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.
"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.
The original Helios Type 69 mic pre/EQ, designed by Dick Swettenham, was "The Sound of Olympic Studios", the legendary recording studio used by Led Zeppelin, The Rolling Stones, The Who, The Faces, Humble Pie and Bad Company to name a few. 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.
The single rack unit Lexicon PCM92 employs the latest DSP technology and a collection of classic Lexicon algorithms to provide an all-in-one processor for live and studio applications, delivering 28 mono and stereo reverbs, delays, and modulation effects, flexible routing configurations, and a comprehensive library of over 700 factory presets. Lexicon claims that the PCM92 represents the most advanced audio processor in its class.The PCM92’s capabilities over traditional reverbs are made possible by the adding of multimode filters in the audio flow, which allows for exact shaping of the reverb while still maintaining the distinctive Lexicon Sound. All reverbs have an infinite switch that allows the reverb to run forever, which is useful for creating unusual backgrounds and sound effects.The Lexicon Room algorithm provides a more flexible alternative to convolution type reverbs. Reflection patterns can be easily selected, scaled and equalised, all while simultaneously passing audio, and the room size can be instantly changed or reversed. The Room algorithm provides the realism of the actual space with the precise control and manipulation that is made possible by a reverb synthesizer.With two channels of balanced XLR or quarter-inch analog I/O and two channels of XLR AES/EBU digital I/O, along with MIDI, wordclock, and ethernet, the PCM92 integrates easily with any DAW and features 24-bit A/D-D/A conversion and 44.1 to 96 kHz sample rates.To supplement front panel navigation for live applications, the PCM92 can be configured and fully controlled remotely through Harman HiQnet System Architect. The foot switch and foot controller inputs offer additional flexibility for the live performer. The Lexicon PCM92 Stereo Reverb/Effects feature set includes: Lexicon® studio-standard reverbs, delays, and modulation effectsComprehensive library of over 700 finely crafted factory presets, including recognizable classics from Lexicon’s immense library of soundsLexicon’s flexible Room algorithm used to generate a vast array of room-related effects for music and postMulti-voice pitch shift to enhance vocalsMono and stereo versions of all algorithmsFull system configuration and control with HiQnet System Architect™44.1 to 96 kHz sample rate, 32-bit floating point processingTwo channels balanced analog I/O (XLR and ¼”)Two channels XLR digital AES/EBU I/OExternal BNC WordclockMIDI in, out, and thruEthernet connectivityFoot controller inputs make it easy to change presets and adjust parameters during live performancesHigh-resolution OLED display is easy to read in all light conditions
Presenting the Grace Design m501, our 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 design, 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.FeaturesFast and musical transimpedance architecturePrecision audio path with 0.5% precision metal film resistorsNew 12 position gold plated rotary gain switchHigh performance output line driver amplifier and HPF amplifierNew Ribbon mic mode (also great for dynamic mics) - Relay bypass of phantom power decoupling capacitors, increased input impedance, and 48V lockoutWide 10-75dB gain rangeEnhanced RFI interference suppressionThree output connectors: XLR balanced, TRS balanced and ¼” unbalancedBombproof laser-etched black anodized frontpanelSealed gold contact relay for Hi-Z input switchingLed indicators for +48V, Ribbon mode, and HPFFive year warranty on parts and laborMade in the USA
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 GlancePCM Reverb Plug-In 7 Algorithms7 legendary Lexicon ReverbsHundreds of brilliantly crafted studio presetsMulti-platform compatibility (Windows XP, Vista, and 7; Mac OSX 10.4, 10.5, 10.6, PowerPC and Intel)Formats that work seamlessly in any VST, Audio Unit, or RTAS compatible DAWGraphical real-time display illustrating the frequency stages of each algorithmVisual EQ section for easy adjustment of both early and late reflectionsPresets can be stored in a DAW independent format which allows custom presets to be transferred between any DAWs.Full parameter control and automationInput and output meters for quick assessment of audio levels going to and from the reverbiLok authorizedEach 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.
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 Low, Low-Mid, Hi-Mid, and High EQ bands with Gain and Frequency controlsLow and High Band “peaking” switchesEQ in/out switchHarrison’s renowned High- and Low-pass filters with sweepable frequencyFilter in/out switch 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.
Primacoustic London 12A Broadway studio kits have been developed for typical rooms with dimensions that are common in both residential and commercial construction. These easy-to-use kits contain everything you need to address room problems such as primary reflections, flutter echo, standing waves and excessive bass. Room designs follow a variant of the LEDE concept (live-end, dead-end) whereby the source or transmit section of the room has greater dampening while the receive section is sparsely treated in effort to retain natural room ambiance. Primacoustic London 12A Kit Consists of: 2x Broadband Panels8x Control Columns12x Scatter Blocks28x Surface Impalers8x Corner ImpalersWall anchors/screws (100)Drill Bit for anchors Each Broadway kit includes a selection of acoustic panels, the corresponding mounting hardware and instructions for easy installation. Broadway panels are made from high density 6lb per-cubit-foot fiberglass panels for upwards to five (5) times greater absorption than offered by typical low cost 1.3lb foam alternatives. This means that you get more absorption with less panels on the wall while assuring a more even absorption curve throughout the frequency range.Broadway panels are completely encapsulated with micromesh on both front and rear surfaces and the sides are resin hardened to create sharp looking architectural lines. The panels are then covered in a durable fabric and offered in choice of three esthetically neutral colors that will easily adapt to the most demanding interior designer needs. For those that have extra artistic flare, the panels may be covered with any breathable fabric without compromising the acoustical performance. Broadway panels have been tested to meet stringent Class-1 fire safety requirements making them suitable for use in public places.Panel mounting is of paramount importance, especially when you decide to relocate the panels to another room or change your work space. Primacoustic Broadway kits come complete with easy-to-mount impalers that screw onto the wall surface on which the panels hang. Unlike foam that is "permanently" mounted using invasive glue, Broadway panels can be removed without having the wall surface completely redressed by a drywall specialist when you decide to make a change. Clean up is simply a matter of filling the holes and repainting. And you can still use your Broadway panels afterwards!Broadway kits present a high performance solution that is attractive, affordable and easy to manage!
When The Beatles recorded "All You Need Is Love", the man behind the board was Eddie Kramer. When Jimi Hendrix recorded "Purple Haze", the man behind the board was Eddie Kramer. When Led Zeppelin recorded "Whole Lotta Love", the man behind the board was Eddie Kramer. When you’re ready to make some music history of your own, get Eddie Kramer behind the board. Featuring Complete all-in-one processing chainsMono-to-stereo and stereo componentsOptimized control ranges24bit/96kHz resolutionSupports RTAS, Audio Suite, VST, AUPC and Mac compatible Waves Eddie Kramer Collection Plugins Kramer Vocal ChannelKramer Bass ChannelKramer Drum ChannelKramer Guitar ChannelKramer Effects Channel Eddie Kramer on the Vocal Channel "When mixing, there’s always a fight going on between the vocal and the rest of the track. For me, what makes a good mix is when all the elements combine seamlessly, complimenting one other. When we designed the Vocal Channel plug-in, we did it in such a way that the user can position the vocals within the mix, and then blend it back into the track, without it popping out, in such a way that it retains all its original clarity and presence." Vocal Channel features 2 flavors: Vocals 1, designed for classic rock vocals with a lot of dynamic range and intensity.Vocals 2, for gentler vocal performances that stay at smoother, steadier levels. Controls Type toggles between the 2 vocal types: Vocal 1, Vocal 2Sensitivity controls input levels.Sensitivity LED indicates the presence of proper levels.Bass controls low frequencies.Treble controls high frequencies.Compress controls amount of compression applied to the signal.Output controls the output level.FX controls the amount of signal sent to the effect.Dly sets the delay time. (Vocal 1 only)Dly Mix controls the amount of audible delay. (Vocal 1 only)Verb Mix controls the relative level of the reverb.Meter Switch toggles meter monitoring between input and output modes.Meter displays input or output. Eddie Kramer on the Bass Channel "The idea behind the Eddie Kramer Bass Channel was to create a fat bass sound with a lot of presence, that cuts through without being overbearing. In general, the low-mid frequencies are accentuated, so it really pops on radio mixes. That way, you can always hear it clearly, even at low volumes on small speakers. The Bass Channel comes in two flavors: Bass1, which is less compressed, for a more dynamic sound that has room to breathe, and Bass2, a more aggressive sound with increased compression." Controls Type toggles between the 2 bass types: Bass1, Bass2.Sensitivity controls input levels.Sensitivity LED indicates the presence of proper levels.Bass controls low frequencies.Treble controls high frequencies.Compress controls the compressor intensity.Output controls the output level.Meter Switch toggles meter monitoring between input and output modes.Meter displays input or output. Eddie Kramer on the Drum Channel "The Drum Channel plug-in really captures the essence of my drum sounds. Whether you’re trying to achieve a huge Zep-like sound with loads of atmosphere, or are going for something more dry and down-to-earth like the Stones, or need a heavily-compressed, squashed, Hendrix-type sound, this is the plug-in that will get you there, quickly and easily. Since there are individual modes for bass drums, snares, and so on, you can really mix things up by creating your own hybrid drum kits, using your favorite elements from each." Controls Type toggles between the 6 drum types: BD, SNR, HH, Toms, OH, Room.Sensitivity controls input levels.Sensitivity LED indicates the presence of proper levels.Compress controls dynamics.Gate controls the gate threshold (available in BD only).Treble controls high frequencies.Bass controls low frequencies.FX controls the effects send gain.Output controls the output level.Meter Switch toggles meter monitoring between input and output modes.Meter displays input or output. Eddie Kramer on the Guitar Channel "The Guitar Channel plug-in features settings for lead guitars and two types of rhythm guitars. When it comes to lead guitar, I want it to become a living, breathing organism, creating a palpable sense of excitement within in the mix. By combining the right amounts of EQ, compression, delay, reverb, and a touch of flange, these 5 elements, working together, make it come alive. For rhythm guitar, I try to get it “in your face” as much as possible, without over-processing the sound with EQ. By adjusting the amount of compression, and not overdoing the amount of space, I make sure that the guitar stays up front, where it belongs." Controls Guitar TypeType toggles between the 3 guitar types: Rhythm 1, Rhythm 2, LeadSensitivity controls input levels.Sensitivity LED indicates the presence of proper levels.Compress controls guitar dynamics.Treble controls high frequency range.Mid controls midrange frequencies.Output controls the output level.FX controls the effects send gain.Depth controls the flange effect depth.Flange Mix controls flange effect mix. (Lead only)Verb Mix controls the relative level of the reverb.Dly sets the delay time.Dly Mix controls the amount of audible delay.Meter Switch toggles meter monitoring between input and output modes.Meter displays input or output. Eddie Kramer on the Effects Channel "For the Effects Channel plug-in, I set out to recreate some of the basic elements that I use whenever I’m painting a sonic picture. Over the years, these elements have evolved to suit a variety of sources and styles, from whispering vocals to screaming guitars and beyond. H-Slap is a shorter delay that emulates tape at 15 inches per second, with some EMT plate reverb at a medium setting.Z-Slap is a longer delay (7 ½ inches per second) with a bit of feedback and a longer setting on the EMT plate. Between the two, you’ll easily find the ideal setting for almost anything you can throw at it." Controls Type toggles between the 2 effect types: H-Slap, Z-Slap.Sensitivity controls input levels.Sensitivity LED indicates the presence of proper levels.Dly controls the time of the slap delay.Dly Mix controls the mix or direct send to the slap delay.Size controls the size and time of the reverb effect.Brightness controls the High Frequency range of the reverb effect.Verb Mix controls the amount of reverb mixed with the signal.Output controls the output level.Meter Switch toggles meter monitoring between input and output modes.Meter displays input or output.
The D.W. Fearn VT-7 has an all Class-A vacuum tube audio path, like all D.W. Fearn products. The gain reduction elements utilize circuitry that duplicates the sound and characteristics of the finest classic vacuum tube compressors, without depending on tubes that are no longer manufactured. The control circuitry is modern solid-state analog. The two channels may be used independently or linked together for stereo. A built in sidechain high-pass filter may be selected to reduce the compression on bass-heavy material. Controls (each channel): ThresholdGainAttackReleaseHarder/SofterSeparate/Link/Link HPF The VU meters may be switched between indicating gain reduction or output level. The VT-7 has a highly-transparent sound, although it can be used more aggressively when needed. Artifacts of the compression process are substantially lower than in most other compressors. The two channels may be used independently, or linked for stereo. In addition, a third position on the Link switch inserts a high-pass filter (HPF) in the sidechain to reduce the gain-reduction sensitivity of the VT-7 to low frequencies. This is a gentle roll-off, nominally at 100Hz. This allows a bass-heavy mix to have a better frequency balance, and prevents heavy low frequency audio from modulating the audio level. (NOTE: The photo of the VT-7 shows an older unit without the Link HPF position.) Like all our products, the VT-7 is designed to process high-quality audio and make it sound even better. The VT-7 is equally adept in tracking as well as as a bus compressor. Even jazz and classical tracks can be sent through the VT-7 for some mild gain control without obvious effect other than a significant improvement in the overall sound of the mix. Performance (no gain reduction, unity gain)Frequency Response: 20Hz - 20kHz +/- 0.5dB -3dB at 5Hz and 65kHzTHD: less than 0.07%Signal to Noise Radio: better than 80dB
Find out why the MicroMain27 is the most sought after speaker in pro audio. Call Vintage King and demo today!The Barefoot Sound MicroMain27 is a groundbreaking new monitor that is in a class all its own. It is quickly becoming the premier choice for top mixing and mastering engineers. The speaker is designed to address the demands of modern recording. It breaks down the barriers between nearfield, mastering and main monitors. No need to have multiple pairs of speakers crowding your studio; no need to guess what the mastering engineer is going to hear. The MM27 is compact and powerful, truly a "nearfield on steroids." While exceptionally neutral and designed for critical listening, the MM27 is still very capable of rocking the house. It redefines the definition of a main monitor -- a "Micro Mainª" monitor. The only monitor you may ever need. 1" soft dome tweeter, dual 5" midbass drivers and dual 10" subs housed in compact sealed enclosures yield high linearity and outstanding impulse response. With 500 Watts of power in the subwoofer channel alone, the dual 10" drivers cross over seamlessly from the midbass, reaching down to 33Hz and rolling off at 1/4 the rate of ported designs to reveal much more deep bass information. Since the sub motor structures are locked together the opposing forces cancel out and the cabinet remains rock steady even at very high output levels. There is no need for bass management in 5.1 systems because the MicroMain27 is truly a full-range monitor. The speaker can be placed either vertically or horizontally using the included pedestal (9" L x 13/8" W x 21/4 H). Hand Crafted in San Francisco, CaliforniaSpecifications Description 3-way active monitor with integral subs Controls Input level attenuator, Twt/Mid/Sub amplifier mutes, crossover voicing select Input Impedance 50k Ohm Frequency Response 38 Hz - 20 kHz (+/- 1.5 dB) Bass Response -3 dB @ 33 Hz Q = 0.707 Slope: 12 dB/octave Cabinet 32 liters total internal volume, Sealed sub enclosure, Individual sealed midbass enclosures, Machined aluminum front baffle plate Crossover Frequencies 110/2500 Hz Tweeter 1" soft dome, Rear waveguide chamber Power: 60 W Dual Midbasses 5" poly cones, Sealed rear waveguide enclosures Power: 250 W Dual Subwoofers 10" aluminum cones, 1" peak-to-peak linear excursion Power: 500 W AC Power Input Nominal 115 VAC or 230 VAC selectable Power Consumption Idle: 30W, Maximum: 675W Weight Speaker: 71 lbs each (32 kg) Shipping: 79 lbs each (36 kg) Dimensions HxWxD 20.5 x 9.5 x 15.25 inches (521 x 241 x 387 mm) Check out this review*Note* The MM27 will have no problem operating at 100VAC in the 115VAC setting.
1" soft dome tweeter, dual 5" midbass drivers and dual 10" subs housed in compact sealed enclosures yield high linearity and outstanding impulse response. With 500 Watts of power in the subwoofer channel alone, the dual 10" drivers cross over seamlessly from the midbass, reaching down to 33Hz and rolling off at 1/4 the rate of ported designs to reveal much more deep bass information. Since the sub motor structures are locked together the opposing forces cancel out and the cabinet remains rock steady even at very high output levels. There is no need for bass management in 5.1 systems because the MicroMain27 is truly a full-range monitor. The speaker can be placed either vertically or horizontally using the included pedestal (9" L x 13/8" W x 21/4 H). Hand Crafted in San Francisco, California
The Standard Audio Level-Or is an API © 500-Series rack compatible JFET limiter / distortion processor. The design was inspired by the Shure Level-Loc PA limiter, popular for its trashy, ultra compressed sound. The extended features of the Level-Or allows it to achieve what the Level-Loc does and much, much more. In "Level" mode, the Level-Or acts very much like the Level-Loc. An additional, faster release time has been added for even more flexibility and a broader range of compressed sounds. In "Crunch" mode, the Level-Or sets itself apart from anything else like it. By sweeping the input level over the range provided, a huge palette of sounds can be reached from slight harmonic enhancement, to aggressive crunch, to buzz-saw like distortion, to complete and utter destruction of the original source material. Between these two available modes, there is a huge array of sounds that can be produced from this unit. Simple drum loops can be taken from boring to unique and drum room mics can be crushed or distorted to taste adding size and character to the overall sound. Bass, guitar, and keyboard sounds can be manipulated, taking on completely new and different sounds than the original source material. *Note: Due to the nature of the design of this unit, it is very aggressive sounding; with noticeable pumping, distortion, and some noise. We consider this a good thing! Features Transformer balanced input Electronically balanced, line level output stage capable of driving balanced loads as low as 600 ohms. Two modes of operation: Level – Sound of unit is dominated by JFET limiter Crunch – Sound of unit is dominated by harmonic distortion from the discrete transistor-based, Class A amplification stages Additional release time added that is twice as fast as original release Output level control to enable the input of the Level-Or to be driven hard without clipping the device that the Level-Or is driving (DAW input, Tape Machine, Console Line Amplifier, etc). ¼" line input on front panel
The BAE 1023 is a 10-series micpre/eq module based on the famous 1073. lt is totally hand-wired using the same Carnhill (St Ives) transformers and inductors as the vintage modules. It has the exact same mic/line preamp as the 1073 but with significantly more frequencies in the mid and hi sections: Mid: 160Hz, 270, 360, 510, 700, 1K6, 3K2, 4K8, 7K2, 8K2, 10K. Hi:10K, 12K, 16K, 20K, 24K.
The BAE 1023 is a 10-series micpre/eq module based on the famous 1073. lt is totally hand-wired using the same Carnhill (St Ives) transformers and inductors as the vintage modules. It has the exact same mic/line preamp as the 1073 but with significantly more frequencies in the mid and hi sections:
No other hardware required ... At last Apogee makes the direct connection to Pro Tools | HD systems a reality. The Apogee X-Series-HD option card allows for direct connectivity between Pro Tools HD™ systems and the AD&DA-16X, Rosetta 800 and Rosetta 200. The result is an incredibly powerful combination of Digidesign's™’ industry leading software and a new generation of superior Apogee conversion and clocking hardware that provides a workstation that will excite Apogee and Pro Tools™ fans alike. Features: Primary HD port for connecting directly to HD Core or Accel cards. Expansion port for daisy-chaining multiple Apogee devices (AD-16X, DA-16X, Rosetta 800 & Rosetta 200) Compatible with the latest versions of ProTools May be used along side similar interfaces (such as Digidesign 96 I/O, 192 I/O, Prism Dream ADA-8)
The API 512C is a mic/line/instrument preamp designed to provide a low noise, unusually good sounding front end for all types of audio systems. Sonically, it is at the top of the "Mic Preamp List", regardless of price. Offering low noise (-129 EIN) and 65 dB of gain, the 512C includes phantom power, switchable polarity, -20 dB pad and Mic/Line or Instrument selector. Front panel XLR and 1/4 inch connectors combined with rear panel mic access allows for additional flexibility when installed in an API LunchBox, Six Pack, 10 position vertical rack, a 2 position horizontal rack, or an API console. What makes the API 512C unique is its long evolution from the original 1967 era 512, the first modular mic pre, to the current full featured 512C, while preserving the original sound character that made it so much a part of the early days of recording. Offering high headroom and a wide variety of inputs and input access ponts, it is ideal for unusual and demanding applications. Imagine a situation where only a few preamps are needed, yet the smallest available console has a proportonally "small" mic preamp, making it useless for the demanding application, or conversely, imagine where you need a large number of preamps, and a console of sufficient inputs and quality would be too large to transport or rack mount. The 512C hits the spot with its quality and famous tone. Expand, combine or downsize at any time without trade-ins or product obsolescence. In addition, the 512C's sound and performance exceeds most "console mic pres" in every respect. The beauty of the entire API 500 Series is its long term flexibility and lasting value when needs change over time. With a range of mounting frame options, the 512C will be a valuable asset to your performance critical applications. The 512C Mic/Line Preamp makes use of the 2510 and 2520 op-amps and therefore exhibits the reliability, long life, and uniformity which are characteristic of API products. Features Mic Preamp with 65 dB of gain Front and Back Panel Mic Input Access Line/Instrument Preamp with 50 dB of gain Front and Back Panel Line/Instrument Input LED VU meter for monitoring output level 20 dB pad switch, applies to mic/line/instrument 48v Phantom switchable power Traditional API fully discrete circuit design Uses the famous API 2520 Op-Amp 2 Year Warranty (labor) 5 Year Warranty (parts) We double the standard API one year warranty (parts and labor) on this item.
What makes the API 512C unique is its long evolution from the original 1967 era 512, the first modular mic pre, to the current full featured 512C, while preserving the original sound character that made it so much a part of the early days of recording. Offering high headroom and a wide variety of inputs and input access ponts, it is ideal for unusual and demanding applications.
Imagine a situation where only a few preamps are needed, yet the smallest available console has a proportonally "small" mic preamp, making it useless for the demanding application, or conversely, imagine where you need a large number of preamps, and a console of sufficient inputs and quality would be too large to transport or rack mount. The 512C hits the spot with its quality and famous tone. Expand, combine or downsize at any time without trade-ins or product obsolescence. In addition, the 512C's sound and performance exceeds most "console mic pres" in every respect.
The beauty of the entire API 500 Series is its long term flexibility and lasting value when needs change over time. With a range of mounting frame options, the 512C will be a valuable asset to your performance critical applications.
The 512C Mic/Line Preamp makes use of the 2510 and 2520 op-amps and therefore exhibits the reliability, long life, and uniformity which are characteristic of API products. Features
The Helios Type 69 Mic Pre/EQ Module Born Again! Re-manufactured classic 3 band eq-pre. Discrete. Switchable lo/mid. Fixed top. Hi-pass. Unlike anything you have ever used (especially if you are stuck in API-Neve land) you will find the Type 69 a unique departure. These hand built units have a Sowter input transformer based on the highly desirable Lustraphone (Olympic) transformer models. The mids are so funky and sweet---- guitars and snare drum's from heaven; woody and sweet! 700 to 2k to die for! Lo's are Pultec-ish. Hi's are sweet and musical. Extremely open Mic-Pre!In mic shootout's, these are as open as the king daddy Telefunken V76m. Transform your life. The Legendary Helios Electronics Ltd has resurrected with much anticipation and patience, the Olympic Type69 Series EQ as used by the Beatles, Rolling Stones, Bob Marley, Led Zeppelin and so on. Helios became known as the musician's choice of recording consoles. Brought in line with modern expectations, these units are built to military specifications with supervision of gear fanatic and music producer Tony Arnold. Endorsed by the late Dick Swettenham, of the original Helios Electronics Ltd, to maintain and service all of the original consoles, Tony later purchased the original company, including all of its circuits and parts. The Olympic Series is a classic 3-band EQ-Pre module, available in a dual mono 1U rack mount, mono vertical module or 500 Series module. The modern circuit is similar to the original, aside from one vast improvement, the build quality and reliability. The openness and clarity has been upheld with the addition of EQ frequency selections. Tony Arnold explains, "We asked many of the original users of the Helios Desks what they felt was missing, and could we make any improvements. Most wanted the 30 Hz added to the Lo-EQ. In order to meet todays CD standards, some wanted a tight high frequency around 16 k, which has been added without changing the original Helios sound whatsoever."
The Weiss DAC2 is a high performance stereo Firewire based digital to analog and digital to digital converter for usage either standalone or in conjunction with a PC or MAC computer.The DAC2 supports the following conversions:Firewire to analogAES/EBU to analogFirewire to AES/EBUAES/EBU to FirewireFeatures of the DAC2:Inputs: Digital Audio inputs on Firewire (two connectors), XLR, RCA and Toslink.Outputs: Stereo analog output on XLR and RCA. Digital Audio output on Firewire, XLR and RCA.Sampling Rates, Wordlength: 44.1, 48, 88.2, 96, 176.4, 192 kHz at up to 24 Bits.Software: Drivers for Windows™ and OSX™ operating systems.Power Supply: A non-switching power supply is used. The mains voltage can be set to 100..120V or 200..240V.
A reincarnation of the high output, low distortion tapes of the past but with even higher coating thickness and improved magnetic properties for the ULTIMATE PERFORMANCE.
The Prism Sound Orpheus is a FireWire interface for personal recording and sound production, for professional musicians, songwriters, engineers and producers. Orpheus is ideal for music and sound recording, production & monitoring, stem-based mastering and analogue summing. Orpheus provides Prism Sound's renowned performance and sound quality in a dedicated FireWire unit compatible with Windows XP & Vista and MAC OS X 10.4x (Intel & PPC). Orpheus has line, microphone and instrument inputs, good foldback and stereo or surround monitoring capabilities, ADAT and SPDIF digital I/O plus support for external MIDI devices. Microphone inputs include MS matrix processing and high-performance digital sampling-rate conversion (SRC) is available for digital inputs or outputs. Orpheus signal pathEight analogue input channels and up to 10 digital input channels are available (SPDIF on RCA/coax plus ADAT optical) as DAW inputs through the host's audio driver. Similarly, eight analogue output channels, up to 10 digital output channels and stereo headphone outputs can play 22 different channels. For low-latency foldback or monitoring to headphones or main outputs, each output pair (1-2, 3-4 etc) can be driven with an individual local mix of any selection of inputs through the controller applet. All inputs are electronically balanced with automatic unbalanced operation. Outputs are electronically balanced with 'bootstrapping', i.e. level is maintained if one leg is grounded. The Prism Sound Orpheus Features: No-compromise, full Prism Sound audio quality Dedicated FireWire interface ASIO and WDM drivers provided for Windows XP and VISTA Directly compatible with CORE AUDIO on Mac OS X 10.4+ (Intel & PPC) Eight "Prism Sound" AD and DA channels, plus SPDIF, ADAT & headphones Four high-end integrated mic preamps (typ.-130dBu EIN), switchable phantom MS Matrix processing on mic inputs Two instrument inputs Prism Sound "Overkillers" on every channel to control transient overloads Fully-floating (isolated) balanced architecture for optimum noise rejection Mono or stereo input configurations Outputs arranged as stereo pairs, each with individual mixer Low-latency "console-quality" 8-bus digital mixer for foldback monitoring Fader, pan, cut, solo on every mixer channel Dual headphone outputs each with its own front-panel volume control Front-panel master volume control, assignable to selected channels Configurable for stereo, 5.1 or 7.1 or surround monitoring Built-in sample rate conversion (SRC) on DIO channels Prism Sound 4-curve SNS noise shaping on digital outputs State-of-the-art clock generation with proprietary hybrid 2-stage DPLL MIDI in/out ports No-compromise, full Prism Sound audio quality Orpheus makes no compromises on audio quality. It is the result of years of research and development into digital audio conversion and extensive dialogue with Prism Sound's customers. Orpheus The Orpheus design brief was: Get Prism Sound quality conversion and mic preamps into a 1U box at a more accessible price point. Sound quality just wasn't negotiable. Orpheus has the same no-compromise analogue front and back ends, with the same fully-balanced-throughout architecture, the same isolation barriers protecting the analogue from digital and computer interference. Orpheus draws on Prism Sound's years of experience in developing digital audio products, including its range of audio test equipment, adopted by a wide variety of clients across the audio industry from pro-audio to consumer electronics. This experience means that Orpheus is well-behaved both as a computer peripheral and an audio processor. Reliability is vitally important in professional recording. Prism Sound has always made extensive use of precise software calibration techniques in its converters - pots and tweaks are always unreliable, so there are none. The design team has gone to great lengths to minimise noise and interference, in particular hum. All of the analogue circuits have galvanic isolation, while the unit's electronically balanced I/O allows it to handle common mode interference sources as well as enabling trouble-free connection to unbalanced equipment. It is often said that THD+N figures do not always correlate well with the perception of sound quality and this is true - partly because the traditional measures of THD+N or SINAD expressed as RMS figures are rather a broad measure. With this in mind, we have taken great care to make sure that not only is the Orpheus noise and distortion spectrum beyond reproach, but the RMS distortion result measures up to the state of the art. Standards compliant FireWire interface Increasing standardisation is leading to more choice for those wishing to "mix and match" editing and production software with various audio interfaces. Prism Sound has taken on board the increasing importance of native processing power for professional users and the fact that software products for standard PC and Mac platforms have been greatly enhanced in recent years. Prism Sound is probably best known for A/D and D/A converters, not least the ADA-8XR, which already provides a solution for those needing a FireWire interface. However, the flexibility and versatility of the ADA-8XR comes with a higher price tag, reflecting the fact that no other interface provides such exceptional audio performance or can work directly into Pro Tools Core/Processing cards, as well as running a concurrent DSD processor or FireWire interface. The solution was to create a unit that is dedicated as a FireWire interface and is compatible with Windows PC and Apple Mac computers. Orpheus is easy to connect to your computer and to your outboard gear. For Windows XP or Vista users ASIO and WDM drivers are provided, while for Mac OS X 10.4 or later, Orpheus interfaces directly to Core Audio. For both Mac and PC platforms, there is a controller application to configure the unit and control its built-in mixer and other functions. Aside from the monitor and headphone level controls, everything else is operated solely from the Orpheus controller application. The controller software opens on-screen as a separate panel alongside your existing editing software. Orpheus Flexible Inputs and Outputs Our customers told us that, along with FireWire connectivity, many professional users wanted a highly integrated solution with instrument and microphone inputs, and line outputs that could be used for stereo or multi-channel monitoring and/or foldback to performers. Orpheus offers eight analogue recording channels, eight monitoring outputs, stereo digital input and output on a phono connector plus concurrent optical digital I/O ports that can interface to S/PDIF or ADAT data formats, giving Orpheus a maximum capability of 18 concurrent input and output channels plus stereo headphones. Orpheus' eight analogue inputs support various capabilities. Orpheus has four really good mic amps with software-controlled gain in 1dB steps, individually-switchable phantom power - and very low noise and distortion. These inputs are auto-sensing, and support microphone and line input, with digitally-controlled mic gain in excess of 60dB. Two of these inputs also support direct injection (DI) instrument connections with quarter-inch jacks on the front panel. RIAA Equalization can be selected in the controller applet on channels 1 & 2 so that turntables can be connected for archiving or sampling applications. By selecting the input type (Mic or DI) , low- or high-impedance cartridges can be loaded with suitable termination impedances. All inputs have individually-selectable Prism Sound "Overkillers" built in, just as on the higher-priced ADA-8XR, to catch those fast transients. The Overkiller threshold automatically follows the operating line-up level selection (+4dBu or -10dBV). Overkillers are ideal for percussive sounds, particularly drums, where headroom can be a problem. The co-axial digital I/O port can be switched in the Orpheus controller applet between S/PDIF and AES3 formats. This control changes the operating voltage and the Channel Status format and is complemented by two in-line adapter leads that provide external XLR connections for AES3 devices. Other connections include MIDI in and out and wordclock sync I/O. Orpheus can also operate in a stand-alone mode using its ADAT or co-axial digital I/O connections. Once set up using the Orpheus controller applet, the unit can be disconnected from the host computer and used independently. Orpheus will retain its settings when powered down so even if it is switched off, Orpheus can be re-powered and stand-alone operation can continue with the automatically-stored settings. Digital Mixer Our customers also identified a need for a unit that could provide low latency foldback to performers, particularly when tracking and overdubbing. In answer to this need, Orpheus has a powerful built-in digital mixer that can be configured from the host computer to provide foldback feeds to performers, each with their own stereo mix of workstation playback and any of the inputs. The question of latency in computer interfaces, especially USB and FireWire boxes, is an important one. Obviously there are situations where the round-trip latency needs to be really short, like in overdubbing. The problem is that even if the latency on the interface and in the driver is as short as it could ever be, a native DAW is busy with plug-ins and other software and buffer times are probably set long. The only answer is to provide local foldback mixing in the interface. This is not new, and other products feature it, but most local mixers in competitive products are just too basic. Orpheus provides 'console quality' local mixing - every output has its own independent mixer, with channel strips for all inputs and workstation feeds, complete with fader, pan/balance pot, solo and mute buttons, and full metering. Strips can be stereo or mono, and the mixes are dithered with filtered coefficients, just as in a top-end digital mixer. There is a very small residual delay through the A/D and D/A conversion process in the foldback path, mostly from filters used for decimation and interpolation. However, with the low-latency Prism Sound DSP mixer, the worst-case delay through the A/D and D/A path is only 0.5ms and is significantly less at higher sampling rates. This is generally reckoned to be small enough not to be problematic. Although the unit's outputs will mainly be used for monitoring or foldback, the fact that they are of such high quality makes them suitable for a range of other applications such as insertion points, analogue summing or stem-based mastering. Orpheus Flexible Monitoring Professional users demand more sophisticated monitoring capabilities and are getting used to surround sound with HDTV and DVD, so it is becoming important to support surround monitoring setups. As well as wanting great analogue recording channels, the DAW user also needs top-quality monitoring. The eight analogue outputs on Orpheus allow monitor setups from multi-stereo up to 7.1 surround. Orpheus has a real volume knob which can be assigned to any or all of the analogue or digital outputs for use as a control room monitor control. There are two headphone amps, suitable for all types of headphones, each with its own independent volume control. As well as having its own workstation feed and mixer, the headphones can also be quickly switched across the other output pairs, which is handy for setting up. Sample Rate Conversion and Noise Shaping The digital output is equipped with the four Prism Sound SNS noise-shaping curves and includes Prism Sound's renowned synchronous sample-rate conversion, allowing outputs to various external devices at other sampling rates. The sample-rate converter can be used at the outputs as well as the inputs, so as well as dealing with unsynchronized or wrong-rate digital inputs, Orpheus can also generate, say, a live 44.1kHz output from a 96kHz session. Since Orpheus also includes the full suite of the famous Prism Sound 'SNS' noise shapers, you can also reduce to 16-bits at mastering-house quality. Unsurpassed Jitter Rejection In the 1990s Prism Sound pioneered testing of sampling and interface jitter and as a result our digital audio products deliver unsurpassed jitter rejection. Prism Sound digital audio products lock up fast and re-generate ultra-stable clock outputs. Another aspect of the traditional Prism Sound converter that is retained was the clocking - it's just as important as analogue-path considerations sound-wise. So whether it's providing a high-quality master clock for the rest of the room, or dealing with a jittery clock from outside, Orpheus is as rock-steady as its forbears. Support Over the years, Prism Sound's reputation for audio quality has been matched by its reputation for after-sales support and technical advice. Orpheus has the benefit of that support and customers have access to one of the best technical teams in the business. The Last Word We believe that Orpheus delivers exactly what our personal studio customers have asked for - all the performance of a Prism Sound product in a dedicated FireWire unit that handles line, microphone and instrument inputs with good foldback and monitoring capabilities, yet at a more accessible price tag. We are confident that customers using the new Orpheus converter will agree that it sounds as good as it looks. Download Product Manual
The Retro Sta-Level is a replica of the legendary 1956 Gates Sta-Level. The Sta-Level dominated the sound of hit radio in the 1960's. Now these super musical sounding compressors have found their way into today's hits. The Retro Gold Edition celebrates fifty years of the products existence. The Sta-Level uses the coveted 6386 tube in a classic G.E. circuit design. Alternatively, the Gold Edition Sta-Level uses a pair of commonly available 6BJ6 tubes (stock). The customer has the choice to use their own 6386 they provide or the 6BJ6 provided with the purchase of the unit. The choice is yours and the sound is undeniably true to the original either way. The original 6386 compressor developed by G.E., utilized a novel program controlled release circuit. The Sta-Level implementation creates density and consistency while minimizing typical compressor side-effects at up to 40dB of gain reduction! The program path is transformer balanced throughout to cancel control-induced distortion. Push pull circuitry with 6V6 output stage provides wide voltage swing for gain control and line output. The result is that the Sta-Level brings a uniquely familiar intimacy, air and warmth to your tracks. Revisions to the original design include; Tube balance can be instantly aligned without test equipment; Uses 6386 tube or a pair of 6BJ6 tubes; Two or more Sta-Levels can be coupled together for stereo, surround or sidechain operation; XLR transformer balanced input and output; IEC Power Connector; Hand matched audiophile-grade coupling capacitors; High quality, custom wound transformers used throughout; Internal component safety cover. SpecificationsGain Reduction Threshold: Continuously adjustable from +24dBm to -20dBmAvailable Gain Reduction: 40dBTotal System Gain: 35dB (with standard input and output pads)Distortion: 1 percent or less from 0-30 dB gain reductionFrequency Response: ± .25dB from 20-20,000 HzNoise: -70dB or better referenced to gain reduction thresholdOutput Level: Continuously adjustable from -40 dBm to +18 dBmInput Impedance: Accepts low impedance active or transformer balanced 0-600 OhmsOutput Impedance: 600 Ohm transformer balanced, terminated and resistive pad coupledTube Complement: 2-6BJ6), 1-12AT7, 2-6V6GT, 1-6AL5, 1-OB2, 1-5Y3Power Requirements: Three prong IEC - Switchable 115-230 VAC - 50/60 HzMounting Requirements: 3U metal rack 5.25" high, 19" wide, 8" deep
The original 6386 compressor developed by G.E., utilized a novel program controlled release circuit. The Sta-Level implementation creates density and consistency while minimizing typical compressor side-effects at up to 40dB of gain reduction! The program path is transformer balanced throughout to cancel control-induced distortion. Push pull circuitry with 6V6 output stage provides wide voltage swing for gain control and line output. The result is that the Sta-Level brings a uniquely familiar intimacy, air and warmth to your tracks.
Revisions to the original design include; Tube balance can be instantly aligned without test equipment; Uses 6386 tube or a pair of 6BJ6 tubes; Two or more Sta-Levels can be coupled together for stereo, surround or sidechain operation; XLR transformer balanced input and output; IEC Power Connector; Hand matched audiophile-grade coupling capacitors; High quality, custom wound transformers used throughout; Internal component safety cover.
Specifications
The Purple Audio MC77 supercedes the MC76 re-engineered 1176 type FET Limiter. The MC77 recreates the audio circuitry of the revision E 1176, using modern components matched to the original. One significant component is the input attenuator. The input attenuator that was custom made for the original revisions A-F 1176 and for the Purple MC76 by Clarostat was discontinued in 2002. It was a J series T-pad attenuator with two 600Ω "build out" resistors to keep a constant 600Ω load on the source and primary of the input transformer. Clarostat changed carbon manufacturers and the new carbon manufacturer was unwilling or unable to make the part. Purple temporarily stopped making the MC76 to find a solution. We tried several three deck Clarostat 70 series pots in different configurations to achieve the same loading and to match the taper. We found a solution that matched the original; both the originals and replacement parts are drawn in the schematic that you can download below. At the same time, we looked at feedback gathered from the hundreds of satisfied MC76 users. Based on that feedback, we incorporated useful new features into the revised unit, which was designated the MC77. NEW features in the MC77: True Bypass via sealed relay with front panel switch Improved stereo linking with front panel switch Sidechain insert loop or key input with front panel switch Buffered VU meter can monitor input or output level at +4dBu LED meter lights that don't burn out as incandescents do Heavy gauge black stainless steel enclosure 115v/230v operation switchable from rear panel (serial# 600) Features inherited from the MC76: Discrete Transistor Audio Path Electronics Single Element Class A Output Amplifier Transformer Balanced XLR Inputs and Outputs Zener Shunt Regulated Audio Power Supply Compression Ratios - 4:1, 8:1, 12:1 and 20:1 Fast Attack Time - 20 microseconds to 800 microseconds Release Time - 50 milliseconds to 1.1 second Gain of 45dB (full gain with no limiting) Ruggedized Design - PCB stiffener, chassis mount transformers Purple Anodized Aluminum Front Panel 3 year warranty
The LaChapell Audio Model 583E features the same amplifier stage found on the 583S including the Jensen JT-115k input transformer coupled with an ultra-clean transformer-less three-band EQ section with fully sweeping frequency controls and cut/boost settings of +/- 8dB. The EQ can run as an integrated EQ serving the preamplifier or separate as its own autonomous module where both units run independently. Features True Vacuum Tube Amplifier. Jensen Microphone Input Transformer. Nickel/Alloy Output Transformer. Accessible Tube Section. Integrated OR Independent 3-band EQ. Fully “sweepable” EQ Frequencies. Extremely Low THD+N Performance Transformer-less EQ Section Premium Components
Designed to be the perfect level control tool kit for recording, the SSL Dynamics Module is densely packed with control features yet simple to use. The combination of separate Compressor and Gate/Expander section is powerful, graceful and ideal for gentle and unobtrusive level control. Key Features Fully balanced XLR input and output Switched -10dB/+4dB operating level Dedicated compression meter display Over-easy RMS compressor, switchable to peak sensing hard knee Variable Ratio, Threshold and Release External key operation Independently switched Gate/Expander section Link facility for multiple module linking
Launched in 1970, the Neve 1073 is the first choice of leading producers and artists, delivering the unique Neve sound on some of the most famous recordings of the past 30 years. The big, punchy sound of the 1073 compliments any musical genre - from rock to pop, hip-hop to rap, thrash to classical. Handcrafted and completely hand-wired by Neve’s dedicated professionals in Burnley, England, the modern-day 1073 is produced to the exact specifications of the original modules. Considerable resources have been devoted to the acquisition of the original components to ensure the sound remains true. Looking for an outside opinion? Lynn Fuston at EQ Magazine performed an in-depth review of the modern-day 1073 versus a vintage model. We think you’ll find his conclusions quite comforting. The Class A design 1073 microphone preamplifier features 3 bands of EQ, with one fixed high frequency band, two switchable bands with cut and boost capability, and a high pass filter. All Neve channel amplifiers are designed to accept signals from a wide range of microphone and line sources. The Neve 1073 mic pre and EQ combination adds warmth and depth to recordings, brings out subtle ambience, maintains spatial positioning, and more effectively captures a precise image. That’s why the 1073 mic preamp is considered by many to be the very essence of the Neve sound. HighlightsClassic transformer microphone preamp amp (Class A design)3 EQ bandsHand-built and hand-wired to original 1970s designHP filterNeve designed hand-wound transformers Both inputs are transformer balanced and earth freeMicrophone Input: Gain +80db to +20dB in 5dB steps.Line Input: Input impedance 10k ohms, gain +20dB to -10dB in 5dB steps.Output: Maximum output is >+26dBu into 600 ohms.Output is transformer balanced and earth freeDistortion: Not more than 0.07% from 50Hz to 10kHz at +20dBu output(80kHz bandwidth) into 600 ohms.Freq Response: +/-0.5dB 20Hz to 20kHz, -3dB at 40kHz. EQ Out.The 1073 can be purchased as single or multiple units. Each 1073 module has been designed to perfectly retrofit into the 80 Series Neve Classic consoles as part of a channel strip or can be mounted in a custom rack. The available rack sizes are 3U (which accommodates 2 modules horizontally) or 5U (which accommodates 8 modules vertically). Neve "Classic" Outboard - Genuine Neve product hand built in UK using all of the original components, nothing new in 30 years
Shadow Hills Optograph This Single channel discrete optical compressor, in the API 500 series form factor, takes its cues and much of its design from the optical stage of its behemoth Big Brother, the Shadow Hills Mastering Compressor. The Optograph's front end is balanced by an original Jensen input transformer. The output utilizes our specially recreated Nickel transformer. The essence of the compressor's character comes from our custom electroluminescent cell and discrete op-amp combination. The high quality circuit is pristine enough for bus compression, and yet has a unique character reminiscent of an antique world class compressor, that has just now been discovered. It's also perfect for squeezing drum busses and massaging vocals. It's the perfect glue for tracking and mixing anything that needs to sound more finished, more like a record. Just like The Mastering Compressor, the Optograph's threshold and make up gain are controlled by discrete attenuators, made from custom twenty-four position Elma switches. The tactile feel and quality of these controls are unrivaled. They are of the highest quality and will last forever. Built into the side chain, are a very powerful series of filters: 90 hertz, 150 hertz, 250 hertz, and band pass. By engaging a specific filter, you choose at which point the onset of compression occurs. First position, nothing below ninety triggers compression. Second position, nothing below one-fifty. Third, nothing below two-fifty. The fourth position is a musical bandpass filter. In this position, compression is triggered by the program's mid-range frequency content, ignoring the highest and lowest frequencies. The filters are amazingly useful for shifting the focus of what should be more compressed and creating compression curves on purpose. Another unique feature for sculpting the overall sonic character is the Transformer Desaturate mode. In this mode we cancel out any distortion and frequency non-linearity’s caused by the output transformer. The result is a pure, almost transformer-less sound, whilst still receiving the benefits, of limiting transients that can only come from magnetics. There is also a VU meter for the display of gain reduction. The Optograph is enclosed in a twelve gauge steel chassis, has an engraved front panel and custom made knobs. Features: Pristine sound quality and unique character Switchable sidechain filters Transformer Desaturate Detented switches Engraved front panels