Vpp
Instrument Input Impedance: 1 MOhms
Gain Range: 27 dB
Frequency Response: 20 Hz - 20 kHz +/-1.5dB
Input Noise: -103 dB(A), -97 dB RMS
Stereo Headphone Output
Connector: 1/8" TRS Jack. Tip: right / Ring: left / Shield: ground
Headphone Level Control numbered thumbwheel
Headphone Level Power 100mW into 50 Ohms

Stereo Headphone Output
Connector: 1/8" TRS Jack. Tip: right / Ring: left / Shield: ground
Headphone Level Power 100mW into 50 Ohms

MIDI Input/Output
Connector: 2x2.5mm TS Jack
Other Input/Output
DC barrel: 1x DC barrel. For power and device charging.
Device Connector: mini-DIN

Common
Conversion: 24-bit A/D, 24-bit D/A
Sampling Rate: 44.1 kHz, 48 kHz, 88.2 kHz and 96 kHz

Power:
- USB bus powered when used with a computer
- Battery powered (2xAA included) when used with an iOS device
- DC power supply (not included, iRig PSU 3A sold separately)*

Package includes
iRig Pro I/O universal Audio-MIDI interface
Lightning to Mini-DIN connection cable with charging features (60cm/23.62")
USB-A to Mini-DIN connection cable (60cm/23.62")
USB-C to Mini-DIN connection cable (60cm/23.62")
2.5mm TRS male to MIDI female adapter (10cm / 3.9")
2 x 1.5V AA Battery
Velcro strip

Dimensions
Size: 43mm/1.69" x 127mm/5" x 37mm/1.46"
Weight: 120g/4.23oz (batteries excluded)

Notes: Deluxe version of the impressive iRig Pro Quattro features the same high-quality portable interface with the added bonus of a full complement of accessories, including mics and travel case.

Supplier's Notes:
***Note: This version of the iRig Pro Quattro I/O comes with 2 x Mic XY stereo microphones, foam windscreen, and a carry case. ***

Go pro all the way

For a little more, you get a lot with the iRig Pro Quattro I/O Deluxe edition. The value of the iRig XY stereo mic capsules alone is worth the upgrade. In addition to the pair of mics you'll get the optional power supply, which you can also use to keep your iOS device charged during extended sessions. Plus you'll get a custom carrying case to protect and transport your Quattro I/O and XY mics wherever your field work takes you.

Pro recording and performing, in the palm of your hand

iRig Pro Quattro I/O is a professional-quality 4 in / 2 out portable audio and MIDI interface packed with everything needed for field recording and content creation, for serious sound and flexibility - anywhere.

Small enough to fit in the palm of your hand, the iRig Pro Quattro I/O still packs in four high-quality, low-noise mic preamps; instrument, RCA and TRS inputs; plus XLR balanced, 3.5mm stereo and headphone outputs - and full MIDI I/O!

iRig Pro Quattro I/O works right out of the box with iPhone, iPad, Android, Mac and PC, but can also serve as a stand-alone mic preamp or line mixer, with a built-in limiter and direct monitoring so you're always guaranteed the perfect take.

Packed with a swiss-army-knife's worth of tools, iRig Pro Quattro I/O ensures you're ready for anything, anywhere.

As a mobile interface

Connected to a computer or mobile device, the iRig Pro Quattro I/O will work as a high-quality, multi-channel audio and MIDI interface with a full array of inputs and outputs so you'll always be prepared for any recording or need. Fully MFi-certified, no adaptors are required for iPhone or iPad for reliable field performance on any device.

As a broadcast / podcast interface

Enable STEREO mode and iRig Pro Quattro I/O mixes all sources down to stereo for streaming or recording. Monitoring is easy thanks to a dedicated headphone output with independent volume control and direct monitoring. Or use MONO mode for streaming to social media; Loopback lets you duet your favourite videos easily.

As a field recording interface

With an onboard microphone and optional stereo mic capsules, iRig Pro Quattro I/O turns your iPhone, iPad, Android device into a full-featured, professional field recorder.

As a stand-alone mixer

iRig Pro Quattro I/O also works as a preamp and mixer: just power it via battery, USB or DC IN and use the full set of inputs/outputs to capture and mix multiple sources to your DSLR camera with professional quality.

Always perfect recordings with the safety backup channels

In STEREO and MONO mode, inputs 3 and 4 can be used as safety channels: recording the same signal as input 1 and 2 but reduced by 12dB as a backup in case the sound source unexpectedly overloads the main channels.

Audio quality without compromise... wherever you are

Up to 24-bit, 96 kHz sample rate Ultra-low-noise mic preamps with full +48v phantom power Professional-quality, transparent A/D and D/A conversion Built-in limiter, for input 1-2, to protect against clipping Multitrack, stereo or mono internal mixdown.

Dimensions

Size: 160mm/6.29" x 90mm/3.54" x 40mm/1.57"
Weight: 340g/11.99oz (without batteries)
Cable length: 60cm/23.62"

Box contents

iRig Pro Quattro I/O
1/4" thread adapter
4x AA Batteries (alkaline)
iOS Lightning to mini-DIN cable
USB-A to mini-DIN cable
2.5mm jack to MIDI female adapter
2 iRig MIC XY stereo microphones
Windscreen
9V PSU
Travel case

Notes: The MiniMod Multi Burst Envelopes module generates variable length bursts of repeating envelopes that can change over time, with CV control of all functions.

The BURSTS control selects a "Burst Sequence" of between 1 and 40 individual envelopes. The envelope shape of a burst is chosen with the rotary switch, or it can be varied using an external CV, so that it is even possible to chose a different envelope shape for each individual burst in a sequence.

The width of the envelope (i.e. the attack and decay time) is varied with the P-WIDTH control and this can also be CV controlled as a sequence is progressing, so it is possible to vary the envelope speed of each individual envelope burst while a Burst Sequence is running.

The repeat time of individual bursts in a Burst Sequence is set with the LFO Rate control. The LFO rate can be varied with an external CV as a sequence is progressing, so the gap between envelopes can speed up or slow down over time. An external clock can also time the bursts in a burst sequence instead of, or in addition to the internal LFO. Bursts can also be manually incremented with the CLK push button.

The SLOPE control allows Bursts to start quietly and build in level to maximum, or they can start at full level and then reduce in level as the Burst Sequence progresses. In the centre position all Bursts are of equal level.

Specifications

Module Width: 12hp
Module Depth: 26mm (Including power lead)
Current Usage: 28mA Positive, 12mA Negative
Individual Envelope Length: 20msecs to 2.5 seconds
Internal clock rate: Off, 0.24Hz to 51Hz, or every 4.2 seconds to every 20msecs
Envelope waveshapes: Sine, LogDown, LogUp, ExpoDown, RampUp, Triangle, RampDown, Pulse
Repeats: 1 to 40
Slope: bipolar, sloping up from 4%->100%, and sloping down from 100%->4%

Notes: Atlas Compressor

The Atlas Compressor offers a comprehensive collection of compression circuits that perfectly capture the tone and feel of the industry's most sought-after compressors. Straight out of the box, Atlas delivers six different styles of compression, ranging from classic stompbox compressors like the now-discontinued Diamond Optical Compressor to high-end rack units such as the snappy 1176 compressor and the smooth and gentle LA2A optical compressor. Atlas also features advanced dual-band compression, which provides an independent set of controls for both the high and low frequencies. Each compression engine has onboard controls for Threshold, Ratio, Attack, Release, Output, Tone, and Blend, allowing for fine-tuned control over the compact One Series hardware.

A History Lesson in Compression

The Atlas is packed with so many different types of compression that it could serve as a tour guide through the history of compressor equipment. First stop: classic stompbox squashers such as the legendary the Dan Armstrong Orange Squeezer and the excellent (but discontinued) Diamond Compressor. Next stop: universal and ubiquitous studio rack gear like the smooth and gentle LA2A optical compressor and the ultra-responsive 1176 FET compressor. Finally, we're taking you to all-out nerd territory with Peak, RMS, VCA, and Dual compression, and an onboard Expander circuit... the Atlas has it all and does it well.

Optical Rack

The Optical Rack compressor engine is based on the legendary LA2A studio opto compressors from the early 60s. Optical compression controls its output level with light. Increasing the incoming audio level in the original units caused an electro luminescent panel to shine brighter. A photo resistor monitors that light panel and decreased the audio output level, as the panel got brighter. The result is a very smooth and gentle attack that imparts a full-bodied and natural sound on guitar, bass, or vocal. This compressor sounds great with clean strummed or arpeggiated guitar lines as well as acoustic guitars.

Studio 76
The Studio 76 compressor is based on the 1176 studio rack compressors of the late 60s and early 70s. This solid state FET compressor is known for its full body and super-fast attack. All of the attack times in 1176 compressors clock in at less than a millisecond. Studio 76 is a great compressor for fast and aggressive country pickin' (think Johnny Hiland), slap bass, slide guitar and much more.

Cubic Zirconia

The Cubic Zirconia was inspired by opto compressors that use an LED in place of the LA2A's electro luminescent panel. This technology is usually what you find inside effects pedals. Using an LED as an optical compressor's light source creates faster attack and release times, which generate artifacts that tend to make the compression sound a bit dirtier. This is a great sound for playing tight and funky riffs on the guitar.

Studio snap

A very aggressive and modern sounding compressor. It's based on hard-knee, VCA feed forward type compressor. It uses two compressors in parallel to handle both RMS level controls ("Root Mean Square" compression means that the compressor will react according to the "average loudness") AND the peaks. The combination of these two compression approaches gives STUDIO SNAP its character.

Dual band

The Atlas has two independent compressors that can be routed in a number of different ways. They can run in Series Mode (one running into another). They can also run in Parallel Mode and be mixed together with a Crossfade control. The Dual Band compressor runs in Band Split Mode, which is a version of parallel routing where one compressor controls only the higher frequencies (everything above 333Hz) and the other controls the lower frequencies (everything below 333Hz). With the Dual Band Compressor the high frequencies release faster than the lower frequencies, so as you sustain a chord the low frequencies remain compressed while the second compressor releases the high frequencies. The result is a beautiful blooming effect that accentuates high harmonics. This is a great compressor for slow ringing chords or lead lines with long sustained notes.

Includes 9v DC power supply (300mA / Negative Tip)
Includes 1/8" to 1/4" cable for Neuro Mobile App connection.
Power Consumption - 165mA
USB port - Connects to Mac or Windows PC for firmware updates and the forthcoming in-depth effects editing software.
Brushed anodized aluminum housing.
Dimensions: L: 4.5 in./11.4cm x W: 2.75 in./7cm x H: 2 in./5.1cm (including knobs)

Notes: Wave Expander bridges the gap between the Melody Miracle and Touch Tones synthesizers, and the rest of the synthesizer world. Control other synths by sending CV and gate signals via Wave Expander, and dial in the right amount of control with onboard CV tuners. Outputs are visually indicated with the Chrominancer color window at the center of Wave Expander, bringing cosmic insight to the signals at hand.

Melody Oracle and Touch Tones expander module
Use AMP synthesizers to control external gear
12 outputs (6 CV/6 Gate)
CV Range: 0V to +12V
Chrominancer color window

Notes: Unique touch-activated instrument with built-in speaker, allowing you to trigger notes with one hand and modulate the timbre via touch pads. Unorthodox tones, unorthodox interface. Lots of fun.

Supplier's Notes:
Touch Tones is a touch activated auto harmonic synthesizer, with a 3 octave tuning range, XOR logic modulator, Harmonicon auto harmony generator, and two onboard bandpass filters. The left side of the Touch Tones holds three capacitive touch pads for playing notes, while the right side contains a set of 7 touch pads to modulate the different parameters sequencing the synthesizer.

Three octave range plus XOR modulator.
Harmonicon automatic harmony keys play a perfect 5th of each tuned ROOT note.
Dual Luma filters for active timbral shaping via light control.
Wear the Touch Tones for Function or Fashion with the included chain lanyard.

Notes: Hue is an audio colour processing effect, in 6HP eurorack format. Hue is comprised of a chain of 5 effects (Drive, Filter, Tape Saturation, LoFi, Compressor). Each effect imparts a particular coloration, tone, distortion, and/or texture to the source audio. With Hue, Drums bang with more intensity & melodies morph into lush LoFi tones.

Hue's goal is to simply colour your sound. The initial concept was born of a debate about the techniques and processes used to make drum machines sound big, bold, and delicious. The sounds that tug at the hearts of boombap, LoFi, and subsequently modbap, enthusiasts are those that have great texture, lush degradation, soft distortion, and big bold strokes of colour. The classic beloved drum machines were often processed with outboard gear, recorded to tape, pressed to vinyl, played in big booming systems, sampled, and resampled and on and on. Ultimately, those are the sounds that become nostalgic and reminiscent of all that we love about classic LoFi boombap production. Hue is Modbap's attempt at bringing that vibe into eurorack with a cute little 6hp package.

The effects chain audio signal path is as follows; Drive, Filter, Tape Saturation, LoFi, Compressor. The panel layout allows for the DJ style filter knob to sit in first position for ease of tweaking. Drive boosts and lightly distorts the signal, while shift+Drive adjusts the Drive tone. The Filter is a low pass filter to the left and a high pass filter to the right. Tape effect is intended to impart cassette tape saturation, while shift+Tape adjusts the intensity. LoFi adjusts bit depth, while shift+LoFi adjusts sample rate. Lastly, the one knob compressor mini pot serves as the final glue in the signal path that brings it all together.CV inputs for Drive, Filter, Tape Saturation, and LoFi opens loads of creative possibilities. Hue is a textural beast when creative modulation is thrown at it.

Hue puts the power to shape and transform your sound at your fingertips. It's great for beefing up drums and equally as magical on melodic content. Hue can be the glue that brings it all together. It also, pairs well with Trinity and Osiris. All in all, Hue exemplifies the tagline "Made for eurorack, dope enough for boombap!"

Tech Specs

Width: 6HP
Depth: 28mm
Power: 50mA +12V, 50mA -12V

Features
5 effects (Drive, Filter, Tape Saturation, LoFi, Compressor.)
4 CV inputs for modulating the effects
Mono in and mono out
6HP

Notes: A fresh twist on a simple 303-style acid synth, featuring a triangle/sawtooth/square wave VCO, multi-mode resonant filter, simple AD envelope and a VCA which can be overdriven. Excellent value.

Supplier's Notes:
ADDAC started this module with the idea of developing a complex drum source however, somewhere along the process, it was clear how much better it performed as a synth voice and this lucky accident was embraced.

It features a VCO with a [FREQUENCY] and [FINE TUNE] knob plus a dedicated CV Input and Attenuator knob (tunable over 4 octaves).

The VCO waveform output is achieved by selecting either a Triangle or Saw through a switch. the selected waveform can then be mixed/balanced against the square wave. The resulting mix is then sent to the Filter.

The Filter features a [CUTOFF] and [RESO] resonance knob plus a Cutoff CV Input and Attenuverter knob.

A 3 position switch is used to select the filter type: Highpass, Bandpass or Lowpass.

The resulting output is then sent to the VCA.

The VCA features an Input with [INPUT GAIN] knob which at maximum can amplify the incoming signal by a factor of 2.

This is a very important control, it accepts any signal Trigger, Gate or CV. Whatever input is plugged in the signal is then fed through an AD with a very short attack and controllable decay through the [VCA DECAY] knob plus the CV Input and Attenuverter knob. The resulting slewed signal is then used to control the VCA gain.

This signal is also sent to the CV OUTPUT as well as the LED monitor.

The Accent input adds to the Input signal creating a different amplitude output.


The CV OUTPUT is normalled to the Frequency and Cutoff inputs.

Tech Specs:
9HP
4cm deep
80mA +12V
80mA -12V

The [INPUT GAIN] knob

Usually Attack/Decay envelopes have a maximum voltage of +5v, no matter if the input gate is +5v or above the AD will clip at +5v. In this case, this clipping method was not used, and instead ADDAC allowed the incoming voltage to determine the maximum AD voltage, meaning that if a +5v gate is present then the AD maximum voltage will be +5v but if a gate of +10v is sent then the AD maximum voltage will be +10v.
This also means that with higher input voltages the decay, although falling at the same speed, will be longer than with lower voltages as it has a longer range to go back to 0v.

As previously mentioned, The [INPUT GAIN] knob can amplify the incoming input by a factor of 2, allowing to use a standard +5v gate or envelope and being able to make the resultant AD go up to +10v.

The AD signal is responsible for opening the VCA. Up to +5v the VCA will open to unity gain above this value the VCA will start to amplify and eventually saturate and distort. This saturation will add harmonics to the signal which will change its gentle timbrical nature to a more unique and peculiar timbre that will make the module shine in Acid contexts.

Adding high levels of Resonance or even filter self oscillation in combination with high VCA saturation will create even more harmonics that ADDAC highly encourage the user to explore.

Notes: The DISTURBANCE is an extreme filter, flanger, and phaser with modulation like you've never heard before.

DISTURBANCE's controls allow for a wide range of sounds to be sculpted perfectly to your needs. The 'Tensity' knob changes the feedback amount and structure of all effects, with two totally different intensity options to choose from for each disturbing modulation. The Filter allows for deep swells and piercing resonance with highpass and lowpass options.

The Flanger creates a classic modulation sound with options for watery, multi-tap delay matrixes or a screaming flange with distorted feedback. Last but certainly not least, the Phazer mode is a psychedelic 8-stage phaser sound with options for mellow filtered scoops on one side and combination phaser/flanger on the other.

But that's not all - the pedal introduces a brand-new way to control an effect: The custom designed modulation core lets you pause the LFO where it stands and listen to the strange world of frozen modulation. Not only does this new system control its own sounds, the DISTURBANCE includes a CV OUT jack which lets the modulation control signal be sent to whatever device dares let it in.

Go forth and DISTURB and bend, break, and twist your sound!

Notes: Great value from 4ms's twist on the Stereo Triggered Sampler module, this time capable of playing and manipulating stereo or mono WAV files from an SD card.

Supplier's Notes:
The Sampler is a variation of the Stereo Triggered Sampler.

Record, play and loop stereo .wav files with an SD Card. Includes the same 2022 Sampler Pack from the Stereo Triggered Sampler.

Features:

The Sampler plays stereo or mono .wav files from an SD Card. Samples are arranged into banks by placing .wav files into folders (each folder is a bank). A maximum of 10 samples can be loaded in a bank. There is a maximum of 60 banks (600 samples maximum total).
Sample files are limited to 4GB each, which is over 6 hours at 44k/16b/stereo.

Hundreds of samples included on microSD card
Extremely quiet, low noise and low jitter design
Stereo or Mono output

A knob and CV jack selects the Sample to play, and the Start Position knob and CV jack control where in the sample playback will begin. The Length knob and CV jack determine how much of the sample is played back: from tiny grains, to percussive hits, to longer loops... or just the entire file.

Pressing Play or firing a trigger into the Play Trig jack will play the sample. Holding the Play button down for a second will toggle looping.

There's also a Reverse button and jack for backwards playback, and an End Out jack for synchronizing loops, or cascading sample playback with other events.

To record samples, hold Play and Reverse for two seconds. Press Play again or fire a trigger into the Play/Rec Trig jack to start and stop recording. Each recording will create a new stereo 48kHz 24bit .wav file in the current bank's folder. Recordings can be as long as your card has the capacity for: a 256GB card is 18 days of continuous recording (432.9 hours). Audio data is split into maximum 4GB wav files on the microSD card.

Play button/trigger jack: Tapping the button or receiving a trigger starts/re-starts the sample
Sample CV/knob: select which sample within the channel's bank
Pitch CV and Pitch knob: playback pitch (-10 to +4 octave range)
Start Pos CV/knob: position in the sample to begin playback (loop start point)
Length CV/knob: playback time (loop end point)
Length knob at max: entire file is played
Length knob 50%-99%: 200ms - 5s of the sample is played (beginning at the Start Position)
Length knob <50%: Percussive decay envelope is applied to the playback (attack-only envelope if playing Reverse)
Length <1%: Tiny grains are played (can be used to sweep through sample file data manually)
Reverse button/jack: toggles forward/backwards
Bank button: selects channel's bank
Reverse + Start Pos button/knob combo allows for setting volume of playback
Maximum file size for playback: 4GB per file (6.7 hours per file at 44.1kHz/16b/stereo)
microSD Card is hot-swappable when Busy light is off: no power-down required to change cards.
Samples that are pre-loaded into memory can continue playing while card is swapped.

File Organization:

Using a computer, the intuitive automatic bank loading features make dragging and dropping your entire sample library a snap. No files need to be renamed. Just create folders and put up to 10 samples in each. If you want a folder to be associated with a particular colour of the Bank button, name the folder a colour name (such as "Blue - Loops").

Ships with a high-speed 16GB microSD card, loaded with hundreds of samples created for 4ms, from musicians such as Richard Devine, Wobbly, Moor Mother, Robert Aiki Aubrey Lowe, Joseph Pailo, Russell E.L. Butler, Hugo Paris & Marc Kate, Paul Barker, Baseck, Animal Collective, Cristian Vogel, Jay E, Marcus Fischer, Chris Carter, Daedelus, Nero Bellum, Anna Sitko, Thomas Fang, Bogdan Raczynski, and more (Sample List)

Intuitive bank/colour organization system; no need to rename your wav files
Record new samples directly to a bank with the Record Input jacks...
... or drag your folders of sound files onto the microSD card, and each folder will become a bank.
Folders starting with a colour name will be loaded into that colour bank
Example "Blue - Frooty Loops" will load into the Blue bank
Sub-folders are ignored, only folders in the root directory are scanned for wav files
Works best with UHS Speed Class 3 cards with |3| symbol printed on the card

Specs:

Electrical and Mechanical Specifications:
Module Size:
16 HP Eurorack format module
10-pin Eurorack power header
1.3" (33mm) maximum depth with power cable
Power consumption:
+12V: 135mA
-12V: 41mA
+5V rail: not used
Connect red stripe of power cable towards the bottom of the module, which is marked with a white stripe and the words -12V and POWER
Audio Inputs:
20Hz to 10kHz (+/-0.1dB); -6dB rolloff @ 20kHz
21V peak-to-peak maximum before clipping
Audio Outputs:
0Hz (DC) to 22kHz with maximum -1.8dB difference between file data and output
+10.5V to -10.5V maximum output
Trigger inputs:
Latency: 0.7us typical

Included with the Sampler:
16GB micro SD card (class 10)
SD adaptor
10-to-16 pin power cable
M3 Knurlie screws

Notes: Pan and cross fade up to three signals with AJH's versatile Triple Cross. Four different modes allow you to set up the three channels in numerous ways, for both audio and CV processing.

Supplier's Notes:
The Tone Science Triple Cross module is a design collaboration between electronic musician/composer Ian Boddy and AJH Synth.

It features three independant cross fader and panner channels, and it can be used with both audio and CV signals. Each of the channels can be configured in one of four different ways:

1) One In, One out Linear VCA

2) Two in, One out crossfader

3) One in, Two out panner,

4) Two in, Two out cross fader & reversing panner combined.

Audio rate modulation can also be used on the CV inputs to create Ring modulation and Cross modulation type sounds. Each channel has a dedicated Fade control, CV input, and CV Level attenuverter. These can be used together to bias incoming CV signals.

The L1 and L2 outputs from channels 1 and 2 are normalised to channel 3's inputs, which also have their own level controls to balance incoming signals. This allows very interesting, complex fading and panning between four different input sources, to either Mono or Stereo outputs..

The Triple Cross is very versatile, yet relatively simple to understand and use. However, as they say "knowledge is power", so do spend a little time reading about how to configure the channels in the four different modes, and how to combine all three channels together to create a super-powerful Four to Two to One cross fader.
Each of the three channels can be configured in four different ways, and all three can be used simultaneously. They do not all have to be in the same mode, so we could for example use channel 1 as a cross fader, channel 2 as a Panner for completely different signals, and channel 3 as a Linear VCA, again for different signals.

Specifications

Effective CV Range for CV-1, CV-2 and CV-3 is 0 to +5V with attenuverter fully clockwise or 0 to -5V with attenuverter fully anticlockwise.
Maximum permissible CV Range for CV-1, CV-2 and CV-3 is -12 to +12V
IN 1A to IN 3B Input levels -5V to +5V, levels up to +/-10V are possible however some clipping may occur.
Module Width: 14hp
Module Depth: 26mm (Including power lead)
Current Usage: 80mA Positive, 72mA Negative

Notes: The Tone Science Triple Cross module is a design collaboration between electronic musician/composer Ian Boddy and AJH Synth.

It features three independant cross fader and panner channels, and it can be used with both audio and CV signals. Each of the channels can be configured in one of four different ways:

1) One In, One out Linear VCA

2) Two in, One out crossfader

3) One in, Two out panner,

4) Two in, Two out cross fader & reversing panner combined.

Audio rate modulation can also be used on the CV inputs to create Ring modulation and Cross modulation type sounds. Each channel has a dedicated Fade control, CV input, and CV Level attenuverter. These can be used together to bias incoming CV signals.

The L1 and L2 outputs from channels 1 and 2 are normalised to channel 3's inputs, which also have their own level controls to balance incoming signals. This allows very interesting, complex fading and panning between four different input sources, to either Mono or Stereo outputs..

The Triple Cross is very versatile, yet relatively simple to understand and use. However, as they say "knowledge is power", so do spend a little time reading about how to configure the channels in the four different modes, and how to combine all three channels together to create a super-powerful Four to Two to One cross fader.
Each of the three channels can be configured in four different ways, and all three can be used simultaneously. They do not all have to be in the same mode, so we could for example use channel 1 as a cross fader, channel 2 as a Panner for completely different signals, and channel 3 as a Linear VCA, again for different signals.

Specifications

Effective CV Range for CV-1, CV-2 and CV-3 is 0 to +5V with attenuverter fully clockwise or 0 to -5V with attenuverter fully anticlockwise.
Maximum permissible CV Range for CV-1, CV-2 and CV-3 is -12 to +12V
IN 1A to IN 3B Input levels -5V to +5V, levels up to +/-10V are possible however some clipping may occur.
Module Width: 14hp
Module Depth: 26mm (Including power lead)
Current Usage: 80mA Positive, 72mA Negative

Notes: Part of Xaoc's Leibniz Binary Subsystem, the updated Drezno ADC/DAC module builds on the original with higher resolution converters, allowing you to use it for pitch quantisation.

Supplier's Notes:
Drezno II is an improved version of the original Drezno module. It is an essential member of the 8-bit Xaoc Devices Leibniz Binary Subsystem. The Leibniz subsystem is a family of modules that operate on 8 binary (two-state, on/off) signals. These signals are arranged as 8 bits of digital data and may represent analogue signals and voltages, or they may be whatever you want them to be (clocks, gates, rhythms, etc.).

Drezno features an ADC (analogue-to-digital-converter) and DAC (digital-to-analogue-converter), allowing it to act as an analog front-end of the system. In other words, Drezno allows one to convert analogue signals and CVs into digital numbers, as well as to obtain analogue signals back from some other digital numbers (or 8 binary signals). Between these two conversions there is a whole universe of digital shenanigans waiting to happen resulting in all kinds of signal transformations and modulations.

Both (ADC and DAC) sections of Drezno can work independently, or they can be linked. Input and output range of analogue signals may be adjusted using Gain and Offset sliders. Individual bit outputs (for the ADC) and inputs (DAC) allow for extensive cross-patching and the creation of complex waveforms and rhythms. A very fast clock (up to 2MHz) at which the conversions are performed may be replaced with any external clock (for each section separately).

Compared to its predecessor, Drezno II uses new converters of much higher resolution which results in much lower noise floor and higher accuracy. The lower bits are much more stable than before. It is easy now to use Drezno for quantizing pitch voltages with accurate results with 128 or 256 semitone steps. Potentiometer ranges are factory calibrated for precise chromatic scale. A new switch on the front panel allows the user to select the calibrated range of 10Vpp or 20Vpp, so that the lowest bit or the second bit correspond to 1 semitone.

Technical details

Eurorack synth compatible
12hp, skiff friendly
Current draw: +80mA/-50mA
Reverse power protection

Notes: The SoundForce Chorus 6 is 16HP Juno style BDD chorus with plenty of CV inputs. It is using the Xvive MN3009 and the same audio path as the original circuit. It can do the classic Juno I, II and I+II modes. I+II modes are available from both the Juno 6 and Juno 60, each in mono (as the original) or stereo, giving you 4 I+II options. But the manual and external modes can take the module away from sweet and pretty sound territories right into weird and noisy soundscapes.

Some tech-talk:

A chorus effect is created by mixing a dry signal with a delayed signal. The delay time is modulated by an LFO to create more movement. To create some stereo width, the left and right modulation are 180 degrees out of phase. Exactly like the chorus in the Juno 6/60/106, the delay is made with MN3009 BBD chips. Also as in the original design, filters are added both before and after the BBD chips in an attempt to hide some of the artefacts of this very crude IC.

To get the audio "moving" through the chain of MOSFETs in the BBDs, driver signals are needed. This used to be done with the MN3101 BBD driver chip. Unlike the MN3009, there is no clone available of this ancient IC and this was a chance to add some modernity to the design. The drivers were replaced by a modern STM32 micro-controller along some extra ICs to convert the 3.3V logic to the negative voltage needed by the MN3009.

In order to recreate a genuine Juno-type modulation, the BBD driver signals of my Juno-106 were sampled over a few periods. Using the timer peripherals of the STM32, a replica of the BBD driver signals can be sent out to the MN3009's. Although the driver signals only follow a very basic triangle LFO shape and could have been generated using direct synthesis in the firmware, this method gives an exact copy of the original driver signals including all the potential imperfections.

Implementing a digital programmable chip also allows a digital-style interface with tactile switches and LEDs as in the original Juno interface. The last used mode is saved and loaded when the module powers on. The CV inputs (with the exception of the MIX CV) are acquired by the STM32 ADC and can modulate the BBD driver signals based on parameters in the firmware.

Features:

The SoundForce Chorus 6 has 4 main modes: I, II, I+II, MAN for manual and EXT for external. I and II are the classic Juno modes available in the Juno 6/60/106. Press both I and II to enter the I+II mode. While holding down I and II, press the OFF button to cycle through the 4 I+II options:

- Juno 60 mono - 1hz - 8% depth
- Juno 60 stereo - 1hz - 8% depth
- Juno 6 mono - 8hz - 25% depth
- Juno 6 stereo - 8hz - 25% depth

In I, II and I+II, only the mix controls can influence the sound, all the other parameters are fixed and CV inputs are not active.

Mix control:

Using the slide pot (fader) or the CV input, the mix between the dry input signal and the BBD delays output can be set. The orignal Juno balance is reached when the slide pot is in the middle (without CV connected). The mix control is a VCA-based crossfader and is 100% analog which means it can withstand really high audio rate frequency modulation.

MAN mode:

The manual mode gives you the opportunity to control every parameter of the chorus and gives you acces to the LFO that modulates the BBD delay times. Here is what you can tweak:
- time slide pot: sets the start delay time
- rate slide pot and CV input : sets the frequency of the delay time modulation
- depth slide pot and CV input: sets the depth of modulation around the start time
- mix slide pot and CV input: see above
- LFO wave: selection of different waveforms to modulate the delay time
- range: selection of 3 different frequency ranges for the LFO (S: 30Hz to 500Hz | M: 1Hz to 100Hz | L: 0.016Hz/1min to 10Hz)

In manual mode, the EXT IN CV input can be used to change the LFO wave, range or add additional modulation of the delay start time.

EXT mode:

In external mode, the BBD delay times can be modulated using an external signal through the EXT IN cv input. Plug in a CV source from your eurorack case into the EXT IN, set manually the starting delay time using the time slider and use the EXT ATT slide pot to attenuated the incoming CV signal (equivalent to setting the depth). In external mode, the depth/rate sliders and CV inputs are not active.

Stereo inputs:

The Chorus 6 can be used with either mono or stereo input signals. When the right channel is unconnected, the left channel is sent to the right jack so it can be used with mono signals. When using both inputs, each input signal is going through its own BBD delay line.

BBD delay time range:

The Chorus 6 gives you in the MAN and EXT modes a delay range that it is about doubled compared to the I/II modes. The original Juno chorus was using only a very limited range of the BBD chip and really stayed in the sweet spot. Extending to the full spectrum of the MN3009 yields pretty horrible high pitch noise at audio rate modulation, so the range has been limited to a usable section. When you apply high frequency modulation with great amount of depth, the delay time will be pushed in the extremities of its time range and quite nasty artifacts will appear in the form of squeaks and whines. Please note this is a characteristic of the BBD. If you want to reduce those artifacts, reduce the depth of modulation. On the other hand, if you are looking to add characteristic and vibe to the sound, audio rate modulation and plenty of depth will help you to get there.

Input levels:

Both Eurorack and Line level signals can be used as input for the Chorus. A small slide switch between the 2 PCBs can switch between Euro and Line. In Euro levels, the input signal is attenuated to an healthy level that is well below the BBD distortion point. Some headroom is also needed as the chorus is a sum of the input signal and the modulated delayed signals. For that reason, when the chorus is turned off the output attenuation is about 50% of the input level. When the chorus is on, the resulting output amplitude is similar to the input level. Line levels are also accepted by the module when the switch is on Line. Please note that Line levels input will be amplified to the Eurorack levels range at the output of the module. So the output levels are Eurorack in both Euro or Line input ranges.

Specifications:
16HP
Depth: 38mm (including power connector)
Power: +12V 76mA, -12V 35mA

Notes: Roucha Legio is a 12 dB /octave state-variable stereo multimode filter available on the Legio DSP+oscillator platform.

Noise Engineering received tons of requests for NE filters over the years, and Roucha Legio is their first dedicated release. The filter features lowpass, bandpass, and highpass settings and tracks frequency via the dedicated encoder and extremely accurate 1v/8va CV input. Roucha Legio's CV-controlled resonance parameter can go from clean to acidy-squelchy for bassline joy. Add CV for filter sweeps that will knock your socks off. Of course, because it's a Noise Engineering module, there's a Fold knob. Add wavefolding to the signal chain before, after, or both before and after the filter.

Noise Engineering have made it easy to carry out dramatic buildups and cuts with Roucha Legio. Just tap the encoder to bypass; tap again to turn the filter on again. Want to automate it? Send a gate into the Bypass input. The filter will remain bypassed as long as a high gate is present.

Roucha Legio is also a DSP and oscillator platform: users can change the firmware on any Legio module, 100% free, to any other firmware in the Legio series.

Dimensions
6 HP
38 mm deep

Current Draw
140 mA +12V
22 mA -12V

Notes: No compromise on safety and security

When it comes to transporting your valuable equipment, there can be no compromise on safety and security. The Reloop Premium Large Controller Case is the ideal choice for DJs who need a reliable and secure case that can accommodate larger controllers such as the Reloop Mixon 4 /Mixon 8 Pro, Roland DJ-8008, Pioneer DJ DDJ-SX3 / DDJ-800, Denon DJ MC7000.

Built to last

Crafted from durable wood and aluminium with a scratch-resistant black hexagon texture, these cases are built to last. They also feature butterfly locks and angular blind cuts to keep your gear safe and secure during transport. Additionally, the case includes a laptop tray with fasteners and a spring-loaded and recessed carrying handle for easy and comfortable transportation.

Fits large controllers

The Reloop Premium Large Controller Case provides DJs a durable and secure solution for transporting larger controllers, ensuring the safety and protection of valuable equipment.

Premium large controller case

Rugged case for common, larger DJ controllers
Integrated laptop tray with fasteners
Two recessed, spring-loaded transport handles & removable blinds for cables

Technical data

Colour: Black
Internal dimensions: 657(W) x 376(D) x 89(H) mm
External dimensions: 751(W) x 474(D) x 198(H) mm
Weight: 11.3 kg

Notes: ***B-STOCK: Box opened, but product is in excellent condition and in perfect working order***


Introducing SUM0, a voltage-controlled oscillator with expanded harmonic possibilities, in a Eurorack format.

The 4 classic waveforms are generated by a meticulously designed oscillation engine.

You can modulate its frequency and its pulse width thanks to its integrated LFO, capable of reaching audible frequencies to achieve expressive modulations with the simple touch of a switch.

Expand the voice of your system by creating complex waveforms and high harmonic content thanks to its output mixer. This mixer also has an input to integrate an external signal to the mix output.

Of course, all its parameters (FM, PWM, SYNC, V/OCT) are at your fingertips through its patch points on the front panel.

And of course, assembled with a load of passion in Bocuma's studio-workshop in Guadalajara.

Nerdy-specs:

22HP
+12V > 12mA -12V > 12mA

Notes: The Skorn da Bask table top module is the same module as the eurorack version but in a hammond 1590BB enclosure and aluminium panel, it can be powered thru a 2,1mm negative pin power jack, compatible with Boss and Ibanez standard power supplies.

Thanks to their routing options, the VCOs can not only be used to generate audio signals, but for modulating the VCF's cutoff frequency as well. Besides processing internally produced saw tooth waves, the filter is also usable for editing sound coming from other modules. Typical Touell Skouarn, the Skorn da Bask distorts like hell. Primary application: The creation of dirty, experimental sound structures and abstract rhythms.

The front panel of Skorn da Bask shows three areas highlighted in red. These zones contain the connectors and control elements for the oscillators. Each VCO comes equipped with a pitch CV input, a tune potentiometer and two switches for defining the routing as well as the frequency range used. In detail, the oscillators can be connected to the audio or CV input of the filter. Thanks to their range switches, each VCO is able to work with audio or low frequency rates. The pitch input does not follow the standard 1v per octave.

After all, Skorn da Bask is not an ordinary oscillator module, but an experimental sound generator.

The low pass filter features a slope of 24 dB per octave. A potentiometer named FreqaNs regulates the circuit's cutoff parameter, Dasson sets the resonance. For processing external signals, there is an audio input. As long as a cable is plugged into this socket, the VCOs are disconnected from the filter's audio path. Third-company modules, like an envelope or LFO, can influence the VCF's frequency via a CV input. For damping modulation signals coming from the oscillators, there is an attenuator knob. The audio output is equipped with a level potentiometer. High settings lead to juicy overdrive effects.

Notes: Digitally-controlled full-analogue path Granular + Multimode VCF

The Grainity brings a unique and never-heard-before concept to the world of analogue VCF.

While being based on analogue filtering only, the Grainity expands and enriches the sound in creating subharmonics, harmonies, unison, flanging effects and formant filtering all at once.

The module presents two sections, the granular filter itself and a typical multimode filter, each with its own output.

A third output offers a mix of both filter paths. Numerous controls and CV allow a very wide palette of results.

What is the Grainity

Granular?

The word granular was chosen for this filter module in relation to the concept invented by the composer Yannis Xenakis, and which is described as assembling small chops of sounds - he called grains - according to a pattern.

This is exactly what the Grainity does, no more no less ... but, digital control allows doing it in ways that are decades away from what tape and scissors allowed at the time.

Technology wise ...

In the granular section, the incoming sound cycle is determined in order to step through various filtering and phase variations of itself according to predefined patterns called structures.

The controls allow alterations to the cycling phase, cycling frequency offset (tuning), structure selection and frequency division of the filters cycling. This comes in addition to the usual filter frequency and resonance controls.

What the Grainity is not

Technology wise ...

Despite relying on a microprocessor, the Grainity is not a digital audio processor. At any time, the sound remains in the analogue domain and goes through analogue circuits from in to out.

The digital processor does not convert or generate audio signals; it is there to control the analogue circuitry and manage the interface and display.

Sound wise ...

Several digital modules offering granular processing rely on the layering of sound grains, which leads to blurry and cloudy textures.

Layering of the grains is a late addition to the concept of granular synthesis and is not a mandatory feature of the concept.

The Grainity does not use delay lines or sampling memory but is nevertheless true to the foundations of the granular synthesis concept.

The granular VCF in the Grainity is unconventional in that, most of times, it is not filtering the sound in the common expectation of removing something. Typically, the granular process adds more than it removes by expanding both the lower range, creating subharmonics, and the upper harmonic contents in multiples ways.

Dimensions
10 HP
21 mm deep

Current Draw
101 mA +12V
59 mA -12V
0 mA 5V

Notes: The Aether is a modulating delay pedal.

The initial vision for the design of The Aether came shortly after the release of the Wardenclyffe Deluxe. The intention was to take design the philosophies and incorporate tonal characteristics that made the Wardenclyffe so special, but this time as a delay pedal.

At the heart of The Aether is a PT2399, tap tempo delay. It has the standard Echo, Feedback, and Time controls you would expect on any delay, but what makes it special is the Color, Saturation, and Degrade Controls. The Color toggle completely changes the character of the Aether and has two voices: Warm Analog and Bright Tape. The Saturation knob is a gain adjustment that overdrives the delay chip. The Degrade toggle adds lofi textures and artifacts to the delay line. The Degrade toggle removes a very vital section of the delay circuitry that keeps the delay stable with longer delay times. As you increase the delay time, the artifacts become more present and lofi.

The Aether uses a simple, but versatile modulation LFO core inspired by the Wardenclyffe, but with more features. The LFO cores sets it apart from other modulated delays out there. The speed of the modulation core can be set 3 different ways: Rate knob, tap tempo tact switch, and Sync toggle. When the Sync toggle is set to its middle position, the LFO can run at whatever speed you want. When the Sync toggle is in the up position it is synced to the delay time. When the Sync toggle is in the down position, it is synced to delay time, but in half-time for slower evolving moduation. The Shift knob determines the depth of the modulation; this has plenty of range in case you want to get really obscene. The Stray toggle changes the LFO from a typical Sine wave to a Slewed Random waveform. Creative exploration of the modulation section of The Aether is strongly encouraged. The way the delay rate and LFO rate interact with each other can yield very creative results.

Notes: *** New features in version 2.0 ***

The new features expand its capabilities meaning it's easier to use whilst making it a more flexible music production tool.

Resampling

Patterns and tracks can be resampled by internally recording the audio output into a new sample. Real-time resampling of your keyboard playing is also possible.

Chopping

Two new voice modes called 'Drum' and 'Chop' auto-slice samples easily with the newly developed Downbeat Divider*. Sliced sections of a sample are then spread across the keyboard making finger drumming possible.

Remix Effect

A beat repeater effect 'R.MIX', added to the master effect options, makes glitchy real-time performances possible.

The new features are designed to work well with each other so you can easily create drum tracks with auto-slicing and record lively performances that are not constrained by the grid of the internal step sequencer.You can also resample samples and tracks for infinite effects processing.

A quirky take on sampling from this 12-bit groovebox inspired by classic 12-bit samplers like the Akai MPC60. Gritty, distinctive sound and some nice sequencing options.

Supplier's Notes:
Retro sampling engine with 12bit Sampler mode

The LIVEN Lofi-12 has a 16bit - 12kHz/24kHz sampling engine with the 12bit Sampler mode that will turn any sound into a pleasing lo-fi sound.

Like the many legendary samplers that were released in the late 80s, this is a genuine low-bit and low sampling frequency sound that cannot be obtained authentically with bit crusher effects and other modern sound degrading tools.

Up to 4 seconds mono sampling per sample slot (2 seconds mono sampling at the 24kHz mode). Start and end point trimming, pitch adjustment, reverse playback, sustain loop, filtering and LFO modulation are all possible.

Easily play samples chromatically across the keyboard

Sounds sampled by the LIVEN Lofi-12 are automatically assigned to the keyboard and can be played easily at different pitches.

The well-known vintage sampler technique of sampling a 33 RPM vinyl at 45 RPM then slowing it down created the gritty, lo-fi sound that is often heard in '90s hip-hop.
The LIVEN Lofi-12 makes it easy to recreate that sound!

80 carefully selected lo-fi preset sounds are included.

Powerful 4-track sequencer with parameter and sound locking
The LIVEN Lofi-12 is equipped with the powerful 4-track step sequencer that has earned the LIVEN XFM a solid reputation.

You can easily create a 4-part song by selecting a sampled sound for each track, and with the Sound Lock function, you can easily switch sampled sounds for each phrase or sequencer step, allowing you to use an almost infinite combination of sounds within a single song.

Sequence up to 64 steps per pattern.

Save up to 64 patterns.

Sub-divide the tempo down to 32nd notes.

Program notes step by step or record them in real time. Additionally, you can also input notes to specific steps while the sequencer is playing to add variation.

The Parameter Lock function allows you to record knob changes into your patterns. This can be recorded live, or programmed in step by step.

The Sound Lock function allows you to record selected sound changes into your patterns. You can use an almost infinite combination of sounds within a single song.

Laid-back knob for creating drunk beats
The Laid-back knob on the LIVEN Lofi-12 allows you to delay the start of the selected sampled sound.

For example, you can slightly delay the snare drum sound in your drum pattern, or use the Parameter Lock function to shift the sound timing later for each step individually.

11 types of Effects each track and 8 types of Master Effect
The LIVEN Lofi-12 features independent effects on all four tracks.
You can select from 11 types of effects from the LIVEN series, and the effect parameters can be automated in steps using the Parameter Lock function.

In addition, 6 types of high-quality Reverbs + Cassette Tape Simulator and Vinyl Simulator are included as the master effect.
The send amount can be adjusted for each track for mixing.

Specifications

Sampler
16bit - 12kHz or 24kHz (switchable) linear PCM sampling (mono)
12bit Sampler mode
Sampling up to 4 seconds per sample (12kHz) or 2 seconds per sample (24kHz)
Auto start sampling function with the input level (input level can be set)
Start/end point, attack, release, pitch, and level adjustment
Reverse playback
Sustain loop settings
Maximum polyphony: 10 voices
128 sample memory slots (16 x 8 banks)
Import and export of a slot or banks of samples (via MIDI)
*Import of audio files from PC/Mac is not supported.


10-note polyphonic mode
Mono mode with adjustable glide time
Legato mode with adjustable glide time
Arpeggiator modes (Up, Down, UpDown, DownUp, Up&Down, Down&Up, Up+1, Up+2, Down-1, Down-2, Random, Play Order)

Filter
Low Pass
High Pass
Band Pass
Each track has its own filter and envelope

LFO
Each track has one LFO which can affect both pitch and filter cutoff with separate adjustable levels
Adjustable LFO Shapes
Adjustable start time delay for each LFO

Sweep
Sweep with adjustable speed, range, and curve

Effects

Track effects

One effect per track
Chorus
Flanger
Tremolo
Delay
Distortion
Bit Crusher
Low Pass Filter
High Pass Filter
Isolator
Tilt EQ
Compressor

Master effects

Reverb send amount can be set independently for each track
Hall Reverb
Room Reverb
Arena Reverb
Plate Reverb
Tunnel Reverb
Infinity Reverb
Cassette Tape Simulator
Vinyl Simulator

Sequencer

4 tracks
Up to 64 steps per pattern
64 patterns
Step length can be set from 1/1 to 1/32
Real-time recording
Directly enter notes for each step when playback is on or off
Enter longer notes (Tied notes)
Metronome and pre-count function
Step copy and paste function
Track copy and paste function
Transpose function
Swing function
Pattern BPM / Global BPM can be set
Pattern chain function

Notes: The Dual EnvVCA is a two-channel analogue linear envelope generator with two fully independent low-noise exponential VCAs. With time ranges from 30 minutes to 8kHz, it can be used as an LFO, slew limiter, ASR/ADSR complex EG, audio-rate oscillator, stereo audio/CV VCA, envelope follower, sidechain processor, and general utility module. The outputs can be attenuated, inverted, and offset without disrupting the VCA, making it a versatile modulation source.

100% analogue

Dual EnvVCA

16HP Eurorack format module
0.95" (24mm) maximum depth (includes power cable)
10-pin Eurorack power header

Power consumption
+12V: 156mA
-12V: 131mA

Audio/VCA
100k input impedance, 1k output impedance, DC-coupled
VCA gain range: -90dB to +0.9dB
DC to 20kHz, +/-0.1dB
VCA response to CV: exponential

Envelope Times
Minimum rise or fall time: ~62us per segment (8kHz max frequency)
Maximum rise time: >10 minutes (typically > 13 minutes)
Maximum fall time: >20 minutes (typically > 40 minutes)

Jacks:
Env jack: Min = -10.2V, Max = +10.1V?, maximum amplitude 9.3Vpp
Trigger jack: rising edge threshold = 2.5V
Cycle jack: rising edge threshold = 2.5V
Follow jack: active range = 0V to +5V

EO* jacks:
EOR (channel A): <80mV low to >4.50V high gate output
EOF (channel B): <80mV low to >4.95V high gate output
Minimum stable pulse width: 1ms. Shorter than this may produce extra 50us pulses
Audio In/Out jacks: -10V to +10V maximum range without clipping

Notes: Moog Music SR Series Cases feature generous padding around a semi-rigid A.B.S. shell, a rugged Ballistic Cordura outer that resists tears and abrasions, and laminated water-resistant zippers to offer on-the-go protection and safe storage for your instrument.

Multiple strap and handle configurations keep your commute comfortable, and multifunction pockets ensure you're organized on arrival.
Designed in conjunction with Subsequent 25
Rugged and road-ready semi-rigid A.B.S shell
Tear-resistant Ballistic Cordura outer
Water-resistant laminate zippers
Also fits Sub Phatty!

Notes: This is the powered version that includes the case as described here, a Strom+ module, a Strom power supply (AC/DC adapter) and a Strom busboard with connector cable. You only need to plug modules into the case and you’re ready to go!

One case fits 84HP of modules with a depth of 70mm.

The cases can be used with an incredible amount of flexibility to swap in an out full rows and create larger setups using the Case Connectors and Case Standoffs.

The side cheeks are made of high-quality Finnish plywood and are coated with a durable transparent melamine film. The black surface is hardwearing and crack resistant as well as easy to clean.

All cases are shipped with a set of 4 rubber feet and 50 x M3 screws.

Notes: The latest and greatest version of one of the most popular microphones around. XLR and USB connections both offer warm sound suitable for vocals, instruments and podcasting.

Supplier's Notes:
The NT1 5th Generation is a revolutionary studio condenser microphone that fuses the classic sound signature of the iconic NT1 with cutting-edge, next-generation technology. It features RODE's groundbreaking (patent-pending) Dual Connect output, offering both XLR and USB connectivity for incredible versatility, a world-first "unclippable" 32-bit float digital output, advanced digital signal processing, and more, all while delivering the same warm, silky character, extremely low noise, and high SPL handling that the NT1 is renowned for. The NT1 5th Generation is a true studio workhorse that excels on everything from vocals and speech, to guitar, piano, drums and percussion - the ultimate studio microphone.

The Iconic Original Studio Microphone, Reborn

The NT1 5th Generation is the latest evolution of the iconic NT1. First introduced in 1991, the NT1 is the world's most popular studio microphone, renowned for its warm, classic sound and incredible versatility. The NT1 5th Generation takes the much-loved and respected core of the NT1 and introduces state-of-the-art features while preserving the legendary sound quality that made the original the go-to microphone for countless creators. A perfect marriage of tradition and innovation.

Game-changing Connectivity

Featuring RODE's groundbreaking, patent-pending Dual Connect output, the NT1 5th Generation can either be connected to an audio interface or mixer via XLR or directly to a computer via USB for studio-grade, plug-and-play recording. Unprecedented flexibility for any studio setup.

Classic Sound on the Cutting-edge

The NT1 5th Generation delivers the same warm, silky character, extremely low noise, and high SPL handling capability that the original is revered for, with improved circuitry for even lower self-noise - an astonishingly low 4dBA, making it the world's quietest studio condenser microphone. When using the XLR output, it delivers the rich, classic studio condenser sound you know and love from the NT1. When using the digital output, you get this same sound plus a slew of cutting-edge features to further enhance your audio for recording pro studio sound directly into your computer.

The Digital Studio of the Future

The NT1 5th Generation features a professional-grade audio interface built into the microphone, which is perfectly optimised to preserve its pristine audio fidelity and also offers a range of highly innovative features for enhanced sound, including the ultra-low-noise, high-gain Revolution Preamp™, high-resolution (192kHz) A/D conversion and advanced digital signal processing. This is so much more than just a studio condenser microphone, it's an all-in-one digital recording solution.

The "Unclippable" Microphone

The NT1 5th Generation features a world-first 32-bit float digital output - a revolutionary recording format that allows you to record everything from a whisper to a scream, the softest guitar to the loudest drummer without having to worry about setting your gain - simply adjust your audio after recording and all the dynamics of your performance will be maintained, with zero risk of clipping or distortion. Gamechanger.

Versatility Like No Other

With its smooth frequency response, extremely low noise, high SPL handling, flexible connectivity options and powerful digital capabilities, the NT1 5th Generation offers versatility like no other studio mic. It excels in any recording application, from vocals and speech to guitars, drums to piano, and can fit into any workflow, from a solo bedroom recording setup to a professional studio.

Craft Your Sound

Craft your sound with advanced APHEXr audio processing, driven by the NT1 5th Generation's powerful on-board DSP. Add compression, a high-pass filter, noise gate, and the legendary APHEXr Aural Exciter™ and Big Bottom™ to your audio with ease using the free RODE Connect and RODE Central companion apps. Dial in the perfect sound with the touch of a button using the intuitive VoxLab™ editor or fine-tune each processor using the advanced editor.

Premium Components, Built to Last

Made from premium components, the NT1 5th Generation is built to the highest standard in RODE's state-of-the-art manufacturing facilities in Sydney, Australia. The HF6 capsule is precision manufactured to sub-micron tolerances to ensure the utmost consistency in tonality and frequency response and is internally shock-mounted for minimising external vibrations. The body is machined from rugged aluminium and is available in black or silver, with high-grade finishes that are extremely resistant to scuffs and scratches, ensuring the NT1 5th Generation is ready to take on the rigours of recording day in, day out.

Everything You Need to Start Recording

The NT1 5th Generation comes with everything you need to start recording in the box, including XLR and USB cables, and a studio-grade shock mount and pop filter for minimising unwanted artifacts in your recording caused by plosives, knocks and bumps.

Unlock More Creative Potential with the RODE App Suite

The NT1 5th Generation is compatible with a range of free RODE apps, including the RODE Central companion app, RODE Connect podcasting app and UNIFY streaming app, all of which allow you to configure settings, access the advanced DSP, update firmware and more.

Specifications

Acoustic & Electrical Specifications

Acoustic Principle: Pressure gradient
Active Electronics: JFET impedance converter with bipolar output buffer
Capsule: 1.00"
Polar Pattern: Cardioid
Address Type: Side
Frequency Range: 20Hz - 20kHz
Output Impedance: 100O
Maximum SPL: 142 dB SPL
Equivalent Noise (A-weighted): 4dBA
Sensitivity: -32 dB re 1 Volt/Pascal (25 mV @ 94 dB SPL) +/- 2 dB @ 1 kHz
Analogue Power Requirements: Via XLR: P48
Analogue Output Connection: 3-pin XLR

Digital Specifications

Digital Power Requirements: Via USB: 5V
Digital Output Connection: USB-C
Bit Depth: 24-bit / 32-bit floating point
Sampling Rate (kHz): 48 / 96 / 192
Processing: APHEX digital signal processing (only available at 48kHz / 24-bit)
Minimum OS Requirements: Windows 10 Mac OS 10.15

Mechanical Specifications

Weight (grams): 308

Dimensions (millimetres):
Length: 52
Width: 52
Height: 189

Box Contains

1 x NT1 5th Generation
1 x SC29
1 x XLR-6
1 x SM6

Notes: Introducing SUM0, a voltage-controlled oscillator with expanded harmonic possibilities, in a Eurorack format.

The 4 classic waveforms are generated by a meticulously designed oscillation engine.

You can modulate its frequency and its pulse width thanks to its integrated LFO, capable of reaching audible frequencies to achieve expressive modulations with the simple touch of a switch.

Expand the voice of your system by creating complex waveforms and high harmonic content thanks to its output mixer. This mixer also has an input to integrate an external signal to the mix output.

Of course, all its parameters (FM, PWM, SYNC, V/OCT) are at your fingertips through its patch points on the front panel.

And of course, assembled with a load of passion in Bocuma's studio-workshop in Guadalajara.

Nerdy-specs:

22HP
+12V > 12mA -12V > 12mA

Notes: Four independent, analogue LFOs in 8hp. Linear voltage control over frequency. Sine or square waveforms.

The Quad LFO gives you four independent high quality temperature stable modulation sources. While audio rate analogue oscillators famously sound more alive, the living quality of analogue modulators is often overlooked. The Quad LFO has a totally analogue sine core, for living, low distortion, accurate, and consistent sine wave outputs. However, if you need a square wave for a clock, to drive a logic input, or for another purpose, you can set one or more LFOs to a square with a jumper on the back.

The Quad LFO provides linear voltage control over frequency, matching our perception of sub-audio rate frequencies. Each oscillator ranges from nearly stopped to around 23 Hz. It is designed for knob twiddling, with accurate potentiometers, aluminium knobs, and 1Hz easily reachable in the centre.

Specs

Size: 8HP
Depth: 32mm
Power Consumption: +12V 51mA, -12V 51mA
Frequency Range: 5+ minutes to 23Hz
Input Range: 0-10V
Input Impedance: 40kO
Output Swing: -5V-+5V peak
Output Impedance: 1kO
Output Drive: 2kO (min), 20kO+ (ideal)

Notes: A multi-purpose module designed to combine noise and modulation sources, slew, sample-and-hold, envelope following and more. Extremely versatile way to add randomness and controlled movement to patches.

Supplier's Notes:
Flurry collects and connects a number of practical modular synthesis tools, focused on clocks, noises, and randomness to create a module that is rife with opportunities for exploration and experimentation. Equally capable as a synthesis source, modulator, clock, or signal modifier; It is bound to find a way to bring new life and dimensions to your synth patches.

Core Functions:

A clock and random pulse source with adjustable and modulatable rate
Accurate pink and white noise sources
Analogue sample & hold and track & hold circuits with very low droop
Analogue slew with adjustable slew time and edge selection (rising edge, falling edge, or both)
Envelope follower
16 noise synthesis algorithms, each with 2 parameters.

One of Flurry's more interesting aspects is its collection of digital noises, which provide a wealth of interesting noises, suitable for either audio or modulation purposes. Each noise type has its own pair of unique parameters to further dial in the desired characteristics, with the primary parameter under CV control.

Most people consider noise to be "undesirable," and product developers expend great effort to design circuits that minimize noise. So why make a module designed specifically to create noise?

Indeed, unwanted noise is undesirable. But not all noise is unwanted. Synthesizing the sound of wind or waves are a couple of the more obvious uses for a noise generator, but the possibilities go far beyond. The crack of a snare hit; the breathiness of a flute sound; an added sizzle to a resonant pad - all are within the sonic domain of noise.

But noise has many other benefits beyond simply being heard. Noise happens to make a wonderful modulator. When noise is used to modulate a filter's cutoff frequency, or an oscillator's pitch or pulse width, then all sorts of raspy, buzzy, gritty timbres are obtained.

Noise is also a key ingredient in sample & hold circuits - a technique most commonly used to generate stepped, random voltages. The circuit works by sampling an input signal's voltage each time you send it a clock pulse, and holding that voltage until the next clock pulse. So, naturally, Flurry also contains the requisite clocking tools! Two of the most common destinations for the S&H output are a filter's cutoff frequency (creating stepped, clocked timbral changes), and oscillator pitch (which produces random notes at clocked intervals).
Of course, you might not always want your voltages to change so abruptly at each clock pulse. Maybe you'd prefer they wobble about gradually and with more grace? To do that, you need a slew circuit, and once again Flurry has you covered - allowing you to select not just the slew time, but also whether the slew is applied to rising voltages, falling voltages, or both. A flick of the switch, and that Slew circuit becomes an envelope follower, enabling Flurry to generate envelopes from input signals (syncing to drum hits are a common use).

With all these features, plus the ability to create random pulses, clock to external pulses, and perform more esoteric track & hold duties, Flurry is a noise lover's toolbox.

Width: 8HP
Depth: 34mm
Power: 116mA @ +12V 30mA @ -12V

Notes: The OpticTone is a combination of a 12dB multi-mode Vactrol Filter and a Vactrol Voltage Controlled Amplifier.

It is a completely original circuit inspired by the singular nature of Vactrols and their fluid and natural character.

The Filter section provides the three fundamental filter types: Lowpass, Bandpass and Highpass.

A Resonance knob with voltage control input is available for improved modulation flexibility. The feedback knob interacts with the Resonance path, tempering it from creamy resonance to wacky self-oscillation.

Filter and VCA run in parallel, both are controlled by the Freq/Amp knob and it's associated voltage-control input. The combination of both signals can be crossfaded with a dedicated knob.

The Freq/Amp is also controlled with direct touch control on the panel via the 3 lines below the knob.

The filter frequency control can be inverted by the Inv Freq switch, providing the possibility to change the filter direction. This feature allows the filter to "close" when the VCA is "opening" or the other way around.

The Ping input allows to generate the well known vactrol filter percussive sounds if a gate or trigger signal is provided. The Ping frequency if controlled by the Freq/Amp knob, ranging from low kicks drums to high pitch bongos.

There is also a Drive knob to take care of your saturation needs. The Overdrive range is wide, from soft textural clipping to heavy distortion.

All voltage controlled inputs have an attenuator-inverter knob.

Features:

Vatrol based 12dB multi-mode Voltage Controlled Filter: Lowpass, Bandpass and Highpass.
Vactrol Based Voltage Controlled Amplifier.
Crossfade between VCF and VCA section.
Voltage controlled Resonance.
Ping Input for percussive sounds generation.
Cutoff Frequency polarity inversion switch. Doesn't change VCA polarity
Overdrive and Feedback controls.
Attenuator/inverter knobs on all voltage controlled inputs
Touch control over Frequency and Amplitude.


Specs:

Width: 10HP
Depth: 33mm
Power consumption: 55mA +12 / 55mA -12
Eurorack Format
Doepfer power connection

Notes: ILLUMINATOR is an eight channel converter that turns Eurorack and analog CV signals into a voltage that drives LEDs and LED strips, creating a unique light show in sync with your live performance. CV signals coming from your music gear are already pulsating with the different parts and sounds of your track, so there is no need for any kind of programming or light operator.

You can focus on playing your set and let ILLUMINATOR take care of the rest, creating a dynamic visual show that follows your tracks down to every note and beat!

No presets, no memory- just your music driving the lights. Create intense strobe-like effects by connecting rapid hi-hats from your analogue drums, or use slow external LFOs for a moody ambience during your ambient sessions.

ILLUMINATOR is compatible with standard 12V LED strips that are affordable and widely available. LED strips are flexible and typically come with adhesive on the back, so it's easy to attach them to any objects and surfaces, including irregular and unusual ones. It can be your Eurorack case, synth stands, walls of the venue, furniture and even your body! This way you can create a unique light show that would otherwise require an expensive (and unwieldy!) lighting setup. Everything you need for your ILLUMINATOR show easily fits into a small backpack for just a couple hundred euros.

ILLUMINATOR can also run an art installation, as LED strips are far from its only application. It can drive anything that accepts DC 12V with a current up to 0.5A. You can hook up anything from motors, electromagnets to old-fashioned filament bulbs. It's the perfect beating heart for your weird mechanical steampunk orchestra. Your imagination is the limit!

ILLUMINATOR accepts all types of CV signals in the range from 3 to 20 volts. Each of the eight channels contains an individual trimmer pot that defines the sensitivity of the channel. It not only adjusts the optimal input range for connected CV sources, but it can also set the brightness of each LED strip. Default factory setting for all trimmer pots is made for the range of 5V, compatible with most gear out of the box.

Applying bipolar CV is also possible but the negative part of the signal will not light up the LEDs (everything below zero will be processed as zero).

CV signals can be connected to ILLUMINATOR using Eurorack mini-jacks or PULSAR-23-like pins and alligator clips. PSU DC IN and all outputs are standard DC sockets 5.5×2.1mm center plus.

Technical specs

Number of channels: 8
Supply voltage: DC 12V
Supply voltage socket: 5.5 × 2.1 mm (center plus)
Output voltage: DC 12V
Output LED socket: 5.5 × 2.1 mm (center plus)
Maximum total power of all channels: 48 watts
Maximum total current of all channels: 4A
Recommended maximum current for one channel: 0.5A
Absolute maximum current for one channel: 1.5A
Input voltage range: 3 - 20 Volts
Input impedance / with correct adjustment of the input sensitivity: > 5.7 kOm / 100 kOm
Input sockets: Eurorack mini-jacks and Pulsar-23 / Ornament-8 pins
Number of mini-jack sockets: 8 (1 for each channel)
Number of Pulsar-23 pins: 9 (1 for each channel + 1 common ground)
LED Items: 8
LED Color: 4000K x 2, Red x 2, Blue x 2, Green x 2
LED Length: 1.5 + 3 m cable
LED Power: 4.8 W/M
LED Voltage: DC 12V
LED IP Rate: IP65
Power supply Input: 100 - 240 VAC, 1.5 A, 50/60 Hz
Power supply Output: 12 DC, 5A
Dimensions: 275 x 70 x 40 mm
Weight: 300 g
Box size: 375 x 310 x 85 mm
Box weight: ~ 800 g

Product dimensions
Illuminator: 275 x 70 x 40 mm
Box: 375 x 310 x 85 mm
Net and gross weight
Illuminator net weight: 300 g
Gross weight: ~ 800 g