The <b>Burl B2 Bomber ADC</b> is the most significant piece of audio gear in your studio, and you don’t even own it yet. You’ve spent a fortune on great mics, great mic pres, and a kick ass DAW, but unfortunately you are missing one key component; an ADC with an audio path worthy of your recordings. <br><br> Open up your best mic pre and look inside. You will see high quality components like tubes, large transformers and high quality capacitors. Open up your ADC and you will find cheap 10 cent capacitors and run-of-the-mill 25 cent op-amps, even when you’ve spent a lot of money on your interface. Most DAW hardware interfaces are designed for inter-connectivity with little attention to the analog audio path, or even the conversion, let alone the tone. <br><br> The B2 Bomber is designed for great specifications, but more importantly, it is designed to give you the ultimate in recording tone. For years now people have been trying to figure out why their digital recordings don’t have the warmth and feel of analog tape recordings. We try using tube mic pres and great compressors, but there is still something missing. There is still that blurriness, that graininess and lack of depth that comes with digital recordings. <br><br> BURL Audio has solved this problem by designing an analog audio path that is complimentary to your mic pre and to the analog to digital process. By using a revolutionary hybrid circuit with a proprietary transformer, the BURL AUDIO BX1, and a discrete class-A, zero feedback, zero capacitor signal path, we have achieved dynamic and tonal balance. <br><br> Using a B2 Bomber in the studio is like taking the governor off of your recordings. And, the B2 is not just for tracking. Mix down and mastering were the main focus of the B2's layout and design. With 30 segment peak amplitude and RMS metering, the B2 lets you know exactly how hot you are running your signal which is critical for mastering. The attenuator settings on the front easily allow you to change input headroom giving you the ability to hit the front end hotter or colder depending on what the material calls for. And don’t be afraid to run the B2 Bomber hot, it only sounds better. Couple that with an extremely low jitter clock source, and you have the B2 Bomber ADC, the most significant piece of gear in your studio. <ul><li>44.1k Hz to 192k Hz, 24 bit, 2 channel ADC </li><li>Proprietary, high definition, BURL Audio BX1 transformer input </li><li>All class A, discrete transistors signal path with zero feedback, zero caps </li><li>Audiophile quality 6 position attenuator with standard headroom settings </li><li>High definition metering with simultaneous RMS and peak indication </li><li>Two AES and one SPDIF output with dual AES wire support </li><li>BNC word clock input with two outputs of extremely low jitter clock </li><li>Frequency response at 48kHz sample rate is 12Hz to 22kHz, +/- 0.1dB </li><li>Frequency response at 96kHz sample rate is 15Hz to 46kHz, +/- 0.1dB </li><li>Frequency response at 192kHz sample rate is 18Hz to 94kHz, +/- 0.1dB </li><li>120dB Dynamic Range (A- weighted), 117dB (no weighting) </li><li>-100dB THD+N (+4dBU = -15dBFs) </li><li>-94 dB THD+N (+18dBU = -1dBFs) </li><li>Rugged, Made in USA design</li></ul>

Burl Audio B2 Bomber ADC

The <b>Moog Music Minitaur</b>’s front panel features a knob-per-function design, for maximum real-time tweakability. There are no internally stored presets, but presets can be created and managed via MIDI. The Minitaur has several new features added to the classic Taurus synth engine, including an external audio input, that expand the basic synthesis and sound design capability of the instrument.  Minitaur fits right into the computer based music production rig or is easily controlled via MIDI controller. <br><br><b>Features: </b> <br><ul><li>100% analog audio signal path based on Taurus I/ Taurus III synthesizers. </li><li>Two Taurus VCOs with variable Glide amount </li><li>Sawtooth (original Taurus) or Square waveform selection for both VCOs </li><li>External Audio Input </li><li>Headphone out </li><li>Two Mixer VCAs for setting the level of the VCOs </li><li>One VCF Moog/Taurus-style Ladder Filter w/ VC Resonance </li><li>One VCA </li><li>Two Minimoog-style Envelope generators for modulating the VCF and VCA: ADSR with the Decay and Release segments controlled by the Decay control, and with the Release segment enabled or disabled with the Release On/Off switch </li><li>One MIDI-syncable triangle wave LFO for modulating VCF/VCOs </li><li>DIN MIDI in and out and MIDI over USB out </li></ul>

Moog Music Minitaur

The <b>sE Electronics Munro Egg 150 Monitoring System</b> is the first in a series of sE Munro 'Active Integrated Monitoring Systems' designed by Andy Munro, James Ishmaev-Young and Siwei Zou.<br><br><b>sE Munro Egg Speaker - A Revolutionary Active Studio Monitor </b><br>The sE Munro Egg incorporates a fundamental rethink of the loudspeaker engineering process. The traditional wooden box has been replaced by a scientifically proven, curved enclosure that virtually eliminates diffraction and resonant effects that distort and smear the original sound.<br><br><b>sE Munro Egg 150 Monitoring System</b><br>The system is complete, with a free standing control unit that delivers a perfect bi-amplified power match to the egg drive units, source select inputs, active analogue crossovers, LF and HF trim pot equalization for room and location set-up compensation and a mid-band control to emulate the mid-range response of both Hi-Fi and NS10 type speakers.<br><br>Above all, the sE Munro Egg 150 Monitoring System is designed to deliver the highest quality sound with the absolute minimum of coloration. In one word - Truth.<br><br><b>Features:</b><br><ul><li>Unique monocoque shell construction (rigid and resonant neutral)</li><li>Near zero diffraction interference (smooth frequency response)</li><li>No internal standing waves (greatly reduced smearing)</li><li>Perfect bass port integration (superb transient response)</li><li>Free standing control unit and integrated amplification with 3m matched speaker cables</li><li>Dual input switching and level control</li><li>Ultra fast and low distortion (4*50W) with 100W/Channel headroom indicator</li><li>Precise 'sweet spot' – unique LED locator guide beams</li><li>Integrated base with adjustable vertical alignment (allows for perfect sweet spot in both vertical and horizontal planes)</li><li>Precision trim pots for bass (LF) and High Frequency (HF) calibration</li><li>Critical Mid Frequency equalization for  ‘Hard’, ‘Soft’ or ‘Reference’ (0) listening</li></ul><b>sE Munro Egg 150 - The Unflattering Active Studio Monitor </b><br>The result is a breath-taking transparency that within seconds of listening reveals a much more defined and open mix, in which the whole balance simply ‘feels’ more complete. Of course certain aspects of some recordings may be shown up to be distorted or not as the engineer intended, but that is the truth, unobscured by coloration. A monitor that flatters to deceive has no place in a professional recording studio and there is no argument for using poor quality speakers to match inferior playback media.  <br><br>Well recorded material, translates to any medium, but the opposite is absolutely not true… if your monitors put a fake ‘sheen’ on everything, how will you ever know what your material sounds like on an honest system..? You may be embarrassed!<br><br>There is also a good argument for a fully analogue monitor. The bandwidth and dynamics of a true analogue system will show up the character of the digital system preceding it. It also removes issues with bit rate conversion and other compatibility matters.<br><br>By using high quality linear power supplies, the sE Egg system maintains maximum transient power up to double the rated value. Bi-amplification further enhances the power headroom and results in the lowest possible distortion values. By using 35V power rails the Egg is capable of high transient sound levels from a relatively modest amplifier compliment.<br><br>The sE Munro Egg 150 is the final product name of the "sE Munro Medium Egg Speaker" announced at Musikmesse and NAMM. The "150" refers to the approximate size of the driver. As more Egg Monitoring Systems are in development, it seemed foolish to term the first sE Munro speaker system as "medium" when it may not be in the fullness of time.

sE Electronics Munro Egg 150 Monitoring System

The <span style="font-weight: bold;">Mäag EQ4™</span> is a one-channel six-band equalizer with AIR BAND™ (shelf boost from 2.5 to 40kHz), compatible with the API 500-6B lunchbox® and 500VPR rack systems.<br> <br> Following its EQ3 and EQ3-D predecessors, the EQ4 provides unparalleled transparency and top end presence while maintaining the true natural sound behind the mix. EQ adjustments are obtained with minimal phase shift and detent controls allow for easy recallable settings.<br> <br> Presented in the flagship 500 Series format, the EQ4 offers the legendary AIR BAND™ and five other sonically superior band passes.<br> <span style="font-weight: bold;"></span><br> <span style="font-weight: bold;">Specifications</span><br> <ul><li>Frequency Response: -2 dB points, 10Hz & 75kHz</li><li>Nominal Input Impedance (XLR): 48 K Ohms, balanced</li><li>Nominal Output Impedance (XLR): 50 Ohms, balanced</li><li>Headroom: +27 dBu</li><li>THD + Noise: < 0.005%</li></ul> *Specifications subject to change without notice.

Maag Audio EQ4

<span style="font-weight: bold;">A Designs Nail Compressor Features</span><br><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<br></li><li>Gain Control (Output)</li><li>Custom-Milled Aluminum Knobs</li></ul>

A Designs Audio HM2 Compressor Nail

The <span style="font-weight: bold;">Acme Opticom XLA-3</span> is an optical audio limiter built to exacting military-style specifications, designed to produce a full range of non-linear, dynamic audio effects.<br> <br> The heart of the XLA-3 is a unique triple optoelectronic circuit that combines the best characteristics of 3 separate compression curves into a single unit. The result: a high-speed optical limiter with tones that range from ‘clean’ to ‘harmonically rich’ to ‘dirty’.<br> <br> The Opticom uses high-speed cadmium-selenide (CdSe) photocells, all-tube circuitry, and military style point-to-point wiring. From the 16 gauge, cold-rolled steel chassis to the high quality components, the XLA-3 is built to provide years of reliable, solid performance that will meet, if not exceed, the exacting demands of the audio professional. Just a few of the standard features: custom-ordered Bakelite analog control knobs, full-sized backlit panel meters, Neutrik/Cliff connectors, heat resistant Micalex tube sockets.<br> <br> <span style="font-weight: bold;">Features</span><br> <ul><li>Made in the US</li><li>Authentic, military style point-to-point circuitry</li><li>Unique Sonic Character</li><li>Versatility</li><li>Serviceability</li><li>Ease of Use</li><li>Competitive Pricing</li></ul>

Acme Audio Opticom XLA-3

Renowned for decades by top engineers as the quintessential box for larger-than-life drum sounds! The ADR Compex F760X-RS may be the best compressor/limiter available for drum room mics and drum bus duties and is also highly sought-after for the character and punch it adds to guitars along with a myriad of other sources... And now it can be yours! <br> <br> The <span style="font-weight: bold;">ADR Compex F760-RS by Q2 Audio</span> reintroduces, at long last, the venerable and highly sought-after dynamics processor of yesteryear. The unique character and spirit of the Compex can be heard on many classic recordings (think of that huge, commanding drum sound on <i>When the Levee Breaks</i> by Led Zeppelin, for instance). With painstaking attention to detail, Q2 Audio has masterfully reproduced the exact circuit design of the original Compex, with a handful of new features and updated components.<br> <br> Functionally, the new Compex has all the same features and settings as the original unit, employing the same super-flexible FET-based compression, limiting and gating. Switchable threshold, compression ratio, and attack/release times are available with all the same settings and a stereo link switch is provided for linking the two channels together for operation in stereo. An all-new external side-chain insert has been added to the new Compex, letting you feed the detector circuit of the compressor with an independent sound source. The side-chain insert can be set to control the gate circuit, instead, by simply switching over an internal jumper. The limit section has a switchable 50us "pre-emphasis" setting in addition to the normal “on” position. Pre-emphasis was originally used for limiting high frequencies to prevent over-modulation during broadcast use. Pre-emphasis boosts the high frequencies entering the limiter side-chain, acting as a de-esser causing limiting action on sibilant high frequencies. The compressor side-chain insert is useful for adding an EQ to the side-chain audio signal to create "vocal stresser" type frequency-dependent compression effects or inserting another audio source for “ducking".<br> <br> Improving upon some of the components and build-quality concerns of the original Compex, a high-quality meter has been employed (replacing the discontinued Sifam™ Director 14 meter) and the original “ABC” cards have been combined to a single PCB to allow for more efficient manufacturing. Also, some slight alterations have been made to the calibration functions to help the unit maintain calibration over a longer period of time. The original potentiometers were an open frame design which became noisy as time wore on, the re-issue has sealed conductive plastic potentiometers to improve the usable life-span.<br> <br> With a wide range of features and functions, the new ADR Compex by Q2 Audio can meet the needs of any dynamic-processing task, adding a character and flavor all its own. Those familiar with the original Compex will be pleased to see how faithfully recreated this reincarnate really is, with a few enhancements and updates to make one of the best compressor/limiters just that much better.<br>  <br> <span style="font-weight: bold;">Features:</span><br> <ul><li>Classic FET based compression/peak limiting/gating, all available individually</li><li>Switchable compression ratios - 1:1, 2:1, 3:1, 5:1, 10:1, 20:1</li><li>Switchable compression threshold in 2dB steps</li><li>Switchable attack times - 250µs, 2.5ms, 25ms</li><li>Switchable release times ranging from 25ms to 3.2s plus "auto" setting</li><li>Peak limiting (100:1) with switchable 50µs pre-emphasis setting (useful for de-essing)</li><li>Expander/gate with 20dB of noise reduction</li><li>Compressor side-chain insert (can be internally modded to control the gate, instead)</li><li>Stereo linkable</li><li>Swiss-made ELMA™ rotary switches</li><li>Conductive plastic potentiometers</li></ul>

ADR Compex F760X-RS by Q2 Audio

The <b>Antelope Audio Eclipse 384</b> is an advanced 384 kHz A/D & D/A converter clocked by Antelope's renowned 64-bit technology and a flexible monitoring system that creates a technological synergy by combining the most prominent Antelope's innovations. It provides mastering and mixing engineers an unprecedented level of productivity, sound quality and ease of use.<br><b><br>64-bit DSP Trinity-level clocks</b><br>The Eclipse comprises 384 kHz A/D & D/A converters clocked by two independent 64-bit DSP Trinity-level clocks. The fully integrated monitor controller employs 0.05 dB accurate gold-plated relay attenuators and provides speaker switching, bass management and cue mix functions with integrated talkback. The Eclipse also includes two dedicated headphone amplifiers and a custom USB interface, as well as two large peak meters on the front panel. The advanced software control panel compatible with both Mac & PC, allows five nameable presets for easy recall of favorite setups.<br><br><b>Studio and live</b><br>Master clocks are becoming more and more popular for use with digital consoles and recording equipment at concerts and live events. By using the Eclipse 384, a live sound engineer is able to provide sync reference for up to four different devices, while using the main D/A for backing material, the A/D with multiple digital outs feeding redundant recording systems, and monitor DAC to check the recording post A/D conversion.<br><br><b>Supreme audio quality and boosted efficiency</b><br>The unique dual-domain clocking system enables analog-based, more natural sounding sample rate conversion. The integrated patching/routing capabilities make monitoring of either analog or digital sources extremely simple, avoiding jitter, distortion and cabling noise. By eliminating the many input and output stages and the various power supplies, that would be present in separate devices, the noise floor can be substantially reduced and the audio quality significantly improved.<b><br><br>Features:</b><br> <br> <b>Clocking</b><br> <ul><li>64-bit DSP Trinity-quality clocking</li><li>0.001 PPM Oven-controlled oscillator</li><li>Two independent sample rates</li><li>Complete Varispeed capability</li><li>10M Atomic clock input</li></ul> <b>Conversion</b><br> <ul><li>384 kHz A/D & D/A converters</li><li>A/D with Dynamic Range of 124 dB</li><li>D/A with Dynamic Range of 129 dB</li><li>Two bypassable A/D inserts</li><li>480 Mbits USB 2.0 custom chip</li></ul> <b>Monitoring</b><br> <ul><li>Three sets of switchable monitor outs</li><li>Second dedicated monitor D/A</li><li>Base Management with LFE output</li><li>Relay attenuators matched to 0.05 dB</li></ul> Complete management via user-friendly Mac/PC/Linux software control panel.<br> <br> Six different presets for convenient set-up<b>.<br></b>

Antelope Audio Eclipse 384

The <b>API 512C</b> 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.<p></p> <p>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.</p> <p>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.</p> <p>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.</p> <p>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.<br><br> <b>Features</b> </p><ul> <li> Mic Preamp with 65 dB of gain </li><li>Front and Back Panel Mic Input Access </li><li>Line/Instrument Preamp with 50 dB of gain </li><li>Front and Back Panel Line/Instrument Input </li><li>LED VU meter for monitoring output level </li><li>20 dB pad switch, applies to mic/line/instrument </li><li>48v Phantom switchable power </li><li>Traditional API fully discrete circuit design </li><li>Uses the famous API 2520 Op-Amp</li></ul> <B>2 Year Warranty (labor) 5 Year Warranty (parts)</B><BR> We double the standard API one year warranty (parts and labor) on this item.

API 512C

The next generation of Shorti Patchbays has arrived; this 2x48 audio patchbay is wired to DB25 connectors. This unit features exceptional flexibility with the <b>Audio Accessories, Inc.</b> exclusive Quick-Switch normalling system located on the rear of the panel. The Quick-Switch normalling allows you to set the individual normals on a per jack pair basis. This enables you to full-normal (FN), half-normal (HN) or non-normal (NN) by sliding the switches into the appropriate position. You also have grounding options: isolated, bussed, or grounds vertically strapped (GVS).<br> <ul><li>2x48x1.5RU Mini audio jack panel</li><li>TRS out to DB25 female connectors</li><li>Pinned out for Pro Tools interface</li><li>User-Programmable Normalling</li></ul> <p> We have an <a href="http://65.61.32.96/documents/patchbay-template.xls">excel template</a> that most people find useful for labeling the db25 shorti.</p>

Audio Accessories Shorti Quick-Switch to DB25 - 1.5

The <b>BAE DMP</b> is a desktop version of the 1073MP. It shares a similar preamp and the same Carnhill (St Ives) transformers as the 1073 and 1084. Now included with the "Bootsy" mod, a Jensen DI that has been added to the DMP.<br><b><br>Features:</b><br><ul><li>Built in PSU for extra portability</li><li>DI in and DI through</li><li>48v for phantom power</li><li>Solid steel chassis</li><li>Same hand-wired modular design that BAE is known for</li></ul>

BAE DMP Desktop Mic Pre

<b>Find out why the MicroMain27 is the most sought after speaker in pro audio. Call Vintage King and demo today!</b><br><br>The <b>Barefoot Sound MicroMain27</b> 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. <br> <p>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). <br><i>Hand Crafted in San Francisco, California</i> </p><br><br>*Note* The MM27 will have no problem operating at 100VAC in the 115VAC setting.

Barefoot Sound MicroMain27 - Pair

The <span style="font-weight: bold;">Black Box Analog Design Vacuum Tube Preamp</span> is an entirely new approach to capturing audio. It is not based on any existing circuit but designed from the ground up, using the best parts and ignoring all of the standard ideas of how a preamp “should work”. The result is an incredibly versatile piece of gear that not only sounds amazing but shatters the idea of what a preamp can do!<br><br><span style="font-weight: bold;">Tonal Control</span><br>Until now, your preamp simply amplified the signal. You essentially get one sound and the ability turn it up or down. The Black Box mic pre on the other hand allows you to drastically shape the response curve of the unit without using any EQ and the associated phase shifts!<br><br>All of the shaping is done at the tubes and from the constantly variable interaction between stages allowing you to dial in virtually unlimited tonal possibilities. Essentially, you have the ability to have full control over how the microphone “hears”! You can find the sweet spot of any microphone and instrument easily and naturally!<br><br>The response curve of the Pentode stage alone gives you a huge amount of control over the tone. Coupled with the independently controlled Triode stage you have virtually unlimited tones at your fingertips.<br><br><span style="font-weight: bold;">All Tube, All Analog</span><br>Black Box Analog Design's entirely analog audio circuit uses only tubes for amplification and is entirely free of op amps, ICs and transistors!<br><br>The Pentode and Triode tube stages are the centerpiece of an entirely point to point, hand wired circuit built on copper boards. From precisely, hand matched resistors and capacitors to custom wound Cinemag transformers , every part is selected and built for the absolute highest quality sound.<br><br><span style="font-weight: bold;">Real Power</span><br>It seems that more and more manufacturers these days are moving to cheap, “wall wart” switching power supplies. Sure they are cheap but we all know that a piece of gear is only as good as its power supply! For that reason, our preamps use a massive toroidal to supply 350v of pure, linear power to get the most out of the tubes. Real power means real sound!<br> <br><span style="font-weight: bold;">Features:</span><br><ul><li>Independently controlled Pentode and Triode tube stages</li><li>Custom wound Cinemag input and output transformers</li><li>Entirely analog audio path (No ICs or Op amps)</li><li>Hand soldered, point to point wiring</li><li>Switchable input impedance done by tapping into the transformer windings</li><li>5 position, gentle and musical roll off</li><li>Linear power supply</li><li>Passive output attenuation control</li><li>48v Phantom power from an entirely separate supply</li><li>Real, amplifier isolated VU meters</li><li>Military spec switches</li></ul>

Black Box Analog Design Preamp

Chandler's newest series of units starts off with the <B>GERMANIUM </B>pre amp. A completely new design by Chandler Limited designer Wade Goeke that uses classic germanium transistors in all class A, transformer balanced, circuits. The use of germanium devices opens a whole new sound palate not found in any pro audio equipment currently manufactured. <P></P>These incredibly smooth sounding transistors were the basis of the earliest transistor designs by Neve (1053, 1055, 1057), EMI ( TG12345 MKI), Telefunken, and Fairchild. Those familiar with these units can attest to the special sound of the germanium tranasistor. The GERMANIUM series runs on +40 volt power, high current, and has a huge +34 output before clipping. The sound is warm and smooth as with many vintage style circuits, but you'll find a whole new world of smooth here as well as a sound that settles in perfectly to your tracks without fighting. Add these new circuits to Chandler's new transformer line that combines the best parts of our favorite trannies, St Ives and Gardners, into new great sounding input and output transformers.

Chandler Germanium Mic Pre

The <span style="font-weight: bold;">Electrodyne 511</span> is a classic two-band, discrete transistor, reciprocal, active inductor-based equalizer using late-’60s/early-’70s design technology. As with the 501, the 511’s custom inductors and output transformer are made by Electrodyne’s original provider to strict factory tolerances as small as two percent. This affords a consistent EQ performance and repeatability from channel to channel that was simply not possible in the 1960s. Smooth performance and EQ response from minimum to maximum gain at all frequencies provides unusually broad sonic and tonal options not experienced since the 1970s. <br> <br> The module’s faceplate sports twin large rotary EQ controls each offering ±12dB of boost and cut with four selectable frequencies per band (LF: 40, 100, 250 and 500Hz / HF: 1.5, 3, 5 and 10kHz). Shelving is available on all frequencies, with peaking offered at 250 and 500Hz in the LF band and 1.5, 3 and 5kHz in the HF band. An EQ in/out switch with accompanying LED rounds out the front panel feature set.<br><br><span style="font-weight: bold;">Specifications</span><br><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

The <span style="font-weight: bold;">Focal Twin6 Be</span> is an active, 3 way, professional near-field/midfield speaker (3 built-in amplifiers - 2x150 +100W rms), comprised of two 6.5 inch (16.5cm) “W” cone sandwich composite drivers, loaded by two large section laminar bass ports and a Focal inverted dome pure Beryllium tweeter.<br><br>Both 6.5” drivers handle low frequencies but only one of the two (selectable) is passing lo-mid frequencies.<br><br><span style="font-weight: bold;">About Focal Professional Studio Monitors</span><br><br><span style="font-weight: bold;">Listen to your music, not to your speakers</span><br>These few words embody the philosophy of the Focal Professional Division. It is crucial for engineers, be it in music production, post-production or broadcast, to be able to completely trust what they are hearing. Our products are designed from the ground up to be professional tools that reproduce the reality of sound without enhancements or degradations.<br><br><span style="font-weight: bold;">Specific tools for specific needs</span><br><br><span style="font-weight: bold;">A speaker that “reveals”</span><br>A tracking engineer needs to be able to capture the exact tone of the instrument he’s recording with his microphones; he needs to discern the slightest shift in microphone position, EQ, or compression on his monitors. A mixing engineer needs to be able to place various instruments and vocals in his mix with precision. The smallest details need to come out clearly on his monitors, i.e., reverbs and spaces, and they need to be reproduced at their proper levels without any artificial alteration of the soundstage.<br><br><span style="font-weight: bold;">A speaker that “translates”</span><br>The mixing engineer also needs to be able to make sure that his mix sounds as good on other speaker systems in other environments. Usually, when a mix is done, the engineer will make a copy of it and go listen to it in a car or on a boom box, just to make sure that everything sounds the same everywhere. In essence, the engineer wants to make sure his mix translates well everywhere it gets listened to. A completely transparent speaker, that doesn’t impart a “color” to the sound, is the best way to insure perfect mix translation.<br><br><span style="font-weight: bold;">Our Speakers “reveal” and “translate” better than others. Why?</span><br>Focal Professional Division designs its products from the ground up to fulfill the specific needs of professional sound engineers: to reproduce sound as naturally and precisely as possible. The truth, and nothing but the truth. Our R&D labs have put all their knowledge and savvy into action to combat any coloration or distortion that could appear throughout all areas of the speaker. Even the slightest coloration can hinder the integrity of the signal’s reproduction. It’s easy to understand how a self vibrating speaker enclosure can blur the original signal, and the same goes for the drivers, the integrated amplifier, the padding, the finish, etc, etc.<br><br>We scrutinize each of these elements with the same care to achieve our goal of absolute transparency.

Focal Twin6 Be (Single)

Presenting the <span style="font-weight: bold;">Grace Design m501</span>, 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.<br><br>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.<br><br>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.<br><br>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.<br><br><span style="font-weight: bold;">Features</span><br><ul><li>Fast and musical transimpedance architecture</li><li>Precision audio path with 0.5% precision metal film resistors</li><li>New 12 position gold plated rotary gain switch</li><li>High performance output line driver amplifier and HPF amplifier</li><li>New Ribbon mic mode (also great for dynamic mics) - Relay bypass of phantom power decoupling capacitors, increased input impedance, and 48V lockout</li><li>Wide 10-75dB gain range</li><li>Enhanced RFI interference suppression</li><li>Three output connectors: XLR balanced, TRS balanced and ¼” unbalanced</li><li>Bombproof laser-etched black anodized frontpanel</li><li>Sealed gold contact relay for Hi-Z input switching</li><li>Led indicators for +48V, Ribbon mode, and HPF</li><li>Five year warranty on parts and labor</li><li>Made in the USA</li></ul>

Grace Design m501

The <span style="font-weight: bold;">Great River 32EQ</span> is a new 500-series version of the EQ and filters from the renowned Harrison 32 Series consoles.<br> <br> 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.<br> <br style="font-weight: bold;"> <span style="font-weight: bold;">The 32EQ has the full features of the 32-series EQ</span><br> <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.<br> <br> 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.”<br> <br> Like all Great River and Harrison products, the 32EQ is designed and built in the USA.

Great River / Harrison 32EQ

The <b>Helios Type 69 Mic Pre/EQ Module Born Again! </b><br>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. <br> <br>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. <br><br>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. <br><br>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."

Helios Type 69 Mic Pre / EQ Module

The new <span style="font-weight: bold;">Mercury M72s (Mk. III)</span> is Mercury's flagship, valve studio microphone amplifier. The M72s is based on the most sought after vintage amplifier module, the ‘Telefunken/ Siemens V72s.’ Which is most famous for being used in REDD.37 consoles, which was used on all the early Beatle recordings by George Martin at Abbey Road Studios in London, England.<br> <br> The Mercury M72s (MK. III) has a new look but same stellar tone and craftsmanship and still proudly built in the USA. We only simplified the look of the front panel by deleting a few vintage inspired, aesthetic extras of the MK II, which is still available as the M72s-CSV (custom shop version), but is still loaded with the same exact tone and warmth utilizing the same exact circuit, same quality parts and same transformers we have used since we started building the first M72s at Mercury Recording Equipment.<br> <br> The Mercury M72s is based on the very rare V72s module used in consoles due to a minimal amount made. The more common V72 amplifier modules (not marked with lower case s, ie V72s) were later modified and have been sold for many years as outboard preamplifiers all over the world. Neither of these amplifier modules have been available new for more than 50 years. So, working units have become extremely rare, very expensive and, in some cases are now in need of much repair. Regardless of cost, the uniquely musical tonal characteristics of these amplifiers has made them the prized possession of many engineers lucky enough to get the vintage modules, and a "secret weapon" for many studio musicians. Mercury Recording Equipment is proud to have a faithful reproduction in the M72s. We have nearly 20 years experience with the amplifiers and feel we have captured not only of how the units behave, but of how they sound overall. We are confident, with the addition of modern features, making the M72s much more versatile, without sacrificing the most important feature: Musicality.<br> <br> <span style="font-weight: bold;">Mercury M72s Features</span><br> The Gain of the Mercury M72s is variable from 28dB to 58dB, in 3dB increments, controlled with a high quality rotary switch. Also, there is an option of a selectable Input Pad of -16dB or -28dB for even more control. Additionally, when the -28dB pad is engaged and it is set at the lowest gain setting (28dB) you can run signal through the M72s to add warmth and tonality to any tracks, mixes, keyboards, drum machines, samples etc... There are also all the modern features we expect on a new piece of equipment: Phantom Power (on/off) , Phase (Polarity) Reversal, and our amazing sounding F.D.I. (FET Direct Input) circuit per channel.<br> <br> The Mercury FDI (FET Direct Input), a proprietary J-Fet circuit, based on a class-A tube topology. The Mercury FDI is designed to reproduce every nuance of a direct recording, while the circuit lets the tube or solid-state character of the amplifier determine the overall tone. The instrument DI signal is sent through the entire microphone pre-amp circuitry, including Mercury’s custom, massive input transformers, so that the individual character of each pre-amp comes through.<br> <br> The Mercury M72s has the rich lows and punchy mids giving you that ‘instant’ vocal tone, a realistic acoustic guitar tone, or add punch to a bass guitar. The same reaction to instruments or source as the vintage module but with slightly more open high end and openness. The Mercury M72s Studio Microphone Amplifier has the ‘vintage’ tone and break up like the original circuit but it is a bit more musical over all (not cleaner, more musical, there is a huge difference).<br> <br> "After building 100's of Vintage V72 packages at Marquette Audio Labs it is nice to know we can continue a tradition that we have been providing for over 10 years now. We are very proud of the Mercury M72s Studio Microphone Amplifiers at M.A.L. and I am personally very pleased with this product which has exceeded my expectations". - David Marquette (Marquette Audio Labs / Mercury Recording Equipment Co. ) "The M72 brings the vocal up in your face in a mix. Not only that, when you're singing, you can distinguish every little nuance." <span style="font-weight: bold;">- Ricky Skaggs</span><br> <br> “Having used the Mercury M72s I now see no need to scour Eastern Europe to search for the last of the original units.” - Joe Chiccarelli "The M72s sounds every bit as good as an original V72s, imparting that nice pillowy softness that is so difficult to get..." <span style="font-weight: bold;">- Pete Weiss, TapeOp Magazine </span><br> <br> <span style="font-weight: bold;">Specifications</span><br> <ul><li>Mic Input Impedance: Approx. 2k</li><li>Suggested Source Impedance: Approx. 200 ohms</li><li>Input Impedance with -28dB Pad: Approx. 4k</li><li>DI Input Impedance: 2M ohms</li><li>Internal Output impedance: 30-50 ohms (depends on gain setting)</li><li>Suggested Minimum Load: 500 ohms</li><li>Max. Mic Signal Input Level (@ 20 Hz): +34dBu with -28dB Pad engaged / +6dBu with</li><li>Pad not engaged</li><li>Max. Output Level: +22dBu</li><li>Frequency response: 20 Hz to 20 kHz +/- 0.5dB</li><li>Harmonic Distortion: All musically related low orders of harmonics, with no</li><li>significant order above 5th.</li><li>Tubes: 2x EF806s, per channel</li><li>Rack Size: 2U</li></ul> <br> <span style="font-weight: bold;">FAQ: Mercury M72s vs. M76m, Whats the difference?</span> <br> Both are multipurpose tools for making music. But they do sound very different.<br> <br> <span style="font-weight: bold;">Tone</span><br> The Mercury M72s is warm and punchy. The M72s has a thick bottom end, a great push in the mid range and a open airy top end. The M72s performance is great on Vocals, Bass, Drums, Acoustic and Electric Guitars as well as room and Over head applications. The M76m is warm, yet open and airy. Compared to the M72s the M76m is overall more evenly "EQ'd". Meaning there is not a push in the low or mids. The M76m shines on Vocals, Bass, Piano and Guitars giving you an amazing of control to shape the amplifier's tone. As well the M76m opens up your room and over head mics. With 70dB (+/-) of total gain and the choice High and Low Input Impedance, so ribbon mics love the the M76m.<br> <br> <span style="font-weight: bold;">Gain</span><br> The M72s has 28dB to 58dB with -16 and -28dB Input Pad and the M76m has nearly 70dB, with the 60dB input gain, and Level Control and the Low impedance selected (+6dB).

Mercury M72s MKIII

<b>2005 TEC Award Nominee</b><br>  <br> Combining a precisely matched octet of discrete bipolar transistors with a laser trimmed DC coupled FET-based output driver stage, the <b>Millennia HV-3C</b> is an entirely "double balanced," extremely wide dynamic range stereo microphone preamplifier intended for demanding acoustic recording. Unlike "textbook" mic preamplifiers, the HV-3C signal path never unbalances or rebalances the original signal — thus maintaining a high degree of signal integrity. Over 20,000 channels of HV-3 micamps are already in use by a who's-who in critical acoustic recording. <p>Unlike preamps using vacuum tubes, IC function modules and audio transformers (all of which can lend their own audio coloration) the Millennia HV-3C is precisely optimized towards capturing the subtlest detail of music and ambiance. With +23 dBu native input headroom, the HV-3C requires no input pads, attenuators, or switch points as found on common textbook preamp designs - thus avoiding yet another source of sonic degradation. Moreover, with its +32 dBu output headroom, ultra-high dynamic stability and linearity is assured. NOTE: The HV-3C is a recently-updated version of the HV-3B two channel mic preamp, with improved mechanical design and cosmetics, and standard 36 step gain control. HV-3C uses the identical preamps as found in the HV-3D 8-ch units. </p><p>HV-3C signal path from input through output is entirely balanced with a minimum amplifier topology. Uncompromising circuit design with meticulous selection and hand-matching of critical components assures consistently transparent audio performance at all dynamic excursions and frequency extensions.</p> <p><b>Features:</b> </p> <ul> <li>High-speed transformerless design</li> <li>23dB input and 32dB output headroom </li> <li>No attenuation pads required Entirely balanced audio path</li> <li>Laser trimmed FET-based output</li> <li>THD+N less than 5 PPM (.0005%)</li> <li>Matched discrete transistor octet front end</li> <li>Precision components throughout</li> <li>Gold audio connectors and switches</li> <li>Standard 36-step gain control (1.5dB per step) </li> <li>Stereo gain matched to .08dB</li> <li>OFC audio wiring, silver Teflon power wiring</li> <li>Ultra-clean torroid power supply</li> <li>Welded 16-gauge chassis</li> <li>Unsurpassed ambience retrieval</li> <li>Effortless lifelike musical performance at all dynamic levels</li></ul>

Millennia HV-3C

Launched in 1970, the <b>Neve 1073</b> 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. <br><br>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. <br><br>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. <br><br>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. <br><br>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. <br><br><b>Highlights</b><ul><li>Classic transformer microphone preamp amp (Class A design)</li><li>3 EQ bands</li><li>Hand-built and hand-wired to original 1970s design</li><li>HP filter</li><li>Neve designed hand-wound transformers </li><li>Both inputs are transformer balanced and earth free</li><li>Microphone Input: Gain +80db to +20dB in 5dB steps.</li><li>Line Input: Input impedance 10k ohms, gain +20dB to -10dB in 5dB steps.</li><li>Output: Maximum output is >+26dBu into 600 ohms.</li><li>Output is transformer balanced and earth free</li><li>Distortion: Not more than 0.07% from 50Hz to 10kHz at +20dBu output</li><li>(80kHz bandwidth) into 600 ohms.</li><li>Freq Response: +/-0.5dB 20Hz to 20kHz, -3dB at 40kHz. EQ Out.</li></ul>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). <br><br><b>Neve "Classic" Outboard - Genuine Neve product hand built in UK using all of the original components, nothing new in 30 years</b>

Neve 1073 CH

The<span style="font-weight: bold;"> Phoenix Audio DRS-1R Mic Pre/D</span>I uses our well proven and loved Class A output stage (DSOP-2), but also has our latest breakthrough in transformerless Class A, Discrete Mic Input Technology which gives a "valve-like" sound. It also incorporates our high input impedance DI circuitry.<br> <br> <span style="font-weight: bold;">Specifications</span><br> <ul><li>Class A (DSOP2) Output specs. Frequency response: 20Hz to 20kHz +- 0.5dB, Maximum Output = +26dBu @ 1kHz, Noise = -90dB @ 20Hz to 20kHz. </li><li>Output connections: XLR's on rear panel of API rack </li><li>Phoenix Audio’s unique Class A, transformerless, True balanced Mic input stage. Microphone input: XLR's  Rear Panels of API rack </li><li>Gain Range (Mic input): -30 to -70 in 5dB steps With 10dB more available on the output fader. </li><li>Gain reduction: -30dB push-button Mic/Line switch (Mic input) </li><li>High Input Impedance DI: Mono 1/4" Jack on front panel </li><li>Gain Meter: LED Metering. (Green = -2dbu, +4dbu & 0dBu, Yellow = +13dbu Red= +16dBu) </li><li> Phantom Power: Switchable phantom on Push-button Switch </li><li>High Pass Filter: on Push-button Switch, 120hz 6db per octave </li><li>Phase Reverse: on Push-button Switch </li><li>Mute : on Push-button Switch </li><li>Frequency Response Mic Input Stage: -0.4dB @ 40Hz, -0.3dB @ 25kHz </li><li>Frequency Response: DI Input Stage: -0.3dB @ 40Hz, - 0.5dB @ 25kHz </li><li>Typical Headroom: +24dB on Mic-Pre stage    </li><li>DI Stage gain: Maximum of 20dB </li></ul>

Phoenix Audio DRS-1R 500

The <b>Prism Sound Orpheus</b> 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. <br><br>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). <br><br>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. <br><br><b>Orpheus signal path</b><br>Eight 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. <br><br> <b>The Prism Sound Orpheus Features:</b> <ul> <li>No-compromise, full Prism Sound audio quality </li><li>Dedicated FireWire interface </li><li>ASIO and WDM drivers provided for Windows XP and VISTA </li><li>Directly compatible with CORE AUDIO on Mac OS X 10.4+ (Intel & PPC) </li><li>Eight "Prism Sound" AD and DA channels, plus SPDIF, ADAT & headphones </li><li>Four high-end integrated mic preamps (typ.-130dBu EIN), switchable phantom </li><li>MS Matrix processing on mic inputs </li><li>Two instrument inputs </li><li>Prism Sound "Overkillers" on every channel to control transient overloads </li><li>Fully-floating (isolated) balanced architecture for optimum noise rejection </li><li>Mono or stereo input configurations </li><li>Outputs arranged as stereo pairs, each with individual mixer </li><li>Low-latency "console-quality" 8-bus digital mixer for foldback monitoring </li><li>Fader, pan, cut, solo on every mixer channel </li><li>Dual headphone outputs each with its own front-panel volume control </li><li>Front-panel master volume control, assignable to selected channels </li><li>Configurable for stereo, 5.1 or 7.1 or surround monitoring </li><li>Built-in sample rate conversion (SRC) on DIO channels </li><li>Prism Sound 4-curve SNS noise shaping on digital outputs </li><li>State-of-the-art clock generation with proprietary hybrid 2-stage DPLL MIDI in/out ports</li></ul><br><br> <b>No-compromise, full Prism Sound audio quality</b> <br> 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. <br><br> 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. <br><br> 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. <br><br> 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. <br><br> 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. <br><br> 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. <br><br> <b>Standards compliant FireWire interface</b><br> 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. <br><br> 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. <br><br> The solution was to create a unit that is dedicated as a FireWire interface and is compatible with Windows PC and Apple Mac computers. <br><br> 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. <br><br> <b>Orpheus Flexible Inputs and Outputs</b><br> 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. <br><br> 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. <br><br> 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. <br><br> 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. <br><br> <b>Digital Mixer</b><br> 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. <br><br> 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. <br><br> 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. <br><br> 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. <br><br> <b>Orpheus Flexible Monitoring</b><br> 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. <br><br> 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. <br><br> <b>Sample Rate Conversion and Noise Shaping</b><br> 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. <br><br> <b>Unsurpassed Jitter Rejection</b><br> 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. <br><br> <b>Support</b><br> 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. <br><br> <b>The Last Word</b><br> 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. <br><br> <a class="bluelinks" href="http://www.prismsound.com/m_r_downloads/Orpheus_Operation_Manual_A4.pdf">Download Product Manual</a>

Prism Sound Orpheus

The <b>ProAc Studio 100 </b>is consistently selected by top recording engineers for near-field monitoring in major studios - ample testimony to its overall sound quality and neutrality at monitor reference levels. <p>Few compact loudspeakers offer such a clean uncolored performance and remarkable transparency. With virtually flat frequency response and negligible distortion, the Studio 100 is one of the most accomplished compact performers on the market today.</p> <p>The bass unit is unique. Manufactured exclusively for ProAc, it has a particularly linear motor assembly, superb magnet and chassis construction and a special center pole plug. When precisely tuned in the cabinet, this driver gives an incredibly natural bass quality and definition, combined with generous power handling.</p> <p>The tweeter is a featherlight one-inch soft dome unit, once again specially manufactured for ProAc. Made from a new impregnated fabric, the dome itself is exceptionally light in construction giving the Studio 100 a distinctively uncolored and transparent high frequency.</p> <p>The high quality crossover network marries the two drive units seamlessly giving a spacious sound stage with almost tangible imagery. Only the finest components are used and the speaker is wired throughout with our own high-quality multi-strand wire. The cabinet itself is made from a composite material with walls of differing thicknesses and a new and more efficient heavy damping material ensures that the cabinets are practically inert.</p> <p><br>Although the Studio 100 can be shelf-mounted, high mass stands with good rigidity are preferable for optimum results. The full potential of these thoroughbred designs will only be realised through the use of the highest quality partnering equipment.</p><p><span style="font-weight: bold;">Specifications</span><br></p><ul><li>Nominal Impedance: 8 ohms</li><li>Recommended Amplifiers: 30 to 150 watts</li><li>Frequency Response: 35hz to 30Khz</li><li>Sensitivity: 88db linear for 1 watt at 1 meter</li><li>Bass/Midrange Driver: 6 1/2" treated cone with special center pole plug.</li><li>Tweeter: 1" (25mm) soft fabric dome with ferrofluid and rear loading. Mirror image offset.</li><li>Crossover: Finest components on dedicated circuit board. Multi-strand oxygen-free copper cable throughout. Split for optional bi-wiring and bi-amping.</li><li>Dimensions: 16" (406mm) high x 8" (203mm) wide x 10" (254mm) deep</li><li>Weight: 24 lbs (11kg) /cabinet</li><li>Mode: Stand-mounted</li><li>Grille: Acoustically transparent crimplene</li><li>Finish: Available in the following real wood veneers: Black Ash, Mahogany.</li></ul><p><br><span style="font-weight: bold;">ProAc Studio 100 Users</span><br></p><ul><li>Bose Automotive Group</li><li>Jack White</li><li>Rich Costey</li><li>John O'Mahony<br></li><li>Brendan Benson</li><li>Michael Marquart</li><li>Peter Frampton</li><li>University of Michigan</li><li>Michael A Sapone</li><li>Bob Ludwig (Gateway Mastering)</li><li>Greg Calbi (Sterling Sound)</li><li>Tony Maserati</li><li>Rick Rubin</li><li>Joe Gastwirt</li><li>311</li><li>Neil Diamond</li><li>John Scofield</li><li>Bill Frisell</li><li>David Sanborn</li><li>Genesis/The Farm</li><li>Jean Marie Horvat</li><li>Kevin Killen</li><li>Michael Brauer</li><li>Ron Saint Germain</li><li>Husky Huskolds</li><li>Troy Germano Studios</li><li>Ryan Hewitt</li><li>Craig Street</li><li>Bob Ezrin</li><li>Mike Campbell</li><li>EastWest Studios</li><li>Blackbird Studios</li><li>Ryan Freeland</li><li>Richard Dodd</li><li>Ben Fowler</li><li>Neil Dorfsman</li><li>Metallica  </li><li>Red Hot Chili Peppers</li><li>Avatar</li><li>Right Track</li><li>Looking Glass</li><li>Bear Tracks</li><li>Hit Factory Miami</li><li>Chung King</li><li>Village Recorder</li></ul>

ProAc Studio 100 Pair Color-     - Select - Black Ash Mahogany

The <B>Purple Audio MC77</B> 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. <BR><BR>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. <BR><BR><B>NEW features in the MC77:</B> <UL> <LI>True Bypass via sealed relay with front panel switch <LI>Improved stereo linking with front panel switch <LI>Sidechain insert loop or key input with front panel switch <LI>Buffered VU meter can monitor input or output level at +4dBu <LI>LED meter lights that don't burn out as incandescents do <LI>Heavy gauge black stainless steel enclosure <LI>115v/230v operation switchable from rear panel (serial# 600) </LI></UL><B>Features inherited from the MC76:</B> <UL> <LI>Discrete Transistor Audio Path Electronics <LI>Single Element Class A Output Amplifier <LI>Transformer Balanced XLR Inputs and Outputs <LI>Zener Shunt Regulated Audio Power Supply <LI>Compression Ratios - 4:1, 8:1, 12:1 and 20:1 <LI>Fast Attack Time - 20 microseconds to 800 microseconds <LI>Release Time - 50 milliseconds to 1.1 second <LI>Gain of 45dB (full gain with no limiting) <LI>Ruggedized Design - PCB stiffener, chassis mount transformers <LI>Purple Anodized Aluminum Front Panel <LI>3 year warranty</LI></UL>

Purple Audio MC77

The <b>Retro Doublewide</b> is a tube compressor module designed from the ground up for the 500 Series. It's authentic Retro compression for the masses. The Doublewide delivers the tube compression sound you expect because it is exactly that. While not precisely like any other tube compressor, the Doublewide methodology takes after the Sta-Level. As such, it excels at processing delicate sources like bass and vocals. It's all Retro and is built with the same quality components in our U.S. shop alongside our other products. <br><br> The tubes and transformers in a fully complementary push-pull topology deliver sonic characteristics typical of a full sized tube compressor. The Doublewide incorporates four NOS 6BJ6 pentodes in two tube gain stages as well as two high-quality Cinemag transformers.  Input and output are fully floating and transformer balanced for use with line level signals. A hard-wire bypass switch allows for quick evaluation of the compression signature.  Compression attack and recovery times are continuously variable.  Like the Sta-Level and 176, the Doublewide has two timing characteristic modes; Single and Double.  Double mode provides a dynamic program-controlled attack and recovery timing. With these two modes, the Doublewide is versatile in the studio, with full control of weight and punch versus transparency.<br>  <br> The Doublewide is designed to provide the user with years of reliable service. Special circuitry eliminates excess current inrush and component stress when power is applied. Power required is 180 mA (6 watts), which is well within a two-slot power allowance. The stainless steel chassis provides good ventilation to minimize heat build-up. The tubes are self-biased and balanced without the need for user adjustments and there are no internal controls. A meter-zero adjustment is accessible from the side of the unit to compensate for tube variations.  Tube replacement is straightforward and the tubes are readily available.<br> <br> <b>Features:</b><br> <ul><li>Single channel tube compression in a two-slot 500 Series module</li><li>Continuously variable input and output levels with fixed threshold</li><li>Continuously variable attack and recovery time</li><li>Single and double time-constants</li><li>Fully complementary push-pull signal path</li><li>Cinemag transformer balanced, fully-floating input and output</li><li>Hard-wire bypass switch with gold-plated contacts</li><li>Gold-plated edge card connections</li><li>Stainless steel chassis construction</li><li>Side-accessible meter zero trim</li><li>Uses new-old-stock U.S. tubes</li><li>Authentic U.S.-made Simpson gain reduction meter.</li><li>Hand built and tested in the U.S.</li></ul> <b>Specifications:</b><br> <ul><li>30dB of available gain reduction</li><li>Signal to noise ratio greater than 76dB</li><li>Flat frequency response within .5 dB from 20-20,000 Hz</li><li>Total harmonic distortion 1% or less from 0-25dB gain reduction</li><li>Input level –15 to +24 dBm 600 Ohm impedance</li><li>Output level –20 to +12 dBm 600 Ohm impedance</li><li>Power consumption 180 mA (6 watts)</li><li>Tube complement 6BJ6 x 4</li></ul>

Retro Instruments Doublewide

With input and output transformers designed and implemented by Mr. Rupert Neve, the high-voltage 72V topology found in the <span style="font-weight: bold;">Rupert Neve Designs Master Buss Processor (MBP)</span> will integrate flawlessly with virtually any system. Additionally, the MBP incorporates mastering grade detented pots throughout to fine tune its revolutionary dynamics, tone, and stereo field controls. This new topology is a significant evolution of Mr Rupert Neve’s classic designs with appreciable benefits to headroom, dynamic range, distortion, noise, slew rate, bandwidth, and accuracy while still providing for the sweet, musical performance that has been a part of countless recordings.<br> <br> <span style="font-weight: bold;">The Compressor</span><br> The MBP’s two compressor sections allow virtually limitless possibilities in dynamics for either dual mono or stereo sources, with controls for ratio, threshold, attack, release, blend, side chain HPF, limit and make up gain. With the stereo link control engaged, Ch. A settings act as the master control for convenient operation. When engaged, the compressor section can be used in both feed-forward and feed-back modes to provide a transparent modern response (feed-forward), or a smoother more musical vintage response (feed-back). Peak mode alters the compressor’s attack to react to peak transients with a roughly .1ms response time. When the Peak switch is disengaged, the compressor responds to the RMS signal in conjunction with the attack and release settings. SC HPF inserts a high pass filter at 250 Hz into the side chain to deal with intense low frequencies that may skew the response of the VCA with certain songs and instruments. “Blend” creates 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.<br> <br> 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 to another (for instance, a voice-over keying the compressor to duck a background music track).<br> <br> <span style="font-weight: bold;">The Limiter</span><br> The Portico II Master Buss Processor also features an extremely versatile, transparent and musical limiter. At first glance, one might scoff at the single knob operation, however this limiter is extremely intelligent, knowing how to appropriately respond to the various signals presented to it. Our new Adaptive Release Technology is behind this revolutionary performance. Using a blend of release time constants, this limiter will simultaneously respond quickly to transient material (such as the “snap” of a snare drum) and slowly to more sluggish signals (such as a bass guitar). This configuration allows the limiter to grab a transient and let go just an instant later, while also dealing with more constant signals in a slower, more musical way. In this manner, the MBP Limiter can provide a much more aggressive amount of limiting than typically possible, while maintaining the essential character of the music and remaining free of the modulation distortion usually found in a fast acting limiter.<br> <br> Typically there is a tradeoff between how fast limiter can react and the amount of modulation distortion in the lower frequencies. This is due to the lower frequencies finding their way into the side chain signal, triggering the compressor on and off very quickly, which ends up modulating the overall signal. This is interesting to look at with sine waves, but sounds quite undesirable with music. The MBP does not have this tradeoff, and one is able to have the best of both worlds: a quick, snappy response while maintaining the integrity and smoothness of the low end. In addition to the adaptive time constant circuitry, the release time is also varied with the position of the knob. As the knob is turned counter clock-wise, the release time is increased accordingly, as typically one would want a longer release time with a larger amount of reduction.<br> <br> The new limiter found in the MBP is designed to respond as fast as .03 mS in order to reduce the first half of a 20 kHz waveform over the threshold. It has a “medium knee” initial ratio and within 3 dB of the threshold attains a better than 10:1 ratio. A soft clipper circuit catches transients that may have been in the “knee” when the threshold knob is set quite high. Both the limiter and soft clipper are switched out of circuit with the knob is fully clockwise. The release times are fully automatic and adjust depending both on the average depth of limiting and the duration of the transients above the threshold. The limiters share the same discrete, class-A gain module and VCA with the compressors, so using the Limiter does not introduce more stages that the music would have to pass through. This combination of features provides exceptionally transparent limiting, and often allows twice as much gain reduction compared to other limiters before objectionable artifacts become apparent.<br> <br> <span style="font-weight: bold;">The Stereo Field Editor</span><br> The stereo field editor on the MBP takes traditional M-S techniques to new heights with width, depth and corresponding bandpass filters.<br> <br> The width control enables the user to increase or decrease the width of a stereo image (wide/mono) and adjust the amount of ambiance inherent in the recording. As the width control is rotated toward wide, the amount of difference material is boosted, often resulting in more ambient material, and accentuated stereo reverbs. Conversely, the stereo field is contracted when rotated to mono, and, if the left and right channels are highly coherent (i.e. both channels include closely similar material that is in phase), this mono content is enhanced. If the phase of one of the input channels is then reversed the mono content may be virtually eliminated. Because the amount of effect the width control has is entirely dependent on the amount of stereo information in the original source material and the interplay between the stereo field editor′s other controls, listening and experimentation are essential for the best results.<br> <br> The depth control of the MBP adjusts the spatial positioning of elements in the sound stage. Center-panned elements like solo instrument or vocal can be brought forward in a mix, in relation to supporting instruments. In many cases, these same elements may be virtually eliminated without adversely affecting the music bed. Used in conjunction, the depth and width controls effectively alter the perceived room ambience and dimension.<br> <br> To fine tune the SFE, there are individual filters that allow a fine tuning of what information is reintroduced from the width and depth circuits, thus tailoring each effect to a specific bandwidth. For example, if one wanted to increase the amount of low frequencies in the center image, engaging the SFE Depth and Depth EQ, set to LF, would filter out everything in the Mid signal except what is below the filter point (in this case, 250 Hz), and once reintroduced to the original would result in a perceived increase in the low frequencies in the center image. It is also possible to do the same thing with the Width EQ, except instead of boosting the width, cutting it, which removes low frequencies from the Sides, tightening up the low frequency perception in the center. Using the Width EQ again, this time set to HM (or LM as the case may be), increasing the amount of band-passed Side information can provide a wonderful spreading of instruments, reverberation and background vocals, giving the illusion that the sounds are spread further out, enveloping the listener.<br> <br> Another technique available on the SFE is routing the Mid and Side signals to the Channel A and Channel B compressors, respectively. Now whatever amount of Depth or Width is introduced is first routed to the compressors, allowing the user to utilize the compressor features on the Mid and Side information. Now it is possible to not only increase the side information, but to utilize the compressor to bring up some of the low level side information, or allow the user to tame an overly expressive lead singer. With the addition of using the EQ section on the Depth and Width, a wide range of tools is available to the engineer.

Rupert Neve Designs Portico II Master Buss Processor

The Mono is the single channel API 500 series version of the GAMA (Golden Age Microphone Amp), with all the capabilities, features, and quality of the original. The Shadow Hills Mono Gama is centered around our custom discrete op-amp. This no-compromise design exudes hugeness, fidelity, punchiness and depth, across the full frequency range, without becoming veiled or choked. <br><br>The preamplifier input is transformer balanced and utilizes an original Jensen input transformer. We couldn't decide which vintage output transformer we liked best, so we had our favorites perfectly recreated, and included them all. You can change the output transformer by cranking the knob, thus changing the tonal capabilities to compliment to whatever material you are recording. This allows you to audition different settings to find the perfect match for microphone or direct input without patching and un-patching other preamps, and possibly alleviating the need for EQ. Choose between Nickel, Discrete, or Steel settings. Each transformer position is stellar unto itself, but the capability of switching between them is like having three great pre's in one. <br><br>All the standard Mic-pre features are present: phantom power, phase reverse, a twenty dB pad, and direct input. We have also included the ability to pad and reverse the phase of the Direct Input. <br><br>Quality is unmatched. Engraved front panels, a twenty four position Swiss-made attenuator for gain control, custom made knobs, and engraved serial number plates, set the build apart. <br><br><b>Features</b> <ul> <li>Superb sound quality and character </li><li>Switchable output transformers </li><li>Stepped attenuator for gain control </li><li>Phase reversible and paddable DI </li><li>Engraved front panels </li><li>75 ma current draw per module </li></ul>

Shadow Hills Industries Mono GAMA Mic Pre

Building on the success of X-Desk, the <span style="font-weight: bold;">SSL X-Panda</span> packs a versatile feature set into its compact frame. As a standalone product it serves as a compact, high quality 24 input mixer that can be used for tracking (in conjunction with external mic pres) or master mix summing. In conjunction with an SSL X-Desk, X-Panda introduces 24 additional channels, which via the ‘X-Desk Link’ connection, feed directly into the X-Desk Master Buss, Cue and Aux Send/Return system. In conjunction with an X-Rack summing system, X-Panda adds faders and additional inputs to create a versatile mixer. X-Panda can also be used with any other mixer to expand available channel count.<br> <br> <span style="font-weight: bold;">Features</span><br> <ul><li>Highly versatile compact analogue mixer with SSL SuperAnalogue™ benchmark audio quality</li><li>24 Ch Expander for X-Desk or any other analogue mixer</li><li>Adds long throw 100mm faders and 24 Ch Inputs to an X-Rack summing system</li><li>Stand alone 24 input summing mixer</li><li>8 mono fader channels with main and Alt inputs giving 16 mono inputs</li><li>4 stereo fader channels with Stereo Inserts and Stereo Direct Outputs</li><li>Stereo Cue, and FX1 & FX2 mono Aux send controls</li><li>Heavy duty self illuminated Solo/cut Switches</li><li>Compact desktop design with removable 19” rack ears</li><li>Rear panel connections via DB25 D-Sub</li></ul> <span style="font-weight: bold;">Channel Architecture</span><br> X-Panda Mono channels feature a Main and Alt input with a +/-20dB input Level Trim control with a centre detent at unity gain, tri-colour signal indicator LEDs, Phase Invert, an Insert switch and it is also possible to source the Direct Channel Output post fader. Each channel has a Stereo Cue section with dedicated Level and Stereo Pan controls. When tracking this is traditionally used to create a monitor feed to the Stereo Cue Buss Output but an Alt button flips the Stereo Cue section controls to controlling the Alt input Level & Pan for mix down. Each channel features independent FX1 and FX2 send level controls. Self illuminated heavy duty Solo and Cut buttons and precision 100mm faders complete the channel strip.<br> <br> X-Panda Stereo channels feature Input Level control, along with individual L&R signal indicator LED’s, separate L&R Phase Invert and a true Stereo Insert point. The Stereo Cue section is dedicated to Cue Buss functionality (there is no Alt input on stereo channels.) FX send controls are identical to the mono channels but take a summed mono feed of the stereo source. Each stereo channel has a Stereo Balance control and in addition to feeding the Mix Buss each channel has its own dedicated Stereo Direct Output which can also be sourced post the channel fader.<br> <br> <span style="font-weight: bold;">Buss Architecture</span><br> X-Panda features a stereo Mix Buss. When linked to an X-Desk using the ‘X-Desk Link’ connection, the X-Panda Mix Buss feeds directly into the X-Desk Mix Buss (the X-Panda Stereo Cue Buss also feeds directly into the X-Desk Stereo Cue Buss). When linked to an X-Rack fitted with a Master Module using the ‘X-Rack Expansion’ connection, the X-Panda Mix Buss feeds directly into the X-Rack Master Module main Mix Buss (the X-Panda Stereo Cue Buss also feeds directly into the X-Rack Master Module Cue Buss). When used as a standalone product the X-Panda main Mix Buss is set to sum at unity gain enabling it to act as a summing mixer without the need for a Master Level control. When Linked to an X-Desk or an X-Rack configured as a summing system, the X-Panda and X-Desk or X-Rack Solo systems are also linked so that Soloing any channel mutes all other channels on all connected units. It is possible to cascade up to 8 X-Desk and X-Panda units using the ‘X-Desk Link’ system, enabling very high channel count systems to be configured.<br> <br> On the X-Panda, the FX1 and FX2 send system is globally switchable between Pre and Post Fader operation. When linked to an X-Desk using the ‘SSL Unite’ connection, the X-Panda FX Busses feed the Master Section of the X-Desk and are routed to the Master Send Level Controls of the X-Desk. When used as a standalone system the X-Panda FX Buss outputs are set at unity gain so function as a traditional Aux send system, but without a Master Send Level control or a dedicated Aux Return (if an FX Return is required a Stereo fader could be used).

SSL X-Panda

The <span style="font-weight: bold;">Telefunken AR-51</span> 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.<br><br>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.<br><br>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

Originally designed as a "distortion box" to simulate distortion in valve amps, the <B>Thermionic Culture Vulture</B> has found lots of uses beyond this. We know that some owners use them on drum loops, vocals, piano sounds and even across entire tracks (it is a stereo unit). <P>We have refined this so that distortion figures are reduced to only 0.2% at lowest to about 99.9% (at least that's what our distortion meter tells us!). Predominant distortion can be changed from even to odd harmonics with a simple switch. We know it's recently been rented to mix a JAMES BOND movie! <P><B>Key Features:</B> <P>Warm sounds gently or create a noise like a 200 watt guitar stack with all the drivers slashed.<BR>Independent channel operation <UL> <LI>Odd or even harmonic distortion, or combination of both. <LI>All valve design free from solid state additives. <LI>High impedance line input or instrument inputs. <LI>4 & 7 kHz filters <LI>Overdrive & bypass switches</LI></UL> <B>Some Cultured users...</B> <BR> Because of their unique blend of flexibility, musicality, and usability, brought about as a result of being hand-built by some of the worlds leading pro audio valve equipment designers and engineeers, Thermionic Culture products attract their fair share of high profile, industry leading producers and engineers from the world of recording technology. Here is just a handful - why not add your name to the list? <UL> <LI>Tony Maserati - Culture Vulture <LI>Black Dog Productions - Earlybird 2.2, Pullet, Phoenix Mastering Compressor, Culture Vulture. <LI>Al Schmitt - Phoenix <LI>Big Life Music - Phoenix Master Compressor <LI>Steve Dub Jones - Earlybird 2 <LI>The Dairy - Phoenix <LI>Al Stone - Culture Vulture <LI>ROBERT CARRANZA - PRODUCER FOR LOS LOBOS, SUPER GRASS, JACK JOHNSON - CULTURE VULTURE <LI>PABLO MUNGUIA - ROBBIE WILLIAMS - PHOENIX & CULTURE VULTURE <LI>STEPHEN STREET - KAISER CHIEFS - CULTURE VULTURE <LI>JAMES FORD - PRODUCER - ARTIC MONKEYS - CULTURE VULTURE <LI>GROOVE ARMADA - EARLYBIRD & CULTURE VULTURE MASTERING - " Both myself and Tom Findlay bought the Earlybird and the Culture Vulture Mastering, they're like your first mobile phone, once you've used them for a week you wondered how you ever liver without them!" Andy Cato - Groove Armada. <LI>Larry Hammel - Producer, Deepwave Music Productions, Earlybird 2.2 and Phoenix <LI>Peter Katis <LI>Chemical Brothers - Culture Vulture Mastering MV & Phoenix<BR> "The vulture it distorts synths in a truly beautiful way-we used to acheive this effect thru a long,convoluted process now we can get this great sound at a flick of a switch!" - Tom Rowlands <LI>Peter Gabriel - Real World Studios - Culture Vulture <LI>Alan Moulder -Producer/Engineer The Jesus & Mary Chain, Swervedriver, My Bloody Valentine,Depeche Mode, Nine Inch Nails Downward Spiral, U2's Pop. <LI>Mike Caffrey <LI>Steve 'Dub' Jones - Engineer/Producer - Chemical Brothers, Madness. Culture Vulture Mastering MV & Phoenix <LI>Tom Findlay - Groove Armada - Earlybird-2, Phoenix <LI>Pietro Taucher - Engineer for Sharrie Williams <LI>Paisley Park <LI>Tie Dye Keith <LI>Jerry Harrison and Eric Thorngren (Talking Heads) <LI>Peter Katis <LI>Jimmy Boyle <LI>Kevin Bacon ( Axis Studios, recent credits include Jaimeson ) <LI>Angel Mountain Studios <LI>Overtones <LI>Andy Strange ( recent credits include Robbie Williams ) <LI>David Ives <LI>Bernard Butler <LI>Tony Hoffer ( Beck , Turin Breaks ) <LI>Edwyn Collins <LI>Mick Jones <LI>The Libertines <LI>Mark 'Spike¹ Stent ( U2, Madonna, Oasis ) <LI>Portishead <LI>Ken Nelson ( Badly Drawn Boy, Coldplay ) <LI>Dave Gilmour <LI>The Prodigy <LI>Hugh Jones <LI>Strong Room Studios <LI>Air Studios <LI>Mute Records <LI>Ross Hogarth - "Mondo distorto! dude, i like your box a lot...really liked the musicality of it!" <LI>San Francisco Soundworks <LI>“La Source” Mastering suite - Paris France <LI>Yves Jaget <LI>Bertrand Blais producer of Etienne Daho <LI>Wisseloord Mastering

Thermionic Culture Vulture

The first Weiss product with analog only inputs and outputs!<br> <br> <b>Applications:</b><br> <ul><li>Recording</li><li>Mixing</li><li>Broadcast</li><li>Mastering</li><li>Live</li></ul> <b>Microphone Preamplifier:</b><br> <ul><li>Transformer-balanced, discrete class A design</li><li>Current-feedback amplifier architecture for low distortion and wide bandwidth across the full gain range</li><li>Very low noise for a wide range of source impedances and across the full gain range</li><li>15 dB to 60 dB gain, 3 dB steps</li><li>20 dB pad to accommodate line levels</li><li>Extensive metering</li></ul> <b>High-Pass Filter:</b><br> <ul><li>Switchable to 40 Hz, 80 Hz or off</li><li>2nd order Butterworth design</li></ul> <p><b>De-Esser:</b></p> <ul><li>Advanced mixed signal design: direct signal path fully analog, side chain digital</li><li>Precision low distortion MDAC as the variable gain element</li><li>Low latency side chain with 500 kHz sampling rate</li><li>Frequency, bandwidth and threshold front panel controls</li><li>Further parameters (ratio, attack and release time) via DI switches</li></ul> <b>Output Driver:<br> </b> <ul><li>Transformer balanced design</li><li>Level control</li><li>Polarity switch</li><li>High maximum output level (+27 dBu) and output current</li></ul>

Weiss A1

Simplicity is the hardest design principle to follow. It’s at the heart of Wunder's approach to the CM67.<br> <br> Seductively simple, the <b>Wunder CM67</b> offers the presence you’d expect from Wunder Audio underpinned by the latest technology and ideas in PSU design.<br> <p>The K67 capsule, which since 1960 has become the most common and most copied condenser capsule in the world, utilizes a two-piece backplate, allowing the diaphragms to be tuned individually and then matched for equivalent response.</p> The tube is a vintage Telefunken EF86, a pentode wired into this circuit as a triode. It uses a 6.3V filament voltage, and is wired to the PCB via a Teflon socket.<br> <br> The transformer in the CM67 is somewhat unique. It makes use of an elaborate feedback network known as a tertiary winding, which creates negative feedback through the capsule to the tube for its HF de-emphasis. It creates a roll-off in the HF response in order to compensate for the natural HF peak due to the original design of the K67 capsule.<br> <br> It takes a very special transformer to accomplish this task while simultaneously achieving such an impressive sound. Some U67 recreations offer a quick fix to simulate the sound of the original, but most of the modifications that attenuate the top end achieve a less than desirable result. This is not exactly a good trade-off. <br> <br> The CM67 is perfect for vocals where a smoother sound is desired as the 67s are prized for their “warmth.” The CM67 is also a terrific all-purpose microphone, as it works exceedingly well in a close-mic situation. The CM67 is excellent for drums (kick, snare, overheads, room) and guitar amp mics. Because it has a very nice top and bottom, and being very natural in the way it brings out the best from any source, the CM67 is a serious pro studio workhorse.<br> <br> Designed without compromise, Wunder Audio’s new tube microphone Power Supply Unit is the result of complete creative and engineering freedom. Its proprietary design features a high voltage path that runs constant current and balances constant current against a shunt regulator. The result is a completely clean voltage to the microphone free of any RF interference.<br> <br> The CM67 is supplied with its dedicated Power Supply Unit, shock mount, Tuchel microphone cable and quarter sawn oak case.

Wunder Audio CM67