What's New at Vintage King Audio

What's New

The Grimm Audio CC1 Master Clock is the lowest jitter Word Clock generator and distributor available, providing 16 super high quality Word Clock outputs. The CC1 can slave to incoming Word Clock and distribute the resulting dejittered Word Clock signal. The CC1 can slave to incoming AES/EBU and pass through a reclocked/dejittered AES/EBU signal, providing an ultra-low jitter digital source for feeding your favorite D/A converter. “Owing to a radically redesigned discrete crystal oscillator, clock stability betters that of even the best test equipment available.” Grimm Audio’s extensive research into the correlation between jitter and sound quality brought to light that emotional response to music is vastly more sensitive to jitter than previously realized. Attention turned from “making jitter low” to achieving the most stable clock possible. This research turned up a surprising array of previously underestimated performance factors like power supply noise, oscillator control circuit noise and low-level crosstalk. Owing to a radically redesigned discrete crystal oscillator, clock stability betters that of even the best test equipment available. To sync to an external reference the CC1 sports a PLL offering an astounding 90dB of jitter suppression at 10Hz, further improving at 60dB/dec. The CC1 offers base rates of 44.1kHz and 48kHz and 1/2/4 multiples in separately selectable groups, in addition to an AES/EBU reclocker for cleaning the digital feed to your DA converter. Like all Grimm Audio products, the CC1 features striking styling and solid build quality. Features <ul><li>Ultra-low jitter clock source</li><li>16 word clock outputs</li><li>AES/EBU reclocking</li><li>Sample rates from 44.1kHz up to 192kHz</li><li>Two independent sample rate groups</li></ul>

Grimm Audio CC1 Master Clock

FXpansion BFD Eco gives you easy access to the some of the best sounds in the BFD2 library, recorded at London's Air Studios. The included sounds are full of dramatic levels of detail and mojo... Anything but sterile, BFD Eco is imbued with a musical character that sets it apart from the homogenized, flat sounds found elsewhere. Included are over 40 presets that showcase BFD Eco's powerful, punchy-sounding mixing section. The EQ and dynamics/effect processing allow you to sculpt the raw sounds and lush stereo ambience for any situation. DCAM circuit-modelled compression, filtering and drive provide hardware-style tone and grit, while Overloud's Breverb plate algorithm can be used for artificial reverb effects. BFD Eco also contains a Groove section containing over 1500 drum patterns to drag into your DAW, or to sequence on the built-in Drum Track. Preview patterns in sync with your project and use the humanization effects to adjust the feel to your needs. You can even export a stereo mixdown from the Groove engine - ideal for loops and breakbeats. Other advanced features that set BFD Eco apart from the competition are its customizable keymapping, electronic drum support, the ability to combine kit-pieces from different expansion libraries, and an interface perfectly balanced between ease of use and powerful flexibility. BFD Eco also can be upgraded to BFD2 - both are fully preset-compatible to enable a seamless transition.

FXpansion BFD ECO

As a music production professional, there are certain things you just can’t live without: A good room. Good mics. Good monitors. Good music. And Waves. You need versatile mixing tools that bring out the best of each and every performance. You need the warm, incomparable sound of vintage compressors for punch and power. You need precision-modeled EQs that add color and character. And you need state-of-the-art mastering processors to give your projects that final shimmer and shine. You need Waves Horizon. Includes over 50+ Waves Plug-ins <ul><li>AudioTrack</li><li>C1</li><li>C4</li><li>Center</li><li>CLA-2A</li><li>CLA-3A</li><li>CLA-76 Blackie</li><li>CLA-76 Bluey</li><li>DeBreath</li><li>DeEsser</li><li>Doppler</li><li>Doubler</li><li>Enigma</li><li>GTR 3</li><li>H-Comp</li><li>H-Delay</li><li>IR-L</li><li>L1 Ultramaximizer</li><li>L2 Ultramaximizer</li><li>L3 Multimaximizer</li><li>L3 Ultramaximizer</li><li>L3-LL Multimaximizer</li><li>L3-LL Ultramaximizer</li><li>L3-16 Multimaximizer</li><li>Liner Phase EQ</li><li>MaxxBass</li><li>MaxxVolume</li><li>MetaFlanger</li><li>MondoMod</li><li>Morphoder</li><li>MV2</li><li>Paz Psychoacoustic Analyzer</li><li>PS22 Stereo Maker (TDM only)</li><li>PuigChild 650/670</li><li>PuigTec EQP-1A</li><li>PuigTec MEQ-5</li><li>Q10 Paragraphic EQ</li><li>Q-Clone</li><li>Renaissance Axx</li><li>Renaissance Bass</li><li>Renaissance Channel</li><li>Renaissance Compressor</li><li>Renaissance DeEsser</li><li>Renaissance EQ</li><li>Renaissance Reverb</li><li>Renaissance Vox</li><li>S1</li><li>SoundShifter</li><li>SuperTap</li><li>TransX</li><li>TrueVerb</li><li>Waves Tune LT</li><li>UltraPitch UM225/UM226</li><li>V-Comp</li><li>Vocal Rider</li></ul>

Waves Horizon - TDM

The Trident A-Range is a legendary piece of equipment. As only thirteen A-Range consoles were ever made, it remains a holy grail for sound engineers and producers around the world that hope to create works of art equal to those that were originally recorded on the A-Range — David Bowie's Ziggy Stardust, Queen, Elton John, the list goes on... The Softube Trident A-Range plug-in is the first and only native and TDM plug-in that has been fully endorsed by original designer Malcolm Toft and Trident Audio. The plug-in is a model of a channel on one of the original 13 Trident A-Range consoles. We chose to model channel 15 from the console at Sweet Silence Studio 'B' in Denmark, producer Flemming Rasmussen's favorite channel for recording lead vocals. Flemming also used it for the solo and rhythm guitars on Metallica's "Ride The Lightning" and "Master of Puppets", and Ritchie Blackmore's guitar on the Rainbow albums "Difficult to Cure" and "Bent out of Shape". The A-Range channel features four bands of equalization and high and low pass filters. The sound of the bands is often referred to as "colorful" and "musical". When running a hot enough signal through the original unit and boosting the bands, it is possible to get it to distort, the result is a hairy and effective saturation that is just right in some situations, and just wrong in others! In Short <ul><li>Created together with original designer Malcolm Toft</li><li>Each and every component carefully modeled</li><li>Same sweet sounding EQ filters that made the original legendary</li><li>Interaction between the different bands, making this equalizer one of its kind</li><li>Saturation control</li><li>Very CPU friendly</li></ul> Controls Authentic User Interface An authentic aubergine color (well, as good as it gets on a computer screen...) and the exact same controls as the real thing (okay, we added the Saturation and the Output volume). And then we removed a couple of controls that weren't relevant for computer recording. All right, we changed a few things — but it still looks good! Saturation The distortion of the original desks is of course captured by the plug-in. Instead of using an input volume to drive the signal, we chose to use a Saturation knob instead — so that you don't have to compensate the Output volume all the time. Just set the Saturation to OFF if you don't need it, which will also make the native version more CPU efficient. Low and High Pass filters The ultimate problem solver. Make sure to use these a lot. If you need even more filtering, just press multiple buttons. Developed by Softube and Trident The Trident A-Range plug-in was developed by Softube in co-operation with Malcolm Toft and Trident. "I am pleased to say that this is an incredibly accurate software recreation of my original A-Range design." Professor Malcolm Toft Original designer of the A-Range console at Trident Audio Developments

Softube Trident A-Range Equalizer - TDM

Solid State Logic X-Patch is designed to deliver the flexibility of plug-in style routing to boutique analogue processing. Developed from technology at the heart of SSL’s acclaimed Matrix console, X-Patch provides a 16x16 SuperAnalogue™ routing matrix that can be Ethernet controlled remotely from a standard computer. SSL’s Logictivity™ Remote Studio Browser application provides Set-up, Configuration and Preset Storage and makes the X-Patch the perfect tool to incorporate analogue processing into any production studio environment. X-Patch can function as a simple ‘X-Y Router’ or as a ‘Matrix’ to create complex processing chains. This allows for analogue processing to be built into favourite processing chains and then easily placed into signal paths, for example, favourite Mic Pre, EQ & Dynamics processors recalled as the perfect vocal chain at a single stroke. In a studio environment X-Patch speeds up workflow and aids creativity by reducing the constant physical re-patching required when using only a patch bay for routing. Up to six X-Patch units can be controlled in parallel from a single instance of the X-Patch Logictivity Browser software. SSL X-Patch is also a powerful stage or studio live performance tool for guitarists. Once configured, X-Patch can operate independently of the Studio Browser application and host computer and has MIDI connections. Connection of a standard MIDI Foot Controller to X-Patch creates a versatile stand alone analogue routing/switching system for on-stage boutique instrument processing, or amplifier selection at a fraction of the cost of comparable professional on stage effects selection systems. Key Features <ul><li>Inserts analogue processors into production workflow like plug-ins</li><li>Suited to studio production and live performance applications</li><li>Insert ‘chains’ can be created, saved and recalled</li><li>Browser software controls routing and set up</li><li>Ethernet connection to Mac or PC host computer</li><li>16 x 16 pure analogue routing matrix</li><li>Rear panel audio connection via 4 x 25 way D-Sub’s</li><li>Front Panel XLR inputs and outputs for channels 1 & 2</li><li>SSL SuperAnalogue™ transparent audio performance</li><li>MIDI connectivity enables use of Programmable Foot controllers</li><li>+4dB and -10dB operating levels facilitate use of guitar pedals etc</li></ul>

SSL X-Patch

The Telefunken AR-51 is designed in the tradition of the classic mics of the past, utilizing the same circuit design as the classic Telefunken ELA M 251E. The AR-51 employs new cost-efficient manufacturing methods to meet a price point suitable to any recording studio, from major World Class facilities to voiceover rooms and home project studios. The new R-F-T AR-51 utilizes only premium components, including a vintage New Old Stock (NOS) tube that has been rigidly tested for noise and microphonics, as in the ELA M 251E. The AR-51 also features a globally-sourced power supply and capsule, both rigorously tested prior to installation. As with all products in the R-F-T range of Telefunken Elektroakustik microphones, the circuit board has been designed for superior current handling, permitting the amplifier to have full access to the necessary "power on demand" for handling low frequency and transient information. Modifications have been made to the amplifier circuit to optimize the performance of the capsule and a signal path that includes the same European manufactured output transformer found in every ELA M 251 E built since 1960. At the Telefunken Elektroakustic studios in South Windsor, CT, audio engineers have successfully tested a wide range of applications for this new large diaphragm microphone. The microphone is especially suitable for recording acoustic and electric guitars, piano, percussion, for drum overheads and close miked drums, as well as for both male and female vocals. The perfect complement to the company's acclaimed AK-47, the AR-51 is a significant addition to any studio's microphone locker.

Telefunken Elektroakustik AR-51

A Designs Nail Compressor Features <ul><li>Dual Mono with Stereo Link</li><li>Hybrid Design (Tube and Solid State)</li><li>12AT7 Tubes</li><li>LED Meters switchable with Link mode</li><li>Mix Control to adjust how much compression you wish to add or subtract</li><li>Attack</li><li>Release</li><li>Threshold (Gain Reduction)</li><li>Hard Threshold</li><li>High-Pass Key Filter </li><li>Gain Control (Output)</li><li>Custom-Milled Aluminum Knobs</li></ul>

A Designs Audio HM2Compressor Nail

With the introduction of the True Systems PT2-500 mic pre and DI, True Systems continues its contortionist legacy of packing immense-sounding microphone preamplification into infinitesimal spaces. Descendant of the TRUE Systems Precision 8, which squeezes eight audiophile-grade preamps into a single rack space, the diminutive PT2-500 conforms to the modular 500 Series format. Like all the other preamps in the TRUE Systems family, the PT2-500 delivers stunning detail, rich analog depth, and a robust, reliable design. But the PT2-500 is 500 module sized, for audio professionals who want everything in a rack ready to use, over and over again without buying additional power supplies and multiple racks. The PT2-500 features up to 70 dB of high-headroom, low-noise microphone gain. An exceptionally high-quality direct input (DI) calms edgy piezos and provides a THRU jack for amping or effects. Switchable phantom power, 80 Hz high pass filter, and polarity reverse give the unit appropriate flexibility, even in its tiny, 500 Series space. Like all TRUE Systems preamps, the PT2-500 boasts a frequency response that reads like a misprint, 1.5 Hz to 600 kHz (-3 dB), ensuring that the audible and usable ultrasonic ranges are well within its linear response. “More so perhaps than any other mic pre manufacturer, TRUE Systems is a natural fit for the 500 Series,” said Brad Lunde, president at TransAudio Group, the U.S. distributor for TRUE Systems. “Engineers have been asking for the PT2-500 for years as the 500 Series format is quickly becoming the de facto protocol for modular gear.”

True Systems PT2-500

The JDK Audio R22 dual channel compressor features two channels of API's patented award winning compressor circuit in a rack mount unit with internal power supply. This is the same compressor circuit originally designed into all ATI Paragon mixing consoles. Each channel includes the patented 'THRUST' switch to protect the sensitive high frequency content of the audio signal even under the most vigorous of compression ratios. The perfect companion to your R20 mic-preamp and R24 EQ, the R22 compressor provides comprehensive easy-to-use control of the audio signal with custom VU metering of both output and gain reduction. The two channels can even be linked for use as a stereo compressor with true RMS power summing of the left and right signals. Features <ul><li>Patented THRUST circuit</li><li>Variable Threshold, Ratio, and Make-up Gain controls</li><li>Switchable metering of output level and gain reduction</li><li>Above threshold LED indication</li><li>Switchable Hard or Soft Knee compression</li><li>Linkable for stereo operation with true RMS power summing</li><li>Fully balanced XLR and 1/4" inputs and outputs</li></ul>Specifications <ul><li>Inputs: Balanced XLR and 1/4" jack (1/4" normaled priority)</li><li>Outputs: Balanced XLR and 1/4" parallel jacks</li><li>Input Impedance: 15KOhms Balanced</li><li>Bandwidth: +/- 0.5db, 20Hz - 50kHz</li><li>THD+N @ 1kHz, +4dBu: <0.005%</li><li>Maximum Level: +19dBu</li><li>Signal-to-Noise Ratio: -88dBu (comp in), -92dBu (bypass), -106dB</li><li>Crosstalk: <84dB @ 20kHz</li><li>Stereo Link: True RMS Power Summing</li><li>Meter: -20 to +3 VU Output Level, 0 to -15 dB Gain Reduction, switchable</li></ul> Compressor Controls <ul><li>Threshold Range: -40dBu to +15dBu</li><li>Ratio Range: 1:1 to 10:1</li><li>Makeup Gain Range: 0dB to +20dB</li><li>Hard or Soft Knee switchabel</li><li>Flat or Thrust side chain filter switchable</li><li>Attach Time: Program and Control Adaptive, 10mSec to 40mSec</li><li>Release Time: Program and Control Adaptive, 30mSec to 400mSec</li><li>Current: 93mA, 10.8VoltAmps</li></ul>

JDK Audio by API R22 - Dual Channel Compressor

With eight channels of Focusrite pre-amplification and a built-in 24-bit / 96 kHz ADAT output, OctoPre MkII is Focusrite’s powerful input upgrade for your Pro Tools system, or any digital audio workstation. It’s also a great expansion for any analogue or digital console, or hard disk recorder. The digital output allows users to make the most of often-neglected ADAT inputs; ideal for expanding the number of mic-pre inputs on your interface. Connect OctoPre MkII to your audio interface’s ADAT input to create a high quality, multi-channel recording solution, ideal for tracking drums, as well guitars, keyboards, vocals and more. Features <ul><li>Eight High Quality Award-winning Focusrite Pre-amps The best sounding multi-channel mic-pre in its class.</li><li>Built-in eight channel 24-bit / 96 kHz digital output Make the most of your interface’s ADAT input. OctoPre connects via its built-in lightpipe output, with a ‘real world’ dynamic range performance of 110dB.</li><li>Mic-pres optimised for tracking drums OctoPre MkII is designed not to clip, with pre-amps optimised to handle extreme levels from sound sources like kick drum and snare.</li><li>Two quality Hi-Z instrument inputs OctoPre MkII’s first two channels feature DI’s, ready for you to plug in your Guitars and Bass, instantly accessible on the front fascia.</li><li>5 LED input metering on every channel OctoPre MkII’s fast-acting 5-LED input meters provide clear reference of input levels, helping you to avoid analogue or digital clipping.</li><li>-10dB Pads on every channel OctoPre MkII is drum-ready, with 10dB pads to stop hot signals from overdriving the microphone pre-amp.</li><li>Direct outputs on every channel For routing and flexibility, especially in the live environment. Each mic-pre can be routed to the mixer’s analogue channels, with two mirrored ADAT outputs (44.1) left free to send to digital recorders.</li><li>48V Phantom power available on every channel Switched in two groups (1-4 and 5-8) every channel is capable of supplying phantom power, for use with most condenser microphones.</li><li>State-of-the-art Clocking Solutions OctoPre MkII's internal clock delivers a jitter performance of < 10 PPM. For external clocks, a BNC connector is provided and industry-standard JetPLL™ jitter elimination technology is employed.</li></ul>

Focusrite OctoPre MKII

Dramastic Obsidian 500 Features <ul><li>TXIO Enhanced Transformer Balanced I/O</li><li>Precision Stepped Controls To Accurately Recall Settings</li><li>VCA Feed-Forward Compression</li><li>Selectable Internal High Pass Filter</li><li>Expansion Port For Additional Features</li><li>Link Multiple Modules Together</li><li>Selectable Pad Settings</li><li>Superior Imaging</li></ul> Specifications <ul><li>Ratio settings: 1.5:1, 2:1, 4:1, 8:1, 10:1, 12:1</li><li>Attack dontrol Range: 0.1, 0.3, 1.0, 3.0, 10, and 30 mSec</li><li>Release control range: 0.1, 0.3, 0.6, 1.2 sec, AUTO and Lo-Fi</li><li>HPF settings: 30Hz, 60Hz, 105Hz, 125Hz, 125Hz, 185Hz, 330Hz</li><li>Makeup Gain Output: 12db</li><li>1 Year Limited Warranty</li></ul>

Dramastic Audio Obsidian 500

The Little Labs VOG analog bass resonance tool is the latest creation from Little Labs, and is Little Labs first entry in the popular API 500-Series module format. The VOG was originally designed to capture the chest resonance of vocalists or voice over artists, enabling them to still have a proximity type effect without having to be so close to the mic, hence the name, VOG, or Voice of God. However, when I heard it on kick drum and bass, it became clear this would be the go to device to make these instruments sound huge. The VOG lets users sweep a sharp peak resonance from 20 to 300 Hz, while anything below the peak is rolled off at a steep -24dB per octave. This lets users focus on the low end they want, while eliminating low-end mush and unnecessary woofer excursion. A subharmonic can be the focus, or the fundamental, or upper harmonics. On a kick drum, for example, the overall effect of turning VOGs frequency knob sounds like someone is tightening or loosening the drumhead. In a mix, VOG allows users to place the bass spectrum instruments so they do not interfere or get in the way of each other.

Little Labs VOG Analog Bass Resonance Tool

Supersize your sound with the new Digidesign Pro Tools Instrument Expansion Pack — the ultimate professional virtual instrument collection for Pro Tools®. Featuring the latest, most incredible versions ever of all five groundbreaking Avid virtual instruments, the Pro Tools Instrument Expansion Pack includes 55 GB of samples, loops, and sound generators — including 16.5 GB of all-new content — for everything from the meanest grooves to the most sophisticated orchestral passages. And with the unbelievable price of $499 for the bundle, the Pro Tools Instrument Expansion Pack offers Pro Tools users access to the most powerful virtual instruments available for Pro Tools at a savings of nearly $1,000 (US)! <ul><li>Play and shape any instrument sound with the Structure® sampler workstation </li><li>Produce world-class drum tracks with Strike®, the ultimate virtual drummer </li><li>Add vintage flavor using emulations of classic electric pianos in Velvet® </li><li>Throw together killer loop-based beats and grooves in real-time with Transfuser™ </li><li>Explore new sounds or re-create your favorites with the Hybrid™ synthesizer</li></ul> The Pro Tools Instrument Expansion Pack includes the following instruments: Structure — Professional Sampler Workstation Structure redefines the art of sampling, allowing you to freely create, sculpt, edit, and refine nearly any sound imaginable, from simple acoustic instruments to highly complex soundscapes. The latest Structure 1.1 sampler comes with 17.5 GB of high-quality content, including 5 GB of new orchestral sounds, Nashville Signature drum kits, and a custom grand piano. Create ultra-realistic orchestral performances with special new MIDI control features, take advantage of new Kontakt 3 and Giga sample format support, gain new EXS modulation matrix support, and enjoy new database enhancements for optimal usability. Strike — The Ultimate Virtual Drummer Strike is a revolutionary instrument plug-in that makes it easy to create professional drum performances with uncanny realism and unbelievable human feel. Pick an acoustic or electric kit or instrument — or build a kit from your own samples/sounds — and tell your drummer how to play. The latest Strike 1.5 instrument includes new features and more content (30 GB total!) to provide you with even more beat versatility. Use your own sounds with new WAV and AIFF sample import, gain more efficiency with the enhanced Style editor, discover over 10 GB of new drum kits and percussion elements, explore 55 new styles (for a total of 205 styles), and more. Velvet — Vintage Electric Pianos Whether you’re inspired by ’60s soul, ’70s classic rock, or ’80s “glassy” sounds, Velvet delivers stunningly realistic emulations of your favorite electric pianos, from vintage classics to a massively popular digital synth. The latest Velvet 1.3 e-pianos plug-in features a new reverb effects module with three reverb types (Ambience, Spring, and Room) and introduces nine new FM tines that deliver unique sonic characteristics, including classic ’80s-inspired “glassy” digital piano sounds. Transfuser — The Ultimate Groove Creator Transfuser is an award-winning real-time loop, phrase, and groove creator that lets you easily create, manipulate, and perform music on the fly. It comes with 4 GB of high-quality drum sounds and loops to get your started, or toss in your own loop library content. The latest Transfuser 1.3 workstation opens up even more sonic possibilities than ever before, adding a new bass module inspired by a classic bass line synth, 1.6 GB of new audio content for a total of nearly 4 GB of included loops and sounds, and a 6-band parametric EQ that lets you accurately perform frequency surgery on your sounds. Hybrid — High-definition Synthesizer For sound buffs who love to get their hands dirty, Hybrid combines the warmth of classic analog waveforms with digital wavetables, enabling you to re-create sounds you remember, or create something no one’s heard before. The latest Hybrid 1.6 synth delivers a new Multi-Square waveform to generate even more unique sounds; introduces Blue, White, Mod, and Crackle waveforms to the noise oscillator to offer more texture versatility; and includes 256 all-new patches for fast and easy sound production in the studio or on stage.

Digidesign Pro Tools Instrument Expansion Pack

Sonic Studio Amarra is the audiophile solution designed to provide the best music reproduction achievable from a computer based system. Amarra represents a true integration of audio production and audio playback. Coupled with the convenience of iTunes, the world's most popular music delivery system, Amarra provides a seamless integration of multiple sample rate and format music formats. Amarra Software Features <ul><li>Optimized playback for a sound that rivals CDs and Vinyl </li><li>Automatic Hardware Sample Rate Management</li><li>Advanced dither and digital volume (as set thru iTunes)</li><li>Plays all high resolution PCM formats up to 192kHz</li><li>Uses iTunes for compressed and rights managed music</li><li>Integration with Apple Remote and iTouch</li><li>Support for Snow Leopard, Leopard and Tiger versions of OS X</li></ul> Precision Sonic EQ With the advanced resolution Sonic EQ, your sound can be tailored to your particular environment providing the best sound possible. Sonic EQ employs a minimal phase topology and double precision operations to ensure inaudible results while preserving the fidelity of the most delicate acoustic performances. Digital Volume and Dither Amarra comes with a very high quality digital volume control with a unique automated dither which is only applied when the volume has been adjusted. Sample Rate Conversion The Sample Rate Converter in Amarra is based on our professional version and comes from our friends at Izotope (www.izotope.com). As a background process you can select sound files and convert them to/from sample rates up to 192 kHz. (release 1.2 Q1'10) Hardware Integration <ul><li>When used with a Model Four the following features are enabled</li><li>Amarra controlled Analog Volume Control</li><li>Integration with DSP based Sonic EQ for room adjustment</li><li>Routing for external components (CD, Phono, DVD, etc)</li></ul>

Sonic Studio Amarra Computer Music Player

The world of Live sound is going digital. Just like studios made the transition from hardware to software, today the same change is taking place in Live sound. And Waves is leading the way. A true Live sound breakthrough, Waves MultiRack is a software host that lets FOH and Monitor engineers run multiple simultaneous instances of the same Native Waves plug-ins used in recording studios and mixing rooms the world over. Armed only with a laptop, an I/O box, and MultiRack, you now have the power to shape your Live sound with unprecedented precision, and do away with rack after rack of heavy effects units. Just imagine: The world’s best-sounding reverbs, equalizers, compressors, limiters, and delays at your fingertips, without the limitations of hardware, and at a fraction of the cost. With easy setup and advanced preset capabilities, MultiRack delivers all the flexibility and portability of software, with sound quality and convenience that beats hardware. <ul><li>Take Studio Sound to the Stage and Back Again</li><li>Quick & Simple Setup</li><li>A Fraction of the Cost of Hardware</li><li>Simple Routing: No Cabling, No Rack Mounting, No Trucking</li><li>Full Recall, Full Control</li><li>Set and save presets and snapshots per song and song sections</li><li>Windows & OS X Compatible</li></ul>

Waves MultiRack

The Elysia Mpressor Plugin is the as close as it can get emulation of our famous creative compressor. Its all discrete circuitry and its special character have been translated into software in all the painstaking details by the specialists from brainworx. The result is an outstanding universal compressor as well as a dynamics effect machine which will significantly enhance the potentials of your host software. The mpressor is available in RTAS, VST, AU and TDM formats. The plugin has all the great functions of the hardware which can be combined in many ways: Auto Fast for attack makes life easy even on longer attack settings. The Anti Log function offers an alternative release curve that can be engaged for instant pumping effects. In addition to standard ratios, the mpressor also provides negative values. Combine this with its extremely fast attack times, and you will be able to create completely new signal envelopes. The Gain Reduction Limiter regulates the virtual control voltage and keeps even the most extreme settings under control – completely independent from the threshold and ratio settings. The additional filter is a perfect fit for changing the compressed signal in its sound character. The possibility to control the mpressor by an external sidechain input to get things grooving makes for further flexibility. Whatever you are looking for: a high grade sum compressor, a flexible tool for single instruments or an inspiring dynamics effect processor – the mpressor delivers. Discover all the options. Try it now! True Emulation The compression behavior and the sound character of the hardware have been recreated in a complex process to match the hardware as close as possible. Transferring a complex analog hardware into digital code is not exactly trivial, especially if the model is a completely discrete design like mpressor. The first important task in a project like this is to fragment the electronic circuitry into separate functional blocks. These blocks are translated into software step by step after which they will be reunited to become a virtual product. This first result is measured very accurately and then compared to the hardware, which leads to an extensive and very detailed matching process. The work on the graphical user interface (photography, retouching, rendering) takes place at the same time. The final stage is the calibration of the behavior of all the controllers in order to give the software the ‘feel’ of the real thing. Finally, the finished code is ported to different plugin interfaces (RTAS/VST/AU/TDM) and packed into installation routines. Oversampling In order to achieve the best results also on lower sample rates, the mpressor plugin uses an increased internal resolution in these cases. The mpressor plugin benefits from higher sample rates in two ways: In the first place, it can react to changes in the source signal faster, which is especially important if a short attack time is set. Secondly, the generated virtual control voltage and therefore the compression behavior of the compressor becomes more precise because there are more measuring points available. The mpressor plugin employs the oversampling technique in order to enjoy these advantages even if lower sample rates are used. This means that the basic sample rate of a project is multiplied by a certain factor inside the plugin without the need to set the complete project to a higher frequency. This method consumes a certain amount of CPU power, but the acoustic result speaks for itself. The mpressor plugin uses oversampling according to the following rules: <ul><li>Project sample rate lower than 50 kHz: 4x oversampling</li><li>Project sample rate lower than 100 kHz: 2x oversampling</li><li>Project sample rate higher than 100 kHz: no oversampling</li></ul> Mousewheel Support Setting parameters with the mouse can be pretty annoying – for this reason the controllers of the mpressor can be easily moved with the mouse wheel. You do not necessarily have to click and drag the controllers of the mpressor. Instead, try making your settings with the alternative mousewheel control without clicking on the specific controller first. The following shortcuts provide some further comfort: Fine mode <ul><li>VST: Shift + mouse wheel</li><li>AU: Shift + mouse wheel</li><li>RTAS/TDM: Ctrl/Cmd + mouse wheel</li></ul> Standard setting <ul><li>VST: Ctrl/Cmd + mouse click</li><li>AU: Alt + mouse click</li><li>RTAS/TDM: Alt + mouse click</li></ul> Linear/Circular <ul><li>VST: Alt</li></ul> Auto Fast Originally developed for the alpha compressor, this switchable semi automation ensures a perfect attack on the basis of the value set by the user. The attack parameter is a very crucial factor for the processing of a compressor. Therefore choosing the right setting is very important, but depending on the dynamic progress of the source material this can be a difficult task – no matter if you are processing single channels or complete mixes. If a very short time is chosen, the compressor is able to catch the short peaks, but on the other hand the sustaining signal will also be processed, which might result in audible distortion. Longer attack settings reduce distortion significantly, but then the compressor is too slow for fast impulses. This is where the Auto Fast function comes into play. For example, if you set the attack to 80 ms and then engage the Auto Fast mode, the attack time will be shortened automatically on fast and loud signal impulses. The compressor reduces the signal quickly and prevents it from slipping through. Then the attack time directly and automatically returns to its original setting. In Auto Fast mode the compressor can be very fast, but only when it is really needed. This function influences the attack parameter on short and loud impulses only; in all other cases the original setting of the controller has priority. Anti Log This alternative characteristic of the release curve follows an antilogarithmic course instead of the linear one and produces a much more audible result. It is the time constants and especially the release parameter that decide if the processing of a compressor is obvious or unobtrusive to the ear. Mastering applications, for instance, require a discreet performance as a general rule. Here you will find mostly logarithmic or linear release curves. It is characteristic of a logarithmic release that the time constant shortens when the amount of gain reduction increases. The advantage of this behavior is that short and loud peaks (e.g. drums) have a fast release time, while the remaining material is processed with a slower release time. But if intentionally striking and creative compression is the goal, it definitely makes sense to turn things upside down. In the Anti Log mode of the mpressor the curve behaves just the other way round: If the threshold value is passed and compression sets in, the release time will be longer at the beginning. If the input signal starts to decline, however, the release time will become faster as a result. A special circuitry makes this behavior independent from the absolute amount of gain reduction. No matter if the compressor reduces 10, 15 or 20 dB, the curve will always stay the same at the beginning and will only become faster at the end. With this feature you can create many exceptional compression effects just by the push of a button! Negative Ratios The characteristic curve bends and goes back down! Heavy pumping, backward sounds, etc. – perfect for very cool compression effects. Negative ratios – what exactly does this mean? To get a better understanding for this function, it makes sense to realize what the ratio control of a ‘normal’ compressor does: <ul><li>1:1 the signal remains linear, there is no compression going on</li><li>2:1 after crossing the threshold, an increase of 2 dB at the input will be compressed to an increase of 1 dB at the output</li><li>∞:1 after crossing the threshold, the output signal is constantly held at the threshold level without reacting to further increases at the input (limiter)</li></ul> At a negative ratio, the characteristic curve bends and returns back down after crossing the threshold. The louder the input signal becomes with a setting like that, the lower the output signal becomes – perfect for groovy compression effects. To get a grip on the extreme ‘destruction’ this can cause, engaging the Gain Reduction Limiter is just the right idea. Beyond infinity – made possible by the mpressor ;-) Audio Filter The mpressor plugin of course also features another popular function of the hardware version: the easy yet very flexible Niveau Filter. This filter is a specialist in changing the overall sonic character of a signal with ease. It features two controllers per channel and is capable of producing convincing results flexibly in no time at all. Whenever a classic shelving filter would be too limited and a fully parametric filter would be too much, the Niveau Filter is the perfect tool. Its main function is to change the proportions between high and low frequencies. The principle is quite similar to a pair of scales: Dependent on the gain setting around a variable center frequency, the high frequencies are boosted whereas the low frequencies are attenuated (or vice versa). By simultaneously boosting and cutting the selected frequency areas, it is much easier to influence the character of a track (bright vs. dark) compared to using other types of equalizers. The center frequency can be shifted continuously between 26 Hz and 2.2 kHz or between 260 Hz and 22 kHz respectively (when the x10 switch is activated). The characteristics of the filter change in the extreme positions of the EQ Gain controller: the fully counter-clockwise setting will produce a low pass filter; fully clockwise position will result in a high pass filter. The overall level can drop quite noticeably then, but this can be corrected with the gain controller easily. External Sidechain You want to create frequency dependent compression or have it accented by the groove? The sidechain function makes it all possible! The external sidechain enables the compressor to control its processing totally independent from the audio material running through it. If the SC Extern switch is active, compression will not be controlled by the signals from the regular audio inputs anymore, but by different signals which are fed into the additional sidechain input connectors. If, for example, a duplicate of the input signal is processed with an equalizer and then fed into the sidechain input, the result will be frequency-dependent compression. Another example is to send the bass drum of a drum machine into the sidechain input in order to achieve nice groovy compression that is pumping in time with the music. The creative options are almost infinite. Compression can be exactly on time or totally against it, which can of course be varied on the fly. Single instruments can be given more space in a mix according to its rhythm. All of a sudden, static sounds become vivid and sound really interesting! Important: Your host software must support sidechain functionality in order to use this feature of the mpressor plugin! Gain Reduction Limiter This novel limiter is not placed in the audio path as usually, but restricts the virtual control voltage of the plugin instead. A specialty of the mpressor is the Gain Reduction Limiter for the virtual control voltage of the plugin. This limiter is not placed in the audio path where you would usually find it, but in the control path of the compressor. When it is activated, it limits the virtual control voltage according to the setting of the GR Limit controller. This means: No matter how high the input level might become – the amount of gain reduction will never exceed the value which you have set. For comparison, imagine a fader on a mixing console with your hand moving the fader to ‘play compressor’. If now the fader was limited by a piece of duct tape at 10 dB, for example, it could only reduce the signal up to this value. If the input level dropped below this limit, the fader would be moved up correspondingly. However, if the input signal got even louder, the fader could not be moved down any further because of the duct tape limit, and then the output signal would become louder again in correspondence with the input signal. Loud parts in an arrangement can keep their dynamics, as they will not be compressed beyond the limit of the Gain Reduction Limiter. Some very nice special effects like ducking or upward compression can be achieved with it easily by only reducing the quieter parts without changing the original dynamics at the same time. System Requirements With regard to its performance the mpressor is not very demanding. Please continue reading for more information on hardware and software requirements. You need the following to run the mpressor plugin: <ul><li>Computer running Mac OS or Microsoft Windows. The following specs are the absolute minimum; we highly recommend running a machine with better performance</li></ul> Mac <ul><li>Intel or PPC CPU (we recommend at least 1 GHz)</li><li>256 MB RAM</li><li>Mac OS X 10.4 or higher</li><li>Pro Tools 7.0 or higher or a VST/AU compatible host</li></ul> PC <ul><li>Intel compatible CPU (we recommend at least 1 GHz)</li><li>256 MB RAM</li><li>Windows 2000 or higher</li><li>Pro Tools 7.0 or higher or a VST compatible host</li><li>Audio I/O system: sound card, external AD/DA converters or the like</li><li>iLok USB dongle</li></ul>

Elysia Mpressor Plug-in - TDM

The API 527 compressor is a single channel module based on API's 225L discrete channel compressor. The 527 features comprehensive controls including variable attack, release, ratio, and outut gain controls. The unit also includes API's patented 'Thrust' circuit, first offered on the 2500 Stereo Bus Compressor. A 10 segment LED meter is switchable between gain reduction and output level. "Anyone who has used the 225L compressor found in API consoles has expressed a longing for the same kind of flexibility and control in the 500 Series for some time" said Larry Droppa, President of API. "We're delighted to now offer the 527 to complement our vintage 525 compressor, which has been many engineers' favorite compressor over the years.

API 527

The Evol Fucifier is a distortion synthesizer & sound shaper, designed to give you many colors with which to create and shape your palette of sounds. Everything in the signal chain, from the discrete Mic Pre, the Vintage Germanium Preamp, the Analog Filter, and the inductors in the Equalizer to the output transformer, is designed to be overdriven, saturated, overloaded or distorted in a pleasing way. Mix & match the Vintage Germanium Preamp for a little warmth or overdrive, the Tape Saturation overdrive for a little compression and limiting or a bit of signal saturation, the Analog Filter/Crossover (which in itself sounds great overdriven, and can be driven into self oscillation), the Dual Band Distortion which allows you to mix many different types of distortion, and the vintage-style inductor based equalizer to shape the overall tone and color of the sound. Features <ul> <li>Branded real wood front panel, custom finished by hand</li> <li>Color coded, lighted acrylic, audio responsive gain meter knobs to see at a glance where your various gain stage levels are</li> <li>Discrete mic preamp</li> <li>Low noise, high quality balanced line input</li> <li>Gain control of mic, instrument & dry mix gain in addition to control of overall input gain</li> <li>True relay bypass of each effect section using lighted indicator ring metal push buttons</li> <li>The Vintage Germanium Preamp section uses specially matched NOS vintage germanium transistors for added warmth and a great overdrive sound</li> <li>The Analog Filter Section Features: <ul> <li>Variable freq & resonance</li> <li>LFO with variable amplitude and speed, an envelope detector and expression pedal control which can all be used separately or in conjunction to control filter frequency</li> <li>Selectable high pass, band pass, or band pass + high pass filter response modes for the high freq distortion section</li> <li>Separate low pass & high pass outputs on the back panel for processing each band separately with external gear</ul></li> <LI>Dual Band Distortion allows control over:<ul> <LI>Drive, distortion amount, distortion type (6 modes), clipping symmetry & low freq/high freq blend control to blend the two distortion sections in varying degrees</li></ul> <LI>Inductor Based EQ with 5 bands of boost/cut: low, mid, high bands plus a low shelf & high shelf</li> <LI>Speaker output to plug a speaker cabinet into your Fucifier for use as an amp sim or even an amp!</li> <LI>Wet/Dry signal blend control</li> <LI>Transformer coupled, discrete Class A output section</li> <LI>Internal regulated AC power supply (no wall warts!)</li> <LI>2 space, 19" rack mount enclosure</li>

Evol Fucifier

The Radial Engineering Workhorse 5000 is best described as a 500 Series module housing on steroids. To be unveiled at the upcoming AES convention in New York City, this three-rackspace unit accommodates up to eight 500 Series modules. Besides providing ±16VDC and 48-volt phantom power for each of its eight module slots and input, direct out and summing connections on 8-channel D-25 sub jacks (for direct-to-DAW or P.A. feeds), each module has ¼-inch and XLR I/O, mono/stereo link switches and a mix/feed switch for internal patch routing of the output to feed the next module or to the master output section. OmniPort connectors on each input provide module designers with custom options, such as integrating “keying” inputs, a footswitch port or a tuner output on a direct box/instrument preamp. Designed to allow on-the-spot mixing or stereo recording, the mix bus section has pan/level pots and mute switches for each module, rotary main out and aux out controls and a headphone monitoring section. The rear panel has XLR and ¼-inch aux and transformer-isolated main outputs, insert jacks for the master outs and bus in/out jacks for connecting additional sources and/or multiple Workhorse units. The Workhorse derives its main power from a custom-designed universal switching supply that delivers more “juice” than most others. Its external power supply eliminates issues with hum due to magnetic coupling between power supply transformer and the plug-in module’s internal transformer. As an extra measure, additional power filtering has been incorporated to eliminate noise. The power supply is also auto-tracked, which means that should a module fail, other modules will be protected against harmful voltage swings that could potentially cause damage. “The Workhorse is one of the most exciting products we have ever developed,” says Radial President Peter Janis. “Not only has it forced us to challenge ourselves with historic compatibility, but it’s forced us to go back to the drawing board and redesign several modules that were essentially set to go. But once you get into a project of this magnitude, you really cannot go backwards. We truly believe that the Workhorse will set a new standard in analog audio interfacing. We plan to openly publish our design specifications to enable other module producers to take full advantage of the feature-set we’ve incorporated into the Workhorse. Ultimately, if it’s easy to use, fun and spurs on creativity, we will have achieved our objectives.” In addition to the Workhorse, Radial has several lunchbox modules near completion. These include the Radial JDV-LB Class-A feed-forward direct box, the Phazer-LB phase adjustment tool, the Radial JDX-LB guitar amp DI emulator, the Komit compressor-limiter, plus a few more surprises! Like all Radial products, the Workhorse is made from 14-gauge steel and finished in a rugged baked enamel finish.

Radial Engineering Workhorse 5000

Tone. It's what the Rascal Audio Analogue ToneBuss is all about. Not bells and whistles not a bunch of unwanted features you don't need and won't use. Just large, full, detailed, opulent analogue TONE. The kind you'd expect from a classic, all-discrete recording desk of the early 1970's. The Analogue ToneBuss uses discrete, class-A circuitry with custom wound input and output transformers specifically designed to provide the larger-than-life punch and authority of the most coveted vintage signal paths. Additonally, the Analogue ToneBuss supports the instant recall of your DAW by offering minimal, practical facilities, all on logable, rotary switches, so you can use your time for mixing instead of wasting it trying to recall your previous settings. If you'd love your DAW mixes to possess the dynamic richness and spacial definition of those mixed on classic, large-format consoles, then look no further. Rascal Audio's Analogue ToneBuss delivers with simplicity and elegance. The Analogue ToneBuss is available in 16-channel and 24-channel models. Front panel controls include pan switches for channels 1 thru 4 and switchable direct thruputs for channels 5 and 6. The pan controls, each of which affects a stereo pair of inputs, allow the user the option of panning individual signals, such as kick, snare, bass, and lead vocal, to the center of the mix in the analogue domain yielding much more focus and definition while simultaneously allowing the user to incorporate their favorite mono processors during mixing. The unique channel 5 and 6 direct thruputs give users without a 'proper' patchbay the ability to hardwire a couple of their favorite outboard processors to their D/A outputs. When editing simply switch to the 'DIR' position, process and re-record the audio, and then switch back to 'L' or 'R' when it's time for mixdown.

Rascal Audio Analog Tonebuss - 16 Channel

The LILPEQr is Purple Audio's take on the classic program (aka final mix) equalizer. It offers high and low frequency shelving bands, each with three carefully selected corner frequencies to choose from, and a switchable fader knob for overall level control. The LILPEQr emulates the colorful 3D richness and useful responses of our favorite vintage tube (aka valve) program EQs (namely, the Lang PEQ-1 and the Klangfilm RZ062), but in a compact, solid-state format. The LILPEQr will add air, weight, and glue to an already balanced mix. Additionally you can also use the LILPEQr to immediately and intuitively alter the overall tone of an individual source. When the EQ is first switched in (center switch position - "IN" - LED off) your signal is injected passively into the tone network, directly following the input transformer. As with many classic program equalizers, this configuration allows for the ultimate in wide, lush, mix-buss response with extra "glue" and all that richness. Flip the switch one more click (upper switch position - "IN+Level" - LED red) to activate the buffered "Level" control. Placed in between the input transformer and EQ network, these discrete, class A buffers change the response ever so slightly, making the signal seem a bit tighter and more focused. This mode is more useful on individual sources within a mix. Relay true bypass (lower switch position - "BYP" - LED green) allows for easy before and after EQ comparisons, and a simple overall circuit path allows for wide bandwidth so that everything passes through... sounding better! In program mode ("IN"), with both EQ knobs centered, frequency response is +/- 0.5dBu from 20Hz to 20kHz, and extends below 10Hz and well beyond 120kHz before any drooping occurs. Features <ul><li>Big sound, small package</li><li>Major vibe, minimal interface</li><li>High Frequency shelf boost/cut 12dB at 5kHz, 10kHz, or 28.5kHz ("AIR")</li><li>Low Frequency shelf boost/cut 20db at 50Hz, 80Hz, or 160Hz</li><li>Switchable, buffered level control for overall volume adjustment</li><li>Relay True Bypass</li><li>Tricolor LED level meters</li><li>Transformer balanced in and out, with bridging 20kΩ input</li><li>100% Class A, discrete signal path</li><li>Current consumption: 60mA</li><li>Over Current Resettable Fused (prevents any module problem from affecting your rack)</li><li>ROHS - Lead Free </li></ul> LILPEQ-R Pinout <ul><li>Pin 1: Chassis Ground</li><li>Pin 2: Output A +</li><li>Pin 3: Output B + (mult from pin 2)</li><li>Pin 4: Output A -</li><li>Pin 5: Audio Ground</li><li>Pin 6: Output B- (mult from pin 4)</li><li>Pin 7: Input -</li><li>Pin 8: Input -</li><li>Pin 9: Input +</li><li>Pin 10: Input +</li><li>Pin 11: NC</li><li>Pin 12: +15 or 16VDC 60ma</li><li>Pin 13: Audio Ground</li><li>Pin 14: -15 or 16VDC 60ma</li><li>Pin 15: NC (+48VDC) </li></ul>Designed by Eisen Audio for Purple Audio.

Purple Audio LILPEQr 2 Band Program EQ

The famous Tonelux MP1A TILT EQ control in a 19” rack mount unit. The Tonelux TILT includes 8) TILT EQ controls with an IN/OUT switch, and a POLARITY switch on each channel. Channels 1 and 2 have XLR inputs and outputs, and channels 1-8 also come out on D-SUBS. The input and outputs are fully balanced. A very inexpensive way to equalize 8 channels with a unique single knob control.The famous Tonelux MP1A TILT EQ control in a 19” rack mount unit. The TILT includes 8) TILT EQ controls with an IN/OUT switch, and a POLARITY switch on each channel. Channels 1 and 2 have XLR inputs and outputs, and channels 1-8 also come out on D-SUBS. The input and outputs are fully balanced. A very inexpensive way to equalize 8 channels with a unique single knob control.

Tonelux Tilt

The new Slate Pro Audio Dragon is unlike any dynamic audio processor you‘ve heard before. Simply put, the Dragon was created to be the most versatile compressor, limiter, and sound shaping tool that the pro audio industry has ever seen. The Concept The Dragon starts with a classic FET compressor circuit, reminiscent of vintage units from the 60’s and 70’s. But the Slate Pro Audio engineers enhanced the circuitry of the unit with a mastering grade signal path, Jensen input transformer, and a CLASS A output section based around a custom-made output transformer. The result is a compression quality that is so rich and warm, but is also capable of sharp and aggressive tones. But this is only the start of what the Dragon can do. Add Some Character The Dragon includes three ‘character’ settings that change the way the unit sounds and responds. The first setting is BOOM. This setting adds sub bass harmonics in a unique way that when combined with some compression, can make basses sound massive and kick drums pound like you’ve never heard before. The second setting is BITE, which adds a gentle ‘forward’ character that makes the source more defined and clear. BITE is superb on bass, snare, vocals, and guitars. Last, is SHEEN. This character setting adds an airy gloss to the source, reminiscent of classic opto tube compressors. This setting is absolutely stunning on vocals, acoustics, and overheads. Full mixes can gain new clarity and excitement from the sheen setting. All character settings can be used simultaneously! Go Vintage The Dragon's "Vintage" button enables circuitry that emulates a more vintage sound from a classic age of compression. The result is more harmonics, a bit of ‘grit’, and an overall more aggressive sound with tons of attitude that is amazing on electric guitars, bass, and a must-hear on drums. Saturate The Dragon has a three setting saturation selector. This saturation can add warm and rich harmonics or even gritty distortion. On setting 1, it can add just the right of “life” to a dull vocal. On setting 2, it can add presence and warmth to a bass or make guitars sound like they are hitting tape. On setting 3, it can add some rich harmonics to a snare to bring out the sustain and make it come alive in the mix. Wait, There’s MORE The Dragon includes a hi pass filter so you have the option to control the amount of low end that feeds the compressor detection. This is useful when stereo linking two DRAGONS together for transparent mixbuss compression, or for grouped drum compression where you don’t want the drums to ‘pump’. FET Your Mixbuss Classic FET compressors have not been common mixbuss choices due to their excessive color and aggressive nature. However, the Dragon includes a gentle 2:1 setting that sounds amazing on full mixes. With the character, vintage, and saturate options disabled, the signal path is quiet and clean with just the right amount of rich tone. Of course if you need more color or saturation on your buss, it’s just a button or switch away. With settings 2:1, character SHEEN, Saturate 1, and slow attack and fast release, your mix will come alive in an incredible way. Squash It The Dragon is one of the finest drum compressors to ever hit the analog market, and is complete with the famous overcompression “all buttons in” mode called ‘Squash’. This setting is simply stunning on drum room mics, or any other source that you want to explode out of the mix. Due to its extremely hifi signal path, the Dragon’s ‘Squash’ setting can create that larger then life effect without causing a harsh or brittle high end. Mix It Up Last and certainly not least, the Dragon includes a ‘Mix” knob to control the wet/dry ratio of the source. This amazing feature allows you to add some heavy compression, and then back it off gently until it creates the perfect balance for your mix. The ‘Mix’ knob also allows you to parallel compress sources, such as drum busses, without having to mult or buss the drums to additional faders! You’ll find the ‘mix’ knob to be one of the most useful tools in modern analog compression. It works especially well with the saturation and squash features. Overcompressing a vocal and then mixing back the wet signal is the ultimate way to make a vocal pop without losing the subtle nuances. Style The Dragon’s faceplate comes with a beautiful tribal mural designed by world renowned tattoo artist Davey Suicide, and will add a unique beauty to your rack. The striking look matches its remarkable sound. Presets Normally compressors don’t need presets, but with so many options (there is literally thousands of combinations), the Dragon ships with a huge listing of presets made by Steven Slate and some of the world’s top producers, mixers, and even mastering engineers. But once you experience the many possibilities and sounds of the Dragon, you’ll soon enjoy making your own!

Slate Pro Audio Dragon

The Helios Type 69 500 holds true to the original Dick Swettenham design, but with a number of modern updates such as 5db steps on the mic preamp gain switch as opposed to the original 10db steps, additional 16khz EQ frequency selection for today's high-bandwidth digital recordings, and a balanced output for easy interfacing with any digital converter.

Helios Type 69-500

The Electrodyne 501 is a two stage discrete transistor, transformer coupled preamp with active DI, using classic (1969? / 1970?) design technology. Each amp stage is individually optimized for peak performance using Electrodyne factory engineering notes and selected high performance components identical to the originals. The transformers are made by Electrodyne’s original manufacturer to exacting factory specifications. The active DI circuit presents an almost immeasurable load to sensitive musical instrument outputs allowing incredibly accurate capture of the instruments true tone. The output of the DI circuit is designed to directly connect and interact with the mic input transformer to allow maximum tone options. Specifications <ul><li>Maximum Gain: 68db. Adjustable over 50db in 2db steps with two ranges using 20db pad. </li><li>Output level control: Infinitely adjustable from 0 (off) to +6db over unity. </li><li>Input impedance: Microphone, 50 / 200 ohms selectable. DI, over 7 megohms. </li><li>Output impedance: 150 ohms </li><li>Distortion: 0.02%typical over entire gain range. </li><li>Maximum output: +30db at less than 0.8%thd. (+29.5db at 0.06%thd) </li><li>Frequency response: +/- 0.25db, 10hz to 60khz. </li><li>Signal to noise: -80db typ, -60db at absolute maximum gain. </li><li>Clip indicator: Monitors all three amp stages and illuminates when any stage is 3db from clip. </li></ul>

Electrodyne 501 Preamp

The Electrodyne 511 is a classic two-band discrete transistor reciprocal active inductor equalizer using (1969? / 1970?) design technology. The custom inductors and output transformer are made by Electrodyne’s original maker, to strict factory manufacturing tolerances as small as 2%. This allows consistent eq performance and repeatability from channel to channel that was not possible in the 60's. Smooth performance and eq response from minimum to maximum gain at all frequencies, provides unusually broad sonic and tonal options not experienced since the 1970's. Specifications <ul><li>Input: Active balanced discrete transistor impedance converter. </li><li>Output impedance: 150 ohms </li><li>+/-12 db gain range: Equalize or Attenuate </li><li>4 frequencies selectable per band. LF: 40, 100, 250, 500hz. HF: 1.5k, 3k, 5k, 10k. </li><li>Peak/Shelf function on each band. Shelving at all frequencies, Peaking at 250, 500, 1.5k, 3k and 5k. </li><li>Distortion: 0.03%typical. </li><li>Maximum output: +30db at less than 0.8%thd. (+29.5db at 0.06%thd) </li><li>Frequency response: +/- 0.25db, 10hz to 70khz typical. </li><li>Signal to noise: -85db typ.</li></ul>

Electrodyne 511 Equalizer

Using the legendary SPA690 discrete amp blocks, the Inward Connections OPT1A has the exact same Optocell gain reduction circuitry as the revered TSL-3 Vac-Rac tube limiter in a solid state 500 series format. Fully transformer balanced input and output, along with an all discrete design insure the sort of top quality audio you would expect from Inward Connections. Features include <ul><li>Gain reduction control</li><li>Zero adjust trim</li><li>Output level control</li><li>Hi-Pass filter: 250Hz for detector circuit </li><li>Bypass switch</li><li>LED lighted VU meter</li><li>VU meter output/reduction switch</li><li>Link switch</li><li>SPA690 all discrete legendary amp blocks</li><li>Optocell reduction circuitry identical to TSL-3 Tube Limiter</li><li>Balance input and output transformers</li><li>Fits standard 500 series slot configuration mechanically and electrically</li></ul> Specifications <ul><li>Gain Reduction up to 40dB</li><li>Input Impedance >100K ohms balance</li><li>Output Impedance 600 ohms balance</li><li>Frequency Response +/- 0.5dB@20Hz to 50KHz</li><li>Output Signal to Noise -95dB or greater</li><li>THD + Noise .01%@1KHz/+4dBu</li><li>Reserve Makeup Gain +15dB</li></ul>

Inward Connections OPT1A

The Shadow Hills Dual Vandergraph is a direct descendant of the Shadow Hills Mastering Compressor and maintains its reputation of unquestionable quality and exceptional performance. It’s inner workings are developed from the discrete gain cells of the Mastering Compressor which are combined with the Optograph’s extremely flexible Sidechain Filter matrix. This combination provides great possibilities for new recording techniques and the dynamic control. This stereo compressor’s fully discrete audio path is Class-A and features Shadow Hills' custom Iron Transformers. The controls are: Ratio, Filter, Compression and Output Level. Just like the Mastering Compressor, the Compression and Output Level controls are Swiss made twenty-four position attenuators. The Ratio switch, selects between: 1.2: 1, 2.5: 1, 4: 1, 8: 1. In addition to changing ratios, each selection changes preset attack and recover times. The Sidechain Filter Matrix controls the frequency sensitivity of the side-chain. The positions are: 90 hertz, 150 hertz, 250 hertz and Bandpass. <ul><li>Input impedance 20k</li><li>Minimum load impedance 600 ohms</li><li>Stereo</li><li>Fully Class-A </li><li>Discrete audio path </li><li>Discrete gain cells from Mastering Compressor</li><li>Sidechain Filter Matrix from Optograph</li><li>Swiss made discrete attenuators</li><li>Military Spec build quality</li><li>Meets the specifications of the VPR Alliance</li></ul>There is no limit to the number of Dual Vandergraphs that can fit into any 500 series rack.

Shadow Hills Industries Dual Vandergraph

Vintage Tube Amplifiers and Custom-Designed Transformers set the soundstage. True Passive Equalization creates Bountiful and Musical Equalization curves. The new RETRO 2A3 is designed as an ultra High-Fidelity Program Equalizer whose sonic benefits go beyond the equalization process. Accurate to the industry-standard Pultec EQ performance, the RETRO 2A3 adds HF boost frequencies in several new sweet spots. Carefully implemented HF boost circuits capture the signature sound of the original passive equalizers. Use these settings to make individual tracks fit in place. Make a vocal shine and add distinct presence. There are many satisfying applications, most notably the stereo mix buss. A new Subsonic Filter allows you to do peaking low frequency boosts that tame the excessive subsonic energy exhibited in the original design if that’s what you want. Utilizing the interstage transformer, the filter not only reduces the subsonic energy but also provides distinct transformer tonal characteristics for tracking and mixing. The filter has settings of 40 and 90 Hz with a peaking response and sharp cutoff. The 40 Hz setting has no apparent loss of lows as it adds excitement to the 35-40 Hz region that is golden in many listening environments. The smoothing effect is pleasantly apparent throughout the midrange. It is easily switched-out for applications that require tight, accurate low frequency response. In consideration of the limited real estate in your rack, the RETRO 2A3 packs two EQ channels into one 2U space with surprising freedom of movement, ease of adjustment and familiarity. Use the two channels for stereo, separately on independent tracks or cascaded for extended equalization possibilities. The channel separation exceeds 70 dB. The RETRO 2A3 passive equalizer circuits do not rely on amplifier feedback, which can cause a harsh and clinical sound in typical active equalizer designs. By incorporating pure passive Capacitor - Inductor based equalization; the RETRO 2A3 sounds so natural and effortless even with extreme boosts and cuts. We also gave special attention to the Pultec-style bass boost/bass cut method that is essential to a powerful bass and kick drum EQ. This involves simultaneous low frequency boosts and lower-mid cuts that can scoop out the mud and add real punch. Quite simply, there are benefits to having the RETRO 2A3 in line without even equalizing, as the tube amplification has it’s own unique musicality. The RETRO 2A3 Dual Channel Tube Program Equalizer provides a very useful palette of colors and textures ready for tracking, mixing and mastering. Features <ul><li>Industry-standard Passive EQ with authentic feel, controls and equalization curves</li><li>Additional HF Boost Frequencies for more precise control</li><li>Easily recallable 100-position knob scales on boost and cut controls </li><li>A ganged interstage-coupled Subsonic High-Pass Filter provides Peaking LF Boosts with alternate tonal characteristics </li><li>Transformer Balanced and fully-floating 600 Ohm Input and Output to eliminate ground-induced hum and noise</li><li>LF Boost Settings of 20, 30, 60 and 100 Hz </li><li>Complementary LF Cut settings of 20, 30, 60 and 100 Hz</li><li>HF Boost Settings of 1.5 kHz, 3 kHz, 4 kHz, 5 kHz, 6 kHz, 8 kHz, 10 kHz, 12 kHz, 14 kHz and 16 kHz</li><li>Complementary HF Cut Settings of 5 kHz, 10 kHz and 20 kHz</li><li>Vintage Class AB tube amplification for unity-gain make-up</li><li>Equalization Bypass Switches</li><li>Size 2U Height: 19" wide x 3.5" tall x 9" deep</li><li>XLR input and output connections</li><li>IEC power connector</li><li>115/230V 50-60 Hz Switchable AC Mains</li><li>Uses readily available 12AX7(2) and 12AU7(2) electron tubes</li><li>Tube substitutions are possible with internal unity gain calibration and jumper selectable 6 or 12 volt tube configuration </li><li>Design minimizes tube heat generation</li><li>High stability components used throughout</li><li>Steel Chassis built to last </li><li>Manufactured in the USA </li><li>Full factory support</li></ul> Specifications for the Retro 2A3 include: <ul><li>Signal to noise ratio of greater than 76dB</li><li>Channel Separation greater than 70 dB</li><li>Frequency response is flat within 1 dB from 20 Hz-20,000 Hz</li><li>Harmonic Distortion is less than 1% from 20 Hz-20,000 Hz</li></ul>

Retro Instruments 2A3

Following in the footsteps of the MicroMain27, the new MicroMain35 brings revolutionary Barefoot three-way technology into an even more compact enclosure. Dual force canceling 7" subwoofers, a 5" midbass driver and 1" dome tweeter are fused seamlessly into a single source. In just a few years Barefoot Sound has built a reputation for groundbreaking technological innovation executed with uncompromising quality and performance. Barefoot monitors are known for delivering a legendary combination of deep sonic clarity and effortless translation. The Micro Main 35 lives up to everything you would expect from a Barefoot. Specifications <ul><li>Description 3-way active monitor with integral subs </li><li>Controls Input level attenuator, Twt/Mid/Sub amplifier mutes, crossover voicing select </li><li>Input Impedance 50k Ohm </li><li>Frequency Response 35Hz - 30kHz (+/- 3 dB), 37Hz - 20kHz (+/- 1.5 dB) </li><li>Bass Response -3dB @ 35Hz, Q = 0.707, Slope = 12 dB/octave </li><li>Cabinet 19 liters total internal volume, Sealed sub enclosure, Sealed midbass enclosure, Machined aluminum front baffle plate </li><li>Crossover Frequencies 100/2500 Hz </li><li>Tweeter 1" soft dome, Rear waveguide chamber </li><li>Power: 50 W </li><li>Midbass 5" poly cone, Sealed rear waveguide enclosure Power: 100 W </li><li>Dual Subwoofers 7" aluminum cones, 3/4" peak-to-peak linear excursion </li><li>Power: 200 W </li><li>AC Power Input Nominal 115 VAC or 230 VAC </li><li>Power Consumption Idle 20W, Maximum: 375W </li><li>Weight Speaker 42 lbs each (19 kg) </li><li>Shipping: 55 lbs each (25 kg) </li><li>Dimensions HxWxD 14.5 x 9.5 x 14.75 inches (368 x 241 x 375 mm)</li></ul>

Barefoot Sound MicroMain35 (Pair)

The Lexicon PCM Native Reverb Plugin Bundle is the result of several years of research and significant advances in computer processing speeds, that have enabled Lexicon to provide seven legendary Lexicon reverb algorithms with the original hardware levels of sonic quality and function, as native plugins.  The plugins can be used within popular DAWs such as Pro Tools and Logic, as well as with any other VST, Audio Unit, or RTAS compatible host. The bundle includes unique plugin for each algorithm, including vintage plate, plate, hall, room, random hall, concert hall and chamber, with hundreds of versatile and finely crafted studio presets including recognizable classics from Lexicon’s substantial library of sounds. Librarian functionality allows the user the option to take an available preset, adjust the parameters, compare it to the original and then return to the edited preset. Presets can also be saved off in a DAW independent file format allowing a user to easily move custom presets to any DAW. Features at a Glance <ul><li>PCM Reverb Plug-In 7 Algorithms</li><li>7 legendary Lexicon Reverbs</li><li>Hundreds of brilliantly crafted studio presets</li><li>Multi-platform compatibility (Windows XP, Vista, and 7; Mac OSX 10.4, 10.5, 10.6, PowerPC and Intel)</li><li>Formats that work seamlessly in any VST, Audio Unit, or RTAS compatible DAW</li><li>Graphical real-time display illustrating the frequency stages of each algorithm</li><li>Visual EQ section for easy adjustment of both early and late reflections</li><li>Presets can be stored in a DAW independent format which allows custom presets to be transferred between any DAWs.</li><li>Full parameter control and automation</li><li>Input and output meters for quick assessment of audio levels going to and from the reverb</li><li>iLok authorized</li></ul> Each reverb algorithm can be run in mono, stereo or a combination of the two. The user interface features a ‘pro or go’ mode which enables the user to easily access nine of the most logical parameters for customization, but also provides the ability to transition deeper into the algorithm to edit the full matrix of parameters. Input and output meters allow a user to quickly assess the audio levels going to and from each algorithm, while the eq section makes it possible to visually dial in the eq settings for both the early and late reflections of the algorithm. Each algorithm is complemented with three different real time displays providing graphical insight into what is happening inside the different frequency stages of the reverbs. The PCM Native reverb plugin bundle is a fully functional cross-platform program that is compatible with Windows XP, Vista, and 7 along with MAC OSX 10.4, 10.5, 10.6, Power PC and Intel based. The bundle is native only, and requires iLok authorization.

Lexicon PCM Native Reverb Plugin Bundle

The world of DAWs doesn't need another channel strip. What it needs is a better way of working with the essential tools you need for all of your tracks and busses. That's why we designed Alloy. Alloy provides the key dynamics and sound shaping effects you'll use again and again, in a self-contained and completely configurable interface. Add the iZotope Alloy to your tracks and busses and bring them to life with six processors designed to add character to every element of your mix. Alloy gives you exceptional sound quality, vintage emulation balanced with digital precision, time-saving presets and workflow features, and a forward-thinking interface that puts just the controls you need at your fingertips. Equalizer As essential as EQ is to mixing, why do so many EQs force you to work with tiny graphs or clumsy knobs? Alloy's powerful 8-band EQ combines exceptional sonic quality with precise controls and a large spectrum overlay that gives you the best visual feedback of any EQ available. <ul> <li> Enter EQ settings visually or with precise numerical values </li><li> Choose from Low Shelf, High Shelf, Peak, Lowpass, Sharp Lowpass, Highpass and Sharp Highpass modes for each band </li><li> Soft-saturation emulates analog EQs for a natural sound when pushed hard </li><li> Alt-click and use your mouse to solo and zero in on problem frequencies </li></ul> Exciter Alloy's Exciter features new Harmonic Scaling technology that lets you add gentle saturation to your tracks by emulating the sounds of tube pre-amps, tape, and other hardware devices in a unique new way. It's perfect for adding subtle character to everything from vocals to strings to drums. <ul> <li> Blend between different Harmonic profiles including Tube, Tape, Transistor and more with an XY pad </li><li> Switch to Multiband mode to apply color to different frequency ranges independently </li><li> Unique spectrum meter shows you which frequencies are being affected by the Exciter </li><li> Integrated stereo width control allows you to narrow or widen different frequency bands </li></ul> Transient Shaper Shape percussive sounds and more with Alloy's simple and powerful Transient Shaper module. Add more “stick” to a snare, make a kick drum less boomy, or even apply Transient Shaper to hammered or plucked instruments to fine tune their attack character. Try Multiband mode for unmatched shaping of more complex material like drum loops and busses. <ul> <li> Perfect for shaping drums and other percussive sounds </li><li> Concise controls let you boost or cut the attack and sustain portions of transients with adjustable time values </li><li> Adjust sounds in different frequency ranges with powerful Multiband mode </li></ul> Dynamics Smooth out vocals or slam your drum buss with a Dynamics module that combines the best characteristics of analog and digital compressors. While many “vintage emulated” dynamics plug-ins are useful on only some material, Alloy can switch between tasks with ease. And with advanced sidechaining and a multiband mode this is sure to become one of the most versatile mixing tools in your arsenal. <ul> <li> Vintage mode combines a forgiving multi-stage release curve with program dependent characteristics; Digital mode provides clean, linear compression </li><li> Gate/Expander provides gating and even upward compression effects </li><li> Sidechaining lets you trigger the Compressor or Gate from external sources– even to individual frequency bands in Multiband mode </li><li> Crosschaining lets you trigger dynamics processes on one band with audio from another, for completely new gating and ducking possibilities </li><li> Two dynamics stages for parallel compression effects </li><li> Soft Knee, RMS, and Auto-gain modes; single or multiband operation </li></ul> De-Esser Tame vocal sibilance and harsh high frequencies in other instruments. Alloy's De-Esser lets you zero in on sibilant frequencies with a simple-to-use spectrum graph. It's perfect for transparently reigning in vocals, cymbals and other high frequency problem spots. <ul> <li> Spectrum meter with solo control helps you set the De-Esser quickly and easily </li><li> Multiband mode affects just the sibilant frequencies; Broadband mode affects the whole signal based on the sibilant band. </li><li> Control over De-Ess attack and release times </li></ul> Limiter Alloy's Limiter module acts as a zero latency loudness maximizer for busses and tracks, automatically adding gain as you pull down the threshold. Or use it to keep a lid on your levels with a simple and efficient Brickwall limiter. <ul> <li> Brickwall mode for keeping peaks under zero at all times; Soft mode for more forgiving, smooth limiting </li><li> Phase Tools panel lets you optimize the symmetry of vocal and instrument waveforms to get more overall loudness </li><li> Zero-latency mode ensures your tracks and busses stay in sync, even in hosts without effective latency compensation </li></ul> MacroPresets Some plug-ins come with five or ten presets. Alloy comes with over 150 presets designed to give you the perfect starting point for a broad range of mixing tasks. MacroPresets customize Alloy's display to show you just the sliders and meters you need for the task at hand, giving you powerful preset signal chains with simple controls, making each preset feel like its own custom-tailored plug-in. <ul> <li> MacroPresets combine the most useful parameters and meters for a preset into one useful view. </li><li> Over 150 presets included covering individual instruments, ensembles, busses, utility tasks, broadcasting & podcasting, post production and more. </li><li> Create or customize your own MacroPresets that show custom configurations of controls in Alloy's Macro screen. </li></ul>

iZotope Alloy

The AEA A840 is different from other microphones. One listen and the AEA A840 stands out with its big, clean sound. It’s an ideal spot mic with its tight pattern, smooth extended top, and less proximity bass boost than the R44 and A440. The A840 effortlessly handles a broad range of applications from accent to ensemble. AEA A840 Features <ul><li>High Output</li><li>Compatibility with any preamp and input source</li><li>Low self noise of 17.5 dB (A)</li><li>Fast, accurate transient reproduction</li><li>High SPL capability of 141 dB</li><li>Single-diaphragm with well-controlled, native figure-8 polar pattern</li><li>Wide-band response from below 20 Hz to beyond 20 kHz</li></ul>The AEA A840 and other AEA Big Ribbon mics are excellent digital recording tools. Digital has different limitations than analog. The digital recording process is capable of preserving razor sharp details for generations. So it makes sense to use microphones that fit well into the final mix from the start. AEA Big Ribbon mics with their smooth, natural and easy-to-EQ sound have enjoyed a rebirth as digital recording has blossomed. Over 30 years of servicing ribbon mics taught AEA what users like, and they all like the big sound of the RCA 44. Preserving and reintroducing that sound became an AEA mission. The AEA R44 and A440s, robust, heavy, and expensive to manufacture, are now in daily use worldwide. But that’s just the beginning of the story. How can AEA preserve and update that Big Ribbon sound in a more affordable package using new technologies? The A840 uses the same big ribbon as the A440, is 5 pounds lighter, and less bulky. Due to its active JFET impedance buffer circuitry the new A840 is insensitive to lower preamp input impedances. The high output sensitivity reduces noise and allows long cable runs.This allows you to use your favorite tube preamp without compromising your classic ribbon sound. Designed for accent and solo work, its bass proximity effect is less pronounced and the upper 10 to 20 kHz octave is stronger than the A440. Why do the AEA Big Ribbon mics sound so different when compared to other microphones? Classic condenser diaphragms are radially stretched. They typically have sharp, narrow resonances in the 8 to 12 kHz range, the sound quality we refer to as “tizz”. AEA Big Ribbon mics use a long rectangular diaphragm clamped only at the ends and tensioned lightly with a fundamental resonance below 20 Hz. Less than 80 millionths of an inch thick, AEA's ultra low mass transducer has extended bass, excellent transient response and few resonances. It’s one of the most sensitive and accurate transducers possible. Many have commented that AEA mics record exactly what your ear hears. Excellent headroom and low distortion allow close micing with an A840 without sounding brittle. Additionally the smooth, non-resonant high end provides engineers considerable EQ flexibility. Try it, you’ll enjoy it.

AEA A840 Active Ribbon Microphone

The Bock 5-ZERO-7 is a new approach in high end microphone design for it combines new and old: a brand new unique and patent pending elliptical capsule married to a vintage inspired mic amplifier and power supply. Designed to be the ultimate in studio vocal microphones, the 507 compares favorably to the best vocal microphones, past and present. What makes it different? The new Bock/Cardas elliptical capsule design strives to resolve the chronic in-band resonance issues of both large and small diaphragm microphones. Round capsules have a relatively low frequency in band resonance, typically around 1K. Small diaphragm mics usually have a much higher in band resonance, typically about 15K or higher. The elliptical capsule offers the best of both, avoiding the constant distance "edge to center" of round capsules, reducing in band resonances dramatically and expanding the application possibilities. The ZERO capsule also has many conventional attributes such as the time tested 6 micron gold sputtered diaphragm used in the BOCK 251 and edge termination design. The capsule is hand built in very small quantities in Germany. The 5-ZERO-7 also offers many other unique features. A large body microphone, it uses a 3 layer headgrille to prevent moisture (and therefore dirt) from becoming a continuing issue in the sound and long term life of the capsule and microphone. A unique tube mic amplifier was designed of very high gain, vintage 47 inspired, with David Bock hand selected parts. No negative feedback or filtering is used in this microphone to affect sound in any way. An extraordinary US handmade BOCK design four section transformer is employed to complete the audio path. The custom BOCK power supply is designed just for this microphone and combines the best of the old with new and modern parts. It is a special inductor based design, something not seen in modern production microphones today. It offers the best combination of long term power, low noise floor and high end mic sonic performance. How do I use it? The 507 microphone itself is a large body type, to give singers and artists a familiar feel in the studio to vintage mics they know and love. The fixed cardioid nature of the microphone guarantees it will always be in its most popular setting without switches and other devices to decrease long term reliability. The microphone sounds very big, utilizing a carefully crafted proximity effect for that extra deep intimacy preferred by artists and engineers alike. This sound is expected to be especially popular on male vocalists but will be expected to outperform most other mics on females as well. There will be a wide variety of color available with distance, so this will increase it’s flexibility. You will find that the 5-ZERO-7 brings something new to voices, something unavailable in any other microphone, past or present, at any price. The Bock/Cardas Elliptical capsule can be fine tuned to more specific applications such as orchestra, etc.

Bock Audio 507

Quested V3110 Features<ul><li>Quested’s highest-rated self-powered monitoring system</li><li>3-way design with clean LF response to 30Hz from custom, Linear-travel long-throw 10” driver </li><li>1kW Combination of Class A/B and Class D amplification</li><li>World-renowned, low-fatigue soft-dome HF and Mid-range technology</li><li>Unique current-driven floating drive stage Class A/B power for HF and Mid drivers.</li><li>High-efficiency “Ultra Cool” 700W Class D amplifier for exceptional LF dynamics.</li><li>HF Trim +/-1dB</li><li>Mid Trim +/-1dB</li><li>Versatile LF Compensation -6/0/+3/+4 dB</li><li>Optimized 24dB/Oct crossover slopes for greater drive unit efficiency</li><li>Very high power density for cabinet footprint and size</li><li>Will outperform other 12”-based monitor designs</li></ul>

Quested V3110 - Pair

The Great River 32EQ is a new 500-series version of the EQ and filters from the renowned Harrison 32 Series consoles. The 32EQ incorporates the original specifications and with support directly from the original designers at Harrison Consoles it is guaranteed that the prized characteristics of the original were maintained in the new design. The 32EQ has the full features of the 32-series EQ <ul><li>Low, Low-Mid, Hi-Mid, and High EQ bands with Gain and Frequency controls</li><li>Low and High Band “peaking” switches</li><li>EQ in/out switch</li><li>Harrison’s renowned High- and Low-pass filters with sweepable frequency</li><li>Filter in/out switch</li></ul> An internal jumper provides selection of the “vintage” feedback design, or a non-feedback option. The Harrison 32-Series console was the world’s first 32-bus “inline” recording consoles. They became a staple among recording studios and were the basis for many console designs (Harrison and otherwise) that followed. Countless hit records were produced on Harrison consoles during the birth of modern pop productions, including Abba, Sade, Queen, Janet Jackson and Michael Jackson. The 32C console was used by Bruce Swedien in the recording and mixing of Michael Jackson’s Thriller, the best-selling album of all time. Gary Thielman said, “For many years, Harrison has had requests for our prized analog products in a smaller form-factor. During that time, we kept hearing great things about Dan Kennedy and Great River Electronics. Their products are superbly made and they are as fanatically supportive towards their customers as we are to our own. We realized that we had an opportunity to launch a product that the world has been requesting, while continuing forward with our passion which is building large-format consoles.” Like all Great River and Harrison products, the 32EQ is designed and built in the USA.

Great River / Harrison 32EQ

For recording and mixing engineers, the greatest challenge will always be to capture and transform disparate 2-D elements into an engrossing 3-D vision that elicits the same raw emotions and compelling vibrance as the live performance. In designing the Rupert Neve Portico II Channel Strip, every possible attention has been paid towards not only producing the most accurate, sweet sounding audio topologies, but also enabling engineers to boldly shape or fine tune tracks to the limit of their imaginations. A self-powered 2U channel module comprised of a fully- featured mic preamplifier, 4-band EQ, compressor-limiter, “texture”control and level metering, The Portico II Channel includes a bevy of new features including: Variable Silk / Silk+ Texture control, a fully tunable “de-esser”, multiple VCA filtering and detection options, a transient-optimized swept HPF and parallel compression blending. With its simple yet powerful topologies and extensive feature set, every element of the Portico II: Channel is geared towards empowering recording artists to realize these visions. Input Section Input signals may be derived from any of three sources; Microphone, Line or DI. The Microphone input has a 10K Ω non-reactive input resistance that handles a full 26dBu signal without the need for a pad. A primary Gain control provides 66dB of gain in 6dB increments for easy recall with a +/- 6dB Trim for fine adjustment. A transient-optimized HPF continuously covering the range from 20-250Hz is included. In addition the usual 48V Phantom Power switch, Phase Inversion, Signal Present indicator and a Mute switch are included. The pre-amp is a TLA non-reactive design that precludes the loading of microphones with limited driving capability and which maintains a 2 dB noise figure over a wide range. Inserting a plug into the DI input, selects a 3M Ω Discrete FET DI circuit that includes the transformer-coupled pristine topography that optimizes both the microphone and DI inputs, allowing low impedance microphones and high impedance sources like electric guitars and basses to deliver their optimum performance. Primary Gain and Trim controls are fully operational. A “Thru” jack on the front panel that duplicates the unaffected source signal allows the use of an external amplifier if required. A “Line” switch selects the independent Line XLR non-reactive input circuit that includes the HPF and Level trim controls. Texture One of the key developments in the Portico II is the new Texture section. Building on the “Silk” mode found in the Portico Series mic pre-amplifiers, Texture is designed to adjust the actual amount of harmonic music content from the source material, in effect, providing countless tonal options in one device. The texture section features distinct modes, controlled by a potentiometer from barely audible to dramatic!. With that said, we have placed great care insuring that the musical integrity of source material itself will not be compromised by these harmonic controls (these are not to be confused with controls from other manufactures that may “incinerate”, “demolish” or “destroy” source material), and we believe they can be confidently modified to fit the personality of any song, instrument or engineer. Like the original Portico, Silk mode works by reducing negative feedback on the output transformer and adjusting the frequency response to more closely resemble Mr. Rupert Neve’s vintage designs. “Silk +” mode pushes this technique still further, achieving a more harmonically rich sound. Like the original Portico, Silk mode works by reducing negative feedback on the output transformer and adjusting the frequency response to more closely resemble vintage designs by Mr. Rupert Neve. Velvet mode pushes this technique further, by increasing saturation on the output transformer to achieve a more colored sound. With that said, we have placed great care insuring that the musical integrity of source material itself will not be compromised by these harmonic controls (these are not to be confused with controls that may “incinerate”, “demolish” or “destroy” source material), and we believe they can be confidently modified to fit the personality of any song, instrument or engineer. Equalizer The EQ topologies in the 4-Band Equalizer of the Portico II combine the best-loved sound of several classic designs, but are also capable of extremely fine surgical adjustments as well as dramatic boosts or cuts. By default, the EQ section follows the mic preamplifier and HPF, and precedes the compressor, but this sequence may be reversed by engaging the Post Comp switch The Low Shelf has an independent engage switch, Classic Peak/ Accelerated Shelf selection, with +/- 12dB level adjustment, and selectable turnover frequencies of 35, 60, 100, and 220 Hz. The Fully Parametric LMF and HMF bands have a joint engage switch, providing +/- 12dB level adjustment, continuously variable “Q” from 0.7 to 5 and continuously variable frequency ranges from 70 to 1.4KHz and 700Hz to 14KHz. In addition to its standard capabilities, the HMF band can also be used to precisely tune a new De-Esser circuit. The De-Esser can be varied from off to full effect, and uses the Frequency, and “Q” controls to tame harsh sibilance in vocals and instruments with an independent high-mid limiter. Even when the De-Esser is on, the EQ level may still be used. The High Frequency shelf has independent engage switch, Classic Peak/ Accelerated Shelf selection, +/- 12dB level adjustment, and selectable turnover frequencies of 4.7, 6.8, 12, and 25kHz. When two Portico II Channels are used side by side, the EQ section can also excel at stereo material. Compressor A further array of tools are used to craft dynamic response. The Portico II’s compressor section is exceptionally flexible. Like the Portico 5043, the compressor a choice of Forward Feed / Return Feed modes, either providing a transparent modern response, or a smoother more musical vintage response. Along with FF/FB modes, there are also 1:1 to 40:1 ratio controls and threshold from -30dBu to +20dBu, together with Attack from 20 to 75ms (.1 ms in fast mode), Release from 100ms to 2.5s, and Make up gain from 0 to 20dB with a Stereo Link. The Portico II compressor however incorporates a number of new ways to tame or enhance dynamics. HPF to SC inserts the High Pass Filter into the side chain to deal with intense low frequencies that may skew the response of the VCA with certain songs and instruments. The “Fast” switch alters the compressor’s attack to react to transients with a roughly .1ms response time. When “fast” is disengaged, the compressor responds to the RMS signal in conjunction with the attack and release settings. “Blend” works by creating a parallel mix between the compressed and dry signals. By mixing the compressed and dry signals, it is possible to increase the volume of quieter elements in the source material (for instance, delicate snare brushing on a track with much louder hits), while maintaining a natural dynamic feel for the louder elements. To further control the side chain, there is also an insert “send” and “return” that may be paired with an external EQ or other filters for additional manipulation. The “return” may also be used as a “Key” input for ducking one signal under another. (for instance, a voice-over keying the compressor to duck a background music track). By connecting the “link” jack on the back of the Portico II to another Portico II link jack and engaging the Link buttons, the compressor section may be used for stereo material.

Rupert Neve Designs Portico II Channel Strip

The name says it all - Waves Vocal Rider is designed to match your vocal tracks to the rest of the mix and then keep the relative volume throughout the song. All you do is instantiate it on your vocal track, bus a summed instrumental mix to the side-chain, and hit play. The plug-in fader then takes over, keeping the relative volume of the vocals and the rest of the mix constant. And, unlike compression, Vocal Rider doesn't crush the dynamic range or color the vocal track at all. Features <ul><li>Intelligent adjustment of the vocal track based on the dynamics of the music </li><li> Saves time doing the kind of automation you would have to do manually </li><li> Writes automatic volume riding to the track as automation for fine tuning </li></ul>

Waves Vocal Rider - TDM

When it comes to sound, image, mystique, and just plain cool factor, few pieces of vintage studio gear rival Tube-Tech’s masterful morsel of Danish sonic sweetness, the CL 1B opto compressor. Even with its mono operation and $3,000 price tag, this piece has been a staple of high-end recording facilities from the day it was first introduced over 20 years ago. From vocals to guitar to drums, world-class recording engineers have consistently turned to the CL 1B for effortless, musical, sonically pure compression. And now this timeless classic can be part of your native DAW rig for about a 10th the cost of the original hardware - without any sacrifice in audio quality or performance. Isn’t there already a CL 1B plug-in available? Absolutely. And guess who designed it? (Hint: Starts with S, ends with E, and the middle letters are oftub). That particular plug-in, however, requires a TC PowerCore or Digidesign TDM system. The all-new Softube Tube-Tech CL 1B is entirely native - Mac or PC, VST/AU/RTAS - and believe it or not, has more features than the versions that require hardware! More features? Surely you jest! For starters, the new CL 1B features the same painstakingly-constructed modeling algorithms as the TDM/PowerCore version. Then Softube doubled up the capabilities and enabled the unit to operate in stereo! (Purists should note thatthe left and right channels are linked for phase accuracy.) Topping everything off is a new external sidechain input that let’s you trigger the compressor from the selected source audio - just the thing for creating classic effects like locking the bass drum compression to the kick. (Note that RTAS and AU hosts support sidechain inputs; please check your VST host to see if this feature is implemented.)

Softube Tube-Tech CL-1B Opto Compressor - Native

The Lipinski Sound L-609 500 series mic pre is based on the same fundamental, discrete design of the patented “Lipinski Square” circuitry. The L-609’s clean signal path contains no integrated circuits (IC) or capacitors and the unit has a 2-step custom input transformer as well as a transformerless, low impedance output. Lipinski Sound L-609 Features <ul><li>"500 Series" Format</li><li>Fundamental, discrete, design based on patented 'Lipinski Square' circuitry</li><li>No capacitors in the signal path</li><li>No Integrated Circuits (IC's) in the signal path</li><li>2 step custom input transformer</li><li>Transformerless, low impedance output</li><li>Pad doesn't degrade signal quality, it only eliminates one stage of amplification</li><li>Low power LED meter that does not affect preamp performance</li><li>Switchable VU/Peak meter</li><li>220K guitar input impedance</li></ul>

Lipinski L-609 Mic Preamplifier

The Lipinski Sound L-629 500 series compressor’s basic principle of operation and ballistics is based on the famous Fairchild Compressor as its extremely fundamental design is rooted in the patented “Lipinski Square” circuitry. This unique circuitry completely eliminates distortion on low frequencies and on low-release time a covetable quality in a compressor of this kind. Lipinski Sound L-629 Features <ul><li>Basic principle of operation and ballistics based on the famous Fairchild compressor</li><li>Patented circuit completely eliminating distortion on low frequencies and on low-release time. </li><li>All other compressors on short release time and on low frequencies changes amplification within single sine wave, </li><li>commonly a major source of distortion.</li><li>Extremely fundamental design based on patented "Lipinski Square" circuitry</li><li>No capacitors in the signal path</li><li>No Integrated Circuits (IC's) in the signal path</li><li>Only 0.04% maximum distortion with the highest (18dB) compression</li><li>Extremely fast attack time - 0.01ms (1/5th 20kHz sine wave)</li><li>Tape Simulation doesn't generate gimmicky even-harmonic distortions. This feature senses and triggers compression on high frequencies (correction within control network)</li><li>Auto gain: additional 5 sec. attack time, and 5 min. release time</li><li>Auto control has ballistics similar to VU meter, it is based on photo resistor, completely eliminates distortion with zero adjustment error - quality on output signal totally independent of input level</li><li>Peak meter: switchable before and post compression</li><li>Hard wiring option for stereo operation </li></ul>

Lipinski L-629 Compressor

Presenting the Grace Design m501, a 500 series module version of the venerable m101 mic preamplifier. Now the signature transparency and detail of our circuit designs is available for your 500 series frame. While the 500 series market has plenty of colored mic preamplifier options, the natural, musical clarity of the m501 makes it a welcome addition to the field. This circuit is for engineers confident with the quality of the source, mic section and placement, and wish to capture it with as little coloration or distortion as possible. The m501 module is a balanced, transformerles preamplifier, with 48V phantom, a 75Hz HPF and a 1/4” HI-Z instrument input. Also standard is our exclusive ribbon mic mode, which raises the mic input impedance, bypasses the input decoupling capacitors and deactivates 48V phantom to protect delicate ribbon mics from damage. Large diaphragm, vintage ribbon or trusty dynamic - the m501 brings out the very best in any microphone and takes the guesswork out of your input chain.

Grace Design m501

The PMC DB1S-AII is the smaller of the two astonishing compact activated models standing just 290mm/11” high offering a supreme level of detail and scale like no other compact monitor. Speed, dynamics and depth of bass are not sacrificed for size. With the combination of Advanced Transmission Line (ATL™) technology and PMC’s ultra clean and highly potent, 200W, DS-001 power module the DB1S-AII offers performance greater than other monitors twice its size. The DB1S-AII is an extremely dependable high resolution monitor and is ideal for situations where space is at a premium but quality must not be compromised. It best suits smaller control rooms and is ideal for a 5.1 system when partnered with a TLE1S powered sub. The exceptional II upgrade The power amplifier section of both DB1S-AII and the larger TB2S-AII have undergone radical improvements with the inclusion of PMC’s all new DS-001, a cutting edge Class D power module. This unit ups the power from the previous module’s output of 100W to 200W and radically reduces distortion producing a significantly cleaner, faster more neutral presentation across the full bandwidth. This innovative, ultra low noise design is not load dependant, as a single feedback loop circuit senses the loudspeaker load and adjusts the response accordingly. This cutting edge technology when partnered with the highly accurate ATL™ (Advanced Transmission Line) bass loading system produces an entirely transparent listening window to judge any piece of audio.

PMC Loudspeakers DB1S-A II (Pair)

The Shure KSM313 is a premium bi-directional ribbon microphone for world-class audio and performance. The Dual-Voice design features discrete front-and-rear side sonic signatures for use with amplified instruments or in intimate vocal applications. Hand assembled in the USA with custom components and proprietary Roswellite™ technology for revolutionary ribbon resilience under extreme conditions. The high-tensile strength, toughness and shape memory of Roswellite™ ribbon material replaces traditional foil ribbons for superior resilience at extreme SPLs. With unique technology and application features, alongside superior materials and construction, Shure KSM Ribbon Microphones are the necessary choice for the world’s most accomplished studio and live performances. Features <ul><li>Bi-directional polar pattern delivers premier audio with superior off-axis rejection</li><li>True Dual-Voice tuned specifically for user’s for choice of response - warm and full for amplifiers, or swivel for bright and flattering vocals</li><li>Revolutionary Roswellite™ proprietary ribbon material replaces traditional foil ribbons with high tensile strength, toughness and shape memory that provides superior resilience at extreme SPLs</li><li>Custom-sourced components for superior presence, minimized signal loss and maximized output.</li><li>146dB SPL across 30 – 15,000 Hz frequency response ideal for capturing fast transients in vocals, acoustic instruments and concert halls</li><li>Legendary Shure quality and superior construction from hand-assembly of machined steel, silver, gold and aluminum materials</li><li>Monocle stand mount custom swivel mount for fully versatile placement and orientation</li><li>Protective wood case included for storage when not in use</li></ul>

Shure KSM313

The Shure KSM353 is a premium bi-directional ribbon microphone crafted for pristine audio in studio and concert hall applications. Proprietary Roswellite™ technology provides revolutionary ribbon resilience and durability under extreme conditions. Hand assembled in the USA from state-of-the art transducers, transformers and metals as the pinnacle of Shure quality for prestigious vocal and acoustic performances. The high-tensile strength, toughness and shape memory of Roswellite™ ribbon material replaces traditional foil ribbons for superior resilience at extreme SPLs. With unique technology and application features, alongside superior materials and construction, Shure KSM Ribbon Microphones are the necessary choice for the world’s most accomplished studio and live performances. Features <ul><li>Bi-directional polar pattern delivers premier, completely symmetrical audio with superior off-axis rejection</li><li>Revolutionary Roswellite™ proprietary ribbon material replaces traditional foil ribbons with high tensile strength, toughness and shape memory that provides superior resilience at extreme SPLs</li><li>Custom ribbon motor assembly tailors bass response without attenuating the overall output for full low and mid ranges, and superior upper range presence from a rising frequency response</li><li>Double-shielded matched full-sized transformer minimizes signal loss and maximizes output while reducing magnetic and RF interference from 90 degree placement, offset relative to the ribbon</li><li>146dB SPL across 30 – 15,000 Hz frequency response ideal for capturing fast transients in vocals, acoustic instruments and concert halls</li><li>Legendary Shure quality and superior construction from hand-assembly of machined steel, silver, gold and aluminum materials, and housed in a pure stainless steel casing</li><li>Spring-loaded suspension shock mount for flexible, fully-isolated stand mounting.</li><li>Upright protective wood case included for storage when not in use</li></ul>

Shure KSM353

The Weiss Powerhouse is a "black box" type universal digital signal processor for audio. Software is downloaded via Ethernet from a host computer. Control of the Powerhouse is also done via the host computer. The host computer typically is a PC or Mac, but can also be a dedicated remote control connected via Ethernet, RS232 or MIDI. The DSP power and memory capacity of the Powerhouse allows for all kinds of signal processing tasks, including multichannel for surround and outboard, live applications, fast convolution, high sampling rate processing for mastering etc. The Powerhouse contains 10 DSP chips which sit on modules and thus can be upgraded to the latest chips available. A 32 bit controller interfaces between Ethernet and DSPs. Large RAM chips offer a total of 160 megasamples for audio storage. On the back of the unit there are 4 slots for audio I/O interface cards. Formats like AES/EBU, analog, Firewire etc. are supported. Synchronization sources can be internal (all standard rates from 44.1kHz to 192kHz) or external via AES/EBU, Wordsync or one of the audio I/O interfaces. External sync sources are not limited to the standard sampling rates.

Weiss Powerhouse

The Gyraf Audio Gyratec X is a true tube stereo compressor based on the classic variable-mu principle. This method - controlling the gain by altering quesient current in differential tube pairs - is much faster than the optical method. The speed is why it was widely used in early limiter devices such as the Fairchild 670 or the Universal Audio 176, but the sonic byproducts of this topology are the main reasons why it is so popular today. The controls on the G10 are as follows: The input level pot is positioned right after the input transformers, thus allowing you to control the input level, the “drive” for the compressor. The Threshold pot controls at what level the compression will set in. Turning counter-clockwize will select a lower start point, resulting in more compression. When turned fully clockwize, the compression is deactivated. The Ratio pot controls the amount of compression in relation to changes in the input level, that is, for the part of the signal that exceeds the threshold value that is set. Setting Ratio fully counter-clockwize will deactivate compression. The Attack pot controls the time the compressor takes to react to a rising input level. The Release pot controls the amount of time it takes from the input signal falling below threshold, to the gain reduction being returned to unity. The Output level pot controls the signal level to the output driver stage and the output. The Bypass switch bypasses the unit completely. The hard way. - A relay simply takes out the active electronics and shorts the input to the output connectors. The G10 is floating, transformer balanced in and out. Input impedance > 10K Ohm, output impedance < 1K Ohm. It is based on a very short true-differential signal path from input to output, using very, very tightly selected PCC189 (7ES8) remote-cutoff dual triodes for the VCA stages, and E88CC (6DJ8) low-noise triodes as cathode followers for the output stage drivers. Both tubes are still fairly easily available throughout the world. Although semiconductors and opamps are used in this unit, they’re confined to powersupply and sidechain functions. At no time will your audio pass through anything but transformers, tubes and passives. So we’re talking REAL tube audio here.. G10 is as standard available as a 230V mains unit - other mains voltages will be made on request (at no extra cost) We have recently needed to change the face plate of the G10, as the panel meter originally used went obsolete from our manufacturer. The panel meter is the ONLY thing we’ve changed, as the G10 seems to be a very fine design indeed - running unchanged for over eight years now..

Gyraf Audio Gyratec X

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