AMS Neve RMX16-500 The AMS Neve RMX16-500 is a high-end digital reverb module in 500-series format, bringing the legendary sound of the original RMX16 from the 1980s into a modern studio workflow. Developed by AMS Neve, it uses the exact same “bit-for-bit” algorithms as the original unit, delivering the iconic sound heard on countless recordings. Originally introduced as one of the first microprocessor-controlled digital reverbs, the RMX16 became a defining tool in music production—especially throughout the 1980s. The 500-series version enhances this legacy with modern DSP, improved noise performance and additional features such as preset storage and wet/dry control. Sound Character The AMS Neve RMX16-500 delivers a rich, dense and musical reverb with a distinctive vintage character. Its most famous feature is the “NonLin” algorithm, which produces the iconic gated reverb sound widely used on 1980s recordings. In addition to classic reverb types (Ambience, Room, Hall, Plate), the unit offers creative effects such as reverse reverb, chorus and delay. It ranges from subtle spatial enhancement to highly characterful effects while always remaining musical and mix-ready. Highlights ✔ classic digital reverb in 500-series format ✔ original RMX16 algorithms (bit-accurate recreation) ✔ iconic 80s gated reverb (NonLin2) ✔ 18 programs including original and expansion algorithms ✔ modern 32-bit DSP with 24-bit / 48 kHz conversion ✔ 100 user preset slots ✔ wet/dry mix control and extended parameter editing ✔ OLED display with intuitive encoder control ✔ significantly lower noise than vintage units ✔ API 500 Series format (3-slot module) Typical Applications ✔ drum and snare reverb (gated / nonlinear) ✔ vocal processing with character ✔ 80s-style effects and retro sound design ✔ mixing and post-production ✔ creative FX (reverse, freeze, modulation) Technical Specifications ✔ device type: digital reverb (500 series module) ✔ DSP: 32-bit processing ✔ converters: 24-bit / 48 kHz ✔ programs: 18 (NonLin, Hall, Plate, Room, Echo, etc.) ✔ presets: 100 user slots ✔ dynamic range: >100 dB ✔ I/O: depends on 500-series rack ✔ format: API 500 Series (3 slots) ✔ configuration: mono-in / stereo-out

Bricasti M10 The Bricasti M10 is a dedicated hardware remote controller designed for the legendary Bricasti M7 reverb. It allows full control of one or multiple M7 units directly from the listening position, eliminating the need to access rack-mounted gear. Built with the same design philosophy as the M7, the M10 features a solid milled aluminum chassis, high-quality controls, and an intuitive interface. It connects via RS422 and can control multiple units simultaneously, making it ideal for complex studio setups and immersive audio environments. Function & Workflow The M10 is not an audio processor but a dedicated control surface: full control over all M7 parameters operation from the sweet spot multi-unit control capability identical interface to the M7 Especially in larger studios, it significantly improves workflow and efficiency. Highlights ✔ dedicated hardware remote for Bricasti M7 / M7M ✔ control of up to 8 units ✔ RS422 connection (stable, low latency) ✔ identical interface to M7 ✔ solid milled aluminum construction ✔ 8-channel metering ✔ powered via M7 (no external PSU required) ✔ ideal for large and immersive setups

Bricasti M7 The Bricasti M7 is a reference-class stereo digital reverb and widely considered one of the best hardware reverbs ever made. Designed by former Lexicon engineers, it was built specifically for high-end studio, mixing, mastering, and post-production environments. Unlike convolution reverbs, the M7 uses advanced algorithmic processing to generate highly realistic and musically flexible spaces. Powered by multiple DSP processors, it delivers extremely dense, natural, and immersive reverb textures. Its combination of modern DSP design, dedicated analog output stage, and linear power supply ensures top-tier audio performance and reliability. Sound Character The M7 is known for its exceptionally natural, dense, and three-dimensional reverb: Rooms: realistic and believable Tails: smooth and cohesive Depth: outstanding spatial placement Blend: integrates into the mix instead of sitting on top Often described as “space in a box”, it can place sounds convincingly into acoustic environments rather than simply adding effect reverb. Highlights ✔ reference-class stereo digital reverb ✔ algorithmic true stereo processing (not convolution) ✔ over 200 high-quality presets ✔ extremely dense and natural reverb tails ✔ up to 24-bit / 192 kHz processing ✔ powerful multi-DSP architecture ✔ dedicated analog output stage + linear PSU ✔ extensive parameter control ✔ MIDI control + optional remote ✔ 19” rackmount (1U)

Bricasti M7M The Bricasti M7M is the modular “mainframe” version of the legendary M7 reverb, designed for professional studio and post-production environments requiring scalable setups. It is sonically identical to the M7, but removes the front panel controls and is operated via the external M10 remote. This design allows multiple M7M units to be controlled from a single interface, making it ideal for large-scale or immersive workflows. Sound Character The M7M delivers the exact same reference-grade reverb sound as the M7: highly realistic spatial reproduction extremely dense and smooth tails precise depth and imaging seamless integration into mixes Its algorithmic processing creates real-time spaces with exceptional musicality and realism. Highlights ✔ identical sound engine to the Bricasti M7 ✔ mainframe version without front panel controls ✔ requires M10 remote for operation ✔ true stereo algorithmic reverb ✔ over 200 presets + user memory ✔ high-performance multi-DSP architecture ✔ premium analog output stage + linear PSU ✔ scalable multi-unit system ✔ ideal for immersive audio setups ✔ 19” rackmount (1U)

Bricasti System 1 The Bricasti System 1 is a complete high-end reverb solution consisting of the M7M reverb mainframe and the M10 remote controller. It is designed for professional studio, post-production, and immersive environments requiring maximum flexibility and scalability. At its core, the system delivers the same legendary sound as the M7, while separating processing and control into two dedicated units. This allows for expandable multi-unit setups and streamlined workflows. System Architecture System 1 includes: M7M Reverb Mainframe → same sound engine as M7, no front panel M10 Remote Controller → full control from the listening position The remote connects via RS422 and is powered directly by the reverb unit. Sound Character The System 1 delivers the classic Bricasti M7 sound: highly realistic spatial reproduction extremely dense and smooth reverb tails precise depth and imaging seamless integration into mixes Its real-time algorithmic processing avoids convolution, resulting in a more natural and musical response. Highlights ✔ complete reverb system (M7M + M10) ✔ reference-class true stereo reverb ✔ identical sound to the M7 ✔ over 200 presets + user memory ✔ control of up to 8 units via one remote ✔ scalable for larger systems ✔ high-end DSP architecture + analog stage ✔ stable RS422 remote connection ✔ handcrafted in the USA

Drawmer 1976 Stereoprozessor 1976 – Stereoprozessor mit 3-Band-Sättigung und Manipulation der Stereobasisbreite Der Drawmer 1976 ist ein 3-bandiger Stereoprozessor für Sättigungseffekte und zur Bearbeitung der Stereobasisbreite, der mit dem Ziel entwickelt wurde, Aufnahmen und Live-Sound mit dem Charakter und der Klangfärbung von klassischem Equipment aus den 1970er Jahren zu veredeln. Die komplett analog erzeugte Sättigung des 1976 bietet eine Authentizität, die Plug-Ins nur anstreben, aber nicht erreichen können. Hier wird steriler und harscher Digitalklang durch Fülle, Weichheit und Wärme ersetzt. Die drei Sättigungsbänder im 1976 basieren auf den hochgelobten Schaltungen zur Sättigung und Manipulation der Stereobasisbreite in unserem digitalen Masteringprozessor DC2476 und fügen dem Klangbild die typischen Artefakte hinzu, die man an klassischen Geräten so schätzt. Sättigung eignet sich perfekt, um Musik Klangfülle hinzuzufügen und diese von einer kalten, kargen Landschaft in ein warmes, helles Panorama zu verwandeln. Im Unterschied zu anderen Sättigungsprozessoren bietet der 1976 gleich drei Bänder zur Kontrolle der Stereobasisbreite und Sättigung. Mit seinem einfachen, aber leistungsstarken und zudem stereophon ausgelegten Regler-Set arbeiten Sie Tiefe und zusätzliche Transparenz in Ihrer Mischung heraus. Sättigung kann den Sound einer Musikproduktion tatsächlich verbessern, indem sie dem Klang durch subtile Verzerrungen wohlklingende Obertöne und zusätzlich Charakter verleiht. Der 1976 erlaubt diesen Klangcharakter in vollständig analoger Form zu generieren. Mit drei unabhängig regelbaren Sättigungsbändern bietet der 1976 beeindruckende Bearbeitungsmöglichkeiten, die sich mit konventionellen 1-bandigen Modellen nicht erreichen lassen. Ohne eine Stereoverbreiterung springt die Mischung den Zuhörer nicht an – und genau dafür lassen sich die drei Bänder des 1976 nutzen: Sie erzeugen eine neue Dimension, die eine Mischung von einer zweidimensionalen Klangfläche weg in ein umfassendes, dreidimensionales Klangerlebnis verwandelt. Im 1976 lässt sich die Stereobasisbreite besonders einfach mit den zugehörigen Reglern für die Tiefen-, Mitten- und Höhenbänder anpassen. Der Vorteil einer unabhängigen Kontrolle über die Stereobasisbreite für jedes Frequenzband besteht darin, den Höhenbereich breiter aufzufächern, ohne gleichzeitig den Bassbereich zu beeinflussen.  Eigenschaften des Drawmer 1976 Stereoprozessors:     •    Leistungsstarke analoge Sättigung über drei Bänder zur Erzeugung analoger Wärme für digitalen und analogen Kläng     •    Akustische Veredelung von Aufnahmen mit dem Klangcharakter der 1970er Jahren und Entfernung der klinischer Härte aus digitalen Aufnahme     •    3-bandige, variable Kontrolle über die Stereobasisbreite für eine perfekt abgestimmte Tiefe und Präsenz    •    Vollständig variable Übergangsfrequenzen zwischen den Bändern gestatten eine maßgeschneiderte Bearbeitung mit individueller Sättigung innerhalb des Frequenzspektrums    •    Echter Stereobetrieb mit einem Regler-Set für beide Kanäle    •    Betriebsmöglichkeit in Mono und Stereo
    •    Der Monoschalter erlaubt die Erkennung möglicher Probleme bei Stereomischungen    •    Integriertes lineares Netzteil mit einstreuungsarmem    •    Gewohnt hohe Drawmer-Fertigungsqualität mit robustem Stahlblechgehäuse und Aluminiumfront.    •    Von Drawmer in Großbritannien entwickelt und gefertigt. Technische Spezifikationen des Drawmer 1976 Soundprozessors:     •    Eingangsimpdanz: > 20k Ohm    •    Eingangspegel, max.: +18dB     •    Ausgangsimpedanz: <100 Oh     •    Ausgangspegel, max.: +21dBu (10k Ohm Lastimpedanz)    •    Frequenzgang: 20Hz bis 20kHz +/-0,2d     •    Verzerrung (THD+N) @ 1kHz/ 0 dB: 0,006%    •    Trennfrequenzen: 70 Hz bis 1,5 kHz, 800Hz bis 15kH     •    Spannungsversorgung: 230V oder 115V, 15V     •    Abmessungen: 482mm (B) x 44mm (H) x 202mm (T)    •    Gewicht: 2,6 kg

Echo Fix EF-X3 Desktop The Echo Fix EF-X3 Desktop is a premium fully analog tape echo unit in a desktop format, combining classic tape delay technology with modern features. Inspired by legendary units such as the Roland Space Echo and Maestro EP-3, it delivers authentic delay effects along with analog chorus and spring reverb. Built with durability and sound quality in mind, the EF-X3 is designed for professional studios, sound designers and demanding musicians. Sound Character The EF-X3 delivers the classic organic tape echo sound with all its desirable characteristics: ✔ warm, slightly saturated delay repeats ✔ natural modulation from tape movement ✔ rich, three-dimensional depth Its integrated BBD chorus and spring reverb provide additional tonal options, from subtle enhancement to pronounced vintage textures. Highlights ✔ true analog tape echo (no DSP) ✔ integrated BBD chorus and spring reverb ✔ three playback heads + additional “sound-on-sound” head ✔ individual outputs per playback head for stereo setups ✔ FET preamp section (can be used as a signal booster) ✔ CV and expression control for speed and feedback ✔ balanced inputs and outputs ✔ auto tape stop to extend tape life ✔ solid desktop construction Typical Applications ✔ vintage tape delay for recording and mixing ✔ sound design and creative processing ✔ guitar, synth and vocal effects ✔ re-amping and analog signal processing ✔ stereo delay setups using individual head outputs Technical Specifications ✔ device type: tape echo / analog multi-effect ✔ effects: tape delay, BBD chorus, spring reverb ✔ signal path: 100% analog ✔ delay time: approx. 56 – 800 ms depending on head configuration ✔ connections: line/instrument input, XLR I/O, head outputs, CV/expression ✔ controls: speed, feedback, EQ (bass/treble), chorus, reverb ✔ power supply: external (24 V) ✔ format: desktop

Echo Fix EF-X3 Tolex The Echo Fix EF-X3 Tolex is a version of the EF-X3 tape echo housed in a classic tolex-covered enclosure, combining vintage aesthetics with modern, reliable analog engineering. It is a fully analog tape echo unit featuring integrated BBD chorus and spring reverb, designed for professional studio and live applications. Built with durability and sound quality in mind, the EF-X3 combines classic tape echo character with enhanced functionality and reliability. Sound Character The EF-X3 Tolex delivers the classic organic tape echo sound with rich analog depth and movement. Typical characteristics include: ✔ warm, slightly saturated delay repeats ✔ natural modulation from tape movement ✔ lively and “breathing” sound texture The integrated analog chorus and real spring reverb add further tonal depth and creative possibilities. Highlights ✔ true analog tape echo (no DSP) ✔ classic tolex enclosure with vintage look ✔ integrated BBD chorus and spring reverb ✔ three playback heads + sound-on-sound head ✔ individual outputs per playback head for stereo ping-pong effects ✔ FET preamp section for tonal coloration ✔ CV and expression control for speed and feedback ✔ balanced inputs and outputs ✔ auto tape stop for extended tape life ✔ robust, road-ready construction with tolex finish Typical Applications ✔ vintage tape delay for recording and mixing ✔ sound design and creative processing ✔ guitar, synth and vocal effects ✔ re-amping and analog processing ✔ stereo delay setups using individual head outputs Technical Specifications ✔ device type: tape echo / analog multi-effect ✔ signal path: 100% analog ✔ effects: tape delay, BBD chorus, spring reverb ✔ delay time: up to approx. 800 ms depending on head configuration ✔ playback heads: 3 + 1 (sound-on-sound) ✔ connections: line/instrument input, XLR I/O, head outputs, CV/expression ✔ controls: speed, feedback, EQ (bass/treble), chorus, reverb ✔ power supply: external (100–240 V) ✔ format: desktop (tolex enclosure) ✔ weight: approx. 12 kg

Echo Fix EF-X3R The Echo Fix EF-X3R is the rackmount version of the EF-X3, a fully analog tape echo system designed for professional studio and post-production environments. It combines classic tape echo technology with modern features and is built for installation in standard 19" racks. Like all EF-X3 models, the EF-X3R uses real tape processing and enhances it with analog BBD chorus and spring reverb, resulting in a highly versatile effect unit with authentic vintage character. Sound Character The EF-X3R delivers the classic organic tape echo sound with rich analog depth and movement. Typical characteristics include: ✔ warm, slightly saturated delay repeats ✔ natural modulation from tape movement ✔ rich, three-dimensional sound The integrated chorus and spring reverb expand the sonic palette, from subtle enhancement to strong vintage textures. Highlights ✔ true analog tape echo (no DSP) ✔ 19" rackmount version of the EF-X3 ✔ integrated BBD chorus and spring reverb ✔ three playback heads + sound-on-sound head ✔ individual outputs per playback head for stereo setups ✔ FET preamp section for tonal coloration ✔ CV and expression control for speed and feedback ✔ balanced inputs and outputs ✔ auto tape stop to extend tape and motor life ✔ optimized for rack integration Typical Applications ✔ tape delay for recording, mixing and mastering ✔ permanent installation in studio racks ✔ sound design and creative processing ✔ guitar, synth and vocal effects ✔ re-amping and analog processing ✔ stereo delay setups using individual head outputs Technical Specifications ✔ device type: tape echo / analog multi-effect ✔ format: 19" rackmount ✔ signal path: 100% analog ✔ effects: tape delay, BBD chorus, spring reverb ✔ delay time: up to approx. 800 ms depending on head configuration ✔ playback heads: 3 + 1 (sound-on-sound) ✔ connections: line/instrument input, XLR I/O, head outputs, CV/expression ✔ controls: speed, feedback, EQ (bass/treble), chorus, reverb ✔ power supply: external

Empirical Labs EL7X FATSO The Empirical Labs EL7X FATSO is a unique analog stereo processor that combines tape emulation, compression, saturation, and transient shaping in a single unit. Designed as an alternative to analog tape machines, it brings the desirable characteristics of tape, tubes, and Class-A circuitry into modern digital workflows. It features a digitally controlled analog signal path, blending multiple non-linear processes to make digital audio sound warmer, denser, and more musical. It is specifically designed to reduce the “cold” and harsh nature of digital recordings. Sound Character The FATSO is not just a compressor—it’s a character processor. It combines: Soft clipping & saturation for analog-style transient rounding Warmth circuit to dynamically smooth high frequencies Transformer emulation for low-end richness Compression for glue and density The result is a thicker, warmer, and more cohesive sound with reduced digital harshness and increased perceived loudness. Highlights ✔ 2-channel analog tape emulator and compressor ✔ combines saturation, soft clipping, compression, and filtering ✔ “Warmth” circuit for dynamic high-frequency control ✔ transformer emulation (“Tranny”) for harmonic depth ✔ multiple compression modes incl. “Spank” and “Eleven” ✔ stereo link for consistent stereo processing ✔ ideal for mix bus and tracking ✔ enhances perceived loudness and analog feel ✔ 19” rackmount design (1U)

Eventide H9000 Der Eventide H9000 ist das Flaggschiff unserer Multieffektprozessoren und setzt unsere ungebrochene Tradition fort, der Pro-Audio-Gemeinschaft die leistungsstärksten Signalprozessoren der Branche zur Verfügung zu stellen. Als Höhepunkt einer nahezu fünfzigjährigen Forschungs- und Entwicklungsarbeit ist der H9000 mit 1.600 einzigartigen Algorithmen bestückt, die sowohl geschätzte Klassiker als auch die neuesten und besten von Eventide bereithalten. Mit seiner Fülle an mitgelieferten Effekten garantiert der H9000 auf Jahre hinaus neue Entdeckungen und Inspiration.   Eigenschaften   Vier Quadcore-ARM-Prozessoren: 16 DSP-Engines Analoge Audioein- und -ausgänge: 8 Kanäle USB-Audio: 16 Kanäle AES/EBU, ADAT, S/PDIF: bis zu 8 Kanäle Audio-Netzwerk-Optionen: 32 Kanäle pro Erweiterungskarte. Drei Erweiterungs-Slots für optionale Audioschnittstellen. Dante und MADI verfügbar Fernsteuerbarkeit: Fernsteuersoftware für Mac/Windows sowie als DAW-Plug-In FX Chains: Innovativer Ansatz zur Konfiguration und Verwaltung komplexer Effekte Direct I/O: Digitales Audio-Interface der Spitzenklasse Umfassende Live-Steuerung: MIDI, Expression-Pedale, Aux-Schalter und USB-Controller Preset-Transfer: Speichern und Laden von Presets über USB-Sticks Netzwerkanbindung für Steuerung und Software-Updates: LAN- und WLAN-Konnektivität

Eventide H9000 Avid ProTools Expansion Board The user-installable expansion card allows Pro Tools HDX or HD Native users to connect the H9000 directly for up to 32 channels of direct I/O with Pro Tools|HD via its Primary DigiLink port. It can alternatively be daisy-chained with other Pro Tools interfaces via the Secondary DigiLink port with the included 12’ mini-Digilink cable. The H9000 can either sync to an external clock via loop sync/word clock, or the H9000 can be the clock source. Concurrent with the release of the Pro Tools|HD Expansion Card is the release of version 1.3 operating software for the H9000.

Eventide H9000 Dante Expansion Board Audinate Brooklyn II Dante Module based design 32 In / 32 Out operation at all supported H9000 sample rates Sample Rate Clock Sources:  Brooklyn II Module, Dante network, H9000 AES67 Support Either 2 Primary and 2 Redundant Networks Connections or 4 Primary Network Connections

Eventide H9000 MADI Expansion Board Supports full AES5 compatibility Sample Rate Clock Sources: H9000 or the selected MADI RX Stream 32 In / 32 Out operation at at all supported H9000 sample rates Optical, Coax or ExpressCard (RME proprietary) connections.

Eventide H9000R Der Eventide H9000 ist das Flaggschiff unserer Multieffektprozessoren und setzt unsere ungebrochene Tradition fort, der Pro-Audio-Gemeinschaft die leistungsstärksten Signalprozessoren der Branche zur Verfügung zu stellen. Als Höhepunkt einer nahezu fünfzigjährigen Forschungs- und Entwicklungsarbeit ist der H9000 mit 1.600 einzigartigen Algorithmen bestückt, die sowohl geschätzte Klassiker als auch die neuesten und besten von Eventide bereithalten. Mit seiner Fülle an mitgelieferten Effekten garantiert der H9000 auf Jahre hinaus neue Entdeckungen und Inspiration.   Eigenschaften   Vier Quadcore-ARM-Prozessoren: 16 DSP-Engines Analoge Audioein- und -ausgänge: 8 Kanäle USB-Audio: 16 Kanäle AES/EBU, ADAT, S/PDIF: bis zu 8 Kanäle Audio-Netzwerk-Optionen: 32 Kanäle pro Erweiterungskarte. Drei Erweiterungs-Slots für optionale Audioschnittstellen. Dante und MADI verfügbar Fernsteuerbarkeit: Fernsteuersoftware für Mac/Windows sowie als DAW-Plug-In FX Chains: Innovativer Ansatz zur Konfiguration und Verwaltung komplexer Effekte Direct I/O: Digitales Audio-Interface der Spitzenklasse Umfassende Live-Steuerung: MIDI, Expression-Pedale, Aux-Schalter und USB-Controller Preset-Transfer: Speichern und Laden von Presets über USB-Sticks Netzwerkanbindung für Steuerung und Software-Updates: LAN- und WLAN-Konnektivität

Freqport FreqInOut FO1 The Freqport FreqInOut FO1 is an innovative hybrid audio solution that allows external analog outboard gear to be used seamlessly as plug-ins within a DAW. As a dedicated hardware insert interface, it bridges classic studio hardware with modern digital workflows while eliminating complex routing, latency issues and cabling overhead. The compact desktop unit features 4 analog inputs and outputs and connects via USB-C. Combined with the Freqport plug-in (VST, AU, AAX), external hardware appears directly inside the DAW like a native plug-in, with full control over routing, gain and configuration. Its integrated hardware matrix and automatic latency compensation enable simultaneous use of multiple analog processors—from compressors and EQs to creative effects—without the limitations of traditional insert setups. Sound Character The FO1 is designed for maximum transparency and accurate signal integrity. Thanks to high-quality ESS converters, the signal remains clean, detailed and uncolored even after multiple A/D and D/A stages. The focus is not on coloration, but on neutral, phase-accurate transmission, ensuring the full character of connected outboard gear is preserved. This makes the FO1 an ideal interface for integrating high-end analog processing into digital production environments. Highlights ✔ hardware insert interface for integrating outboard gear into a DAW ✔ 4 × analog inputs and outputs (balanced TRS) ✔ USB-C connectivity for direct computer integration ✔ plug-in control (VST3, AU, AAX) with full recall ✔ latency-compensated routing via integrated hardware matrix ✔ configurable as 4 mono or 2 stereo insert channels ✔ parallel processing (dry/wet) and M/S mode ✔ high-quality ESS 32-bit converters up to 192 kHz ✔ expandable via cascading (up to 8 channels) ✔ simplified workflow without complex routing ✔ USB bus-powered or optional external PSU Typical Studio Applications ✔ integrating analog compressors, EQs and effects into DAWs ✔ hybrid mixing workflows with hardware outboard ✔ insert processing on tracks, buses and master ✔ parallel processing (compression / saturation) ✔ expanding existing audio interfaces with additional I/O Technical Specifications ✔ device type: hardware insert interface / hybrid audio system ✔ analog I/O: 4 × inputs / 4 × outputs (¼″ TRS) ✔ audio resolution: 32-bit ✔ sample rates: up to 192 kHz ✔ dynamic range: approx. 120 dB (D/A) / 116 dB (A/D) ✔ routing: integrated hardware matrix with latency compensation ✔ plug-in formats: VST3, AU, AAX ✔ connectivity: USB-C ✔ power: USB bus or external 12 V PSU ✔ expansion: cascading of two units supported ✔ form factor: desktop / half-rack

Freqport FreqInOut FO1 The Freqport FreqInOut FO1 is an innovative hybrid audio solution that allows external analog outboard gear to be used seamlessly as plug-ins within a DAW. As a dedicated hardware insert interface, it bridges classic studio hardware with modern digital workflows while eliminating complex routing, latency issues and cabling overhead. The compact desktop unit features 4 analog inputs and outputs and connects via USB-C. Combined with the Freqport plug-in (VST, AU, AAX), external hardware appears directly inside the DAW like a native plug-in, with full control over routing, gain and configuration. Its integrated hardware matrix and automatic latency compensation enable simultaneous use of multiple analog processors—from compressors and EQs to creative effects—without the limitations of traditional insert setups. Sound Character The FO1 is designed for maximum transparency and accurate signal integrity. Thanks to high-quality ESS converters, the signal remains clean, detailed and uncolored even after multiple A/D and D/A stages. The focus is not on coloration, but on neutral, phase-accurate transmission, ensuring the full character of connected outboard gear is preserved. This makes the FO1 an ideal interface for integrating high-end analog processing into digital production environments. Highlights ✔ hardware insert interface for integrating outboard gear into a DAW ✔ 4 × analog inputs and outputs (balanced TRS) ✔ USB-C connectivity for direct computer integration ✔ plug-in control (VST3, AU, AAX) with full recall ✔ latency-compensated routing via integrated hardware matrix ✔ configurable as 4 mono or 2 stereo insert channels ✔ parallel processing (dry/wet) and M/S mode ✔ high-quality ESS 32-bit converters up to 192 kHz ✔ expandable via cascading (up to 8 channels) ✔ simplified workflow without complex routing ✔ USB bus-powered or optional external PSU Typical Studio Applications ✔ integrating analog compressors, EQs and effects into DAWs ✔ hybrid mixing workflows with hardware outboard ✔ insert processing on tracks, buses and master ✔ parallel processing (compression / saturation) ✔ expanding existing audio interfaces with additional I/O Technical Specifications ✔ device type: hardware insert interface / hybrid audio system ✔ analog I/O: 4 × inputs / 4 × outputs (¼″ TRS) ✔ audio resolution: 32-bit ✔ sample rates: up to 192 kHz ✔ dynamic range: approx. 120 dB (D/A) / 116 dB (A/D) ✔ routing: integrated hardware matrix with latency compensation ✔ plug-in formats: VST3, AU, AAX ✔ connectivity: USB-C ✔ power: USB bus or external 12 V PSU ✔ expansion: cascading of two units supported ✔ form factor: desktop / half-rack

Freqport FreqTube FT-1 The Freqport FreqTube FT-1 is an innovative analog-digital hybrid processor that brings real tube saturation directly into your DAW. As a hardware-based effect with plug-in control, it combines classic analog tone shaping with modern digital workflow and full recall capability. The system consists of a compact desktop unit with integrated tube circuitry and a software interface that allows the FT-1 to operate like a plug-in within your DAW. Via a simple USB-C connection, audio can be processed through real analog circuits without complex routing. Inside, four selected vacuum tubes (2× E83CC/12AX7 and 2× 12AT7) provide a wide range of tonal options, from subtle saturation to aggressive distortion. Sound Character The FT-1 delivers true analog tube harmonics with exceptional flexibility. Its sound ranges from subtle warmth and harmonic enhancement to heavy saturation and creative distortion. One of its key strengths is the ability to bring elements forward in a mix without introducing harshness. Transients remain intact while harmonic content is enhanced, making it suitable for both transparent mix enhancement and bold sound design. Highlights ✔ real analog tube processor with plug-in integration ✔ four tubes: 2× E83CC/12AX7 + 2× 12AT7 ✔ USB-C connectivity for direct DAW integration ✔ 32-bit / up to 192 kHz audio resolution ✔ up to 2 stereo or 4 mono instances simultaneously ✔ eight assignable hardware controls ✔ fully controllable via VST, AU, AAX plug-in ✔ adjustable harmonics (even/odd), drive and overdrive ✔ integrated hardware filters for tone shaping ✔ full parameter recall within sessions ✔ no traditional analog routing required Typical Studio Applications ✔ analog saturation within digital DAW workflows ✔ vocal and instrument enhancement ✔ drum bus saturation and mix glue ✔ creative sound design with tube character ✔ hybrid setups without complex cabling Technical Specifications ✔ device type: analog hardware processor with plug-in integration ✔ tubes: 2× E83CC/12AX7, 2× 12AT7 ✔ audio resolution: 32-bit ✔ sample rates: 44.1 – 192 kHz ✔ dynamic range: approx. 120 dB (DAC) / 116 dB (ADC) ✔ channels: up to 2× stereo or 4× mono ✔ control: software plug-in + 8 hardware encoders ✔ connectivity: USB-C ✔ plug-in formats: VST3, AU, AAX ✔ form factor: desktop (optional rack mount)

Pope Audio Dirty-DC Desktop Analogue Chorus Module   The Pope Audio Dirty-DC is essentially the Dirty-C in a standalone desktop unit. It features the exact same circuit/sound and controls with an external 24 VDC power supply. Utilizing an original BBD MN-3007 chip, (the same chip used in the original RE-501 chorus), the Dirty-DC perfectly replicates the lush, lo-fi sound of the original, albeit in a much smaller (and more affordable) package. It features pseudo-stereo L&R out puts like the RE501 (really a full wet against a full dry). An instrument output allows you to use the Dirty-DC in a guitar pedal chain. Use as a reamp output if the balanced line input is chosen. It also features a footswitch input for “IN/OUT” control and an expression pedal input for the modulation Speed control.   Features:   Same circuitry as the Dirty C in a convenient desktop format Genuine Bucket Brigade MN-3007 chip as used in original RE-501 Chorus Fully analog path Fully balanced I/O Hand-built in the UK

Sherman Dual Filterbank 2 Die Sherman Dual Filterbank 2 beherbergt zwei identische Filterbank 2-Einheiten in einem flachen 19"-Gehäuse, das auch einfach auf dem Tisch gelegt werden kann. Alle Ein- und Ausgänge befinden sich auf der Frontplatte, was unzählige Verknüpfungen zwischen den beiden Geräten, aber vor allem auch mit anderen Tischgeräten einfach macht. Über 'link'-Buchsen lassen sich die beiden Filter verknüpfen und parrallel bespielen. Wie auch die einfache Rackversion haben hier beide LFOs auch einen Wellenform-Ausgang.

Sherman Filterbank II classic Die Filterbank 2 von Sherman ist ein Musikinstrument in deutlicher Abgrenzung zur einfachen Klangformung eines "gewöhnlichen" Filters. Mit zwei parallel oder seriell verwendbaren sehr komplexen Filtern, LFO, Hüllkurven, einer satten Verzerrung und diversen modularen Verknüpfungsmöglichkeiten kann mit der Filterbank 2 schon aus dem einfachen Rauschen eines losen Kabels ein satter Beat generiert werden. Das Gerät ist für unbegrenzte Klangformungsanwendungen die richtige Antwort, wenn man es erst einmal verstanden hat. In einem ungewöhlichen Gehäuse und mit einem sagenhaft übersichtlichen Manual ist die Filterbank 2 unerwartet funktional und auch noch ausgesprochen schön. Unabhängige Tiefpass-, Bandpass- und Hochpassfilter mit Resonanz-, Frequenz- und ADSR-Regler (positiv - negativ) Diese sind umschaltbar zwischen parallel und seriell Das zweite Filter hat einen 6-Oktaven-Schalter und ist mit dem ersten Filter synchronisierbar 1 ADSR-Generator für diese Filter mit einem externen Ausgang zur Verwendung mit mehreren Filterbänken oder modularen Systemen, ausgelöst durch das eingehende Signal, durch eine zweite Schallquelle oder durch ein Gate 1 FM-Eingang für die Filter, internes oder externes Audio, LFO oder CV 1 VCA-Overdrive auch mit AR-Generator 1 Ringmodulator-Eingang, internes oder externes Audio, LFO oder CV

Sherman Filterbank II COMPACT Die Filterbank 2 von Sherman ist ein Musikinstrument in deutlicher Abgrenzung zur einfachen Klangformung eines "gewöhnlichen" Filters. Mit zwei parallel oder seriell verwendbaren sehr komplexen Filtern, LFO, Hüllkurven, einer satten Verzerrung und diversen modularen Verknüpfungsmöglichkeiten kann mit der Filterbank 2 schon aus dem einfachen Rauschen eines losen Kabels ein satter Beat generiert werden. Das Gerät ist für unbegrenzte Klangformungsanwendungen die richtige Antwort, wenn man es erst einmal verstanden hat. In einem ungewöhlichen Gehäuse und mit einem sagenhaft übersichtlichen Manual ist die Filterbank 2 unerwartet funktional und auch noch ausgesprochen schön. Die kompakte Version ist viel kleiner und hat alle Anschlüsse auf der Oberseite. Unabhängige Tiefpass-, Bandpass- und Hochpassfilter mit Resonanz-, Frequenz- und ADSR-Regler (positiv - negativ) Diese sind umschaltbar zwischen parallel und seriell Das zweite Filter hat einen 6-Oktaven-Schalter und ist mit dem ersten Filter synchronisierbar 1 ADSR-Generator für diese Filter mit einem externen Ausgang zur Verwendung mit mehreren Filterbänken oder modularen Systemen, ausgelöst durch das eingehende Signal, durch eine zweite Schallquelle oder durch ein Gate 1 FM-Eingang für die Filter, internes oder externes Audio, LFO oder CV 1 VCA-Overdrive auch mit AR-Generator 1 Ringmodulator-Eingang, internes oder externes Audio, LFO oder CV

Sherman Filterbank II single rack Die Filterbank 2 von Sherman ist ein Musikinstrument in deutlicher Abgrenzung zur einfachen Klangformung eines "gewöhnlichen" Filters. Mit zwei parallel oder seriell verwendbaren sehr komplexen Filtern, LFO, Hüllkurven, einer satten Verzerrung und diversen modularen Verknüpfungsmöglichkeiten kann mit der Filterbank 2 schon aus dem einfachen Rauschen eines losen Kabels ein satter Beat generiert werden. Das Gerät ist für unbegrenzte Klangformungsanwendungen die richtige Antwort, wenn man es erst einmal verstanden hat. In einem ungewöhlichen Gehäuse und mit einem sagenhaft übersichtlichen Manual ist die Filterbank 2 unerwartet funktional und auch noch ausgesprochen schön. Unabhängige Tiefpass-, Bandpass- und Hochpassfilter mit Resonanz-, Frequenz- und ADSR-Regler (positiv - negativ) Diese sind umschaltbar zwischen parallel und seriell Das zweite Filter hat einen 6-Oktaven-Schalter und ist mit dem ersten Filter synchronisierbar 1 ADSR-Generator für diese Filter mit einem externen Ausgang zur Verwendung mit mehreren Filterbänken oder modularen Systemen, ausgelöst durch das eingehende Signal, durch eine zweite Schallquelle oder durch ein Gate 1 FM-Eingang für die Filter, internes oder externes Audio, LFO oder CV 1 VCA-Overdrive auch mit AR-Generator 1 Ringmodulator-Eingang, internes oder externes Audio, LFO oder CV

Solid State Logic Fusion The Solid State Logic Fusion is a high-end analog stereo outboard processor designed for mix bus and mastering applications. It combines multiple processing tools in one unit, allowing engineers to add musical polish and character to their mixes. Fusion features six analog processing sections: Vintage Drive, Violet EQ, HF Compressor, Stereo Image, Transformer, and Master Insert. These modules can be used individually or together to shape and enhance the overall sound. Its flexible signal path makes it ideal as a final processing stage in both analog and hybrid workflows. Sound Character Fusion delivers musical, warm, and refined processing. Rather than aggressive processing, it focuses on subtle enhancements that improve depth, width, and harmonic richness. It can range from transparent polishing to more noticeable coloration and saturation. It is particularly effective on mix buses, adding cohesion, clarity, and analog depth. Highlights ✔ analog stereo outboard processor for mix bus and mastering ✔ 6 processing modules in one unit ✔ vintage drive for harmonic saturation ✔ violet EQ for musical tone shaping ✔ HF compressor for high-frequency control ✔ stereo image for width adjustment ✔ transformer stage for analog coloration ✔ master insert for external gear integration ✔ fully analog signal path ✔ ideal for hybrid workflows Typical Applications ✔ mix bus processing (final polish) ✔ mastering with analog enhancement ✔ stereo bus processing ✔ enhancing DAW mixes ✔ hybrid studio setups Technical Specifications ✔ device type: stereo analog processor ✔ circuit: fully analog ✔ format: 19” rack ✔ channels: stereo ✔ modules: drive, EQ, HF comp, stereo image, transformer ✔ insert: stereo insert loop ✔ processing: analog