Signal Processors

This small digital SPDIF converter converts optical signals into electrical signals and vice versa. The direction of the conversion is set using a dip switch.

AudioFuse X8 OUT is an ADAT expander that adds 8 high-quality line outputs to existing audio interfaces and expands the connection options of existing setups to the highest studio quality.The AudioFuse X8 OUT is an indispensable tool for producers, sound engineers and live musicians who are looking for additional high-quality analogue outputs. By expanding digital-analogue connectivity, the AudioFuse X8 OUT provides more channels to support complex signal paths for live setups or external audio workflows in the studio. It increases the flexibility and scalability of your studio without sacrificing audio quality and fulfils the highest sonic demands.Specifically, an 8-channel ADAT-to-analogue output expander allows you to send audio signals to external equipment (hardware effects, modular systems, mixing consoles, etc.), add additional speakers and control a variety of synthesizers via DC-coupled outputs.Efficient expansion:The AudioFuse X8 OUT provides 8 additional line outputs via ADAT, expanding the connectivity of your existing audio setup without the unnecessary cost and hassle of another full-featured audio interface.Professional audio quality:The AudioFuse X8 OUT supports 8 channels of ADAT to analogue conversion at the highest quality up to 96 kHz and uses the best components on the market to ensure that your recordings and outputs maintain the highest fidelity.Flexible use:The AudioFuse X8 Out is designed as a half-rack unit and allows flexible use both as a desktop unit and for mounting two AudioFuse X8 units in a 19-rack.Robust construction:The AudioFuse X8 OUT was developed for tough studio and live use and has simple controls on the front, a robust mains connection and stable jack outputs.

Awaken the sound of colour:Let the sound colour drip over your beats, melodies and sounds. The Analog Heat +FX is an all-round sound colouring and manipulation device. It offers a wide colour palette of processing options. A real treat for the ears. Whatever the source, and wherever the destination. Let the sound be your canvas.The Heat-Block:Add warmth, depth and character to your sound with eight fully analogue stereo distortion circuits, ranging from a subtle clean boost to total annihilation. Shape and process your material with the multimode filter. Choose from 7 filter types such as classic low-pass and high-pass, as well as band-pass, notch or peak. By turning the controls of the adjustable 2-band EQ, you can fine-tune, add or subtract highs and lows. You can interrupt the sound with the noise gate.The digital effects:Let yourself drift with the Warble effect - let nostalgic magnetic media guide your inspiration. Continue this journey with a touch of chorus. Create rich, haunting soundscapes with delay and reverb. Give the sound a pinch of bit and sample rate reduction - for that special touch of lo-fi glitchyness. Use the compressor to limit or fine-tune the dynamic range of your signal. Bass focus nails the stereo field down to mono at the selected crossover point, making the bass fatter and more focussed.Flow:Rearrange the effects to experiment and find your own signal flow, with your own special timbres. Add depth, dimension, complexity and colour to your music - the switchable signal path allows you to showcase all its nuances. The options are endless. Save your explorations, no matter how wild, in one of the 512 preset memory locations - for instant recall.Modulation:Experiment and have fun with a range of modulation options, each with two purposes. Take the incoming sound and shape it into a modulation with the envelope follower. Let the parameters wiggle and weave in a coordinated or chaotic way using the 3 LFOs - all with their own waveforms and modulation destinations. Be complex, be simple. It's up to you. Discover Analog Heat +FX - an analogue and digital dynamic sound processor.

The DM2 T.N.T is an active inline preamplifier based on the design of the sE DM1, but offering even more gain and less noise with greater flexibility. The integrated, recessed switch allows you to select a clean, transparent gain of a moderate 15 dB or an explosive 30 dB - more headroom and an extremely clean signal, with an ultra-high impedance input and eight selectable load settings to perfectly match any microphone. It is suitable for direct connection to the output of a passive ribbon or dynamic microphone, eliminating the need for an additional cable.Switchable, clean gain from a moderate 15 dB to an explosive 30 dB:Two amplifier stages provide even greater flexibility with different microphones, designed for neutral, balanced sound quality and extremely low noise. The uncompromising design amplifies the signal regardless of load impedance or cable length. The DM2 T.N.T can be perfectly combined with any passive microphone, boosts the signal, reduces the occurrence of interference and provides a significantly better level for the preamplifier of the audio interface / mixer - all without noise and colouration!Get the best out of the microphone thanks to the variable input load:Conventional inline preamps tend to load the transducers of passive microphones due to their electronic design, resulting in a degradation of sound quality and output level. The ultra-high impedance input of the DM2 T.N.T promises the best possible signal quality, regardless of the load of the microphone. The selectable impedance of the DM2 invites experimentation - it can be adjusted from an unconventionally low 50 to an astonishingly high 10 M to match the preamp to any microphone and achieve a clean recording with the lowest possible noise. The DM2 T.N.T was developed to bring out the best sound characteristics of each microphone.Sophisticated Class A electronics with additional output stage:With a sophisticated electronic design and specially selected high-quality FET transistors, the DM2 T.N.T has the lowest self-noise in its class and enables effortless studio-quality recordings. Unlike other commercially available inline preamplifiers, the DM2 T.N.T delivers a constant gain of +15 dB / +30 dB regardless of the connected load. An ideal tool for sound engineers, the DM2 T.N.T's output impedance is also the lowest in its class, making it easy to handle long cable runs while reducing susceptibility to RF interference, hum and noise.Slim, robust all-metal housing with premium finish:Even in the most demanding environments, the DM2 T.N.T scores with a stage-ready and sleek all-metal design, robust construction, high-quality manufacturing standards and a gold-plated XLR connector for reliable lossless signal connection, whether on stage or in the studio. With its exceptionally practical, phantom-powered design that requires no external power supply, the sE DM2 T.N.T is the ultimate game changer in your pro audio arsenal.

ARC Studio is a standalone room correction solution for stereo monitoring systems, It combines a hardware processor, the ARC 4 software and a precise MEMS Measurement microphone. The completely new algorithm corrects unfavourable acoustic conditions that could otherwise lead to an incorrect mix.The system thus enables a natural and transparent sound that guarantees reproducible results.The quality of the monitoring path is crucial. ARC Studio offers a dynamic range of 120 dB(A) between analogue input and output, a frequency response of 0.5Hz to 40kHz and ultra-stable clocking for infinitesimally low jitter. This ensures sonic transparency and accuracy that can only be found in professional audio interfaces and mastering-grade audio converters Mastering-grade audio converters.The connection to studio monitors is made via balanced XLR connectors, which are equipped with relay-based hard bypass to prevent unwanted background noise.

An innovative combination of powerful software and intuitive professional DJ hardware.:The revolutionary 3-in-1 RMX-1000 system comes with editing software, innovative performance hardware and VST and AU plug-ins and offers completely new possibilities for working with effects and samples.It combines the versatile possibilities of studio effects and beatboxes with the outstanding ease of use of professional DJ equipment. Intuitive performance interfaces of our top mixers and effects units - such as the Isolator FX of the DJM-2000, the X-Pad of the DJM-900nexus and the multi-FX chaining of the EFX-1000 - have been further developed and combined in one unit.In addition, numerous parameters can be customised using the RMX-1000's clear and intuitive controls. However, the device is not only incredibly easy to use, but also extremely flexible. You can use the Pioneer remixbox editing software to create unique sound variations, customise the RMX-1000's hardware to suit your personal performance style and load your own sample banks onto the device.The RMX-1000 also acts as a USB controller for the included VST and AU plug-ins, allowing DJs to work with these plug-ins as intuitively as studio DJ mixes. Hours of editing sessions are a thing of the past, and studio productions open up a whole new world of possibilities.X-Pad with exclusive pitch control:The drum samples can be called up using the individual sample buttons on the X-Pad.You can load sample sounds, use the Quantise button to apply each tap to the next beat and use the exclusive Pitch button to change the pitch of each repeated beat to give the performance even more character.

AudioFuse X8 IN is an ADAT expander that adds 8 high-quality line inputs to existing audio interfaces and expands the connection options of existing setups to the highest studio quality.The AudioFuse X8 IN is an indispensable tool for producers, sound engineers and live musicians who are looking for additional high-quality analogue inputs. By expanding analogue-digital connectivity, the AudioFuse X8 IN offers more channels for instruments and external equipment. It increases the flexibility and scalability of your studio without sacrificing audio quality and fulfils the highest sonic demands.In particular, an 8-channel analogue-to-ADAT input expander allows you to expand your synth and instrument setup so that all your equipment is connected and ready to record in your DAW at the same time.Efficient expansion:The AudioFuse X8 IN provides 8 additional analogue line inputs and improves the connectivity of your existing audio setup without the unnecessary cost and hassle of another full-featured audio interface.Professional audio quality:The AudioFuse X8 IN supports 8 channels of premium analogue-to-ADAT conversion at up to 96 kHz and uses the best components available to ensure your recordings maintain the highest fidelity.Flexible use:The AudioFuse X8 IN is designed as a half rack unit and allows flexible use both as a desktop device and for mounting two AudioFuse X8 units in a 19-rack.Simple operation:Front panel controls provide direct access to channel management, making setup and adjustments straightforward. Coloured LEDs also provide visual feedback for level monitoring.

With the 551 Inductor EQ, Rupert Neve Designs is simultaneously taking a step forwards and a step backwards. Backwards in the sense that it draws on well-known 1970s designs and forwards in the sense that the 551 optimises the vintage circuits with the best components of today.The 551 was built around the custom transformer specially developed by Rupert Neve for the 500 format. The 3-band EQ offers an HF shelf/peak band, an LF shelf/peak band and a parametric mid band as well as an 80Hz LoCut filter.As with the legendary vintage predecessors of the 10XX series, the mids of the 551 are equalised using a coil filter. The advantage of an inductor circuit over a capacitor circuit is the incredibly musical sound it develops when it is driven into saturation. This allows the 551 to add unrivalled punch and personality to the signal that goes far beyond standard frequency processing.Q-factor:Basically, the filter quality of the mid band depends on the gain setting. In "MF Hi Q" mode, the Q factor of the mid-band is increased or the filter quality is reduced.

The DM1 DYNAMITE is an active inline preamplifier that is connected between a passive ribbon or dynamic microphone and a microphone preamplifier. The DM1 offers a gain of +28dB, making it the perfect solution for amplifying the signal of a passive microphone, reducing the occurrence of interference and significantly increasing the preamplification - without adding noise or colouration With a sophisticated Class A design and specially selected high-quality FET transistors, the DM1 has the lowest self-noise in its class. In addition, the dedicated output isolation amplifier offers a constant gain of 28 dB, regardless of the connected load. Its output impedance is also the lowest in its class, allowing it to run long cable runs while reducing susceptibility to RF interference, hum and noise.The roadworthy and slim all-metal design, robust construction and high manufacturing standards ensure reliable operation even in demanding applications on stage in difficult environments.

Supports uncompressed digital audio stereo input signals, e.g. PCM audio. Does not support compressed audio formats such as Dolby Digital, DTS, 5.1, etc. To operate this converter on your HDMI TV or on your digital home theatre system, you must set the sound output in its on-screen configuration menu to PCM or other uncompressed formats with the above-mentioned sampling rates.

The ML-16D is a compact, high-quality analogue to Dante analogue converter for the bidirectional transmission of 16 analogue audio channels via a Dante network. It has space-saving 25-pin D-sub connections which, in conjunction with optionally available connection modules, serve as balanced inputs and outputs for analogue signals with line level. Two Ethernet connections (Dante Primary / Secondary) enable redundant operation via two separate networks.The converter has indicators for the operating status and the current sampling rate as well as overload LEDs for each input and signal LEDs for each input and output. Switches on the rear panel allow you to set the required maximum input/output level. All other settings such as sampling rate, channel routing or latency are accessible via the free "Dante Controller" computer application.Dante is a network technology from Audinate for high-performance distribution of audio signals over standard IP networks. It supports the simultaneous transmission of hundreds of uncompressed audio channels with extremely low latency and precise synchronisation.

The DM2 T.N.T is an active inline preamplifier based on the design of the sE DM1, but offering even more gain and less noise with greater flexibility. The integrated, recessed switch allows you to select a clean, transparent gain of a moderate 15 dB or an explosive 30 dB - more headroom and an extremely clean signal, with an ultra-high impedance input and eight selectable load settings to perfectly match any microphone. It is suitable for direct connection to the output of a passive ribbon or dynamic microphone, eliminating the need for an additional cable.Switchable, clean gain from a moderate 15 dB to an explosive 30 dB:Two amplifier stages provide even greater flexibility with different microphones, designed for neutral, balanced sound quality and extremely low noise. The uncompromising design amplifies the signal regardless of load impedance or cable length. The DM2 T.N.T can be perfectly combined with any passive microphone, boosts the signal, reduces the occurrence of interference and provides a significantly better level for the preamplifier of the audio interface / mixer - all without noise and colouration!Get the best out of the microphone thanks to the variable input load:Conventional inline preamps tend to load the transducers of passive microphones due to their electronic design, resulting in a degradation of sound quality and output level. The ultra-high impedance input of the DM2 T.N.T promises the best possible signal quality, regardless of the load of the microphone. The selectable impedance of the DM2 invites experimentation - it can be adjusted from an unconventionally low 50 to an astonishingly high 10 M to match the preamp to any microphone and achieve a clean recording with the lowest possible noise. The DM2 T.N.T was developed to bring out the best sound characteristics of each microphone.Sophisticated Class A electronics with additional output stage:With a sophisticated electronic design and specially selected high-quality FET transistors, the DM2 T.N.T has the lowest self-noise in its class and enables effortless studio-quality recordings. Unlike other commercially available inline preamplifiers, the DM2 T.N.T delivers a constant gain of +15 dB / +30 dB regardless of the connected load. An ideal tool for sound engineers, the DM2 T.N.T's output impedance is also the lowest in its class, allowing it to handle long cable runs with ease while reducing susceptibility to RF interference, hum and noise.Slim, robust all-metal housing with premium finish:Even in the most demanding environments, the DM2 T.N.T scores with a stage-ready and sleek all-metal design, robust construction, high-quality manufacturing standards and a gold-plated XLR connector for reliable lossless signal connection, whether on stage or in the studio. With its exceptionally practical, phantom-powered design that requires no external power supply, the sE DM2 T.N.T is the ultimate game changer in your pro audio arsenal.

The ADI-2 FS is a compact and extremely flexible 2-channel hi-end AD/DA converter. The device offers first-class AD/DA conversion from/to SPDIF, AES/EBU and ADAT at up to 192 kHz in a half-19 inch housing. Its ability to handle SPDIF, AES/EBU and ADAT equally gives it outstanding compatibility. Whatever it is - just plug it in!Balanced analogue inputs and outputs, monitoring via headphone output, SteadyClock FS for optimum converter function, 3-stage hardware adjustment of input and output levels, as well as the additional D-D and A-A modes make the ADI-2 FS a universal tool and versatile problem solver, which with its exceptional sound quality is also ideal as a complement to other high-quality converters.The ADI-2 FS features two servo-balanced line analogue inputs via XLR/jack combo socket, balanced XLR and unbalanced jack outputs that can be used simultaneously, an adjustable high-power stereo headphone output on the front, an ADAT-compatible optical SPDIF I/O and a coaxial SPDIF (RCA, AES/EBU compatible). These extensive connection options allow high-quality and flexible signal transmission without compromise.The analogue inputs use the exquisite circuit technology of the award-winning ADI-2 Pro, offering excellent noise and distortion values as well as a very wide transmission range. Three different input and output levels are available to guarantee optimum adaptation at the analogue level. The ADI-2 FS also features SteadyClock FS, RME's latest clock technology with minimal inherent jitter and maximum jitter suppression. It guarantees the best sound characteristics of the AD/DA conversion, completely independent of the quality of the reference clock used. The ADI-2 FS also utilises technology from the ADI-2 Pro for the converters: the AK557x family and the AK4490 not only offer 192 kHz sample rate and low distortion, but also a signal-to-noise ratio of 120 dBA.However, the ADI-2 FS not only offers optimum sound, but also optimum digital connectivity. Its coaxial RCA SPDIF I/O is AES/EBU compatible thanks to switchable channel status (cable adapter required). The optical TOSLINK I/Os work up to 192 kHz, and can also be used as ADAT I/Os even at 192 kHz. It is also possible to use SPDIF input with ADAT output or vice versa in AD/DA and D-D mode at any time.The clear font panel layout with unambiguous labelling, functional buttons with informative status displays, a level indicator with 6 LEDs and the tidy rear panel make the device easy to use.

With its opulently dimensioned power supply unit and versatile connection options, the Rupert Neve Designs R10 Rack offers everything that professional users of the 500 format could wish for.Performance you can trust:The R10 Rack delivers more than 150% of the power a 10-slot rack actually needs to provide sufficient voltage for even the hungriest modules. To make it easier to monitor the power supply of the modules, the rack is equipped with an LED meter that displays the current power consumption of the modules.The internal power supply has been designed with great care and is double shielded to prevent unwanted magnetic influences in the individual slots. Thanks to the great attention that Rupert Neve's team has paid to the design of the proper power supply, the respective modules used can realise their full potential and sound just as the developers intended.Solid steel housing and flexible mounting:The housing of the R10 rack is made of solid steel. The nuts for fastening the modules can be moved freely in a rail so that they can be easily and flexibly adapted to the width of the respective module. To ensure that the R10 can be used for as long as possible, all connections and slots are manufactured to the highest quality standards.Versatile linking and improved appearance:Using link jumpers, the 500 series VCA buses can be activated in pairs in channels 1-2, 3-4, 5-6, 7-8, 9-10. This makes it possible for two compressors (e.g. Portico 543) to share one VCA and function as a stereo compressor. Blank panels 510 are optionally available to cover free slots.

Bricasti Design Ltd., a new company founded by former Lexicon employees Casey Dowdell and Brian Zolner and based in the USA, specialises in the development of reverbs. The Model 7 sets new standards - it combines maximum musicality and ease of use in a processor-controlled reverb unit. The modern and highly qualified design is based on a platform consisting of six DSPs with the latest state-of-the-art technology.The separate analogue section and the specially developed linear transformer power supply guarantee optimum analogue sound quality.The exceptionally robust housing is made of stainless steel. The specially crafted aluminium front panel features a classic, easy-to-read display. A small number of switches and pots ensure simple operation and clarity; a well thought-out design for the challenges of today and tomorrow.Every control and detail of the M7 has been carefully selected and moulded. Recognising the best designs of its predecessors and a passion for advancing the science of reverb, the M7 offers a palette of sounds that clearly eclipse anything that has gone before in quality and expression.

Scarlett OctoPre Dynamic offers eight high-quality second-generation Scarlett microphone preamps. These are complemented by eight analogue compressors that provide very musical soft-knee compression for subtle dynamics control. With the "More" button you can double the ration and realise more aggressive compression effects. The eight channels of precision Focusrite converters capture every sonic detail of your performance with clarity. If you have an ADAT output on your interface, eight of these channels with D/A converters are available as balanced line outputs. With optical ADAT input and output, Scarlett OctoPre Dynamic is the ideal extension for your studio. The two instrument inputs have sufficient headroom to handle even very loud guitar pickups. If you add a touch of compression, you have the dynamics of electric guitars and basses even better under control.Scarlett OctoPre Dynamic is the ideal solution to enhance any high-quality multi-channel recording system. With its high headroom mic, line and instrument inputs and analogue compressors on each channel, it's perfect for recording drums, guitars, keyboards, vocals - and more. The Scarlett OctoPre Dynamic's eight outputs can be fed either directly from the mic preamps, which is very handy for live use, or from the ADAT inputs. This gives you eight analogue outputs for your DAW in the studio.

The MixDream allows both the central integration of analogue equipment into digital production environments and stereo summing, i.e. the mixing of audio tracks on an analogue level.Conceptually, the reduction to high-quality analogue summing without panorama controls and faders offers the great advantage of being able to retain the entire computer automation. A MixDream can sum 16 audio tracks to one stereo signal. If more than 16 tracks are to be summed, any number of MixDreams can be operated together.The MixDream is based on discrete Class A technology with 60-volt operation (+/-30V). Newly developed circuits based on the latest analogue components ensure extremely high slew rates, a low noise level of -97dBu (A-weighted, 16 channels open) and a high dynamic range of over 125dB. The MixDream therefore easily reaches the technical level of the best consoles.

elysia xmax Masterbus processor - Precision and sound quality for mixing & masteringThe elysia Masterbus processor is a first-class solution for mixing and mastering applications. Equipped with a high-quality Class A analogue circuit, this device offers exceptional transparency and dynamic processing. Thanks to its M/S multiband compression with separate low, mid and side bands, it is ideal for shaping the mix with maximum control. The variable soft clipper, inspired by the legendary alpha compressor, provides more loudness without causing clipping on external A/D converters.Full control over the stereo imageWith the VariLink function, the bands can be flexibly linked together, allowing targeted processing of the frequency spectrum.Of particular interest to mastering professionals: the variable crossover frequency between the low and mid bands allows precise control over the bass range, while the integrated side-band mono maker (Lowmo) concentrates the low-end on the centre of the stereo image - for a powerful and balanced sound foundation.Intuitive operation & high-end displayThe global release control allows quick adjustment of the compression behaviour, making processing intuitive and efficient. This is complemented by a high-quality analogue display that provides precise information on signal level, clipping, tape compression and soft clipping. The alternative "Punch" mode, which generates a more assertive compression, also ensures powerful sound design.Versatility for studio & stageThe elysia Masterbus processor not only shines in the studio - it is also perfect for live applications at the FOH mixing console to optimally balance different volumes. The Tone control also serves as a unique shelving EQ and allows final adjustment of the overall sound.

The Aurora defined the standards in transparency, reliability and value stability for professional A/D D/A converters and is at the centre of professional recording studios around the world. The completely redesigned Aurora(n) continues this legacy. The new member of the Lynx family builds on the features that made the Aurora famous. The innovative design means that the Aurora(n) sounds better, is more flexible, fits better into new applications and is therefore even more suitable as the centrepiece of your recording studio. The Aurora(n) delivers the detailed recording and playback you expect today, while being expandable and ready for whatever the future holds for your studio. We've designed Aurora(n) to be the last audio converter you'll buy! Simply because we've listened to you and built a precision audio tool that's exactly what you want.

The first generation Passeq was the first passive equaliser to provide three separate frequency ranges for boost and cut. With 12 switchable frequencies per band, i.e. 36 frequencies for the cut and 36 frequencies for the boost, the Passeq was the most powerful passive EQ filter network of all time. The same number of boost and cut frequencies is now also available with the new PASSEQ Mastering Equaliser. However, the selection of frequencies has been changed and the Q factor, or quality factor, has been individually re-tuned for each selectable frequency using an optimised coil, capacitor and resistor combination. Each boost and cut frequency now has an individually optimised Q factor.The entire passive filter (variable resistor, capacitor and coil) delivers a very nice sound characteristic. In addition to the choice of components, the charging behaviour of the capacitors and the saturation behaviour of the coils play a major role in this. The resulting relative inertia compared to potentially very fast active filters is the reason for a pleasant, musically favourable sound characteristic. We would choose smoothness and transparency as well as strikingly silky highs and pithy basses as a suitable description for our perception.During fine-tuning, through component selection, in countless listening sessions, the focus was on achieving the most musical sounding curves possible, which need not shy away from comparison with a Pultec EQ from the 1950/60s, for example. But without all the disadvantages of a 60-year-old original, such as the very high noise floor, but above all the very limited frequency selection.The mid-range bands in particular are now tuned more harmonically for sum processing than on the previous model. The mid range cut band now starts working from 200Hz. We have had new coils specially made for this purpose.Another highlight of the new PASSEQ is therefore the HF+ band, which has been extended by the frequencies 25kHz and 35kHz and sounds so incredibly good that you won't want to switch it off.The output level can now be raised or lowered by 10dB using the large rasterised output control in order to address subsequent devices at the perfect working level. This is particularly important for use in mastering.The gain, Q (HF +) and output parameters of the PASSEQ mastering equaliser can be set in 41 steps using rasterised potentiometers (ALPS RK27 "big blue"). All other parameters are switched via 12-position switches. A perfect recall of all settings is therefore easy to realise thanks to these switches and the detented potentiometers,As with all new SPL mastering devices, the new PASSEQ also features auto baypass. Each of the two channels can of course be deactivated or activated manually using the large illuminated buttons. However, thanks to the auto bypass, this can also be done automatically in a freely selectable time window.

The 6176 Vintage Channel Strip combines the coveted all-tube sound of the legendary 610 mic preamp developed by Bill Putnam with the characteristic FET compression of our 1176LN, making it the ultimate channel strip for recording.The warm, smooth sound of this microphone preamp circuit, which can be heard on recordings from Frank Sinatra to Van Halen, and the unique character of the compressor are two of the main reasons why the 6176 can be found in studios all over the world.

The Crescendo is the first microphone preamplifier to work with an internal operating voltage of 120V.This makes it absolutely unique, as the 120V technology means that this preamplifier is virtually impossible to overdrive. The Crescendo shines with absolute signal fidelity and clarity. Rediscover your microphones, because they have never been amplified like this before.