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WHAT IS A DEMO ITEM?
A demo item simply means we've had it out of the box to demonstrate to another customer on site or sent to their premises to hear in their own studio for a few days, therefore it is no longer factory sealed. All demo items we sell come with a full manufacturer's warranty and have been evaluated by our factory authorized technical staff to ensure your "demo" purchase is in perfect working order before we ship it to you. No need to worry as you will have peace of mind knowing that our return policy applies to all demo items.
HOW DO I ORDER A DEMO ITEM?
To place an order or inquire about a product, simply click on the item and then click "add to cart". If the item says "Call or email for best price" please contact our Sales Team at 1-877-653-1184 and press 3 or email us.
If outside the United States please call 001-248-591-9276
AVAILABILITY
All "Demo Deals" items are offered on a first-come, first-served basis. Offer is limited by availability and once a demo item has sold out, the listed price is no longer available. Our demo deal list is updated weekly. For up to date and immediate stock status please call and speak with a sales rep.
The API 550 discrete four-band EQ played a major part in the history of music recording in the USA during the 60's and 70's. Originally conceived for use in API's consoles, this latest 550 "B" version is a continuation of the original 1967 550 through the 550A with one additional filter band and several new frequencies. Incorporating API's exclusive circuitry and proprietary components (such as the legendary API 2520 op-amp), the 550B artfully blends the past with the present. So many hit records still depend on the unique 550 sound, that the 550B will be an invaluable tool you will turn to again and again. The API 550B—unlike any other you will ever use. Many EQs today offer a huge assortment of complex features, but the API 550B provides exactly the right number of controls to the professional engineer. Deceptively simple at first glance, the 550B's four EQ bands are overlapped significantly to aid in dual roles as problem solver and sweetening device. Each band offers 7 switchable filter frequencies spanning 4 to 5 octaves, selected through experience by a "who's who" list of the industry's most proficient engineers. This re-issue design has been taken from the original blueprints and spec control drawings from the API archives.
Making use of "API Proportional-Q", a design introduced by API in the 60's, the 550B intuitively widens the filter bandwidth at minimal settings and narrows it at higher settings—without the need for additional bandwidth controls. This feature minimizes the "phase-shift" sound found in many equalizers. The reciprocal nature of the 550B enables the user to "undo" what has been done previously.
The benefits of the API 550B are most obvious to those who work with EQ on a continuous basis. The 550B's ability to affect or reverse tonal modifications is perhaps the most important feature. If major tonal restructuring is required, the extraordinary headroom made possible with API's 2520 op-amp offers predictable and gentle "analog" performance under duress. With a surprisingly wide range of tonal variations, you can do no better than the API 550B.
Features
4 bands of famous-sounding equalization Each band offers 7 API-selected frequency centers Reciprocal and repeatable filtering 12 dB of boost/cut per band EQ bands 1 and 4 offer shelf/peak switching "Proportional Q" narrows filter Q at extremes Traditional API fully discrete circuit design High headroom: +30 dB clip level Re-issue of 1967 API 550 EQ with an extra band 2 Year Warranty (labor) 5 Year Warranty (parts)
We double the standard API one year warranty (parts and labor) on this item.
The ADAM S3A is one of ADAM’s most famous models and is found in many of the most prestigious recording studios all over the world. The successor of this world famous model is the S3X-H, a monitor second to none within its size and price point.
Unlike the S3A, the new X-model incorporates a 4” HexaCone™-midrange as part of the redesigned construction to further improve the musically most important midrange. This new design provides more detail with improved radiation characteristics while presenting a musically coherent sound.
To produce deep, tight bass both woofers now work within the same frequency range and radiate symmetrically. This allows for a problem free installation in any multi-channel application.
Optional digital input
All SX-models (except the S1X) can be optionally retrofitted with a 24 bit/192 kHz D/A Converter featuring an AES/EBU (XLR) and a SPDIF (RCA) inputs as well as a R/L/Mono switch.
The front panel
Six controls can be found on the new front panel. Input sensitivity is handled by two controls. One control allows for coarse settings (-20dB to +8dB), while the second is used for fine tuning in 0.5dB steps, ranging from -1.5dB to 2dB.
There are two controls for the lower frequencies. One is an equalizer at 80 Hz that boosts bass response sometimes desirable in popular music. The second is a shelf filter that allows the engineer to alter the frequencies below 150 Hz.
High frequencies can also be adjusted by two different controls. The first alters the high frequency volume (-2dB to +2dB), while the second is a shelf filter for frequencies above 6 kHz. Both the high and low shelf filters can raise or lower ±4 dB progressively within their frequency band.
The I/O Modules are the sound and soul of Symphony I/O, and represent 25
years of Apogee R&D and design excellence. These powerful Modules
deliver Apogee’s core competency, superior sounding digital audio
recording. With 5 available I/O Modules, you can choose from standard
configurations or customize the following: channel count, analog to
digital input, digital to analog output, mic preamps and digital I/O.
You compose your Symphony I/O.
Apogee Symphony I/O 16 Analog Out + 16 Optical In Module Features
Inputs
Optical: Supports ADAT, SMUX & SPDIF on 1 Toslink connectorADAT: 16 channels 44.1-48 kHz SMUX: 8 channels 88.2-96 kHz SPDIF: 4 channels up to 192 kHz Coax: Supports SPDIF on 1 RCA connector
Outputs
Analog OUT: 16 balanced outputs, 2 DSUB 25-pin connectorsOptical: Supports ADAT, SMUX & SPDIF on 1 Toslink connectorADAT: 16 channels 44.1-48 kHz SMUX: 8 channels 88.2-96 kHz SPDIF: 4 channels up to 192 kHz Coax: Supports SPDIF on 1 RCA connector
Avid 96 I/O Features
Supports up to 16 channels of high definition I/O
High-quality audio performance at a new, low price point
8 channels of 24-bit/96 kHz analog I/O (TRS) 8 channels of ADAT optical I/O (TOS-Link)
1 AES/EBU I/O pair (XLR)
1 S/PDIF I/O pair (RCA)
Expansion Port that allows for direct connection of another 96 I/O
Legacy Peripheral port that allows for connection of 888|24, 882|20, 1622, and 24-Bit ADAT Bridge I/Os
Avid Pro Tools|HD Accel PCIe
supercharges any Pro Tools|HD System. HD Accel cards result in higher
voice counts across all sample rates. Affords nearly twice the raw DSP
power as HD Process cards, not to mention additional Pro Tools 6.2
software optimizations, further enhancing the overall power of Pro
Tools|HD Accel systems.
Get the absolute highest audio quality and unprecedented I/O flexibility with Avid HD I/O ,
a high-performance Pro Tools HD Series audio interface that’s designed
to completely integrate with Pro Tools|HD systems. While its sleek look
resembles its prior generation interfaces, HD I/O has been completely
redesigned on the inside with high-quality electronics, giving you
pristine, best-in-class A/D and D/A conversion and the lowest possible
latency. Plus, with three configurations to choose from and a modular
design, you can expand and customize the interface for your connection
needs.
Features
Whether you work with music or sound for picture, HD I/O gives you
professional-grade audio quality, high performance conversion, and the
flexibility to integrate with a variety analog and digital gear, so you
can quickly adapt the interface to meet whatever need comes your way.
Pristine audio clarity and ultra low latency through
state-of-the-art A/D and D/A conversion, premium analog circuitry, and
meticulous attention to design Adapt the interface for your needs—choose from three configurations: Get a balance of analog and digital I/O with HD I/O 8x8x8 Maximize your analog I/O with HD I/O 16x16 Analog Go all digital with HD I/O 16x16 Digital Versatile digital I/O on all interfaces, with built-in sample rate conversion Expand or customize the I/O with analog or digital option cards Completely integrates with Pro Tools|HD for full input, output, and routing control Track hotter signals and smooth out sounds with Curv, a new built-in soft-knee limiter Keep things in sync with dedicated Word Clock and Loop Sync I/O Keep an eye on your levels easily through 32 4-segment metering LEDs Robust 2U rack-mountable chassis Connects to Pro Tools|HD through a DigiLink Mini connection (cable and adaptor included)
Hear What You’ve Been Missing
A sonic improvement over its audio interface predecessors, HD I/O
features premium A/D and D/A conversion, so you can achieve higher audio
fidelity, extensive dynamic range, super-low jitter, and the absolute
lowest possible latency in your sessions. With certain configurations,
you also get built-in sample rate conversion and a soft clip feature for
worry-free recording. And using Curv, a new built-in soft-knee limiter
that catches even the fastest transients, you can track hotter signals
and smooth out inconsistent levels from incoming sources.
The Connections You Need—Now and Later
HD I/O comes in three configurations, so you can choose the interface
that best suits your connection needs. What’s more, you can expand your
analog or digital I/O by simply installing an HD I/O option card into
any empty bay (HD I/O 16x16 Analog cannot be expanded, as all four bays
are occupied).
HD I/O 8x8x8
The perfect choice when you need a balance of analog and digital I/O for
your music or post-production projects, with the option to expand your
I/O.
High-quality analog I/O: 8 analog inputs (using DB25) 8 analog outputs (using DB25) Versatile digital I/O: 8 channels of AES/EBU I/O (supports 192 kHz single-wire and dual-wire) 8 channels of TDIF I/O 8 channels of ADAT I/O (supports S/MUX II and IV)
Additional digital I/O: 2 channels of AES/EBU I/O (supports 192 kHz single-wire) 2 channels of S/PDIF I/O 8 channels of ADAT I/O (supports S/MUX II and IV)
One empty bay for I/O expansion
HD I/O 16x16 Analog
The ideal choice for music productions—get the maximum complement of
analog inputs and outputs (can be customized by replacing a card, but
not expanded).
High-quality analog I/O: 16 analog inputs (using DB25) 16 analog outputs (using DB25)
Additional digital I/O: 2 channels of AES/EBU I/O (supports 192 kHz single-wire) 2 channels of S/PDIF I/O 8 channels of ADAT I/O (supports S/MUX II and IV)
HD I/O 16x16 Digital
The ultimate choice for post-production studios, and ideal if you need
to interface with a digital mixer, dubber, or other digital
infrastructure, and want the option to expand the I/O.
Versatile digital I/O: 16 channels of AES/EBU I/O (supports 192 kHz single-wire and dual-wire) 16 channels of TDIF I/O 16 channels of ADAT I/O (supports S/MUX II and IV)
Additional digital I/O: 2 channels of AES/EBU I/O (supports 192 kHz single-wire) 2 channels of S/PDIF I/O 8 channels of ADAT I/O (supports S/MUX II and IV)
Two empty bays for I/O expansion
The Ultimate in Interface Flexibility
As your needs grow or change, you can easily add more analog inputs,
analog outputs, or digital I/O by installing an option card. You can
install one option card in HD I/O 8x8x8 and up to two in HD I/O 16x16
Digital. While HD I/O 16x16 Analog comes with the maximum four cards,
you can customize your I/O by swapping out any card.
HD I/O AD Option: Adds 8 high-fidelity analog inputs (using DB25)HD I/O DA Option: Adds 8 high-fidelity analog outputs (using DB25)HD I/O Digital Option: Adds 8 channels each of AES/EBU (supports 192 kHz single-wire and dual-wire), TDIF, and ADAT (supports S/MUX II and IV) I/O
Work with Pro Tools and More
HD I/O and all other Pro Tools HD Series interfaces completely integrate
with Pro Tools|HD, so you get maximum performance, dependability, and
control of your inputs, outputs, and routing right from your Pro Tools
software interface. You’ll also get higher production quality and
greater workflow flexibility than with previous Pro Tools interfaces.
And if you want to use your interface with Pro Tools|HD and other DAW
software, you can do that too, as all Pro Tools|HD systems support Core
Audio and ASIO drivers.
The BAE 1073 modules are
authentic reproductions of vintage Neve™* 1073 console modules. They fit
racks for vintage Neve modules as well as fitting into vintage Neve A
Series consoles.
The BAE 1073 preamp yields a really fat sound. It seems to add
subharmonics to the low end which makes the bottom end sound huge. It
smooths the top end which gives the highs a silky sheen. As well the
1073 compresses transients a little bit. It is a smooth, fat, larger
than life sound.
The B.A.E 1073 equalizer is very broad-banded and musical. It has a
hi-pass filter with four frequencies- 50 Hz, 80 Hz, 160 Hz and 300 Hz.
The low frequency band is shelving and has four frequencies - 35 Hz, 60
Hz, 110 Hz and 220 Hz. The low end of this equalizer is so tight that
you can boost 220 Hz and not muddy up the sound. The mid band has six
frequencies - 360 Hz, 720 Hz, 1.6 kHz, 3.2 kHz, 4.8 kHz and 7.2 kHz. The
bandwidth is broad so you can boost or cut a lot without making things
sound processed or unnatural. The hi band is shelving and is fixed at 12
kHz. It is smooth and silky and a little goes a long way.
*NEVE IS EITHER A TRADEMARK OR REGISTERED TRADEMARK OF AMS-NEVE LTD, AND BAE IS NOT AFFILIATED WITH SUCH ENTITY.
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
Features
Strong steel chassis - not aluminum
Shielded wires on individual connectors for serviceability - no PCBs
XLR inputs and outputs
Remote power supply keeps power transformer interference away from modules
One year warranty
Built custom by Sound Anchor for Barefoot MM27 Stand Specs:
Base: 21"x17"x2" (LxWxH) Stand Height: 56" Tweeter Height Adjustment: 13" to 54.5" Maximum clearance Under Speaker (vertical): 44.25" Maximum Clearance Under Speaker (horizontal): 49" Speaker Mount Tilt: 5 degrees up or down Sound Anchors Stand Mounting Procedure 1. Remove the mounting bracket from the stand and place it on the side of the speaker cabinet. Line up the mounting holes, insert the bolts and finger tighten. If the bolts are too long and bottom out before the bracket is snug against the speaker cabinet, STOP. You may have the wrong size bolts. Contact Vintage King. 2. Secure the speaker mounting bracket to the stand at desired height. 3. Attach MM27 handles the side of cabinet opposite to the side the mounting bracket will attach to. Then set the speaker horizontally on the floor. * If your cabinet doesn't have mounting bolt holes on both sides, then set the speaker horizontally between two stacks of books straddling the subwoofer to prevent it from touching the floor. 4. Tilt the stand down and line up the mounting bracket holes with the cabinet bolt holes. 5. Insert bolts and tighten until the mounting bracket is snug against the speaker cabinet. Do not over tighten! If the bolts are too long and bottom out before the bracket is snug, STOP. You may have the wrong size bolts. Contact Vintage King. * A good way to ensure you do not over tighten the bolts is to put the long end of the Allen wrench into the bolt and turn it using the short end. Tighten the bolt using no more force than you can apply to the short end of the wrench while holding it in the tips of your thumb and index finger. 6. Carefully tilt the entire assembly (speaker and stand) back up so it stands vertically.
The Benchmark DAC1 USB (USB DAC)
is a reference-quality, 2-channel 192-kHz 24-bit digital-to-analog
audio converter featuring Benchmark's Advanced USB Audio™ technology,
UltraLock™ clock system, and HPA2™ headphone amplifier.
Engage the Music with the DAC1 and DAC1
USB
The DAC1 USB is a mastering-quality, 2-channel 192-kHz 24-bit
digital-to-analog audio converter. For more then 10 years, engineers and
audiophiles have come to rely on Benchmark for critical digital audio
conversion. They consistently find themselves engaged in the music in a
way that is rarely achieved with digital converters.
Benchmark has achieved unprecedented sonic purity in their converters
through intelligent engineering, eliminating digital distortion
artifacts, and maintaining a pristine analog delivery path. The DAC1 USB
continues this tradition of excellence.
Preserving the Classic DAC1 Heritage
The pristine audio path of the award-winning Classic DAC1 has made it
the 'Benchmark' of standalone D-to-A converters. The DAC1 USB preserves
the exact topology of the audio path of the Classic DAC1, while adding
some of the most frequently requested features.
The DAC1 and DAC1 USB both include Benchmark's jitter-immune UltraLock™
clock system and Benchmark's 0-ohm HPA2™ headphone amplifier. The DAC1
USB features Benchmarks AdvancedUSB technology to ensure
bit-transparent, high-resolution playback directly from your computer.
The DAC1 USB also includes a programmable mute function that
automatically mutes the main outputs upon insertion of a headphone plug.
Other features that are exclusive to the DAC1 USB include a
programmable headphone gain range, an automatic 'Standby' feature, and a
high-current output stage designed to drive even the most difficult
loads.
Jitter-Immune UltraLock™
Digital interconnect cables, electro-magnetic interference, and many
other variables introduce jitter into the digital audio. Jitter presents
a major problem to most D-to-A converters. Jitter is a type of clock
error that, if not properly addressed, can cause the D-to-A to misfire.
The result of these misfires is a non-musical, digital distortion.
Many modern (and expensive) converters suffer from severe jitter-induced
distortion. Jitter is NOT a problem for the DAC1 and DAC1 USB, which
achieve jitter immunity by utilizing the proprietary UltraLock™
clock-recovery system.
With the UltraLock™ clock-recovery system, the digital-to-analog
conversion-clock in the DAC1 and DAC1 USB is totally isolated from the
clock of the digital audio input. This clock-recovery topology
outperforms even the most well-designed two-stage PLL designs. Using
state-of-the-art Audio Precision testing equipment, no jitter-induced
artifacts can be detected with the DAC1 or the DAC1 USB. Any signal that
can be decoded by the USB or AES/EBU receivers will be reproduced
without the addition of any measurable jitter artifacts. The bottom line
is this: Benchmark converters will consistently and faithfully deliver
truthful audio with no jitter-induced artifacts, no matter what
variables are present.
AdvancedUSB Audio Technology
The USB interface of the DAC1 USB features Benchmark's unique
AdvancedUSB technology. This USB interface is completely
bit-transparent. In other words, every sample of digital audio is
transferred from the computer to the DAC1 USB without modification. The
DAC1 USB is natively compatible with Windows 7/Vista/XP/2000 and Mac OS
X. As a native device, there are no drivers or other invasive software
required to install or configure. The DAC1 USB is a true plug-and-play
DAC USB, and is designed to begin playback immediately after the unit is
connected to a USB port for the first time.
Unlike previous native USB audio interfaces, Benchmark's AdvancedUSB
Audio technology supports 24-bit audio at sample-rates up to 96 kHz. It
is very important to have a 24-bit USB path, even for 16-bit playback,
because certain computer-related functions can increase 16-bit audio to
24 bits. If a 16-bit USB DAC is used, 8 bits of every digital audio
sample would be truncated. Truncating digital audio causes severe,
non-musical digital distortion. Benchmark's AdvancedUSB interface
ensures a full 24-bit digital audio path to avoid truncation distortion.
HPA2™ Headphone Amplifier
The DAC1 and DAC1 USB feature the HPA2™ - Benchmark's signature
high-current, 0-Ohm headphone amplifier. The HPA2™ is arguably the
ultimate reference headphone amplifier. It will deliver the full rated
performance of the DAC1 and DAC1 USB to the headphones. The HPA2™
maintains less than 0.0003% THD+N under full load. The DAC1 and the DAC1
USB both have two 1/4" headphone jacks on the front panel. The
performance of the HPA2™ remains consistent even when two headphones are
being driven simultaneously.
What is it? The higher end BOCK AUDIO 251 (formerly Soundelux E251C) is treasured by users for its extraordinary up close cardioid performance on premier voices and its distance performance on piano, ensembles and orchestras. Many of our customers have asked for a vocal mic only version at reduced cost and this drove development of the BOCK 151. The 151 is a microphone designed to sound as close as possible to the flagship BOCK 251 up close but cost 30% less. What makes it different? The BOCK 151 Microphone uses a proprietary BOCK hand made German capsule, a proprietary BOCK hand wound vintage style audio transformer, a NOS tube in the mic amplifier and a BOCK hand built custom outboard power supply. The BOCK 151 adds a unique "brite/normal" switch which delivers two sounds in one mic. “Bright” allows unfiltered extended high frequency response that defines a modern high end condenser; “normal” offers a less bright, more vintage style high frequency response more forgiving to sibilant singers and or voices that need a darker treatment. The BOCK 151 offers enduring qualities impossible to fully describe in writing. While Chinese condensers try to impress with lots of top and bottom end on first listen, they lack midrange so critical in defining a great vocal. The BOCK 151 built-in midrange presence curve, so important in “cutting through” a dense mix, is impossible to obtain through EQ. Time has proven this vocal sound must come from the mic itself. The unique BOCK AUDIO "brite/normal" switch adds long term versatility. The high frequency response is extended but never gets harsh or gritty. The BOCK 151 Proximity Effect (bass gets stronger with closeness) is naturally powerful without being too boomy or bottom heavy. Midrange is smooth and does not require post EQ. Best of all, the BOCK 151 has a way of lifting the singer’s vocal to a higher level when they hear themselves for the first time on this great microphone. The BOCK 151 is perhaps the biggest bargain in lead vocal microphones available on the market. How do I use it? The BOCK 151 primary use will undoubtedly be female and male vocals. Use a pop filter and get up close for that intimate “big vocal” sound. A few inches away, the proximity effect recedes and the mic gets flatter in response. At distance, the BOCK 151 is amazing on instruments and ensembles. In front of a drum set it sounds like you are standing there; on stringed instruments it captures the transients and complex harmonic structure that spell the difference between good and great. Features: Cardioid-only version of the renowned Elux 251 Time-proven tube and transformer circuitry - all electronic components are point-to-point wired Each individual component in every microphone has been pre-screened for its sonic quality Power supply is true to the original, unregulated supply Excellent bass simultaneously occurring with an open, airy top end Fulfills the quest for high frequency response that is not only present, but is also natural Allows vocals to sound naturally dominant without being overbearing No midrange overemphasis and buildup when stacking vocal tracks No honkyness or grunge Provides liquid tone, even when recorded digitally Sounds great in every mixing studio and on every stereo system, before and after mastering System Includes 6 pin Tuchel cable, AC cable, wood box, shock mount, and shipping system. System Includes 6 pin Tuchel cable, AC cable, wood box, shock mount, and shipping system.
The BOCK 195 (formerly Soundelux U195 FET) is custom built, hand made cardioid patterned, phantom powered, large diaphragm FET microphone. In designing the 195, David Bock set out to create an extreme value in studio mics that could satisfy a wide range of demands for highly experienced engineers that own many more expensive microphones, yet need additional mics within a budget. It delivers superb results on lead vocal tracks as well as on instrument and guitar amp tracks, providing remarkable fullness and presence with exceptionally low background noise. What makes it different? Unique to the BOCK 195 is the switches that enable you to customize the sound for the application at hand. The combinations of tone switches allows for unmatched versatility in a professional studio mic. Three different switches are offered: a "Mode" switch, a “PAD” switch and a “LOW CUT” switch. The Mode switch offers two modes of operation: FAT and NORM. FAT is a low end boost between 10Hz and 400Hz for that distinctive sound of older tube classic cardioid condensers. NORM is extended flat response with slightly elevated high end typical of FET condensers. The 195 PAD reduces the mic sensitivity by 10dB for louder sources such as a guitar amp. The LOW CUT reduces frequency response by 10dB at 20 Hz to clean up the bottom end of vocals, bottom heavy source or a boomy room. The U195’s unique amplifier circuitry is key to why a 195 sounds so different in use compared to other FET mics. It offers superior response to wider dynamics while avoiding the typical transistor low-level high-order harmonic distortion commonly found in other FET mics. The BOCK 195 also contains a unique to the industry output transformer—many times the size of competitors. Audio professionals know a large and well designed transformer is the key to wonderful analog sound. Our proprietary transformer design takes up half the room in the mic body! We did this not just to prove that bigger is better, but to increase the low frequency headroom beyond that of other transformer coupled mics. How do I use it? The 195 accepts a broad spectrum of acoustic input levels, from soft vocals to heavy drums, yet also offers extended bandwidth response. The versatility afforded by the switches make the BOCK 195 an ideal choice for recording voice, percussion (tom-toms, in particular), and acoustic guitar. High SPL tolerance allows use of the 195 in front of electrical guitar amplifiers, where the FAT Mode response adds extra oomph to the guitar tracks. Many engineers will find uses for the 195 limited only by their imaginations and ever-changing session demands. One thing is assured: you will find your BOCK 195 to be one of your most useful mics for many years to come!
The essence control circuit design is based on the Buzz Audio SOC-1.1 Stereo Optical Compressor but the main audio path utilises our BE40 and BE50 True Class A amplifiers along with high spec Lundahl input and output transformers. The input transformer is directly coupled to the Buzz proprietary Opto gain reduction element (completely passive input), amplified and then balanced via the output transformer - a very simple but gutsy signal path! The essence control circuitry utilizes the Differential Drive Side Chain (DDSC) topology that was developed for the SOC-1.1. This system, which processes positive and negative audio waveform in seperate paths, provides the incredibly musical characteristic you will hear with this unit. The DDSC design also provides for a much faster attack time compared to most other LDR based optical compressors. Simple side chain equalizers have been provided to allow the user to adjust the sensitivity of the compressor at high and low frequencies. In addition, a side chain insert point is provided for more complex EQ and compressor "keying". The essence consumes 2 slots within a 500 Series rack. The input/output (I/O) connectors associated with the first slot carry the main audio path. The I/O connectors associated with the second slot are used for a side chain insert point (send and return). The insert can be switched in/out of circuit and monitored via front panel switches. PLEASE NOTE - RACK COMPATIBILTY Each Buzz Audio Essence module uses 2 rack spaces and draws 120mA +/- 15-18 volts DC, as supplied by rack power supply. Most currently manufactured rack systems will easily power a full compliment of essence units, but some older rack systems may have insuffcient power, if in doubt, please check with your rack manufacturer. We recommend the API Lunchbox and VPR500 racks. Features API* 500VPR Series rack module format - plug in and play! Advanced analogue technology - not a clone or copy of products from the past. Powerful fully discrete hi-bias, wide bandwidth, differential Class A amplifiers. Passive transformer coupled input, transformer coupled output Side chain insert point Side chain monitor selector On board side chain EQ Unique Differential Drive Side Chain design May be linked to multiple essence units within one rack Switchable 10 segment VU meter NOTE: Due to some misinformation during the design of the potion and essence 500 series compressors, the card edge connection for the linking of our 500 Series compressors is incorrect. The potion and essence have PIN 11 designated as the link connection, when in actual fact the API* standard is PIN 6. This error can be easily remedied with a simple wire link soldered between PIN 6 and PIN 11 of the potion and/or essence edge connector. Click here for details. * API is a registered trademark of Automated Processes Inc
The Buzz Audio Potion is a True Class A FET based Compressor designed to fit the API* 500 VPR Series rack frames. The potion will suit those looking for a audio compressor with "attitude" and is particularly useful in rock, hip hop and modern music styles where a somewhat aggressive character adds to the energy of the track. Drums, electric guitars, vocals and the mix bus will all benefit from the potion's rich harmonic sound - potion - the harmonic energizer!
Although the potion uses a FET (Field Effect Transistor) as the gain reduction element (like a well known classic limiter), the control circuitry (or side chain) is a completely new Buzz Audio design with some real innovative techniques to achieve an original compression characteristic.
Included in this new circuit is an element we have nick named "Ready To Rock" (RTR). This circuit automatically alters the Release time setting of the compressor depending on the attack depth and prevents the compressor from over shooting and undesirable "pumping" when using faster Attack settings. A front panel LED indicates when this is function is active. (Actually, RTR stands for Release Time Reduction, but call it what you will!).
As you can see from the specifications, the FET itself produces a relatively large amount of harmonic distortion compared to Optical and VCA compressors, but this distortion is the sound that will add energy and richness to your tracks.
The potion is capable of of very fast attack (less than 50uS at F attack setting) and when set to 20:1 Ratio, we believe it approaches the performance of "look ahead" digital limiters within the constraints of the analogue domain.
Click here to hear the potion in action on a snare drum track.
The audio section of the potion consists of our unique True Class A BE40 and BE50 amplifiers coupled with Lundahl input and output transformers for a fast yet full sonic signature. The input signal is passively coupled directly to the FET element via the input transformer, amplified and balanced via the output transformer - a simple signal chain.
In addition to the Gain, Drive, Attack and Release controls, we have included a % Mix control which allows the user to mix the uncompressed (input) signal with the compressed (output) signal providing yet another means of obtaining an original sound on your tracks.
The potion consumes 2 slots within the 500 Series frame. The audio in/out connectors of the first slot carry the main signal path whilst the in/out connectors of the second slot are utilized as a side chain insert point where external equalization can be applied to alter the compressor sensitivity at different frequencies. This insert can also be used for compressor "keying". A switch on the front panel allows the user to monitor the side chain insert and switch it in and out of circuit. Two potions' can be linked for stereo (mix bus) operation.
Rack Compatibility
The essence power consumption of 120mA +/- 15-18 volts DC (as supplied by rack power supply) is higher than other 500 series modules, but because this demand is spread over two slots (60mA each) it should work fine in all available racks.
Features:
API* 500VPR Series rack module format - plug in and play!
Advanced analogue technology - not a clone or copy of products from the past.
Powerful fully discrete hi-bias, wide bandwidth, differential Class A amplifiers.
Passive transformer coupled input, transformer coupled output
Side chain insert point with monitor.
FET gain reduction element for harmonic character.
VERY fast attack.
Unique RTR Release Time Reduction circuit.
May be linked to another potion for stereo operation.
Switchable 10 segment VU meter
NOTE: Due to some misinformation during the design of the potion and essence 500 series compressors, the card edge connection for the linking of our 500 Series compressors is incorrect. The potion and essence have PIN 11 designated as the link connection, when in actual fact the API* standard is PIN 6. This error can be easily remedied with a simple wire link soldered between PIN 6 and PIN 11 of the potion and/or essence edge connector. Click here for details.
Before modern semiconductor technology, audio filters and equalizers were made with passive components - capacitors and wound coils of wire that formed inductors (or chokes). Audio frequency chokes were bulky and expensive so these days it's easier (and cheaper) to use an electronic circuit to simulate the inductors. However, we have discovered that the difference in sound between the real thing and a simulator is quite noticable. So with the tonic equalizer we have brought you the best of both worlds, utilizing both real chokes and electronic filters in perfect harmony.
The Buzz Audio Tonic offers 3 bands of equalisation with a versatile sweep midrange section and fixed two frequency high and low bands. The tonic also has a high pass filter utilizing another real choke. The signal path of the tonic is all True Class A discrete transistor with the output balanced via a Lundahl* LL1517 high spec transformer. The tonic is designed to fit into the popular API* 500 Series modular racks and requires a rack to function.
The Tonic high band is switchable bell or shelf and features two frequency settings of 5kHz and 11kHz. The low band is also switchable bell or shelf with two frequency setting at 120Hz and 60Hz with both bands providing +/- 15dB of boost and cut. Thanks to the use of passive choke based filters, users can expect a very tight sounding bottom end and sparkly clear highs when applying boost with these two bands.
The sweep "bell" midrange extends from 75Hz to 1500Hz (x1 mode) or 750Hz to 15000Hz (x10 mode). The midrange "Q" or bandwidth can be selected tight(ish) or broad and utilizes the Constant Amplitude Phase Shift (CAPS) filter circuit developed by Mr Steve Dove. +/-15dB of midrange boost and cut is available.
The Tonic 18dB/oct high pass filter is completely passive with two turn over frequencies (50Hz or 90Hz) selected by the user via jumpers on the circuit board. Completing the picture is the "OL" LED which illuminates when levels exceed +20dBu and the "BYP" switch which completely removes the tonic from the signal path.
Rack Compatibility
The Tonic uses 1 rack space and draws 120mA +/- 15-18 volts DC, as supplied by frame power supply. Power consumption is higher than most other 500 series modules and this may stress some rack power supplies depending on what other modules are fitted. Please check the following list to see if it will work OK in your rack.
API Lunchbox 6B - recommend only two tonic modules are fitted along with other modules.
API 500VPR with L200PS - no problems.
Atlas Pro Audio Revolver - no problems.
OSA Power Rack and Track Pack - no problems if you have the 3A external power supply.
B.A.E. 6 & 11 Slot Racks - not tested, check with B.A.E. for power capacity.
Features:
API* 500VPR Series rack module format - plug in and play!
Real choke based EQ offering superior sonics.
Advanced analogue technology - not a clone or copy of products from the past.
Fully discrete hi-bias wide bandwidth differential Class A amplifiers.
High performance Lundahl* output transformer.
What makes the Chandler Little Devil Preamp
so special? Well, several things. The Class A gain stage sounds like no
other design Chandler’s done. It’s big, yet open and full of life. Just
about any parameter of the sound is changeable including the ability to
change both the Feedback and Bias of the amplifier. The Feedback
control increases harmonics thousands of times over the range of the
control. The nature of clipping and distortion also changes and ranges
from soft Class A clipping to super soft gushy tube type. The sounds
range from a very colored and “over biased” tone to a gently colored and
“in your face” tone. Other features of the Devil Pre include High and
Low input impedance; transformer balanced line in for using the Devil as
a color box on mixes or tracks; low cut; bright switch and more! This
unit has no power-up issues with lunchboxes and rack systems.
Utilizing active balanced circuitry to preserve signal-to-noise ratio and avoid loading or signal loss, the Coleman Audio MS6A also has a separate headphone output with independent level control for direct monitoring and a mono switch to check phase compatibility. This model incorporates XLR connections for all inputs and outputs. Features:
Six balanced stereo inputs on XLR's
One balanced stereo outputs on XLR's
Internal trims for adjustable gain
Mono switch to check phase
Active balanced circuitry throughout
Headphone output with independent level control
The DAD AX24 Converter System is Musical, Correct, and Flexible AX24 allows you to config the number of line and Mic channels that you need at the sample rate currently used. Analog in and output channels can be expanded in pairs of two. Especially configurations like 2 or 4 analog input channels and 6 analog output channels are very suitable for mastering or multi-track productions where only few analog input channels are used simultaneously but monitoring in 5.1. is essential. Features Fully Modular - 224 different configurations Sample rates and I/O structure can be upgraded From 2 to 8 analog Line or Mic input channels From 2 to 8 analog output channels 24 bit PCM sample rates up to 384 kHz DXD sample-rate: 352.8 kHz DSD sample rates: 64fs & 128 fs Digital I/O: MADI, AES3, TDIF, and ProTools Digital DSD formats compatible with MADI and SDIF-3 Integrated Mic pre with analog gain Full remote controllable via RS-422, USB or Ethernet Mic pre controls can be represented on the channel strip of major consoles and work stations 24 bit PCM sample rates up to 384 kHz , DXD and DSD AX24 is based on the Digital Audio Denmark high quality multi bit converter and modulator design, using a 5 bit delta sigma modulator. AX24 offers A/D and D/A conversion in PCM up-tp 384 kHz, DSD at 64fs or 128fs and DXD at a sample-rate of 352.8 kHz. AX24 covers applications for music recording as well as mastering and monitoring in DSD. Mic pre with analog gain in 3 dB steps and digital gain with 0.25 dB accuracy is available for AX24 providing a sonically very transparent integrated Mic pre and ADC design. The Dynamic range is between 118 & 121 dB and the Mic pre equivalent noise floor is at -130 dB. AX24 offers an impressive dynamic range , making it the quietest and most transparent converter on the market. Full remote controllable via USB or Ethernet AX24 is ideally suited for stage box applications or machine room environment, due to the powerful remote control. The remote control interface is RS-422 and can be adapted to USB, Ethernet or other data formats providing a very flexible control structure. The DADman control utilities, provides various control interfaces for MAC and PC This enables the Mic pre controls can be represented on the channel strip of major consoles, and various DAW including ProTools HD Modular Interface structure All DAD-I/O modules are available for AX24 providing interfacing for the most used PCM formats up-to 96 kHz. Additionally the 8 channel AES3 module can interface 192 kHz and 384 kHz on a multiple wire interface.The MADI interface Supports all the sampling formats of the AX24, and is an extremely powerful and capable interface MADI which input and output all the PCM rates up-to 192 kHz as well as DSD and DXD. The interface can be configured to operate in a daisy chain for connecting more converters to the same MADI link. The MADI interface and the converter can be set to access any of the up-to 64 MADI channels in any configuration in blocks of 8 consecutive channels. All "unused" channels are linked trough the interface with very low latency of only a few samples. External synchronization to AES 11, Word Clock, Super clock or video (PAL, NTSC, SECAM) The built-in PLL circuit is able to correct for minor variations in the external sync clock signal, giving a very stable adaptation to the external synchronization signal. Sync alarm If a digital output is chosen with the DA source button, and the digital input for some reason is out of synchronization, the red Sync alarm LED will blink. This function is available to secure that the whole recording system is synchronized. The function can be turned off in the system menu.
The Drawmer D-Clock is a dual input/twenty output word clock distributor with a 16 character blue LCD display providing a reference measurement of the incoming sample rate. Both AES/EBU and BNC word clock inputs feature a zero latency loop-through output with switchable high impedance to maintain the correct level of the digital signal for onward distribution. The D-Clock displays incoming sample frequencies up to 768 kHz to an accuracy of 2ppm (parts per million) with two further modes - sample frequency with ±ppm error and sample frequency with % pull up/down for video users. Sixteen BNC clock outputs are situated on the rear panel with a further four BNC clock outputs on the front panel for quick patching to other digital devices. The word clock outputs can be derived from either a word clock input or alternatively an AES/EBU audio signal from which clock information can be stripped. In the latter case, one of the many mic pre’s with internal A/D conversion but no word clock output can be utilized as the master clock source for the whole digital studio. Features:Inputs: BNC with zero latency loop-through AES with with zero latency loop-through Outputs: BNC loop-through / AES loop-through 20 BNC clock outputs - 33½ impedance 3 sample rate measurement LCD modes accurate to 2ppm Universal voltage filtered mains power input
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 Electrodyne 511 is a classic
two-band, discrete transistor, reciprocal, active inductor-based
equalizer using late-’60s/early-’70s design technology. As with the 501,
the 511’s custom inductors and output transformer are made by
Electrodyne’s original provider to strict factory tolerances as small as
two percent. This affords a consistent EQ performance and repeatability
from channel to channel that was simply not possible in the 1960s.
Smooth performance and EQ response from minimum to maximum gain at all
frequencies provides unusually broad sonic and tonal options not
experienced since the 1970s.
The module’s faceplate sports twin large rotary EQ controls each
offering ±12dB of boost and cut with four selectable frequencies per
band (LF: 40, 100, 250 and 500Hz / HF: 1.5, 3, 5 and 10kHz). Shelving is
available on all frequencies, with peaking offered at 250 and 500Hz in
the LF band and 1.5, 3 and 5kHz in the HF band. An EQ in/out switch with
accompanying LED rounds out the front panel feature set.Specifications Input: Active balanced discrete transistor impedance converter. Output impedance: 150 ohms +/-12 db gain range: Equalize or Attenuate 4 frequencies selectable per band. LF: 40, 100, 250, 500hz. HF: 1.5k, 3k, 5k, 10k. Peak/Shelf function on each band. Shelving at all frequencies, Peaking at 250, 500, 1.5k, 3k and 5k. Distortion: 0.03%typical. Maximum output: +30db at less than 0.8%thd. (+29.5db at 0.06%thd) Frequency response: +/- 0.25db, 10hz to 70khz typical. Signal to noise: -85db typ.
The elysia mpressor is a new tool for creative dynamics processing. On top of the tried and tested standard features, this creative compressor provides several special functions that produce fat and freaky sounds by employing a punchy control behavior, colorations full of character and extreme settings. Besides classic tracking applications, its favorite domains are groovy effect compression and creative sound design.
Each of both linkable channels offers a switchable Auto Fast function that allows very fast yet distortion-free compression without any artifacts. The Anti Log circuit generates explicit compression effects such as pumping and breathing. As the mpressor is a true feed forward design, it also allows negative ratios which result in the most ex-treme over-compression effects. The Niveau Filters can change the character of a track from subtle to striking with the possibility of a continuous frequency shift. The Gain Reduction Limiter is a novel feature that lets the user choose the maximum amount of Gain Reduction he wants to draw from the unit, and this at the twist of a knob. The switchable external sidechain inputs make for further flexibility in terms of compression behavior.
The circuit design of the mpressor is completely based on discrete analog technology. The entire signal processing is implemented with single transistors in permanent class-A mode, and even the sidechain and power supply are composed of fully discrete circuits. An oversized power transformer, capsuled conductive plastic potentiometers, internal temper-ature stabilizers for critical components as well as special current feedback amplifiers are part of the technical finesse. The combination of these elements aims at the maximum reachable signal quality and guarantees clean and powerful sound characteristics even when extreme settings are used.
The creative compressor
Compressors are absolute standard tools in daily studio work, but depending on the specific task they need to be used in lots of different ways. The range goes from mild changes of level during mastering to limiting the dynamic range in recordings to extreme compression to shape sounds envelopes.
Hence the various topologies and large range of compressors in the market today. The mpressor is a typical recording and mixing compressor that can handle a wide array of jobs because of its premium audio quality and precise control characteristics. On top of that, it offers a number of special functions that enable the user to create completely new and very spacy sounds.
This makes the mpressor a truly creative compressor that especially convinces with its elaborate and practical innovations. Reducing options to the essential makes it possible to achieve outstanding results for any audio material at first go.
Control characteristics
The control characteristics of the mpressor offer a great spectrum of settings without generating any unwanted artifacts or noises. Even very fast attack and release settings can be used without any problems. Already known from the alpha compressor, the Auto Fast feature is a guarantee for very fast but undistorted gain reduction processes: If this function is activated, the attack time will automatically become shorter on input signals with fastly rising levels and intense changes of dynamics.
The novel Anti Log function generates an anti-logarithmic progress of the release curve. The release time will at first become slower when there is an ample increase of compression. As soon as the input signal returns to a noticeably lower amount, the release time will be shortened automatically. Depending on the other settings, compression will become much more audible than before as a result. This feature can be used to achieve BPM-dependent dynamic effects, manipulate the ambience parts of a reverb effect return, or process single sources such as vocals or bass.
Gain Reduction Limiter
Another specialty of the mpressor is its limiter for the control voltage. With it, one can set the maximum amount of gain reduction that will not be exceeded as long as the function is active. Normally the threshold and ratio parameters determine the amount of gain reduction which is adapted to the input signal in order to reduce the loudest parts as desired.
If the limiter is set to a maximum reduction value, threshold and ratio can have completely different settings without changing the maximum gain reduction. In this manner, very special variations of control characteristics can be achieved, e.g. applying a strong amount of compression while preserving the dynamics of loud passages at the same time.
Combining the Gain Reduction Limiter with the external sidechain transforms the mpressor into a true ducker. In this application, the music will automatically be turned down when a voice sets in (or just the other way round). Another option is to apply upward compression that only raises the volume of the quieter parts without changing the actual dynamics at all.
External sidechain
The mpressor provides a switchable external sidechain input per channel. Once it is activated, the compressor reacts to the fed in signals and opens up an enormous variety of additional potentials. A typical application is frequency-dependent compression, where an additional equalizer is used to give certain frequencies a stronger or weaker influence on the overall compression.
True ducking effects are also possible as well as very cool groovy sounds, e.g. controlling a stable synth pad with a drum beat fed into the sidechain. The combination of the Anti Log and Gain Reduction Limiter functions result in very unusual effects that sound like an inverted gate: the loud signals are reduced intensely while the quieter parts come in front more obviously.
Niveau Filter
In order to add flexible sonic stamps to compressed signal, there is a Niveau Filter placed after the compressor section which has been designed in the style of the audio EQ of the alpha compressor.
It also boosts the high frequencies beyond a selectable center frequency and cuts the low frequencies at the same time (or, depending on the individual setting, just the other way round). But according to the intended use of the mpressor, we have made the possible settings more extreme for more striking and dramatic effects. Only two controllers produce a great variety of different sounds.
These kinds of settings could not be generated by using standard parametric equalizers. The combination of this filter and the special control characteristics enable the mpressor to give a totally new character to the original signal which can vitally enhance a whole production.
100% discrete
Here is an old elysia saying: If you want to make it right, make it discrete! Of course this is also true of the mpressor, as the combination of very extreme control characteristics and an absolutely convincing quality of sound can only be achieved by using discrete circuitry. Therefore all signal stages were designed to use single transistors, resistors and capacitors, so each of the stages could be perfectly matched to its specific task as an important side effect.
Many parts of the circuitry do without negative feedback, and the complete unit works in constant class-A mode in order to ensure the best audio quality in any situation. Even the sidechain and the power supply are built with the exclusive use of single transistors, resulting in considerably lower noise compared to conventional ICs.
A look inside
The choice of components for the mpressor did not leave the slightest room for compromise – a great part of components which are used in the alpha compressor can also be found in the mpressor. All controllers use high-grade conductive plastic potentiometers. All switching functions are coupled via capsuled relays which are placed at the ideal places in the circuitry in order to keep the signal path reduced to the max.
A special input stage was trimmed to show a similar behavior to that of an input transformer. Furthermore, this stage distinguishes itself by a very good common mode rejection and an extra low capacity, preventing unwanted changes of the input signal effectively. A high-grade class-A output stage delivers adequate current to drive even long cable lines smoothly.
The power supply also benefits from the discrete design, as the voltage regulators produce such a low amount of noise that it can hardly be measured at all. A generously dimensioned toroidal power transformer provides enough current to charge the great number of quality electrolytic capacitors very quickly.
Certain critical components of the discrete circuitry are surrounded by special heating elements made of copper. This keeps important parameters constant and makes them independent from changes in temperature and operating time.
The Empirical Labs EL500 features ultra quiet power supply, instrument line input, flush-mount front panel.
The Gefell M200 cardioid is part
of a series of three modular condenser mics that include the M210
hypercardioid and the M270 omnidirectional. Each mic comes complete with
capsule head and microphone amplifier and capsules are interchangeable
for maximum flexibility. The combined use of ceramics and a hand
stretched gold evaporated Mylar (PE) capsule provides a rich, natural
tone with universal appeal for studio, broadcast, on-stage and location
sound.
Modular mic system Choice of 3 patterns Low noise output stage 16mm gold diaphragm
The M200 cardioid, M210 hypercardiod and M270 omnidirectional mics,
comprise a modular system of mics with interchangeable capsules. Each
microphone has an extended response for extremely accurate reproduction
throughout the audio range. Ideal for recording, broadcast and live
reinforcement where natural and colourless results are needed. With
deluxe wood case.
The Gefell M 300 is a compact
condenser microphone in studio quality. The frequency response of the
transmission factor 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. lt permits a lifelike recording also at various
sound incidence, e.g. announcer in synchronous studios, speaker, blower.
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 also under adverse acoustic
conditions, e.g. in churches.
A new low-noise integrated hybrid circuit and the transformer-less
circuit design guarantee an extremely wide dynamic range by 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 Gefell UM92.1S is a ‘vintage’ large-diaphragm multi-pattern tube microphone. It is equipped with the original M7 capsule. This pressure gradient transducer features a dual-membrane, gold evaporated PVC capsule that is still hand-made in the Gefell factory following the 75-year tradition. The Gefell UM92.1S is the choice for those that want the true sound and character of the original Neumann-Gefell UM57 with the added benefit of improved reliability and lower noise. The UM92 exhibits high sensitivity, excellent signal-to-noise ratio and it has the full sound, tube warmth, and character that are favored by singers and soloists. Design Features The Gefell UM92.1S is a multi-pattern microphone with switchable back-diaphragm voltage on the power supply for choice of figure-8, cardioid and omni-directional patterns. It is also available in a cardioid-only version called the M92.1S. As with all Gefell microphones, mechanical and electrical attributes have been optimized for performance. Careful attention to reducing ‘vortex’ around the mic and capsule is achieved by mounting the M7 capsule on triangulated pedestal. This has the benefit of directing unwanted reflections away from the capsule, reducing comb-filtering and phase disturbance. The smaller housing advantages the user with greater freedom to place the mic in tighter spots while also reducing particle pressure refraction or what is also known as the vortex around the mic. Following tradition, a natural presence rise provides added clarity and detail ‘on-axis’, while the ‘off-axis’ transition is both smooth and musical. The habitual low-frequency proximity effect provides a warm bottom end that is enhanced as the source is brought closer to the diaphragm. The UM92’s internal low-noise vacuum-tube amplifier features a ‘time proven’ design that employs a high-mu triode valve that has been selected for its warm sonic characteristics and the natural even-order overtones that are only possible from a tube. For long cable runs, a filament voltage stabilizer is built into the microphone that allows distances of up to 100 meters (328feet) without fluctuation. Using the UM92 The Gefell UM92 comes equipped with the EA92 deluxe elastic suspension to reduce vibration-borne noise from affecting the microphone. The suspension is secured to the base of the microphone by way of the screw-on, machined housing. This allows the UM92 to be positioned at any angle without worry. The UM92 comes with its own power supply called the UN920. The UN 920 power supply provides the polarization voltage for the capsule as well as DC voltage for the microphone amplifier. The UN920 features a selectable fuse housing that allows both 110V and 220V operation. A front-panel pattern switch allows the user to remotely select the pattern for figure-8, cardioid, or omnidirectional polar response. All connections are made on the rear panel. A 6-pin Tuchel connector and cable, which powers the microphone, is supplied. Output is balanced 3-pin XLR with pin-2 hot as per the AES standard. The 92.1S microphone is available in choice of multi-pattern (UM92.1S) or cardioid only (M92). May be custom ordered in a dark bronze finish. The complete system includes the UM92.1S or M92.1S mic, UN920.1 power supply, EA92 elastic suspension, C92.1 cable and W92 windscreen. Comes in it's own hand-crafted wood box.
During its 12 years in production, the Grace Design m201 mic preamplifier found its way into countless recording facilities around the world, gathering a considerable amount of critical acclaim along the way. Now our classic two channel mic preamplifier has been completely redesigned and transformed into the m201. Like the rest of our line, the new m201 delivers unmatched audio performance, with massive headroom, ultra-wide bandwidth and a very open, musical character. The signal path has been hot-rodded to be fully balanced from start to finish, resulting in a wider dynamic range, while new high-current output drivers enable even longer cable runs without signal loss. This means the m201 can translate a sonic picture of astonishing clarity and detail, which serves to capture the essential character of the music being recorded. We offer the m201 with our state of the art 24-bit/192kHz A/D converter option, from the factory or as a retrofit to existing units. With the A/D converter module installed, the microphone preamplifier section or the M+S matrix can be routed to the A/D converter input directly for standard recording applications. Or, with the additional analog line inputs on the A/D converter module, the mic preamplifier and A/D converter sections can be used separately, which allows for external processing to be inserted between the mic amp and the A/D converter. External synchronization to word clock or Digidesign LoopSync is available, and features our proprietary dual stage PPL with s-Lock for extremely low jitter and rock solid digital stability without the use of sample rate conversion. Our exclusive ribbon mic mode is standard, which shifts the gain range up 10dB while deactivating 48V phantom power, bypassing the decoupling capacitors and optimizing the input impedance. Each channel has an ‘input mode’ rotary switch which selects between a standard 48V phantom input, ribbon mic mode, a front panel DI input or optional DPA high voltage inputs (130V or 190V). Included is our newly designed M+S (mid side) decoder circuitry with a built-in width control. The front panel width control is wired with a precision 12 position rotary switch which provides a ratio range from 100% mid (mono) to 30% mid / 70% side. The ultra-precision summing and difference amplifiers feed a set of dedicated outputs, allowing simultaneous recording of discrete mid+side signals as well as the stereo matrix. The new front panel mounted HI-Z inputs are designed to accommodate a wide variety of high impedance input sources, making the m201 an excellent choice as a DI box which will flawlessly preserve the sound of any plugged-in instrument. Imagine the same resolution and detail as our microphone preamplifiers available for DI recording situations. Whatever the application, the remarkable sonic performance and functionality of the new m201 will help you make the finest recordings of your career.Grace Design m201
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 Great River MP-500NV preamp module for the '500' series rack is now available! The MP-500NV is a professional quality microphone preamplifier designed to re-create the vintage sound characteristics of the early 1970s large consoles. Modern components give the MP-500NV more clarity, punch and performance under normal use. In addition, features like metering before and after the major gain stages let the user drive it hard to greatly expand the range of sonic options. All microphones will benefit from the power of the MP-500NV preamplifier, including dynamic, condenser and especially ribbon types. The MP-500NV uses the same audio and metering circuit board found in the NV series preamplifiers. This unit is designed to fit into a "500 Series" rack using 2 available slots. It receives its power supply from the host rack. The unit is equipped with a HI-Z instrument jack that has its own input amplifier and was designed for DI applications. All specifications and performance details are identical to the NV series preamplifiers with the exception of size, weight, patch jack, -10dBv Out. 125ma current draw.
The Great River MixMaster 20 is a comprehensive analog centerpiece for your DAW based recording system. Tracking, editing, and mixing are all facilitated with this unit. The MixMaster 20 includes four transformer coupled high gain low noise microphone preamplifiers that work with any type of microphone, ribbon, dynamic or condenser. Polarity and phantom power switching is provided on the front panel. Line and instrument duties are handled by transformer coupled input bridging amplifiers with wide gain ranges, suitable for anything from keyboards to tape machines. Each of the four input channels has a balanced insert point as well as four auxiliary output mix sends. Each channel may also be fed to the main stereo mix bus. Mixing duties are handled by the four input channels plus the 16 transformer-less line input channels. Each of these channels has fully controllable level and pan settings to the stereo mix bus. Mix settings can be stored and recalled via the USB port and a GUI compatible with most digital recording programs is in development to make the MixMaster 20 an automated analog mixer. The mixing is done passively, with the summing gain provided by transformer coupled mixing amplifiers. The balanced output of this stage is available as a patch point for the insertion of compressors, equalizers or other signal processing equipment. The return is also transformer balanced and feeds the Penny & Giles stereo fader and output line amplifier, also transformer coupled. Control room monitoring functions are also provided. These include a versatile solo system, multiple speaker system selector and level control, mix replay monitoring and talkback. Talkback feeds the cut outputs and dims the main monitoring system to aid communications and minimize feedback.
The Gyraf Audio Gyratec XIV is
a true tube, all passive, stereo equalizer. It consist of five bands -
each with 11 switchable frequencies, variable “Q”, and boost/cut
selection. It also has an output level trim, and a “hard” relay bypass
control.
The two channels are linked - controlled with one set of knobs. The
“Q”s (filter sharpnes’es) are rather low when boosting, but higher at
cut - approaching a notch on extreme cut settings. This is both a
side-effect of the applied passive technology - as well as quite
desireable in situations where sound integrity is the paramount factor.
The frequencies are selected with special attention to controlability
of the midrange area, and with half an frequency-range overlap to the
adjacent bands.
This equalizer is a very extended version of the classic passive
inductor/capacitor type filters, the most famous being the “Pultec”
units. Even though the classic versions are usefull tools in most
situations, they are somewhat limited in use due to their simple
construction. Our engeineers wanted - as engeineers tend to do - more
of everything.
So we came up with the parallel-passive design, the G14.
It is a truely passive design - based on switching a set of inductors
and capacitors in and out of circuit - combined with an E88CC (6DJ8)
tube-based makeup gain/output stage. Not a single IC’s or semiconductor
in the signal path - this is a real tube unit.
The inputs and outputs are - off course - transformer balanced, for trouble-free interfacing to the outer world.
Input impedance - around 5K Ohms, transformer balanced Output impedance - ~1K Ohms, transformer balanced Max boost/cut - ~10-12dB, depending on “Q” setting Gain trim ~+6 - -20dB Each band individually hard-bypassable Channel tracking within ~1dB
Frequency markings are:
Low: 35-48-60-70-80-100-120-160-190-220-270HzLow mid: 180-230-280-330-400-430-490-550-590-650-750HzMid: 500-570-700-800-900-1K-1K1-1K3-1K5-1K8-2K1 HzHigh mid: 950-1K2-1K4-1K7-2K-2K5-3K-3K5-4K2-5K-6K3 HzHigh: 4K5-5K6-6K5-7K8-9K1-12K-14K-16K-18K-20K-22KHz
The Helios Type 69 Mic Pre/EQ Module Born Again! Re-manufactured classic 3 band eq-pre. Discrete. Switchable lo/mid. Fixed top. Hi-pass. Unlike anything you have ever used (especially if you are stuck in API-Neve land) you will find the Type 69 a unique departure. These hand built units have a Sowter input transformer based on the highly desirable Lustraphone (Olympic) transformer models. The mids are so funky and sweet---- guitars and snare drum's from heaven; woody and sweet! 700 to 2k to die for! Lo's are Pultec-ish. Hi's are sweet and musical. Extremely open Mic-Pre!In mic shootout's, these are as open as the king daddy Telefunken V76m. Transform your life. The Legendary Helios Electronics Ltd has resurrected with much anticipation and patience, the Olympic Type69 Series EQ as used by the Beatles, Rolling Stones, Bob Marley, Led Zeppelin and so on. Helios became known as the musician's choice of recording consoles. Brought in line with modern expectations, these units are built to military specifications with supervision of gear fanatic and music producer Tony Arnold. Endorsed by the late Dick Swettenham, of the original Helios Electronics Ltd, to maintain and service all of the original consoles, Tony later purchased the original company, including all of its circuits and parts. The Olympic Series is a classic 3-band EQ-Pre module, available in a dual mono 1U rack mount, mono vertical module or 500 Series module. The modern circuit is similar to the original, aside from one vast improvement, the build quality and reliability. The openness and clarity has been upheld with the addition of EQ frequency selections. Tony Arnold explains, "We asked many of the original users of the Helios Desks what they felt was missing, and could we make any improvements. Most wanted the 30 Hz added to the Lo-EQ. In order to meet todays CD standards, some wanted a tight high frequency around 16 k, which has been added without changing the original Helios sound whatsoever."
IK Multimedia ARC System
delivers the most advanced solution to acoustical problems for any
DAW-based studio. Combining a professional calibrated measurement
microphone, standalone software that captures sound information and
calculates proper room correction, and a multi-platform plug-in: this
technology will improve how your studio sounds forever.
ARC features the revolutionary Audyssey MultEQ® technology, which
measures acoustical information throughout the listening area in your
studio. It then combines this information to provide an accurate
representation of the room’s acoustical problems. The equalization
solution then corrects for both time and frequency response problems
more effectively and efficiently than any other room correction EQ on
the market. The result is a clear and reliable representation of your
mix. Regardless of the acoustical issues in your studio, what you are
recording, mixing or mastering becomes immediately clear and reliable
and your studio sound will improve forever.
Easy to use, step-by-step, room measurement software The first and only room correction system in a plug-in for most popular DAWs VST, RTAS, AU Includes a calibrated measurement microphone, measurement software and multi-platform correction plug-in Improves clarity, stereo imaging and frequency response, for faster, more reliable mixing Revolutionary Audyssey MultEQ® technology corrects frequency and
phase response not only for the engineer’s ‘sweet spot’, but also
multiple points in the room Step by step setup measurement wizard will have you up and running in minutes A convenient, unique, mobile correction solution for the traveling engineer Sonically 'treat' your room so you can finally trust the sound of your studio For MacOS®X, Windows®
With a warm and upfront low-end, punchy mid-range and open top end, the Inward Connections Magnum
will bring new life to your recording rig. Classic sounding warmth, but
with airiness and headroom for days. Magnum offers something different
than the old standbys. Extra large tone for your lunchbox!
Features
Circuitry features two VF600 all discrete amp blocks +48VDC phantom power switch Phase reverse switch -20dB pad switch Hi-Z low level line input 1/4" phone jack for guitars, keyboards and low level instruments 12-position rotary trim (5dB per step), 25dB to 80dB selectable range Three hi-pass filter selectable switches: 70Hz, 100Hz and 200Hz Output level rotary control Balanced input and output transformers Fits standard 500 series slot configuration, mechanically and electrically Powder coating in Gunmetal color
With a warm and upfront low-end, punchy mid-range and open top end, the Inward Connections Magnum VU will bring new life to your recording rig. Classic sounding warmth, but with airiness and headroom for days. Magnum offers something different than the old standbys. Extra large tone for your lunchbox! Features Circuitry features two VF600 all discrete amp blocks LED VU meter that monitors the output level +48VDC phantom power switch Phase reverse switch -20dB pad switch Hi-Z low level line input 1/4" phone jack for guitars, keyboards and low level instruments 12-position rotary trim (5dB per step), 25dB to 80dB selectable range 70hz hi-pass filter Output level rotary control Balanced input and output transformers Fits standard 500 series slot configuration, mechanically and electrically Powder coating in Gunmetal color
The Inward Connections Nitro is a
fully parametric 500 series 2-band EQ using the new VF-600 amp blocks
for adding warmth and power to your mix. Use for individual tracks where
you want to open the top end up, or to pull out problem frequencies.
Very simple and versatile design. Charge up your tracks with the Nitro!
Features
Low band frequency range 35Hz to 1KHz High band frequency range 560Hz to 16KHz Maximum boost/cut +/- 20dB Range of bandwidth adjustment 0.16 to 2.0 octave Bypass switch VF600 discrete amp blocks Differential balance input transformer less Balance output transformer Fits standard 500 series slot configuration mechanically and electrically Powder coating in gunmetal gray
Specifications
Input Impedance: >20K 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
Add warmth and euphonic tube character to your tracks with the venerable Inward Connections Vac Rac TSL-3. Famous amongst professionals for making a vocal sit in a mix without the need for vocal rides, it imparts a wonderful character without sounding "tubby." The elegant way in which it controls transients makes it a must have in the studio for vocals, bass, and buss duties.Features per channel:
Gain make-up level
Gain reduction control
Bypass switch
Output/Gain reduction switch mode
Zero adjust trim Master Link Switch Large 3"x3" VU meter
Illuminated VU meters
Tubes used in both input channels: Input stage - 6072A, Output stage - 12BH7A
Power Supply: Pure Class A high voltage tube design using 6BM8
Unit dimensions: 3 rack space. 5.25" Height, 19" Width, 11.5" Depth
Rugged steel chassis design
Classic vintage look
Specifications:
Gain Reduction: up to 40db
Input Impedance: >100k ohms
Frequency Response: +/- 0.5 dB @ 20Hz to 20KHz
Output Signal to Noise Ratio: -85dB
THD + Noise:
.015% @ 20Hz (0dB ref.)
.085% @ 20KHz (0dB ref.)
.15% @ 20Hz to 20KHz (+30dB ref.)
Maximum Input Level: +20dB
Maximum Output Level:
+20dBm @ 600 ohms
+34dBv @ 10K
Output Impedance: 600 ohms transformerless unbalanced output
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