GOLDEN AGE PRE73 MK2 WITH CARNHILL MOD 3 Microphone preamp

The PRE-73 Mk2 is the first one-channel low-cost preamp on the market that is using a fully discrete signal path with three separate transformers for the balanced mic input, the line input and the output.

The circuit is very similar to the classical preamp section of the Neve 1073 pre/eq module with a corresponding sound character that is warm, punchy, sweet and musical.
This classical sound has been heard on countless recordings through the years and it is a versatile sound that works very well on most sound sources and in most genres.

The PRE-73 Mk2 has a maximum gain of 80 dB on the mic input and the gain range on the Line input is -20 to + 10 dB. There is also a high-impedance instrument input.

The PRE-73 Mk2 is perfect for almost any signal source, no matter if it's a microphone with a high or low sensitivity (like a passive ribbon mic), a syntheziser or sound module or an electric guitar or bass.

The input impedance of the microphone input can be switched between 300 and 1200 ohms, which makes it possible to affect the tone of most microphones. There is also phase switch and an simple 4-step LED signal level meter.

The output level control makes it possible to make fine gain adjustments and also to overload the main gain stage for more character and then lower the signal to a suitable level before the output stage.

The Mk2 update consists of:
- An unbalanced insert jack (nominal level about 18 dBu) added on the back panel for inserting units like the coming GAP EQ-73, a three band eq.
-  LED meter levels adjusted.
- 600 Ohm termination resistor and jumper added to power supply board (accessible from the inside of the unit).
- Revised Gain switch, turning the know clockwise now increase the gain.
- Reduced noise at high gains by a change to the power supply grounding scheme.

Features
Superlative sound
Studio quality mic preamp
Total control - Input Gain, EQ contour, Output Gain,
Overload indicator
Multiple output connectors
XLR/jack combo input 
Switchable 48 V DC Phantom Power for condenser Mics
Compact, computer friendly package
Needs no external power supply
 
ARX is proud to introduce the e-pre - the ultimate microphone pre-amp for
hard disk recording, designed by audio professionals, not computer engineers.
The all-new e-pre is so much a part of your system that it fits right into a spare
5?" slot in your computer case! It's the perfect tool for all your day-to-day
recording requirements. No equipment racks to plug in, no masses of cabling to
connect - the e-pre is ready and waiting when you need it.
The e-pre is a studio quality microphone pre-amp using the same internal components
as high end mixing consoles, built into a compact, computer friendly package.
It has controls for:
Input Gain - fully variable from 10dB through to 60dB to suit all types of microphones,
plus switchable 48 V Phantom Power to enable condenser microphones to deliver
their best.
A Contour control for EQ sweetening, providing fast, easy access to a whole range
of useful EQ curves
Output level control - fully variable to match the e-pre's output to the input of the sound
card. From -10dB consumer cards through to +4dB for professional sound cards, the
e-pre can handle it.
Multiple output connectors match up with any kind of sound card, including mini jacks,
standard jack, RCA (phono) plugs, and even a connector to link to the internal CD ROM
audio input of your sound card (if fitted).
We've even fitted a 12 VDC input connector on the rear to enable the e-pre to run from an
external power supply if required (eg for use with notebook computers or stand-alone use).

Any way you look at it, the e-pre by ARX is the total professional microphone pre-amp for
all your hard disk recording needs

e-pre Benefits
Professional Balanced XLR mic input
True 48v phantom power
Runs into the Line input of ANY sound card - uses the card you've already paid for
High headroom = clean recordings
Smart EQ
Compatible with non-computer audio equipment
Professional performance and specs in a user friendly package
 

T E C H N I C A L  S P E C I F I C A T I O N S
Input Impedance
Mic 2 K Ohms Balanced; XLR type Pin 1 Audio Ground, Pin 2 +, Pin 3-
Line 25 K Ohms Balanced, ?" TRS jack Tip +, Ring , Sleeve Audio Ground
Phantom Power Voltage Switchable +48 V DC on XLR Pins 2 and 3
Max Input Level Mic +15 dB, Line +20 dB
Input Gain 10 dB Minimum, variable to 60 dB Maximum
Output Level -10 dB variable through 0 dB to +4 dB
Overload Indicator Measured at all Gain points throughout the circuitry
Output Noise -90 dB Unweighted, -97 dB 'A' weighted
(Measured with Gain nominal 20 dB, Output level -10 dB)
Frequency Response 20 - 20 KHz ± 0.25 dB
Distortion
100 Hz 0.0085%
1 KHz 0.008%
10 KHz 0.0083%
(Measured with Mic Gain 20 dB, Output 0 dB)
Power Requirements 12 V DC from either computer supply or external supply
(external power supply not included in e-pre package)
Size 148mmW x 42mmH x 150mm D (5?“W x 1?H x 6“D)
Weight Under 1 Kilo (2.2 lbs)
 

 The OctaMic II provides 8-Channel 192 kHz / 24 bit AD conversion with eight hi-class microphone and line pre-amplification channels, featuring a combination of sophisticated components and approved RME technology.

Lowest distortion, excellent signal to noise ratio and perfectly linear frequency response transmit and amplify the microphone signals truly unchanged.

The OctaMic II includes some significant enhancements compared to the OctaMic:

The balanced TRS inputs of the Neutrik Combo XLR jacks are phantom power-free and can be operated as real line inputs too. The improved design of the input circuits allows for a maximum input level of +21 dBu with a convenient gain range from 6 dB up to 60 dB.
Improved signal to noise ratio (SNR) ADC 107.5 dB
Improved THD, especially at higher gains
Optimised heat dissipation by a new and larger housing with improved convection
Internal wide range power supply with line filter, insensitive to voltage fluctuations
Super-stable, short circuit proof 48 V phantom power


Features

OctaMic II offers 8 balanced XLR mic / line inputs via Neutrik XLR/TRS combo jacks. Each channel contains switches for 48V phantom power, a low cut filter and phase reversal. Amplification can be set between 6 and 60 dB. LEDs for signal, clip, and activated phantom power give a complete overview on the unit's status. When the special Clip Hold mode is activated, any detected clip-state will cause the corresponding LED to flash once per second. With this, the user gets a long-term peak detection, and no longer needs to constantly watch the LEDs. At the same time momentary overloads are still displayed correctly.

Frontside switches include power on/off and output level, for a choice of -10 dBV, +4 dBu or Hi Gain (+19 dBu) as reference level. This unusual feature offers two advantages. First, the reference level can be easily switched to match any of RME's current interface devices, from HDSP 9632 through Multiface up to the renowned ADI-8 series converters. Second, the Signal to Noise ratio is optimized, and the Clip-LED will exactly match the ones of the ADI-8 (2 dB below 0 dBFS).

The balanced line level output signal is available at the back of the unit via 8 stereo TRS jacks. The specially developed, internal hi-performance switch mode power supply lets the OctaMic II operate in the range of 100V to 240V AC. It is short-circuit-proof, has an integrated line-filter, is fully regulated against voltage fluctuations, and suppresses mains interference.

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AD-Conversion

The 8-Channel AD-conversion of the OctaMic II operates at up to 192 kHz. The digitized signal is available simultaneously at the double ADAT output (S/MUX, up to 96 kHz), and at a DB-25 connector (4 AES/EBU outputs, up to 192 kHz). The digital part can be clocked internally (master) and externally via word clock, AES/EBU and SPDIF.

RME's outstanding SteadyClock(TM) ensures perfect AD-conversion, as jitter on the external sync-sources is nearly completely removed. All settings are done via DIP-switches on the back of the OctaMic II. Analog outputs and both digital outputs operate fully simultaneously. The choice of reference level affects the analog outputs only, the signal/clip indication and the AD-conversion will react only to the Gain-pots.


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Tech Specs

8 balanced XLR/TRS mic/line inputs

54 dB gain range

Input impedance: XLR 2 kOhm, TRS 5 kOhm

Analog input level: from -40 dBu up to +21 dBu

Maximum output level: +21 dBu

Output impedance: 75 Ohm

Output level switchable Hi Gain / +4 dBu / -10 dBV

Signal to noise ratio (SNR): 129 dB EIN @ 150 Ohm

THD:
Large frequency range (200 kHz) with special EMI input filtering

Frequency response -0.5 dB: 5 Hz - 200 kHz

Hi-pass filter: 80 Hz, 18 dB/oct.

Line Out: 1/4" TRS (6.3 mm stereo jack), servo-balanced

Phantom power: +48 Volt switchable per channel

Internal wide range switching power supply 100-240 Volt AC

Unbeatable price/performance ratio!

AD conversion

SNR: >110 dB(A)

Supported sample rates: 28 kHz - 200 kHz

THD:
Sync Sources: AES/EBU (also SPDIF coaxial), wordclock, internal

SteadyClock(TM) ensures best sound quality even with jittery external clocks


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Accessories

Digital breakout cable Pro, AES/EBU, D-sub25 to 4x XLR-3 female + 4x XLR-3 male, 1m/3m/6m (ALVA Audio)
Digital D-sub cable, AES/EBU, D-sub25 male to D-Sub25 male, 1m/3m/6m (ALVA Audio)
Optical ADAT Toslink cable 1m/2m/3m/5m/10m (ALVA Audio)
Word Clock cable, 1 x BNC male to 1 x BNC male, 1m/5m/10m (ALVA Audio)

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Add-On

BOB-32    Universal Breakout Box with 2 x 8 XLR to 2 D-sub connectors (supports TASCAM and YAMAHA format), flip-frame case

The PreSonus HP60 is the most flexible and advanced headphone amplifier system available for professional and project recording studios as well as live sound in-ear monitoring systems. Loaded with six independent,
loud and clear headphone amplifiers, the HP60 features dual stereo inputs as well as external input, for each
channel allowing you to mix between three stereo audio streams (mix A, mix B, and external input). Stereo output is also available on each channel to send line level headphone channel mixes to additional headphone amplifiers or monitor systems. Each channel features headphone level, mix control between A and B inputs, external input volume, mute and mono. The HP60 also features talkback via external XLR microphone input. Common set up of the HP60 in a recording session is to send the main control room mix to input A. Send the click track to input B. Send the each band member direct recording input to each channelʼs external input (more me). This will allow each band member to create a mix between the main mix, the click track and themselves. The HP60 is the ultimate headphone amplification system delivering loud and clear headphone mixes for real-world recording situations.

Features:
-Six independent ultra low noise, high output headphone amplifiers (150 mW per channel)
-Two sets of stereo inputs (A and B) with balanced TRS connectors
-Stereo external input point on each channel for "more me” with trim control
-Mix control between inputs A and B
-Talkback with external XLR microphone with control
-Direct stereo line output on each channel