The Acme Opticom XLA-3 is an optical audio limiter built to exacting military-style specifications, designed to produce a full range of non-linear, dynamic audio effects. 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’. 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. Features Made in the US Authentic, military style point-to-point circuitry Unique Sonic Character Versatility Serviceability Ease of Use Competitive Pricing
The predecessor to the Adam A7X , the A7, became the most famous of all ADAM monitors in a very short time. Not only has it been reviewed over three dozen times with outstanding results, it has also received numerous awards. The A7 quickly gained mass appeal, praised in many of the world’s largest internet forums. Still today, the A7 are the reference monitors in many smaller studios. With the A7X, ADAM Audio proudly presents the evolution of a legend. It takes everything that made the A7 such an outstanding speaker to a new level.Driver technology The first aspect that distiguishes the old from the new model is the X-ART tweeter. The ‘X’ stands for ‘eXtended frequency response’ and thus for one of the features of the Accelerating Ribbon Technology that has been drastically improved: the frequency response. It now extends all the way up to 50kHz. In addition, the X-ART tweeter has a higher efficiency and higher maximum sound pressure levels. The perfect integration with the lower frequencies has been achieved with a newly designed 7” midwoofer. It has been redesigned with a much bigger voice coil (1.5”) and is driven by an amplifier with twice the power compared to its predecessor. This combination produces an amazing sound and pressure levels with an almost distortion-free musical reproduction.Amplifiers / controls Each driver has its own dedicated amplifier. A 50W A/B amp is responsible for the X-ART tweeter, while the midwoofer is being driven by a 100W PWM amp. The front panel includes a power switch and a control for the volume that retains the volume setting independently from the on/off switch. On the rear panel are several additional controls: a gain for the high frequencies (± 4dB) and two shelf filters for high and low frequencies. To ensure greater compatibility, there are both XLR (balanced) and RCA (unbalanced) connectors.
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! The ADR Compex F760-RS by Q2 Audio 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 When the Levee Breaks 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. 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". 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. 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. Features: Classic FET based compression/peak limiting/gating, all available individually Switchable compression ratios - 1:1, 2:1, 3:1, 5:1, 10:1, 20:1 Switchable compression threshold in 2dB steps Switchable attack times - 250µs, 2.5ms, 25ms Switchable release times ranging from 25ms to 3.2s plus "auto" setting Peak limiting (100:1) with switchable 50µs pre-emphasis setting (useful for de-essing) Expander/gate with 20dB of noise reduction Compressor side-chain insert (can be internally modded to control the gate, instead) Stereo linkable Swiss-made ELMA™ rotary switches Conductive plastic potentiometers
Originally designed to deliver the truest and most natural gain for
impedance and noise sensitive ribbon microphones, the full-rack AEA RPQ
preamp has proven itself to be the perfect match for any microphone,
ribbon or not. The AEA RPQ500 module is based on this successful
design and provides the same ultra-clean, high-gain signal path that
has earned AEA preamps their great reputation, in a 500 series package.
The RPQ500 has been enhanced with additional features, turning it into
a workhorse tool for your API 500 compatible rack. The RPQ500
delivers the bandwidth and quietness needed for high resolution
recording. JFET circuit topology delivers all the dynamics, subwoofer
bass, and fast transients that your microphones can record. No Load™
input impedance above 10,000 Ω means the RPQ500 will not load down a
mic and change its sound. Low Energy Storage™ circuit design instantly
recovers from overloads for superior dynamic performance. The
CurveShaper™ offers a natural option for sculpting your sound right at
the start of the signal path. Switchable and tunable low frequency (LF)
and high frequency (HF) contour controls allow you to tame proximity
problems and provide HF extension and slope control. The high-frequency
CurveShaper™ excels at adding a touch of air or presence, and the
low-frequency control can remove boxiness and boominess in a fast and
unobtrusive way. The bottom line is a versatile channel strip that will
deliver an unadulterated performance in any application. The
Line/Mic switch bypasses the microphone gain stage and allows the EQ to
be used for tracking with other preamps or during mixdown. Add the
output level control and the RPQ500 becomes a high-quality processing
tool that can be used for mixing and mastering when combined with
summing racks and mixers such as the Purple Audio Moiyn or the Roll
Music Folcrom. The original RPQ with CurveShaper™ was designed
to fully capture every nuance of ribbon microphones: vintage or modern,
passive or phantom powered. Engineers have discovered that the RPQ
also complements their moving coil, tube and solid-state mics. By
virtue of its sonic qualities and versatility, the AEA RPQ500 is the
tool of choice for all microphones whenever a true and pristine signal
path is needed. It will allow the best performance of all your
microphones with a cleaner, more open, and more responsive sound than
you’ve ever heard.
Features:
80 dB of quiet JFET gain High-impedance No LoadTM inputs CurveShaper™ EQ for extended highs and articulate lows Line-level input mode for using the EQ during mixdown Output level control VPR alliance application pending
The Antelope Audio Eclipse 384 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. 64-bit DSP Trinity-level clocks 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.Studio and live 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.Supreme audio quality and boosted efficiency 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. Features: Clocking 64-bit DSP Trinity-quality clocking 0.001 PPM Oven-controlled oscillator Two independent sample rates Complete Varispeed capability 10M Atomic clock input Conversion 384 kHz A/D & D/A converters A/D with Dynamic Range of 124 dB D/A with Dynamic Range of 129 dB Two bypassable A/D inserts 480 Mbits USB 2.0 custom chip Monitoring Three sets of switchable monitor outs Second dedicated monitor D/A Base Management with LFE output Relay attenuators matched to 0.05 dB Complete management via user-friendly Mac/PC/Linux software control panel. Six different presets for convenient set-up.
API 1608 Fader Automation has arrived! Features 1x stereo fader 16 mono faders (can act as group masters) 2x group master faders 3x modules create a 16 channel system (2x 8-channel input fader kits with motherboard and master module with proprietary internal computer, housed under master module within desk) Connects to DAW via MIDI/HUI DAW/plugin write capability Volume/mute/insert on-off automation per channel Simple and intuitive to use Takes only a few hours to install by a competent technician
The API 512C is a mic/line/instrument preamp designed to provide a low noise, unusually good sounding front end for all types of audio systems. Sonically, it is at the top of the "Mic Preamp List", regardless of price. Offering low noise (-129 EIN) and 65 dB of gain, the 512C includes phantom power, switchable polarity, -20 dB pad and Mic/Line or Instrument selector. Front panel XLR and 1/4 inch connectors combined with rear panel mic access allows for additional flexibility when installed in an API LunchBox, Six Pack, 10 position vertical rack, a 2 position horizontal rack, or an API console.
What makes the API 512C unique is its long evolution from the original 1967 era 512, the first modular mic pre, to the current full featured 512C, while preserving the original sound character that made it so much a part of the early days of recording. Offering high headroom and a wide variety of inputs and input access ponts, it is ideal for unusual and demanding applications.
Imagine a situation where only a few preamps are needed, yet the smallest available console has a proportonally "small" mic preamp, making it useless for the demanding application, or conversely, imagine where you need a large number of preamps, and a console of sufficient inputs and quality would be too large to transport or rack mount. The 512C hits the spot with its quality and famous tone. Expand, combine or downsize at any time without trade-ins or product obsolescence. In addition, the 512C's sound and performance exceeds most "console mic pres" in every respect.
The beauty of the entire API 500 Series is its long term flexibility and lasting value when needs change over time. With a range of mounting frame options, the 512C will be a valuable asset to your performance critical applications.
The 512C Mic/Line Preamp makes use of the 2510 and 2520 op-amps and therefore exhibits the reliability, long life, and uniformity which are characteristic of API products.
Features
Mic Preamp with 65 dB of gain
Front and Back Panel Mic Input Access
Line/Instrument Preamp with 50 dB of gain
Front and Back Panel Line/Instrument Input
LED VU meter for monitoring output level
20 dB pad switch, applies to mic/line/instrument
48v Phantom switchable power
Traditional API fully discrete circuit design
Uses the famous API 2520 Op-Amp
2 Year Warranty (labor) 5 Year Warranty (parts)
We double the standard API one year warranty (parts and labor) on this item.
The next generation of Shorti Patchbays has arrived; this 2x48 audio patchbay is wired to DB25 connectors. This unit features exceptional flexibility with the Audio Accessories, Inc. 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). 2x48x1.5RU Mini audio jack panel TRS out to DB25 female connectors Pinned out for Pro Tools interface User-Programmable Normalling We have an excel template that most people find useful for labeling the db25 shorti.
Your dedication deserves the best tools available. Get everything you need to compose, record, edit, and mix—in the highest quality—with Avid Pro Tools 10 software. Whether you work in a professional facility, home studio, or on the road, nothing gives you the quality, performance, reliability, and ease to create like Pro Tools, the most widely used digital audio workstation in the industry.Mix it up—at higher resolution With Pro Tools 10, you can mix multiple audio file formats and bit depths within the same session—including interleaved—without any file duplication to cause project bloat. Plus, with support for 32-bit floating-point file formats, you’ll get higher resolution sound when recording or importing, with more headroom to preserve the integrity of your audio from beginning to end.Drive faster—with better performance Pro Tools 10 is designed with the professional in mind, offering powerful yet easy-to use tools and streamlined workflows that boost your efficiency. Gain better recording and playback performance when working on a laptop, with an external hard drive, or network-attached storage device. Speed up editing and mixing with Smart Tools, Clip Gain, and new AudioSuite workflows. And handle the biggest plug-in heavy mixes with 16k of Automatic Delay Compensation (ADC). Plug into legendary sound Get the sound you want using a wide variety of Pro Tools plug-ins from Avid® and our partners, some of the best audio designers in the industry. Pro Tools comes with over 75 virtual instruments, effects, sound processing, and utility plug-ins—some of which emulate the sounds of classic hardware processors, amps, and instruments, as well as the renowned EQ and dynamics of the System 5, one of the most sought after consoles used to create some of the greatest mixes ever produced. Add more to expand your sonic palette even further.Work the way you want Record, edit, and mix music and sound for picture your way. Pair Pro Tools with your favorite Avid or third-party audio interface to record and monitor vocal and instrument performances. Capture virtual instrument performances using a MIDI keyboard controller. Or record and create with just your computer* and the software alone for ultimate portability. Feature highlightsUnleash your creativity with the award-winning tool-set Get unrivaled sound quality, now with even higher resolution Work the way you want—with an interface or without Create bigger mixes, with up to 96 audio tracks Compose with virtual instruments and MIDI and Score Editors Work faster with Clip Gain, real-time fades, ADC, and more Add multiple audio formats to a session, without conversion Polish mixes with over 75 plug-ins, including the new Channel Strip Speed up mixing with industry-trusted automation tools Get the same features in Pro Tools software Build bigger mixes, with up to 768 audio and 64 video tracks Get highly responsive performance with the extended disk cache Get advanced audio editing, video editing, and automation Mix in up to 7.1 surround
The BAE DMP 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. Features:Built in PSU for extra portability DI in and DI through 48v for phantom power Solid steel chassis Same hand-wired modular design that BAE is known for
Find out why the MicroMain27 is the most sought after speaker in pro audio. Call Vintage King and demo today! The Barefoot Sound MicroMain27 is a groundbreaking new monitor that is in a class all its own. It is quickly becoming the premier choice for top mixing and mastering engineers. The speaker is designed to address the demands of modern recording. It breaks down the barriers between nearfield, mastering and main monitors. No need to have multiple pairs of speakers crowding your studio; no need to guess what the mastering engineer is going to hear. The MM27 is compact and powerful, truly a "nearfield on steroids." While exceptionally neutral and designed for critical listening, the MM27 is still very capable of rocking the house. It redefines the definition of a main monitor -- a "Micro Mainª" monitor. The only monitor you may ever need. 1" soft dome tweeter, dual 5" midbass drivers and dual 10" subs housed in compact sealed enclosures yield high linearity and outstanding impulse response. With 500 Watts of power in the subwoofer channel alone, the dual 10" drivers cross over seamlessly from the midbass, reaching down to 33Hz and rolling off at 1/4 the rate of ported designs to reveal much more deep bass information. Since the sub motor structures are locked together the opposing forces cancel out and the cabinet remains rock steady even at very high output levels. There is no need for bass management in 5.1 systems because the MicroMain27 is truly a full-range monitor. The speaker can be placed either vertically or horizontally using the included pedestal (9" L x 13/8" W x 21/4 H). Hand Crafted in San Francisco, California
*Note* The MM27 will have no problem operating at 100VAC in the 115VAC setting.
The Black Box Analog Design Vacuum Tube Preamp 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!Tonal Control 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! 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! 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.All Tube, All Analog Black Box Analog Design's entirely analog audio circuit uses only tubes for amplification and is entirely free of op amps, ICs and transistors! 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.Real Power 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! Features: Independently controlled Pentode and Triode tube stages Custom wound Cinemag input and output transformers Entirely analog audio path (No ICs or Op amps) Hand soldered, point to point wiring Switchable input impedance done by tapping into the transformer windings 5 position, gentle and musical roll off Linear power supply Passive output attenuation control 48v Phantom power from an entirely separate supply Real, amplifier isolated VU meters Military spec switches
The Burl B80 Mothership is a
multi-channel version of Burl Audio’s critically acclaimed B2 Bomber ADC
and DAC. Pushing the boundaries once again, the Burl Audio B80
Mothership AD/DA converter has up to 80 channel capability. Using a card
based system, and a heavy duty 4U chassis, the B80 Mothership employs
10 card slots with 2, 4 and 8 channel AD/DA cards, all with the B2
quality signal path. Every ADC channel has a BX1 transformer and every
DAC channel has the latest Burl Audio discrete op-amp, BOPA2,
technology. For many who have already embraced the B2 Bomber, this is a
dream come true. Recording an entire band and playing back through 24
channels Burl at a time, is a truly beautiful experience.
With a completely configurable AD/DA, a converter for any situation can
be realized. If you just mix, load up the B80 with 8 BDA8 cards for 64
channels of mind blowing fidelity. If you track live, 6 BAD4 cards will
give you 24 B2 quality inputs, and 4 BDA8 cards gives you 32 channels of
outputs. Rocking.
Digital interconnect is also completely configurable, and controlled by a
swappable motherboard. The first offering of the B80 digital
motherboard, the BMB1, comes standard with two Digi-link connectors for
direct hookup to Pro Tools. Burl Audio does away with the expansion
port, making both primary ports. That means you can play 64 DAC channels
straight out of Pro Tools with one unit! No more cascading units with
multiple word clocks. All channels can run off of one internal master
clock. Good bye word clock generator! Of course if you do need more
channels, multiple units run flawlessly with tight clocking.
The technology that makes this possible is the Burl BackBone. The Burl
BackBone is a high bandwidth backplane delivering incredible digital
audio throughput. The Burl BackBone also delivers large amounts of power
to the analog sections. Master clock travels directly to each converter
eliminating the need for individual PLLs.
Digital ad on cards to follow will be the BMADI and BAES. Because each
card slot can handle a single MADI card, an entire MADI switch matrix
can be realized in one unit. Other future digital cards will include
word clock distribution and time code sync for film and video.
The B80 Mothership is the result of over five years of tireless research
and development. The no compromise, no BS design strategy is
immediately apparent in the build quality. The 4U, steal enclosure is
truly impressive. No plastic, no layers of menus, everything is meant to
be used easily with zero hassle. Configuring a B80 is literally a snap.
All you do is slide the card down its card guide, snap it in the Burl
BackBone, turn the hand screws, and you’re there.
Configuration cards currently consist of the BMB1 with Digi-link, the
BAD4, 4-channel ADC card, and the BDA8, 8 channel DAC card. Cards to
follow are the BMADI card and the BAES card. (available 2011) There are
also plans for film and broadcast grade “Black Face” cards where channel
density and price per channel are of greater concern.
Powering a unit like the B80 Mothership takes some juice, so we designed
the BP250 to drive the 80 channels of class-A analog circuitry. With a
250 watt continuous power capability, and +/- 24 volt rails, a
multi-channel AD/DA converter is finally getting the power it deserves.
For complete control of all of the B80 signal routing, and stand alone
metering, a USB port has been included. This is not an audio interface,
it is strictly for control. A future touch screen control will plug in
here. The flexibility of the B80 design allows for a smooth integration
and upgrade path.
Burl Audio’s intention with the B80 is to create one main chassis that
takes care of virtually any music, broadcast or film recording
situation, revolutionizing pro audio as we know it. In this digital era,
many yearn for the tone of records past, and analog tape is becoming a
lost art. The B80 Mothership clearly bridges this analog to digital gap.
B80 Mothership
4U main chassis BMB1, motherboard 2 Digi-link connectors 1 BNC Word Clock In 2 BNC Word Clock Out 10 card slots All slots accept AD/DA, AES, MADI 2, 4, and 8 channel analog cards BP250, 250 watt, 1U power supply
BAD4, 4-channel ADC
44.1k Hz to 192k Hz, 24 bit, 4 channel ADC Exact same circuit path as B2 Bomber ADC All class A, discrete transistors signal path with zero feedback, zero caps 4 XLR connectivity
BDA8, 8-channel DAC
44.1k Hz to 192k Hz, 24 bit, 4 channel ADC Circuit path based on B2 Bomber DAC Latest Burl Audio BOPA2, OP-AMP All class A, discrete transistors signal path, zero caps DSUB 25 connectivity
Future cards, features
BMADI card with 64 channels at 48kHz, and 32 channels at 96kHz, (Spring 2011) BAES, 16 channel AES/EBU (summer 2011) USB control (summer 2011) BBAD8, Blackface, low-cost, transformerless 8 channel ADC card for film/broadcast BBAD8, Blackface, low-cost, 8 channel DAC card for film/broadcast BWORD, 8 BNC word clock output card for word clock distribution BSUM, summing bus, summing all DAC outputs BCONTROL, control room monitor BMIC2, two channel mic pre / AD BSYNC, time code synchronization
With the release of the 1073 mic preamp/EQ in 1970, and the 1081
released in 1973, Neve’s classic circuitry has consolidated their
position in the professional audio industry. Since then, a desire for
‘cleaner’ recordings using modern circuitry has resulted in classic
circuitry becoming an option – but not the only option.
The development of technology and the digital world cannot be ignored.
With these two mindsets clear, the Custom Series 75 has achieved a world first, the combination of classic circuitry and modern circuitry in one console.
The classic Neve circuitry has been painstakingly re-created, keeping
faithful to the original whilst taking advantage of modern assembly
methods. The original circuits have only been changed where justified by
an improvement in reliability or performance. Rather than running the
length of the console, the voltage summing busses are a mere 150mm long
and they are now balanced, resulting in lower noise and less crosstalk.
Other “charming quirks” of the original circuits such as the level onto
busses varying as more busses are assigned from a channel, have been
overcome. Improved BA338 amplifying stages and legendary BA283 output
stages combined with classic Neve LO1166 output transformers, sound
warm, defined and punchy.
Stereo busses are implemented in both classic (voltage summing) and
modern (current summing) technology and the feed to these can be
selected on a per-channel basis. The 2 busses are combined at the master
fader then passed via (patchable) 2254 compressors and an insert, to
the stereo outputs. There are both modern (transformerless) and classic
(transformer) outputs, providing maximum flexibility of sonic “flavour”.
Naturally the AFL busses are also replicated for correct Solo
monitoring. The eight Group outputs use classic circuitry, with the
added bonus of Stem outputs, inserts and 8 dedicated faders to feed
groups or playback to the stereo busses. A 32 channel console can mix a
total of 80 inputs to stereo.
The 2081 inline channel module features a blend of the best features of
Neve’s legendary 1073 and 1081 modules. The mic preamp and 4 band EQ is
straight from the 1081, while the output amp is based on the 1970′s
BA283, single ended, class A circuitry with a gapped core transformer,
as used in the 1073 and 2254. Five auxiliary sends, fader swap and a
choice of either classic or modern channel output circuitry are
provided.
Two 2254 compressors, four stereo reverb returns, eight recallable
scenes, monitoring of up to 12 sources simultaneously and comprehensive
7.1 monitoring are just some of the features found in the master
section.
Available in a 16, 24, 32, 40, 48, 56 or 64 module chassis, the Series
75 is flexible and able to suit a variety of studio applications from
broadcast to music and even film production.
Designed specifically for longevity, gold plated switches and connectors
have been used for all audio circuits, and all parts have been selected
on their projected availability many years from now. All capacitors in
the signal path are polypropylene film or Rubycon ZLH series
electrolytic, chosen for superb sonic performance and long life.
Ergonomic design allowing the engineer to reach any parameter on the
board from a seated position (up to 32 Channels) make this console a
workstation that is comfortable for long hours, flexible for countless
practical and creative applications, and unbeatable for sonic character
or transparency!
The Neve Custom Series 75 is a console that is here to stay.
Technical Specs
Switchable “retro” (transformer) or “modern” (transformerless) output Voltage and current summing mix busses All capacitors in the audio signal path are either: o WIMA polypropylene film capacitors, or o Rubycon ZLH electrolytic capacitors Gold plated switchers and audio connectors Two linkable mono 2254 compressors (option) Four stereo reverb returns Eight recallable console scenes Up to twelve monitoring sources True stereo to 7.1 monitoring Original mic pre-amp from the Neve 1073 Four band EQ based on the 1073 circuitry & 1081 functionality Five AUX sends (2 stereo & 3 mono) Redesigned BA338 amplifying stages BA283 output stages Neve LO1166 output transformers
Main Features
One year labour and five (5) years parts guarantee From 8 to 64 channels Flexible Master Section position Flexible Channel numbering Fader swap and patchable insert points Wood Panels: Rosewood, Ash, clear and black. Redundancy broadcast power supply Handrest Leather: black, red, blue Optional patch-bay Dual stereo outputs (classic and modern) Fader Automation (from mid 2011) Retrofit HUI automation (from mid 2011) All Direct Transformer outputs, 8 sub-groups In line design with great monitor features
Dimensions
32 Channel Frame
Width: 167cm (65.75 in) Depth: 99cm (38.98 in) Height (without stand): 45cm (17.72 in)
24 Channel Frame
Width: 135cm (53.15 in) Depth: 99cm (38.98 in) Height (without stand): 45cm (17.72 in)
16 Channel Frame
Width: 102cm (40.16 in) Depth: 99cm (38.98 in) Height (without stand): 45cm (17.72 in)
To survive in this modern day audio jungle, the contemporary studio must
be lithe and agile; a facility that can adapt to any workflow:
effortlessly. Tracking, mixing and mastering services oftentimes must be
performed under one roof, utilizing every particle of outboard gear
available.
The Dangerous Liaison realizes
this dream by providing unprecedented access to your outboard equipment.
Connect up to six of your favorite two channel units to the Liaison
(more if you daisy chain, leverage a patch bay or add a Dangerous
Master). Instantly audition any device or combination, change the order,
dial in some parallel processing and then store these customized signal
paths as presets to use again, on demand, anytime. True hard-wire
bypass relays remove the gear entirely from the signal path; this
includes outboard that lacks an integrated bypass or true hard-wire
bypass. Mastering grade components switch instantly and silently,
providing true A/B comparisons without coloration or latency. Pre and
post-processing monitor outputs make it simple to audition between the
original source and the treated product.
The A and B buss paths aren’t just for comparing signal paths. They can
be strung together in series. For example, if your favorite EQ (the BAX
of course) is connected at 1 and a limiter is connected at 5, selecting 5
from Buss A and 1 from Buss B reverses their order.
Parallel Processing allows the original and effected signals to be
blended independently. For example: crush the drums with heavy
compression for tight, explosive impact, while mixing in the original to
retain transients and dynamics.
Features
6 Stereo Insert Loops, Assignable to 2 Stereo Busses Flexible and Potent Parallel Processing Loop Route, Switch, Matrix, Flip & Audition Instantly Recall any Tracking, Mixing or Mastering Configuration Seamlessly Integrates with Dangerous Master
Application Examples
Tracking: craft recall-able patches for cutting Pop vocals, hip hop, voice overs... Kick drums that click, resonate or “thwomp!” Bass lines that are rich, liquid, punchy or fat Parallel processing with a compressor, Dolby A or... Mixing: use it for All the tracking tricks above And all the mastering examples below Mastering: audition the possibilities EQ before compression? Or compression before EQ? Compression ratio at 1:1.5 with a low threshold followed by limiting at 1:20 with a high threshold?
The Dangerous Liaison: unlock the potential of all your outboard on
every session, every time, in precise, repeatable combinations.
Tempest is a collaboration between Dave Smith and longtime friend and fellow instrument designer Roger Linn. Though they've consulted with each other on past projects, Tempest marks the first time a product will carry both the Dave Smith Instruments and Roger Linn Design logos. "If you're going to make a drum machine, who better to have in your corner than Roger Linn?" said Smith, referring to Linn's legacy as inventor of the digital drum machine. Tempest is a professional drum machine that generates its sounds using six of Dave's powerful, highly-acclaimed analog synthesis voices, and uses an innovative, performance-oriented operating system that permits an extraordinary level of control to create, edit, arrange and manipulate beats in real time without ever stopping. Features Each of the 6 analog voices has 2 analog oscillators plus 2 digital oscillators (with a large bank of included samples), Dave's classic analog low-pass filter with audio-rate modulation, an additional high-pass filter, analog VCA with feedback, 5 envelopes, 2 LFOs, an extraordinary variety of analog modulation routings, and stunning sonic quality, warmth and punch. Although optimized for drum sounds, it excels at tuned sounds as well, and even doubles as a 6-voice analog synth. In addition to the 6 direct voice outputs, there are stereo mix outputs and phones outputs, plus 2 inputs for foot switches or expression pedals, MIDI in/out and USB. The performance-oriented operating system, 90 panel controls, and bright 256 x 64 OLED display work together to provide a tightly integrated, non-stop workflow: record a drumbeat in real-time, switch to another drumbeat and use the lit pads to record it using step programming, switch to another drumbeat and record tuned keyboard parts, use the 2 touch controllers to to record real-time note or beat-wide parameter animations, use the generous sound controls to edit any of the drum sounds, tweak the analog effects or drum mix, arrange beats in real time and record the live arrangement into a song, enter/exit Song mode and much more, all without ever stopping play. 16 pressure- and velocity-sensitive lit pads are arranged in a 2x8 configuration, providing intuitive access to all your fingers and providing the ideal compromise between the popular 4x4 pad arrangement (popular for real time programming ) and 1x16 arrangement (popular for step programming) because Tempest does both. The pads can be used to play 32 drum sounds (2 banks), mute/un-mute the 32 sounds on playback, play and arrange 16 beats in real time, play one sound at 16 tunings (in a variety of scales) or 16 velocities, or as 16 time steps for step programming. The ROLL button permits creating drum rolls or repeated groove patterns by varying pad pressure as the beat records, and doubles as a momentary "stutter" effect when the pads are assigned to play beats. Use the Sustain button on tuned parts like a keyboard's Sustain pedal, or to choke drum sounds or drumbeats when the pad is released. Two pressure- and position-sensitive Note FX slide controllers permit real-time recording of note or beat-wide sound parameter changes into the drumbeat as you play. For example, record simultaneous filter frequency, tuning, envelope decay and pan changes for each note, or control similar parameters affecting the entire beat. A variety of unique effects are provided while maintaining a pure analog signal path: 1) Stereo analog compressor and distortion circuits affect the stereo output mix, 2) beat-synced delay is achieved by generating additional delayed note events within the sequencer, and 3) a beat-synced "stutter" effect is created entirely within the sequencer by looping short portions of the drumbeat on demand. The degree of swing timing can be adjusted in real time during playback. Roger used his entire bag of tricks to make the swing sound very human and natural. Compact and portable: 15.4" L x 9" W x 2.5" H
Dramastic Obsidian 500 Features TXIO Enhanced Transformer Balanced I/O Precision Stepped Controls To Accurately Recall Settings VCA Feed-Forward Compression Selectable Internal High Pass Filter Expansion Port For Additional Features Link Multiple Modules Together Selectable Pad Settings Superior Imaging Specifications Ratio settings: 1.5:1, 2:1, 4:1, 8:1, 10:1, 12:1 Attack dontrol Range: 0.1, 0.3, 1.0, 3.0, 10, and 30 mSec Release control range: 0.1, 0.3, 0.6, 1.2 sec, AUTO and Lo-Fi HPF settings: 30Hz, 60Hz, 105Hz, 125Hz, 185Hz, 330Hz Makeup Gain Output: 12db 1 Year Limited Warranty
The Electrodyne 501 is a
two-stage, discrete transistor, transformer-coupled preamp with active
DI based primarily on the modules found in the classic 1608 console.
Each amp stage is individually optimized for peak performance using
detailed Electrodyne factory engineering notes and select high
performance components identical to the originals. In fact, the 501’s
transformers are made by Electrodyne’s original supplier to exacting
factory specifications.
The new preamp’s active DI circuit presents an almost immeasurable load
(over six megohms!) to sensitive musical instrument outputs allowing
incredibly accurate capture of the instrument’s true tone. Furthermore,
the output of the DI circuit is designed to directly connect and
interact with the mic input transformer to permit an extremely broad
spectrum of tonal options.
The faceplate of the Electrodyne 501 features a large rotary gain
control offering up to 68dB of gain – adjustable over 50dB in 2dB steps
with two ranges via a 20dB pad switch – and a smaller output level pot
infinitely adjustable from 0 (off) to +6dB over unity. Additional
switches for impedance selection (50 or 200 ohms), phase reverse, +48V
phantom power and DI (with 1/4" input jack) are also present, as well as
a clip LED that monitors all three amp stages and illuminates when any
stage is 3dB from clipping.Specifications Maximum Gain: 68db. Adjustable over 50db in 2db steps with two ranges using 20db pad. Output level control: Infinitely adjustable from 0 (off) to +6db over unity. Input impedance: Microphone, 50 / 200 ohms selectable. DI, over 7 megohms. Output impedance: 150 ohms Distortion: 0.02%typical over entire gain range. Maximum output: +30db at less than 0.8%thd. (+29.5db at 0.06%thd) Frequency response: +/- 0.25db, 10hz to 60khz. Signal to noise: -80db typ, -60db at absolute maximum gain. Clip indicator: Monitors all three amp stages and illuminates when any stage is 3db from clip.
The Focal Twin6 Be 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. Both 6.5” drivers handle low frequencies but only one of the two (selectable) is passing lo-mid frequencies.About Focal Professional Studio Monitors Listen to your music, not to your speakers 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.Specific tools for specific needs A speaker that “reveals” 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.A speaker that “translates” 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.Our Speakers “reveal” and “translate” better than others. Why? 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. We scrutinize each of these elements with the same care to achieve our goal of absolute transparency.
Gefell M300 Stereo Set - Vintage King Edition Includes:
2x M300 2x 10m XLR cable 2x elastic suspension or 2x standard mounting EH93.1 1x stereo bar for ORTF and XY Aluminum Case
The Gefell M 300 is a studio quality compact
condenser microphone. The frequency response is practically linear for a wide range of sound
incidence and has a smooth treble boost rising to about 3 dB between 6
and 10 kHz. The miniature microphone is well suited for vocal and
instrumental soloists, permitting a lifelike recording in a multitude of settings.
The M 300 is designed for studio applications in radio, television
broadcasting and films; for live performance and recording of
instruments, vocals and speech; and sound reinforcement in the
professional and semiprofessional market - including adverse acoustic
conditions such as churches.
A new low-noise integrated hybrid circuit and the transformer-less
circuit design guarantee an extremely wide dynamic range, with a low
equivalent loudness level and a high reliability in operation. RFI susceptibility is very low.
The microphone is powered by a 3-pin XLR connector with the C 70
microphone cable. The power supply is provided by 48 V phantom powering,
which is internationally standardized as P 48 in DIN 45596 and IEC
268-15. A two-channel powering is possible with the N 200 power supply.
The finish of the miniature microphone is dark bronze.
The Grace Design m102 Optical Compressor takes the sonic performance characteristics from the compressor section of the Grace Design m103 Channel Strip. The unit is based on an optical attenuator, the purest high fidelity gain control mechanism available, and provides gentle limiting to fairly aggressive compression while remaining neutral and transparent. Housed in a 1U, half-rack enclosure, the single channel m102 sports a familiar and elegant industrial design that was derived from the Grace Design m101 Microphone Preamp. Front panel controls include +/- 10dB of input and output level on rotary pots that allows for fine tuning of gain settings. Four additional pots allow the user to control Threshold, Attack, Release and Ratio settings. A three position switch selects Normal operation and the added ability to Link two m102 units for stereo compression or to select a Sidechain input for frequency dependent compression. A 10 segment LED array displays gain reduction levels, and a bi-color LED is provided for signal presence and peak detection. The rear panel includes an AC inlet for the built in universal power supply, XLR and ¼” TRS balanced input and output connections, and a ¼” jack used for linking two m102’s or for use as a side-chain input.
The Great River 32EQ is a new 500-series version of the EQ and filters from the renowned Harrison 32 Series consoles.
The 32EQ incorporates the original specifications and with support
directly from the original designers at Harrison Consoles it is
guaranteed that the prized characteristics of the original were
maintained in the new design.
The 32EQ has the full features of the 32-series EQ
Low, Low-Mid, Hi-Mid, and High EQ bands with Gain and Frequency controls Low and High Band “peaking” switches EQ in/out switch Harrison’s renowned High- and Low-pass filters with sweepable frequency Filter in/out switch
An internal jumper provides selection of the “vintage” feedback design,
or a non-feedback option. The Harrison 32-Series console was the
world’s first 32-bus “inline” recording consoles. They became a staple
among recording studios and were the basis for many console designs
(Harrison and otherwise) that followed. Countless hit records were
produced on Harrison consoles during the birth of modern pop
productions, including Abba, Sade, Queen, Janet Jackson and Michael
Jackson. The 32C console was used by Bruce Swedien in the recording and
mixing of Michael Jackson’s Thriller, the best-selling album of all
time.
Gary Thielman said, “For many years, Harrison has had requests for our
prized analog products in a smaller form-factor. During that time, we
kept hearing great things about Dan Kennedy and Great River
Electronics. Their products are superbly made and they are as
fanatically supportive towards their customers as we are to our own. We
realized that we had an opportunity to launch a product that the world
has been requesting, while continuing forward with our passion which is
building large-format consoles.”
Like all Great River and Harrison products, the 32EQ is designed and built in the USA.
The original Helios Type 69 mic pre/EQ, designed by Dick Swettenham, was "The Sound of Olympic Studios", the legendary recording studio used by Led Zeppelin, The Rolling Stones, The Who, The Faces, Humble Pie and Bad Company to name a few. The Helios Type 69-500 holds true to the original Dick Swettenham design, but with a number of modern updates such as 5db steps on the mic preamp gain switch as opposed to the original 10db steps, additional 16khz EQ frequency selection for today's high-bandwidth digital recordings, and a balanced output for easy interfacing with any digital converter.
Love your 500 Series rack, but can't quite pull off that "high-end" vocal sound you hear in your head? Enter The Brute from Inward Connections. This 500 Series compressor/limiter is a gentle giant, adding substantial warmth and presence to your vocal. It helps bring out a breathy and airy quality while at the same time smoothing out peaks and bringing the vocal up to the front of the mix where it belongs. A monster on vocals but not to be overlooked on guitars, bass, and anywhere else an optical limiter can be of use.
Excels at giving you an up-front vocal sound
Inward Connections: boutique gear handmade in LA
Excels at giving you an up-front vocal sound No frills here, just results. Send a vocal through The Brute and it'll sound like you've been working for days...or using gear many times more expensive. This optical limiter loves bringing vocals right up front in the mix, while retaining a smoothness and airiness, so you get just the right amount of squeeze. A VU meter is a nice addition for the 500 Series format, added to by a 250Hz highpass filter plus a link switch for when you can't get enough of The Brute.Inward Connections: boutique gear handmade in LA Initially one of LA's best-kept secrets, the word about Inward Connections has been sweeping the globe. After all, when you hear guys like Chris Lord-Alge and Eddie Kramer swearing by a brand, you know to pay attention. Dreamed up and designed by Steve Firlotte, a well-known LA tech, the Inward Connections line offers premium options for the discerning ear, all meticulously handmade in Firlotte's Los Angeles workshop.
Inward Connections The Brute features:
500 Series optical limiter
Gain reduction control
Zero adjust trim
Output level control
Highpass filter: 250Hz for detector circuit
Bypass switch
LED lighted VU meter
VU meter output/reduction switch
Link switch
VF-600 all discrete legendary amp blocks
Opto-cell reduction circuitry identical to Vac Rac tube limiter
Balance input and output transformers
Fits standard 500 Series slot configuration mechanically and electrically
Powder coating in Gunmetal gray
Specifications:
Gain Reduction: up to 40dB
Input Impedance: >100K ohms balance
Output Impedance: 600 ohms balance
Frequency Response: +/- 0.5dB@20Hz to 50KHz
Output Signal to Noise: -95dB or greater
THD + Noise: .01%@1KHz/+4dBu
Reserve Makeup Gain: +15dB
The UBK Fatso " I had three goals in mind when I set about to modify what was already one of my favorite pieces in rack number one, Dave Derr's brilliant Fatso Jr." - UBK
1. I wanted to create a collection of fixed-setting compressors that, with the turn of a single knob, could shape and tame any sound you throw at it, and be especially adept with those that are the easiest to screw up: vocals, drums, bass, acoustic and electric guitars, and pianos. 2. I wanted these preset comps to offer up several distinctly different ‘flavors’ of compression, each of which has a ‘grab’ and ‘motion’ that is totally unlike the others, in order to give the modern engineer maximum flexibility in terms of style, attitude, punch, and squeeze. 3. I wanted to make these compressors incredibly smooth and easygoing at moderate settings, but able to go to extremes (and beyond) to create sounds that are surprising, inspiring, and drenched in vibe. No matter how much you dig in, the results should be musical; it might be too much compression for you, but it will never be *bad* compression. “How do I do this?” Once upon a time, compressors came with two knobs, and your compression options were ‘more’ and ‘less’. But make no mistake: this minimalism in design was actually an asset to the guy who had to turn the knobs and push the faders. The engineers who designed these simple boxes paid painstaking attention to every crucial aspect of envelope shaping --- the knee, the ratio, the attack and the release --- and they tweaked and tweezed the mysterious and often chaotic interplay of these parameters until they found their holy grail: a circuit whose action was incredibly musical and whose grab sounded amazing on every source and sound you feed it. “Compression was easy, and it always sounded good.” So it is with the UBK Fatso. In addition to all the tone-sweetening characteristics and features of the classic Fatso --- harmonic thickening, tape-like saturation, independent ‘warmth’ control to soften and tame high frequency harshness --- the UBK Fatso features 3 brand new presets, lovingly crafted by yours truly, tweaked exclusively by ear, and exhaustively tested in the trenches on every sound you’re ever likely to find in your mixes, and a few you probably never will… just in case. Splat - this is my take on the comps built into my favorite 3-lettered vintage console. At modest settings, it stiffens and reigns in drums, focuses a vocal, and enhances the vibe of the original sound while making it easier to manage. But when you dig in, wonderful things begin to happen. Drums develop thwack and hit you in the chest. Vocals get utterly creamy with a pleasing, old school hair. Loops come alive and breathe organically. Electric guitars get thicker, deeper, and stay pinned where you want them. But don’t be afraid of those meters: push it farther, past that red light. 20db of reduction? Don’t worry, keep going. Ever heard your drum buss do that? Didn't think so.Welcome to 'no rules' compression. Smooth - this is classic limiting, with a twist. My intent here was to craft a tracking style limiter that would allow the engineer to shave 6-10db off the peaks of instruments without sounding like much of anything happened, yet things are somehow sweeter, they behave better. When you print elements thru this preset, you’ll find your mixes come together easier and faster, with less eq and compression. Things just ‘fit’, the way they did when we tracked to tape. But again, what happens if you start to abuse this preset in the mix? ‘Fast and smooth’ begets and ‘fast and aggressive’, that’s what. Slap it on electronic drum loops and watch the Germans seethe with envy. Put it on a room mic in parallel, maybe a little Abbey Road really is what that kit needs. Crazy hip hop vocal refusing to behave? Clamp it down, ruthlessly, effortlessly.One compressor does it all. Glue - the bottom end never sounded like this. It is simply unreal how easily this preset will lock the bass in place, punch it up, let all the notes ring clear as a bell, and make it loud while getting it out of the way. Engage the transformer to make the low end sing, even on iPod docks. If gluing the bass were the only thing the UBK Fatso did, it would still be worth the price. But Glue is every bit as versatile and surprising as the rest of this box, especially when pushed. It’s an extraordinary drum compressor, imparting a distinct smack even as it makes the room explode. It’s lovely on acoustic guitars, Rhodes, sitar… whatever.When all else fails, glue the sucker down. In addition to all of these purposeful compressors, the UBK Fatso features Dave Derr’s classic“Spank” for that inimitable SSL-style smack. You can also combine any two presets, or all three, for even more unusual and unexpected results. So grab those knobs, experiment fearlessly, and ignore those meters: your ears will tell you all you need to know.
The Little Labs Redeye 3D Phantom
is the new improved major revision of the popular Little Labs Redeye
direct box / re-amp box. More people are re-amping now than ever before,
and we at Little Labs know re-amping! We made this new major Redeye
revision because we wanted to make sure we continued to make the best
sounding, most flexible, easiest to use, and reasonably priced re-amping
and direct box product available.
Features
Simultaneous vintage transformer direct box and re-amping tool Compare direct and re-amped signal with one button Phantom powered super hi fi buffered or passive direct box inputs What you heard is what you will hear easy re-amping operation Easily interfaces pro gear with guitar gear and vice versa Expandable for daisy chaining redeyes to feed multiple amps or pedals Rack mountable 4 redeyes fit in a 1U space
For easy accurate re-amping the Redeye 3D lets you listen thru your
whole recording chain, from Redeye 3D direct box, to mic pre to DAW (or
tape machine), to Redeye 3D Re-amp, to your guitar amp. In DI (direct
box) mode the instrument / re-amp out on the front of the Redeye 3D
works as a thru signal so you can simultaneously feed your guitar amp
while supplying a signal from the rear xlr to your mic pre and DAW or
tape machine. In re-amp mode the instrument / re-amp out on the front of
the Redeye 3D signal comes from the line level output of the DAW or
tape machine (converted to HI z guitar level signal) to feed your guitar
amp. This simple way of listening thru the chain (re-amp mode) and
being able to bypass the mic pre and DAW or tape machine in the chain
(DI mode) makes level adjustments a snap. This also assures you a
re-amped guitar sound that will be exactly what was heard when laying
down the track.
The heart of the original Redeye was a Little Labs custom wound
transformer. This transformer was chosen for its sonic characteristics
and is made with the same core material and winding technique as the
legendary UTC transformers found in many classic pro audio devices. The
Redeye 3D features two of these excellent transformers for simultaneous
direct box and re-amping use. We have also added to the Redeye direct
box a very high quality, hi fi, phantom powered, high impedance (10MΩ)
instrument buffer. When you are using the Redeye with sensitive passive
or piezo pick ups, this high quality buffer assures no tone change from
loading will occur. For those who use active pick ups or prefer the
sound of a passive direct box, you still have the option of plugging
into the Redeye 3D direct box passively as in the original Redeye. The
Redeye 3D sounds amazing and even if you never used it for re-amping it
is an incredible value just as a direct box.
The Redeye 3D phantom - the now even smarter flexible audio tool to aid you in all stages of your production.
The Lynx Studio Technology Hilo Reference AD/DA Converter System re-engineers and redefines the concept and functionality of the two-channel converter. Essentially three units in one: A/D converter; D/A converter; and headphone amplifier, Hilo offers connectivity, performance and control never before available. Newly designed differential analog stages and bi-linear converter topology provide industry-leading noise and distortion specifications and unprecedented transparency. Digital I/O is extremely versatile with support for AES/EBU, S/PDIF (via coax or optical) or ADAT. The Lynx LT-USB is standard, allowing additional USB I/O and computer connectivity. Hilo is also the first professional converter with touchscreen operation and monitoring. All controls and signal routing is set up using easy to use menus. The setup page shows two stereo input meters and three stereo output meters. Optional screens show two selectable meter sets and even old-style analog VU meters. Future screens (updatable in the FPGA via download) will include further metering options, diagnostic operations and personalization. The internal 32 channel mixer allows any input to be routed to any output. In fact there are three stereo analog output options. Line Out that supports eight standard trim settings, Monitor Out with a volume control, and a Headphone output, also with a volume control. In fact the headphone output has its own dedicated D/A converter with the option of a different mix than the Line and Monitor outputs.Features Full-featured, ultra high performance AD/DA converter 12 inputs and 16 outputs Control and monitoring via 480 x 272 LCD touchscreen Three in one pro-level unit: A/D converter; D/A converter; headphone amplifier Latest FPGA technology allows future enhancements and ability to use LCD technology for added metering, diagnostic, measurement functions Utilizes Lynx Studio Technology’s proprietary SynchroLock™ technology to virtually eliminate jitter Analog, AES/EBU, S/PDIF (via coax or TOSLINK), ADAT, USB inputs and outputs Ships with LT-USB RoHS compliant Designed and manufactured in the USA by Lynx Studio Technology, Inc. **The Lynx Hilo has been improved as of April 2012 and now contains the following revisions**
Additions
Output Mix Routing Page: Mono Mode. Any stereo input can now be set to Mono. For example, when bringing in a single channel on Line IN Left, pushing the Mono button under the meter will allow the same signal to also be routed to both the LEFT and RIGHT Outputs. The Mono button will also merge any LEFT / RIGHT pair to a single mono signal routed to both outputs. It is important to note that this selection is unique for every input/output combination. Overload Indicators. On the Output Mix Routing Page. Above each vertical meter we have added an Overload indicator. This aids in setting levels in the digital domain when routing the signals. “Show/Hide Recall Scene” button added to the Display menu. This button allows the end user to add a button to easily recall and switch between scenes from the VU and Horizontal meters pages. This same button also allows this added Scene Recall button on the Meters page to be hidden. “Power Up State” button in the Tools Menu. Useful for customers who power down an entire rack or studio with one switch. When this button shows “Standby”, Hilo can only be turned on by touching the power switch on the front. When the button shows “On”, Hilo can be turned on by the power switch on the back or by applying AC (with the power switch in the On position.) Headphone “Jack Sense” implemented. Hilo will now detect whether or not a cable is plugged into the Phones jack on the front of Hilo. When nothing is detected, the Rotary Control is only used for the Monitors output level – you do not need to press the knob to toggle between the levels for Headphone and Monitor outputs. Ability to play audio files directly from an iPad. Using Apple’s iPad Camera Connection Kit and aps such as iTunes, Garage Band, Golden Ear and others, Hilo can directly play back iTunes files as well as higher resolution WAV and FLAC files. ADAT implementation Phase 1. Hilo can now play 8 channels ADAT audio via the Optical output. This offers 8 channels at 44.1/48 kHz, and implement s S/MUX for 4 channels at 88.2/96 kHz or 2 channels at 176.4/192 kHz. ADAT inputs will be implemented in a future firmware release. Improvements “Return to Meters” button on the Display page, formerly named “Menu Delay”. “Return to Meters” better describes the function. The selectable time has been changed, to a minimum of 15 seconds (until the Meter screen comes back on) to 5 minutes. A “Never” option allows the user to only get back to meters by pushing the Display button in the lower left corner of the LCD. “About Hilo” in the Info page. Information about the LSlot card now shows the serial number. Level meters on Output Mix Routing Page now correctly show levels from -60dBFS to 0dBFS. Previously the meters showed levels down to -38.5dBFS.
The Mäag EQ4™ 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. 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. Presented in the flagship 500 Series format, the EQ4 offers the legendary AIR BAND™ and five other sonically superior band passes. Specifications Frequency Response: -2 dB points, 10Hz & 75kHz Nominal Input Impedance (XLR): 48 K Ohms, balanced Nominal Output Impedance (XLR): 50 Ohms, balanced Headroom: +27 dBu THD + Noise: < 0.005% *Specifications subject to change without notice.
The Manley Reference Cardioid Microphone shares the same electronic attributes as the Gold Reference Series, but has a center-fixed cardioid-only capsule with a thicker gauge (6 micron) gold sputtered diaphragm. With the similar film thickness and construction, similar high frequency resonance (a little edge), similar proximity effect and pretty good immunity from pops and sibilance problems, our Reference Cardioid more closely recalls how many of the vintage European tube mics such as the beloved U47 sounded like when they were new. Its rich tonal balance and liquid character is consistently admired for instruments such as guitars, drum overheads, saxophone, and especially vocals. With your present mic, if you find yourself leaning on your compressors and boosting 5 or 10K to score a bit more testosterone, then the Reference Cardioid just might be the mic you?re looking for to cure what ails ya. If you seem to be constantly boosting 12-18K and trying to get a clean, intimate sound, then probably the Reference Gold would be the safest bet.
Using the popular module format, the McDSP 6030 Ultimate Compressor
offers ten different compressors. All of these designs are by McDSP -
some completely from the ground up, while others are emulations of
existing gear with unique variations created by McDSP. Each 6030
Ultimate Compressor module is easy to operate, and yet has enough
sophistication for the most discerning professional.
Whatever your style, from smooth tube emulations to aggressive
solid-state designs, the 6030 Ultimate Compressor has a custom-made
dynamic range control module that is just right for you.
Features
Unique twists to classic designs in addition to several completely new designs Multiple compression algorithms in a single compressor Side chain support Analog Saturation modeling Double precision processing Ultra low latency Mono and stereo versions Compressor Emulations
U670
In 2000, a generous (and very trusting) Nashville-based engineer named
Jeff Balding offered McDSP the opportunity to ‘borrow’ his Fairchild 670
compressor to see how it could be modeled. Some of this effort turned
into presets for the CompressorBank plug-in, as well as a mode in the
CompressorBank CB4 plug-in. But there was always more that could have
been done, and now it has. A new set of attack ballistics was created to
accommodate modern production styles, the ‘warmth’ factor was tweaked,
and the U670 was born.
Moo Tube
An all tube design emulation, with some unique twists. Mid range
sensitivity, attack and recovery time ranges, and output frequency
response characteristics were all re-worked by McDSP engineering. And we
like the cow factor.
iComp
The iComp may not operate on your iPad (yet), but this all-original
McDSP design sounds great. Attack and release times are not adjustable,
but are instead automatically updated based on user selected threshold
and ratio control values. Good on songs headed to iTunes.
Opto-C/Opto-L
Not just another electro-optical attenuation circuit emulation – these
Opto models use McDSP designed key signal circuits for an improved
(watch out for jimmies) response from the originals. The Opto models
also use a different release characteristic, because we like to tweak!
British-C
Based on a more ‘traditional’ design, the British-C model offers the
standard compressor control set – threshold, ratio, attack, and
recovery. McDSP added internal smarts to prevent unwanted distortion,
even at extreme compression settings, not unlike other high-end analog
compressors from this part of the world.
Over EZ
The Over EZ module incorporates a smooth knee response with a flexible
control configuration, making it useful in a variety of situations.
SST’76
It may not get you from San Francisco to London within 6 hours, but this
compressor has some super sonic transporting capabilities of its own.
The SST’76 is a fast reactive design that sounds great on drums and
other percussive sources. The solid-state (SST) circuit model uses a
McDSP designed key circuit to boot.
FRG444
Referred to as ‘The Frog’ by McDSP staff, the FRG444 uses a moderately
aggressive compression design for a more, well, aggressive kind of
sound. The FRG444 will not give you warts either. But goose bumps
possibly, especially on big rock drum kits.
D357
The D357 is the most aggressive compressor in the 6030 collection. An
LED style gain reduction meter is used to display the rapid dynamic
range control changes the D357 is capable of enacting on unsuspecting
audio. Use with caution!
Notable McDSP Users
Joe Chiccarelli Mitch Allen Jeff Balding Jay Baumgardner Brent Carpenter Brett Chassen Richard Chycki Cory Churko John Cooper Brett Dicus Patrick Dillett Jack D. Elliot Ray Fabi John Feldmann Sam Fishkin Terry Fryer Jay Gordon Ron Harris Stephen Hart Ross Hogarth Goh Hotoda Joel Jaffe The Jam Shuji Kitamura Gary Lux Bruce MacPherson Brad Madix Roger J. Manning Jr. Buddy Miller Frank Morrone John Paterno Dave Pensado Bob Power ROCAsound Eric Rosse Thom Russo, Jr. Squeak E. Clean Marc Urselli Ken Van Druten Visionary Music Group Joey Waronker Jim Warren Scott Weber Byron Kent Wong Giles Woodhead
The MSC-72 Mic Amplifier
The M72s is based on the most sought after vintage "V series" module,
the ‘Telefunken/Siemens V72s.’ Which is most famous for being used in
REDD.37 consoles used on the early Beatle recordings by George Martin at
Abbey Road Studios in London, England.
Unlike those vintage modules that had a fixed 34dB gain, the gain of the
Mercury M72s is variable from 28dB to 58dB, in 3dB increments. Also,
there is an option of a selectable Input Pad of -16dB or -28dB for even
more control. When the -28dB pad is engaged and it is set at the lowest
gain setting (28dB) you can run line level 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.
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 preamp
circuitry, including Mercury’s custom, massive input transformers, so
that the individual character of each preamp comes through.
The Mercury M72s has the rich lows and punchy mids giving you a instant
vocal tone, or assisting with a realistic acoustic guitar tone, punch to
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 that "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).
Tubes
The Mercury M72s amplifier uses 2x EF806s, per channel
"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." - Ricky Skaggs
“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..." - Pete Weiss, TapeOp Magazine
"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 actually exceeded my expectations." - David Marquette (Marquette Audio Labs / Mercury )
The MSC-72 Equalizer
The Mercury EQ-H1 features a passive EQ circuit with a single ended gain
make up amplifier topology, based on the vintage ‘Pultec EQH’ circuit.
The Mercury EQH2 circuit provides a musically satisfying result
unobtainable with active and parametric EQs, since the EQH2 does not
rely upon negative feed back (and all it's associated phase and dynamic
distortions) to achieve equalization. The EQH2, like all Mercury
products, has transformer balanced (XLR) input and outputs. The only
additions or changes to the original are a much more powerful and stable
power supply and running DC on the heaters, rather than AC.
EQH2 vs EQP2
Simply, the key to the tone of these two circuits is it’s amplifier, and
Mercury EQ-H2 and EQ-P2 have different amplifiers to make up the gain
lost by the passive equalizer circuit, thus it has a different tone.
Meaning the EQH2 and EQP2 sound different on the same application, and
have a different reaction to the same instrumentation or voice.
The EQH2, like the mono version EQ-H1, is more punchy and robust and has
a slight push in the low end and lower midrange. The EQP2, like the
mono version EQ-P1, is more open and silky, and equally adds warm to
highs, mids and lows of your source. Although both sound amazing on any
application and are both multi-purpose tools (equalizers) in the studio,
if you had both you could then choose between applications. The Mercury
EQH2 is warm and fluid but a bit "thicker" than the EQP2, thus shines
on thickening (and pushing forward in the mix) your Vocals, Bass, Kick,
Snare, and Acoustic Guitar etc... The Mercury EQP2 is great for Shaping
and adding ‘air’ to your vocals, Acoustic instruments, Guitar, Piano
etc... and is magical strapped across your two buss to add musicality to
you mix. The Mercury EQ-H1 and EQ-P1 are based on the original Pultec
equalizers which were tools developed to deal with the limitations of
recorded music. Limitations that most often manifest themselves in the
highest and lowest frequencies of the program material. The family of
Pultec EQs were originally designed to bring back the life and
musicality lost in the recording. Whether by accident or genius, nothing
has been able to do it better. The interaction of the passive boosting
and attenuating shelving EQs (not relying on negative feed back), as
well as the transformers, tubes and other amplification circuitry all
add to the incredibly musical character of the product. Working
engineers try other types of equalizers, but always end up coming back
to the Pultec style as the equalizer of choice for those final touches
while tracking or mixing and even mastering.
Tubes
The Mercury EQ-H1 amplifier uses a 1x 12AX7 and 1x 12BH7, per channel.
Frequencies
Low Frequency Select (CPS; Cycles per second): 20, 30, 60, 100, 200 Hz Low Frequency Boost Control: Shelf Boost, 0dB to +13dB Low Frequency Attenuate Control: Shelf Atten. 0dB to -17dB High Frequency Select (KCS; kilocycles per second ): 3, 4, 5, 8, 10, 12, 16 kHz High Frequency Boost Control: Shelf Boost, 0dB to +16dB High Frequency Attenuate Control: 10k Shelf Attenuate, 0dB to -16dB
Metric Halo recognizes that sometimes you already have mic pres you love or don't need pres at all, but you're eager to tap into the phenomenal conversion that's built MH's stellar reputation. They heard and delivered. Meet the Metric Halo LIO-8 FireWire Mac interface and standalone converter, borrowing its converter design from the acclaimed ULN-8 to deliver eight channels of straightforward archival-quality 24-bit/192kHz conversion whenever you need it.Plenty of I/O with fully clickless remote-controlled monitor outputs Offers pristine and colorless 24-bit/192kHz Comes loaded with renowned DSP sound-shaping tools Operates standalone when you need it Expandable to work in tandem with other MH gear Relevant today and tomorrow thanks to upgradability Plenty of I/O with fully clickless remote-controlled monitor outputs The hallmark of any Metric Halo piece of gear is a functional design with pro needs at the forefront. The connectivity of the LIO-8 matches that of the ULN-8, just without the eight preamps. (Keep in mind that if you decide you want to tap into Metric Halo's boutique-quality pres, you can upgrade your LIO-8 to add four or eight pres at any time with a connector board available here at Vintage King.) The LIO-8 offers eight line inputs plut eight channels of AES I/O with eight balances sends and eight line/monitor outputs plus MIDI and word clock connectivity. All of the LIO-8's analog outputs also feature Metric Halo's clickless remote technology, allowing for flexible monitoring applications from stereo up to 7.1 sound.Offers pristine and colorless 24-bit/192kHz Rather than lock you into a color from the beginning, the Metric Halo LIO-8 is designed for transparent performance built with a straight wire gain mentality. Metric Halo's goal was to provide the cleanest and most transparent conversion platform. If you want to put your very character-heavy mic pres in front of it or plug-ins after it to get color, you can dirty it up however you like, but the LIO-8 will ultimately give you what you give it. It's become a popular choice with mastering engineers for that reason.Comes loaded with renowned DSP sound-shaping tools Metric Halo was the first to make portable interface DSP a reality and the LIO-8 carries the tradition nicely. Thanks to its 2d DSP chip, you can take advantage of a bunch of included processing solutions, including Metric Halo's exclusive per-channel Character control that lets you dial in the right flavor and add some color to your inputs. You can also upgrade and get the +DSP license to tap into 100+ additional processing tools from Metric Halo's renowned collection.Operates standalone when you need it When what you need is an excellent standalone converter, the LIO-8 is your ticket. Just boot it in standalone mode and you'll be ready to go. The range of I/O onboard the ULN-8 opens up your options, with line inputs, eight channels of balanced send, eight line/monitor outputs, and eight channels of AES I/O, letting you go from and into just about anything. Expandable to work in tandem with other MH gear When you need a larger-scale setup, the LIO-8 is ready for expansion. It works seamlessly with other ULN-8, ULN-2, and 2882 units even scaling the DSP organically so you don't have to compromise. Relevant today and tomorrow thanks to upgradability Metric Halo is committed to making their new and legacy gear upgradable. After all, why obsolete your own gear? Their track record is proven, with even their earliest-era gear still in use today. To achieve the "future proof" design, Metric Helo's DSP processing card is upgradable and they're constantly making sure the MIO control software and drivers backing the unit up are as good as can be too. Plus, here, you can add microphone preamps to your LIO-8 at any time should your needs change, rather than go out and grab the ULN-8. The bottom line is that Metric Halo is all about protecting your investment and making sure you have an interface that sets a standard today and tomorrow too.Metric Halo LIO-8 features: Exceptional archival-grade converter quality Flexible, integrated analog-domain monitor controller Universal sample rate support Analog, AES, and FireWire interfacing Rock-solid stability: mature, real-world tested Dependable firmware, software, and drivers Instant access to tactile controls Fully digital control for total recall, remote control, and control-surface support Comprehensive precision metering Works with or without a computer Metric Halo’s exclusive per-channel selectable character Integrated ultra-low-latency instantiable processing with plenty of DSP power to run it
The Millennia Media HV-35 features a front panel instrument input, DC coupled ribbon mic with 10dB gain boost setting, 80 Hz rolloff filter, 48V phantom, 15 dB Pad and Polarity flip. The gain control is continuously variable. It will work in any API-compatible 500 Series rack.
The MXL Genesis II takes the first generation Genesis microphone to a new height of performance and versatility. The Genesis II borrows the outstanding qualities of the original Genesis and adds our dual diaphragm capsule and patented warm and bright switch. This switch allows you to change the tonality of the mic to suit your application. The Genesis II also includes the celebrated vintage, new old stock Mullard 12AT7 vacuum tube which is hand-selected and burned in at our California headquarters prior to shipment, ensuring the highest quality. Patented warm and bright switch Distinctive glossy red body and gold grill Vintage tube sound quality Two distinct tonal characteristics Hand-selected new old stock Mullard 12AT7 tube - 10dB pad 150hz, 6dB/octave roll-off switch Includes 7-pin Mogami cable Custom pop filter MXL Micro-fiber cleaning cloth Power supply Shock mount XLR Mogami mic cable 3-year warranty/90-days tube
Launched in 1970, the 1073 is the first choice of leading producers and artists, delivering the unique Neve sound on some of the most famous recordings of the past 40 years. The big, punchy sound of the 1073 complements any musical genre – from rock to pop, hip-hop to rap, thrash to classical. And now it is available for your LunchboxTM
Designed and manufactured in England, the 1073LB retains the unique sonic characteristics of the original 1073 Classic microphone preamplifier by using the same architecture, matching components and original hand-wound transformers. And it delivers it in a modern and portable form-factor that professional producers and engineers demand.
With new features like a fine TRIM control, switchable microphone input impedance, signal presence LED, intelligent protected switching of front combi-XLR input connector and Neve's clever Audio Processing Input design, the 1073LB takes your LunchboxTM to the next level.
Simply install into an empty slot in your compatible LunchboxTM rack, connect your microphone or line level signals and inject that legendary sound into your audio creations.
Highlights
Classic transformer microphone preamp amp (Class A design) Audio path is absolutely 100% discrete and the same design as the 1073 Classic module mic/pre circuit Neve designed hand-wound transformers Both Mic and Line inputs are transformer balanced and earth free Gain knob with Signal Presence LED +5/-10dB level Trim control with integrated phantom power on/off switch Phase, Impedance and Front Input selector switches Front combi-XLR connector with intelligent switching of phantom power Audio Processing Insert design allows processing from adjacent Neve modules in same Lunchbox to be inserted into the 1073LB's pre-output stage Microphone Input: Gain -80db to -20dB in 5dB steps Line Input: Input impedance 4k ohms bridging, gain -20dB to +10dB in 5dB steps Output is transformer balanced and earth free Distortion: Not more than 0.07% from 50Hz to 10kHz at +20dBu output (80kHz bandwidth) Freq Response: ±0.5dB 20Hz to 20kHz, -3dB at 40kHz Crafted in England by Neve engineers LunchboxTM is a trademark of Automated Processes, Inc. Note: The Neve 1073LB will not work with the A Designs 500HR 2 Slot rack or the Empirical Labs EL500 2 Slot 500 Series Rack due the physical size of the slots in those particular racks not being able to accommodate the dimensions of the module. The module will fit in all other new 500 series racks that we carry.
The Neve 1073LBEQ provides a
Neve 1073 EQ circuit in a single 500 Series module. The 1073 is widely
considered the world’s premier mic preamp & EQ module, delivering
the unique Neve sound featured on some of the most famous recordings of
the past 40 years.
The big, punchy sound of the classic Neve 1073 module complements any
musical genre, from rock to rap, jazz to classical. Its lush sound with
classic Neve signature makes it the consummate mic pre/EQ module for
recording vocals, guitars and acoustic instruments of all descriptions.
With the 1073LBEQ module, the highly sought-after classic EQ sound is
now available in a 500 Series form-factor.
Highlights
Legendary three-band EQ design When integrated with 1073LB preamp, the combination is 100% discrete and genuine 1073 mic pre/EQ Rotary controls provide accurate adjustment of the 1073 three-band EQ and high pass filter Signal presence LED illuminates green from a level of -25dB and red from a level of +24dB Switch for EQ IN/OUT function, with LED indication Electronically balanced classic Neve circuits used in Line Input
and Line Output stages for standalone module use. These stages are
bypassed and the 1073LBEQ module is 100% discrete when combined with the
1073LB mic preamp Audio Processing Insert design allows the audio to/from the
1073LBEQ module to be inserted into the audio path of an existing 1073LB
(modules must be fitted in the same rack) High frequency concentric knobs perform two functions: HF shelf control and High Pass Filter control Fully screened module packaged in Mu-metal case eliminates stray electromagnetic fields from Lunchbox PSU or other module types
The 1073LBEQ provides the legendary sound of Neve in a remarkably simple
format. A line level input feeds an electronically-balanced EQ circuit
with three bands of adjustable EQ and a high pass filter, which can easily
be switched in/out. The output is also an electronically-balanced line
level output. A signal presence LED and Neve’s clever Audio Processing
Insert design is also integrated into the module.
The 1073LBEQ module can be inserted between the input and output stages of the 1073LB module, creating a true 1073 classic audio path.
This mono unit is the perfect solution for studios and audio engineers
looking to bring in the very essence of the Neve 1073 EQ sound to their
portable racks.
Note: The Neve 1073LBEQ will not work
with the A Designs 500HR 2 Slot rack or the Empirical Labs EL500 2 Slot
500 Series Rack due the physical size of the slots in those particular
racks not being able to accommodate the dimensions of the module. The
module will fit in all other new 500 series racks that we carry.
Originally designed in 1974, the Neve 2264A mono Limiter/Compressor unit quickly became a legend. Delivering similar facilities and performance as the earlier 2254 Limiter/Compressor type, it did so despite being packaged in a more compact form-factor. Discrete, Class A/B design and transformer-coupled circuits used in the input, innovative bridge-driver design, side-chain and output stages ensured a totally unique sound. The original 2264A units are still sought after, decades after production stopped, and they remain one of the milestone pieces of studio equipment. And now the 2264A is available for your 500 Series rack. Crafted in England by Neve engineers, the Neve 2264ALB retains the unique sonic characteristics of the original 2264A by using the same architecture, matching components and original hand-wound transformers as its classic counterpart and adds new features: a Signal Presence indicator LED and a Slow Attack option. And it delivers it in a modern and portable form-factor which professional producers and engineers demand. Simply install into an empty slot in your compatible 500 Series rack, connect your line level signals and inject the legendary Neve dynamic sound into your audio creations. And it's the perfect pairing for the incredible Neve 1073LB microphone pre amp module. The 2264ALB module can be inserted between the input and output stages of the 1073LB module, creating an all-new Neve classic audio path. Classic… and Revolutionary. Features: Classic mono Limiter/Compressor, available in a 500-series format Transformer-coupled circuits used in Input, Bridge-driver, Side-chain and Output stages Independent Limiter and Compressor IN/OUT selectors BYPASS switch connects input terminals directly to output terminals Slow Attack switch changes between classic 2264A attack settings and new Slow Attack setting for buss compression work Digital LED meter with approximate PPM ballistics indicates 0dB to 20dB of gain reduction Audio Processing Insert design allows the audio to/from the 2264ALB module to be inserted into the audio path of an existing 1073LB (modules must be fitted in the same rack chassis)
The Peluso Microphone Lab P12 is inspired by and styled after
AKG's legendary vintage "C12". It has the 9 polar patterns, controlled
from the power supply. It faithfully reproduces the airy sonic qualities
and the robust low end of the original "C12".
The output transformer is a custom unit made by Tom Reichenbach of
CineMag Transformers.
The microphone comes as a complete system, with Microphone in Wood Box,
Power Supply, 7 Conductor Cable, and Shock Mount, packed in an
attractive Flight Case.
Hand built in the USA.
Technical Data
Type: Condenser Pressure Gradient w/ 34mm edge-terminated capsule
Frequency Range: 20Hz/24Khz
Polar Pattern: 9 - Switchable from omni- to bi-directional
Sensitivity: 11mv/pa
Impedance: 200 ohms
SPL: 136 db
Equivalent Noise: 15 db (A-weighted)
Tube Type: 6072A-M
Power Requirements: Dedicated Power Supply
Size: 45mm x 240 mm
Weight: 750 g
About Peluso Microphone Lab
Each of our microphones are hand-crafted, using the finest components.
All the tubes in our microphones have been tested and hand selected in
our lab to insure the highest quality and lowest noise possible. We
carefully test and listen to each microphone, and all the components of
systems, during construction and prior to shipment. Our commitment is to
bring you the best possible microphone at a fair and affordable price.
We have a full service repair lab with an extensive stock of vintage
parts and a full service diaphragm repair lab. We offer the benefit of
twenty-six years of experience repairing and restoring all brands of
vintage and modern Microphones.
Phoenix Audio DRS-EQ-500 Specifications:
Class A (DSOP-2) Output specs. Frequency response: 20Hz to 20kHz +- 0.5dB Maximum Output Level: +26dBu @ 1kHz Noise: -90dB @ 20Hz to 20kHz. Phoenix Audio’s unique Class A, buffer DSblip stage. Input level range: -10dBu to +10dBu Self-contained on-board “boost” PSU. Gain Meter: LED Metering. (Green = -2dB, +4dbu & +10dbu Yellow = +13dB, Red = +16dbu. High Pass Filter: Roll-off starts at 80Hz @ 12dB per Octave. Gyratory-based EQ Frequency Centres: (4 Bands) High – 15K, 10K, Hi
Mid – 6K, 3K, 1K6, Low Mid – 800Hz, 400Hz, 200Hz, Low – 130Hz, 80Hz,
40Hz EQ Cut/Boost Levels: 16dB Cut/Boost Per Band, all EQ bands are
stepped/detented to 21 positions for easy recall & stereo operations
for mastering or the mix bus. DRS-EQ/500 has an overall output level with 15db of additional
gain even when the EQ is in bypass, allows the output fader to be used
to drive the output stage & transformer to color & saturate your
sound, again even when the EQ is in bypass.
The Prism Sound Orpheus is a FireWire interface for personal recording and sound production, for professional musicians, songwriters, engineers and producers. Orpheus is ideal for music and sound recording, production & monitoring, stem-based mastering and analogue summing. Orpheus provides Prism Sound's renowned performance and sound quality in a dedicated FireWire unit compatible with Windows XP & Vista and MAC OS X 10.4x (Intel & PPC). Orpheus has line, microphone and instrument inputs, good foldback and stereo or surround monitoring capabilities, ADAT and SPDIF digital I/O plus support for external MIDI devices. Microphone inputs include MS matrix processing and high-performance digital sampling-rate conversion (SRC) is available for digital inputs or outputs. Orpheus signal path 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.
The Prism Sound Orpheus Features:
No-compromise, full Prism Sound audio quality
Dedicated FireWire interface
ASIO and WDM drivers provided for Windows XP and VISTA
Directly compatible with CORE AUDIO on Mac OS X 10.4+ (Intel & PPC)
Eight "Prism Sound" AD and DA channels, plus SPDIF, ADAT & headphones
Four high-end integrated mic preamps (typ.-130dBu EIN), switchable phantom
MS Matrix processing on mic inputs
Two instrument inputs
Prism Sound "Overkillers" on every channel to control transient overloads
Fully-floating (isolated) balanced architecture for optimum noise rejection
Mono or stereo input configurations
Outputs arranged as stereo pairs, each with individual mixer
Low-latency "console-quality" 8-bus digital mixer for foldback monitoring
Fader, pan, cut, solo on every mixer channel
Dual headphone outputs each with its own front-panel volume control
Front-panel master volume control, assignable to selected channels
Configurable for stereo, 5.1 or 7.1 or surround monitoring
Built-in sample rate conversion (SRC) on DIO channels
Prism Sound 4-curve SNS noise shaping on digital outputs
State-of-the-art clock generation with proprietary hybrid 2-stage DPLL MIDI in/out ports
No-compromise, full Prism Sound audio quality
Orpheus makes no compromises on audio quality. It is the result of years of research and development into digital audio conversion and extensive dialogue with Prism Sound's customers.
Orpheus The Orpheus design brief was: Get Prism Sound quality conversion and mic preamps into a 1U box at a more accessible price point. Sound quality just wasn't negotiable. Orpheus has the same no-compromise analogue front and back ends, with the same fully-balanced-throughout architecture, the same isolation barriers protecting the analogue from digital and computer interference.
Orpheus draws on Prism Sound's years of experience in developing digital audio products, including its range of audio test equipment, adopted by a wide variety of clients across the audio industry from pro-audio to consumer electronics. This experience means that Orpheus is well-behaved both as a computer peripheral and an audio processor.
Reliability is vitally important in professional recording. Prism Sound has always made extensive use of precise software calibration techniques in its converters - pots and tweaks are always unreliable, so there are none.
The design team has gone to great lengths to minimise noise and interference, in particular hum. All of the analogue circuits have galvanic isolation, while the unit's electronically balanced I/O allows it to handle common mode interference sources as well as enabling trouble-free connection to unbalanced equipment.
It is often said that THD+N figures do not always correlate well with the perception of sound quality and this is true - partly because the traditional measures of THD+N or SINAD expressed as RMS figures are rather a broad measure. With this in mind, we have taken great care to make sure that not only is the Orpheus noise and distortion spectrum beyond reproach, but the RMS distortion result measures up to the state of the art.
Standards compliant FireWire interface
Increasing standardisation is leading to more choice for those wishing to "mix and match" editing and production software with various audio interfaces. Prism Sound has taken on board the increasing importance of native processing power for professional users and the fact that software products for standard PC and Mac platforms have been greatly enhanced in recent years.
Prism Sound is probably best known for A/D and D/A converters, not least the ADA-8XR, which already provides a solution for those needing a FireWire interface. However, the flexibility and versatility of the ADA-8XR comes with a higher price tag, reflecting the fact that no other interface provides such exceptional audio performance or can work directly into Pro Tools Core/Processing cards, as well as running a concurrent DSD processor or FireWire interface.
The solution was to create a unit that is dedicated as a FireWire interface and is compatible with Windows PC and Apple Mac computers.
Orpheus is easy to connect to your computer and to your outboard gear. For Windows XP or Vista users ASIO and WDM drivers are provided, while for Mac OS X 10.4 or later, Orpheus interfaces directly to Core Audio. For both Mac and PC platforms, there is a controller application to configure the unit and control its built-in mixer and other functions. Aside from the monitor and headphone level controls, everything else is operated solely from the Orpheus controller application. The controller software opens on-screen as a separate panel alongside your existing editing software.
Orpheus Flexible Inputs and Outputs
Our customers told us that, along with FireWire connectivity, many professional users wanted a highly integrated solution with instrument and microphone inputs, and line outputs that could be used for stereo or multi-channel monitoring and/or foldback to performers. Orpheus offers eight analogue recording channels, eight monitoring outputs, stereo digital input and output on a phono connector plus concurrent optical digital I/O ports that can interface to S/PDIF or ADAT data formats, giving Orpheus a maximum capability of 18 concurrent input and output channels plus stereo headphones.
Orpheus' eight analogue inputs support various capabilities. Orpheus has four really good mic amps with software-controlled gain in 1dB steps, individually-switchable phantom power - and very low noise and distortion. These inputs are auto-sensing, and support microphone and line input, with digitally-controlled mic gain in excess of 60dB. Two of these inputs also support direct injection (DI) instrument connections with quarter-inch jacks on the front panel. RIAA Equalization can be selected in the controller applet on channels 1 & 2 so that turntables can be connected for archiving or sampling applications. By selecting the input type (Mic or DI) , low- or high-impedance cartridges can be loaded with suitable termination impedances. All inputs have individually-selectable Prism Sound "Overkillers" built in, just as on the higher-priced ADA-8XR, to catch those fast transients. The Overkiller threshold automatically follows the operating line-up level selection (+4dBu or -10dBV). Overkillers are ideal for percussive sounds, particularly drums, where headroom can be a problem.
The co-axial digital I/O port can be switched in the Orpheus controller applet between S/PDIF and AES3 formats. This control changes the operating voltage and the Channel Status format and is complemented by two in-line adapter leads that provide external XLR connections for AES3 devices. Other connections include MIDI in and out and wordclock sync I/O.
Orpheus can also operate in a stand-alone mode using its ADAT or co-axial digital I/O connections. Once set up using the Orpheus controller applet, the unit can be disconnected from the host computer and used independently. Orpheus will retain its settings when powered down so even if it is switched off, Orpheus can be re-powered and stand-alone operation can continue with the automatically-stored settings.
Digital Mixer
Our customers also identified a need for a unit that could provide low latency foldback to performers, particularly when tracking and overdubbing. In answer to this need, Orpheus has a powerful built-in digital mixer that can be configured from the host computer to provide foldback feeds to performers, each with their own stereo mix of workstation playback and any of the inputs.
The question of latency in computer interfaces, especially USB and FireWire boxes, is an important one. Obviously there are situations where the round-trip latency needs to be really short, like in overdubbing. The problem is that even if the latency on the interface and in the driver is as short as it could ever be, a native DAW is busy with plug-ins and other software and buffer times are probably set long. The only answer is to provide local foldback mixing in the interface. This is not new, and other products feature it, but most local mixers in competitive products are just too basic. Orpheus provides 'console quality' local mixing - every output has its own independent mixer, with channel strips for all inputs and workstation feeds, complete with fader, pan/balance pot, solo and mute buttons, and full metering. Strips can be stereo or mono, and the mixes are dithered with filtered coefficients, just as in a top-end digital mixer.
There is a very small residual delay through the A/D and D/A conversion process in the foldback path, mostly from filters used for decimation and interpolation. However, with the low-latency Prism Sound DSP mixer, the worst-case delay through the A/D and D/A path is only 0.5ms and is significantly less at higher sampling rates. This is generally reckoned to be small enough not to be problematic.
Although the unit's outputs will mainly be used for monitoring or foldback, the fact that they are of such high quality makes them suitable for a range of other applications such as insertion points, analogue summing or stem-based mastering.
Orpheus Flexible Monitoring
Professional users demand more sophisticated monitoring capabilities and are getting used to surround sound with HDTV and DVD, so it is becoming important to support surround monitoring setups.
As well as wanting great analogue recording channels, the DAW user also needs top-quality monitoring. The eight analogue outputs on Orpheus allow monitor setups from multi-stereo up to 7.1 surround. Orpheus has a real volume knob which can be assigned to any or all of the analogue or digital outputs for use as a control room monitor control. There are two headphone amps, suitable for all types of headphones, each with its own independent volume control. As well as having its own workstation feed and mixer, the headphones can also be quickly switched across the other output pairs, which is handy for setting up.
Sample Rate Conversion and Noise Shaping
The digital output is equipped with the four Prism Sound SNS noise-shaping curves and includes Prism Sound's renowned synchronous sample-rate conversion, allowing outputs to various external devices at other sampling rates. The sample-rate converter can be used at the outputs as well as the inputs, so as well as dealing with unsynchronized or wrong-rate digital inputs, Orpheus can also generate, say, a live 44.1kHz output from a 96kHz session. Since Orpheus also includes the full suite of the famous Prism Sound 'SNS' noise shapers, you can also reduce to 16-bits at mastering-house quality.
Unsurpassed Jitter Rejection
In the 1990s Prism Sound pioneered testing of sampling and interface jitter and as a result our digital audio products deliver unsurpassed jitter rejection. Prism Sound digital audio products lock up fast and re-generate ultra-stable clock outputs. Another aspect of the traditional Prism Sound converter that is retained was the clocking - it's just as important as analogue-path considerations sound-wise. So whether it's providing a high-quality master clock for the rest of the room, or dealing with a jittery clock from outside, Orpheus is as rock-steady as its forbears.
Support
Over the years, Prism Sound's reputation for audio quality has been matched by its reputation for after-sales support and technical advice. Orpheus has the benefit of that support and customers have access to one of the best technical teams in the business.
The Last Word
We believe that Orpheus delivers exactly what our personal studio customers have asked for - all the performance of a Prism Sound product in a dedicated FireWire unit that handles line, microphone and instrument inputs with good foldback and monitoring capabilities, yet at a more accessible price tag. We are confident that customers using the new Orpheus converter will agree that it sounds as good as it looks.
Download Product Manual
The ProAc Studio 100 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.
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.
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.
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.
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.
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.
Specifications
Nominal Impedance: 8 ohms Recommended Amplifiers: 30 to 150 watts Frequency Response: 35hz to 30Khz Sensitivity: 88db linear for 1 watt at 1 meter Bass/Midrange Driver: 6 1/2" treated cone with special center pole plug. Tweeter: 1" (25mm) soft fabric dome with ferrofluid and rear loading. Mirror image offset. Crossover: Finest components on dedicated circuit board. Multi-strand oxygen-free copper cable throughout. Split for optional bi-wiring and bi-amping. Dimensions: 16" (406mm) high x 8" (203mm) wide x 10" (254mm) deep Weight: 24 lbs (11kg) /cabinet Mode: Stand-mounted Grille: Acoustically transparent crimplene Finish: Available in the following real wood veneers: Black Ash, Mahogany. ProAc Studio 100 Users
Bose Automotive Group Jack White Rich Costey John O'Mahony Brendan Benson Michael Marquart Peter Frampton University of Michigan Michael A Sapone Bob Ludwig (Gateway Mastering) Greg Calbi (Sterling Sound) Tony Maserati Rick Rubin Joe Gastwirt 311 Neil Diamond John Scofield Bill Frisell David Sanborn Genesis/The Farm Jean Marie Horvat Kevin Killen Michael Brauer Ron Saint Germain Husky Huskolds Troy Germano Studios Ryan Hewitt Craig Street Bob Ezrin Mike Campbell EastWest Studios Blackbird Studios Ryan Freeland Richard Dodd Ben Fowler Neil Dorfsman Metallica Red Hot Chili Peppers Avatar Right Track Looking Glass Bear Tracks Hit Factory Miami Chung King Village Recorder
Many manufacturers, including past Purple Audio prototypes, use linear
power supplies for their racks. Traditional wisdom holds that linear
power supplies are ideal for audio and generate less noise and
interference. In many cases, especially with poorly designed "off the
shelf" switching power supplies, this is true. However, in the case of a
10 slot series 500 rack, even a well designed linear power supply
generates a substantial magnetic hum field and far too much heat to
allow for internal mounting. This problem is caused in part by the wide
variation of line voltage across the world. External power supplies are
cumbersome and inconvenient and consideration still has to be taken as
to their placement in relation to other equipment. With the advancement
of switching power supplies it is now possible to have a switcher that
does not compromise audio performance and delivers or exceeds all
specified current needs. Purple Audio has spent years perfecting just
such a power supply. They have comparatively tested all of theirs, and
many of their competitors, modules at full bandwidth (10Hz - 500kHz) on
lab quality over built bench supplies to make sure that the Sweet Ten is
not compromising audio performance in any way. Simply put, your 500
series modules will perform at their highest level in a Purple Sweet Ten rack. Audio performance improvements are considerable.
Features
10 Slot 500 Module Rack Internal 150K switching supply +/- 16VDC 1.5A +48VDC 150ma Second Balanced Output Utilizing Edge Connector 3 & 6 Second Balanced Input Utilizing Edge Connector 7 & 9 Front Handles Locking Female XLR connectors Ninth slot capable of accepting the Purple Audio Moiyn (8x2) Mixer Module (specs to come) No PSU noise in adjacent slots Rear Ground connector Capable of putting out the 500 module current spec 1.3A from 90VAC to 240VAC 50/60hz CE, ROHS, & WEEE Marked
The Sweet Ten top, side, and back panels are finished in trivalent
yellow chromate cold rolled steel instead of painted steel for improved
grounding and shielding. Purple Audio's ground plane motherboards
provide additional grounding. Additionally, the rack motherboard has
.1uf high frequency bypass capacitors at each slot to improve filtering,
especially for modules with DC/DC converters. Functionally, the Sweet
Ten offers 2 inputs and 2 outputs per slot. Many of the Purple modules
already take advantage of second output and the Moiyn mixer module will
use the second input to allow for stereo balanced source injection.
Future uses of these connectors include fader loops on future modules as
well as a few surprises!
The Retro Doublewide 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. 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. 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. Features: Single channel tube compression in a two-slot 500 Series module Continuously variable input and output levels with fixed threshold Continuously variable attack and recovery time Single and double time-constants Fully complementary push-pull signal path Cinemag transformer balanced, fully-floating input and output Hard-wire bypass switch with gold-plated contacts Gold-plated edge card connections Stainless steel chassis construction Side-accessible meter zero trim Uses new-old-stock U.S. tubes Authentic U.S.-made Simpson gain reduction meter. Hand built and tested in the U.S. Specifications: 30dB of available gain reduction Signal to noise ratio greater than 76dB Flat frequency response within .5 dB from 20-20,000 Hz Total harmonic distortion 1% or less from 0-25dB gain reduction Input level –15 to +24 dBm 600 Ohm impedance Output level –20 to +12 dBm 600 Ohm impedance Power consumption 180 mA (6 watts) Tube complement 6BJ6 x 4
The Royer Labs R-121 is a thoroughly modern ribbon microphone designed for today's studio and live environments, exhibiting a flat frequency response and well balanced, panoramic sound field. Its ability to withstand 135 dB SPL @ 30 Hz makes it ideal for applications that were previously considered off limits to ribbon microphones. The R-121 was developed through relatively recent advancements in magnetics, materials and mechanical construction techniques. The ribbon element's smooth frequency response and phase linearity, coupled with sensitivity levels surpassing those of "classic" ribbons, make this microphone an ideal choice for digital recording. The R121 delivers a consistent, natural acoustical performance with stunning realism. Widely used on electric guitars, brass instruments and drums, the R-121's figure-8 pattern also conveys superb ambiance and depth when used for room miking applications, orchestral and choral recordings. The heart of the R 121 is its low mass, 2.5-micron, pure (99.99%) aluminum ribbon element. The R-121’s unique, patented offset-ribbon transducer assembly incorporates rare earth Neodymium magnets in a specially designed flux-frame, which forms a powerful magnetic field while reducing unwanted stray magnetic radiation. The R-121’s offset-ribbon design positions the ribbon element towards the front of the microphone body, which allows for higher SPL handling on the front (logo) side and the option of a brighter response when recording lower SPL sound sources on the back side (3 feet and closer; phase reversed in this position). Features
High SPL capabilities for electric guitar and percussion
No internal active electronics to overload or produce distortion up to maximum SPL rating
Extremely low residual noise
Ribbon element unaffected by heat or humidity
Absence of high frequency peaks, "ringing" and phase shifts
Equal sensitivity from front or back of element
Gold plated XLR contacts Recommended Applications
Close miking
Electric & acoustic guitar
Drum overheads, kick drum (see manual for position), room miking
Percussion instruments
Brass, horn sections
Strings - solo & sections
Acoustic piano
Vocals
Live events – recording and sound reinforcement
With input and output transformers designed and implemented by Mr. Rupert Neve, the high-voltage 72V topology found in the Rupert Neve Designs Master Buss Processor (MBP) 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. The Compressor 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. 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). The Limiter 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. 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. 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. The Stereo Field Editor The stereo field editor on the MBP takes traditional M-S techniques to new heights with width, depth and corresponding bandpass filters. 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. 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. 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. 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.
The sE Rupert Neve RN17 Stereo Pair contains two factory-matched sE RN17, two factory-matched cardioid capsules, two custom shock mounts and a premium flight case with an additional eight canisters to hold the optional supercardioid, omni, hypercardioid and figure-eight capsules.
Featuring extended high frequency performance, coupled with a warm,
deep low end, this mic is simply revolutionary. For all acoustic
instruments, drums overheads, orchestral recording and many other
unique applications, the RN17 is the new gold standard!
Includes :
2x matched sE RN17 microphones 2x matched cardioid capsules 2x shock mounts Premium flight case
The Shadow Hills Mastering Compressor has extraordinary functionality, and also provides mastering grade compression and limiting for tracking and mixing. Essentially it is two compressors in series, per channel, that can act in stereo, or dual mono. First comes our mastering grade electro optical compressor, followed by our discrete Class-A compressor/limiter. Both feed our switchable custom output transformers, Nickel, Iron, and Steel. There is enough gain in each section to overdrive the hottest tape, or clip your converters, if you're into that sort of thing. The input stage is transformer balanced, followed by our fully discrete optical compressor. This section utilizes the same T4B optical attenuator as the LA2A and LA3A, but is optimized for mastering. The second stage is our discrete Class-A VCA compressor. There are no electrolytic caps or IC chips in the audio path. There are six ratios: 1.2 to 1, 2 to 1, 3 to 1, 4 to 1, 6 to 1, and 10 to 1. There is six attack settings: .1, .5, 1, 5, 10, and 30 milliseconds. And there are six release times selectable: .1, .25, .5,8, 1.2, and Auto. There is an insertable filter in the side chain to limit pumping with bass heavy material. Lastly the signal goes through our new Shadow Hills custom transformer-switching network. The first position is Nickel, which is our custom version of a famous L. A. custom console. Next is Iron. In this mode the signal goes through our op-amp and into a Class-A output stage then to our custom Iron transformers, the last transformer position is Steel. These selections allow you to choose between different output transformers that are in effect: clean, colored, and dirty, respectively. The optical section has a hard-wired bypass that completely removes it from the signal path, and the Class-A VCA compressor is also hardwire bypassable, completely removing it from the chain. It is possible to independently bypass both compression sections and have your signal go through the input transformer and transformer selector only. There is also a hardwire bypass for the entire compressor, effectively a strait wire in and out of the box. So besides being a mastering grade, and highly functional buss compressor, it serves as an excellent tracking compressor. For instance, you might call up the chain of the optical compressor, then to the Class-A output and Iron transformer, for vocals, or select just the Class-A VCA compressor with a 10:1 plus the Steel transformer for crushing some room mics, and on and on. The metering can reflect, optical gain reduction, discrete gain reduction, or output level. The Magic eye tube follows the output meter to act as a peak meter. All 17 of the rotary switches are detented. The front panel is engraved. The knobs are bake-lite.Read the Shadow Hills Mastering Compressor Review, Resolution, October 2010
Results 1-48 of 60 1 2