FOCUSRITE PLATINUM V-MAST. PRO VOICE PRE-AMP. CHANNEL STRIP

Discontinued

541.25 incl. GR VAT

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PLATINUM V/MAST PRO With every project, you need to capture that perfect vocal and make the sound your own. VoiceMaster Pro represents a new generation of voice recording equipment.Many of the elements from the award-winning VoiceMaster have been innovated upon. In addition, the unit features a whole host of new processors designed to ensure you […]

 

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PLATINUM V/MAST PRO

With every project, you need to capture that perfect vocal and make the sound your
own. VoiceMaster Pro represents a new generation of voice recording equipment.
Many of the elements from the award-winning VoiceMaster have been innovated upon.
In addition, the unit features a whole host of new processors designed to ensure you
have all you require to make the most of the artist you’re recording.
The award-winning Class A preamp captures every nuance from any source, whilst
latency-free monitoring ensures direct and delay-free mix control. Tools such as the
voice-optimised EQ, Vintage Harmonics and Tube Sound allow you to get creative
with a touch of class, putting your own stamp on every recording.
All of this, combined with the optional 24 bit, 96 kHz digital output board means that
VoiceMaster Pro is the perfect interface between your microphone and your DAW or
other recording media, placing the ultimate tracking solution firmly within your grasp.

Specifications
Mic Input Response
    * Gain range = 0dB to 60dB continuously variable
    * Input Impedance = 1k4 Ohms from 10Hz to 200kHz
    * EIN = 128dB measured at 60dB of gain with 150 Ohm terminating impedance
      and 20Hz/22KHz bandpass filter
    * THD at minimum gain (0dB) = .0004% measured with +16dBu input signal and
       with a 20Hz/22KHz bandpass filter
    * THD at maximum gain (60dB) = .003% measured with a ?36dBu input signal
      and with a 20Hz/22KHz bandpass filter
    * THD at maximum Input level (26.5dBu) = .005% measured with a 20Hz/22KHz
      bandpass filter
    * Frequency response at minimum gain (0dB) = flat at 10Hz and ?2dB down at 200kHz
    * Frequency response at maximum gain (60dB) = ?2dB down at 10Hz and 200kHz
    * CMRR at full gain (60dB) = 80dB

Line Input Response
    * Gain range = -10dB to +10dB continuously variable
    * Input Impedance = 10k Ohms from 10Hz to 200kHz
    * Noise at main output with gain and fader set to unity (0dB) = -94dBu measured
      with a 20Hz/22kHz bandpass filter
    * Signal to noise ratio relative to max headroom (27dBu) = 121dB
    * Signal to noise ratio relative to 0dBfs (+22dBu) = 116dB
    * THD at unity gain (0dB) = .0006% measured with 0dBFS (+22dBu)
    * Input signal with a 20Hz/22kHz bandpass filter
    * Frequency Response at unity gain (0dB) = 0.25dB down at 10Hz and ?3dB
      down at 200kHz

Instrument Input Response
    * Gain range = 0dB to 40dB continuously variable
    * Input Impedance >1MOhm
    * Noise at minimum gain (0dB) = -90dBu measured with a 20Hz/22kHz bandpass filter
    * Noise at maximum gain (40dB) = -78dBu measured with a 20Hz/22kHz bandpass filter
    * THD at minimum gain (0dB) = .006% measured with ?10dBu input signal and
      with a 20Hz/22kHz bandpass filter
    * Frequency Response at unity gain (0dB) = 0.5dB down at 10Hz and ?1dB down at 200kHz
    * Frequency Response at maximum gain (60dB) = 6dB down at 10Hz and ?1dB down at 200kHz

Input Meter
      6 LED meter is calibrated relative to 0dBfs where 0dBfs = +22dBu
      (the maximum level which can be correctly converted by the optional
      internal A/D converter before overload occurs). The meter calibration
      points are as follows:
    * O/L dBFS (converter overload) = +22.4dBu (equivalent value)
    *  0dBFS = +22dBu
    * -3dBFS = +19dBu
    * -6dBFS = +16dBu
    * -10dBFS = +12dBu
    * -20dBFS = +2dBu (average level to allow 20dB of headroom for EQ and Dynamics

High Pass Filter
    * Roll off = 18dB per octave 3 pole filter
    * Frequency Range = continuously variable from 30Hz to 400Hz measured
      at the 3dB down point

Optical Expander
    * Threshold hold range = -20dBFS (0dBu) to ?60dBFS (-40dBu)
    * Expander ratio = 2:1
    * Attack time = 750us
    * Release time = 0.25s to 4s
    * Noise with maximum expansion = -96dBu measured with a 20Hz/22KHz
      bandpass filter
    * Signal to noise ratio relative to max headroom (27dBu) = 123dB
    * Signal to noise ratio relative to 0dBfs (+22dBu) = 118dB
    * 6 LED meter shows downward expansion calibrated in dB increments

Vintage Harmonics
    * Threshold range = -10dBFS (+12dBu) to ?40dBFS (-18dBu)
    * Compression ratio Depth switch out = 1:1.5
    * Compression ratio Depth switch in = 1:3
    * Mid band centre frequency = 3.5kHz
    * High band centre frequency = 20kHz
    * Band Q = 0.2
    * 6 LED meters show the amount of boost at the centre frequency of the band,
      meter calibrated in dB increments

Optical Compressor
    * Threshold hold range = -15dBFS (7dBu) to ?40dBFS (-18dBu)
    * Compressor ratio Hard Ratio switch out = 2.5:1
    * Compressor ratio Hard Ratio switch in = 6:1
    * Attack time Slow Attack switch out = 0.5ms
    * Attack time Slow Attack switch in = 5ms
    * Release time = 100ms to 1s and then auto release mode when
      the release knob is turned fully clockwise. Auto release creates
      a release time dependent upon the average level of the incoming signal
    * Noise = -94dBu measured with a 20Hz/22KHz bandpass filter
    * Makeup gain = 0 to +18dB

Tube Sound
    * Harmonic generation = 2nd, 3rd and 5th harmonics generated from
      fundamental frequency of the incoming signal relative to position of
      the Drive knob
    * THD with Drive knob at cool = 0.6% with a ?10dBFS (+12dBu) 1kHz
       input signal measured with a 20Hz/22kHz bandpass filter
    * THD with Drive knob at warm = 6% with a ?10dBFS (+12dBu) 1kHz
       input signal measured with a 20Hz/22kHz bandpass filter
    * The effect can be monitored visually using the drive led as follows: Blue
       to Green = Mostly 2nd with some 3rd, Yellow to Orange = More 3rd and 5th,
       Orange to Red = Large amounts of all harmonics above the fundamental frequency
    * Tone = 6dB per octave roll off low pass filter
    * Tone frequency Range = continuously variable from 4.5kHz to 30kHz
      measured at the 3dB down point

Voice Optimised EQ
Breath
    * EQ shape = Shelving
    * Peak frequency Shift switch out = 10kHz
    * Peak frequency Shift switch in = 16kHz
    * Gain range = +/-8dB

Mid
    * EQ shape = Peak
    * Centre frequency = 1.3kHz
    * Q in boost mode = 0.5
    * Q in cut mode = 2
    * Gain range = +8 to -12dB

Absence
    * EQ shape = Peak
    * Centre frequency = 3.9kHz
    * Q = 3.5
    * Gain range = 0dB to -10dB

Warmth
    * EQ shape = Peak
    * Centre frequency range = 120Hz to 600Hz
    * Q in boost mode = 0.5
    * Q in cut mode = 2
    * Gain range = +8 to -12dB

De-esser
    * Threshold hold range = -10dBFS (+12dBu) to ?40dBFS (-18dBu)
    * De-esser ratio at centre frequency = 2:1
    * Centre frequency range = 2.2kHz to 10kHz
    * Q of cut = 3.5
    * Attack time = 0.5ms
    * Release time = 100ms
    * Noise = -94dBu measured with a 20Hz/22KHz bandpass filter

Output Meter
    * Calibrated for 0dBFS = +22dBu and indicates the level after the
      output fader being sent to both the internal AD converter and the
      VoiceMaster Pro XLR output
    * O/L LED = is lit when any section of the unit (including the cans output)
      reaches a level greater than 0dBFS (actual level is +22.4dBu)

Weight 5.00 kg

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Description
Content

PLATINUM V/MAST PRO

With every project, you need to capture that perfect vocal and make the sound your
own. VoiceMaster Pro represents a new generation of voice recording equipment.
Many of the elements from the award-winning VoiceMaster have been innovated upon.
In addition, the unit features a whole host of new processors designed to ensure you
have all you require to make the most of the artist you're recording.
The award-winning Class A preamp captures every nuance from any source, whilst
latency-free monitoring ensures direct and delay-free mix control. Tools such as the
voice-optimised EQ, Vintage Harmonics and Tube Sound allow you to get creative
with a touch of class, putting your own stamp on every recording.
All of this, combined with the optional 24 bit, 96 kHz digital output board means that
VoiceMaster Pro is the perfect interface between your microphone and your DAW or
other recording media, placing the ultimate tracking solution firmly within your grasp.

Specifications
Mic Input Response
    * Gain range = 0dB to 60dB continuously variable
    * Input Impedance = 1k4 Ohms from 10Hz to 200kHz
    * EIN = 128dB measured at 60dB of gain with 150 Ohm terminating impedance
      and 20Hz/22KHz bandpass filter
    * THD at minimum gain (0dB) = .0004% measured with +16dBu input signal and
       with a 20Hz/22KHz bandpass filter
    * THD at maximum gain (60dB) = .003% measured with a ?36dBu input signal
      and with a 20Hz/22KHz bandpass filter
    * THD at maximum Input level (26.5dBu) = .005% measured with a 20Hz/22KHz
      bandpass filter
    * Frequency response at minimum gain (0dB) = flat at 10Hz and ?2dB down at 200kHz
    * Frequency response at maximum gain (60dB) = ?2dB down at 10Hz and 200kHz
    * CMRR at full gain (60dB) = 80dB

Line Input Response
    * Gain range = -10dB to +10dB continuously variable
    * Input Impedance = 10k Ohms from 10Hz to 200kHz
    * Noise at main output with gain and fader set to unity (0dB) = -94dBu measured
      with a 20Hz/22kHz bandpass filter
    * Signal to noise ratio relative to max headroom (27dBu) = 121dB
    * Signal to noise ratio relative to 0dBfs (+22dBu) = 116dB
    * THD at unity gain (0dB) = .0006% measured with 0dBFS (+22dBu)
    * Input signal with a 20Hz/22kHz bandpass filter
    * Frequency Response at unity gain (0dB) = 0.25dB down at 10Hz and ?3dB
      down at 200kHz

Instrument Input Response
    * Gain range = 0dB to 40dB continuously variable
    * Input Impedance >1MOhm
    * Noise at minimum gain (0dB) = -90dBu measured with a 20Hz/22kHz bandpass filter
    * Noise at maximum gain (40dB) = -78dBu measured with a 20Hz/22kHz bandpass filter
    * THD at minimum gain (0dB) = .006% measured with ?10dBu input signal and
      with a 20Hz/22kHz bandpass filter
    * Frequency Response at unity gain (0dB) = 0.5dB down at 10Hz and ?1dB down at 200kHz
    * Frequency Response at maximum gain (60dB) = 6dB down at 10Hz and ?1dB down at 200kHz

Input Meter
      6 LED meter is calibrated relative to 0dBfs where 0dBfs = +22dBu
      (the maximum level which can be correctly converted by the optional
      internal A/D converter before overload occurs). The meter calibration
      points are as follows:
    * O/L dBFS (converter overload) = +22.4dBu (equivalent value)
    *  0dBFS = +22dBu
    * -3dBFS = +19dBu
    * -6dBFS = +16dBu
    * -10dBFS = +12dBu
    * -20dBFS = +2dBu (average level to allow 20dB of headroom for EQ and Dynamics

High Pass Filter
    * Roll off = 18dB per octave 3 pole filter
    * Frequency Range = continuously variable from 30Hz to 400Hz measured
      at the 3dB down point

Optical Expander
    * Threshold hold range = -20dBFS (0dBu) to ?60dBFS (-40dBu)
    * Expander ratio = 2:1
    * Attack time = 750us
    * Release time = 0.25s to 4s
    * Noise with maximum expansion = -96dBu measured with a 20Hz/22KHz
      bandpass filter
    * Signal to noise ratio relative to max headroom (27dBu) = 123dB
    * Signal to noise ratio relative to 0dBfs (+22dBu) = 118dB
    * 6 LED meter shows downward expansion calibrated in dB increments

Vintage Harmonics
    * Threshold range = -10dBFS (+12dBu) to ?40dBFS (-18dBu)
    * Compression ratio Depth switch out = 1:1.5
    * Compression ratio Depth switch in = 1:3
    * Mid band centre frequency = 3.5kHz
    * High band centre frequency = 20kHz
    * Band Q = 0.2
    * 6 LED meters show the amount of boost at the centre frequency of the band,
      meter calibrated in dB increments

Optical Compressor
    * Threshold hold range = -15dBFS (7dBu) to ?40dBFS (-18dBu)
    * Compressor ratio Hard Ratio switch out = 2.5:1
    * Compressor ratio Hard Ratio switch in = 6:1
    * Attack time Slow Attack switch out = 0.5ms
    * Attack time Slow Attack switch in = 5ms
    * Release time = 100ms to 1s and then auto release mode when
      the release knob is turned fully clockwise. Auto release creates
      a release time dependent upon the average level of the incoming signal
    * Noise = -94dBu measured with a 20Hz/22KHz bandpass filter
    * Makeup gain = 0 to +18dB

Tube Sound
    * Harmonic generation = 2nd, 3rd and 5th harmonics generated from
      fundamental frequency of the incoming signal relative to position of
      the Drive knob
    * THD with Drive knob at cool = 0.6% with a ?10dBFS (+12dBu) 1kHz
       input signal measured with a 20Hz/22kHz bandpass filter
    * THD with Drive knob at warm = 6% with a ?10dBFS (+12dBu) 1kHz
       input signal measured with a 20Hz/22kHz bandpass filter
    * The effect can be monitored visually using the drive led as follows: Blue
       to Green = Mostly 2nd with some 3rd, Yellow to Orange = More 3rd and 5th,
       Orange to Red = Large amounts of all harmonics above the fundamental frequency
    * Tone = 6dB per octave roll off low pass filter
    * Tone frequency Range = continuously variable from 4.5kHz to 30kHz
      measured at the 3dB down point

Voice Optimised EQ
Breath
    * EQ shape = Shelving
    * Peak frequency Shift switch out = 10kHz
    * Peak frequency Shift switch in = 16kHz
    * Gain range = +/-8dB

Mid
    * EQ shape = Peak
    * Centre frequency = 1.3kHz
    * Q in boost mode = 0.5
    * Q in cut mode = 2
    * Gain range = +8 to -12dB

Absence
    * EQ shape = Peak
    * Centre frequency = 3.9kHz
    * Q = 3.5
    * Gain range = 0dB to -10dB

Warmth
    * EQ shape = Peak
    * Centre frequency range = 120Hz to 600Hz
    * Q in boost mode = 0.5
    * Q in cut mode = 2
    * Gain range = +8 to -12dB

De-esser
    * Threshold hold range = -10dBFS (+12dBu) to ?40dBFS (-18dBu)
    * De-esser ratio at centre frequency = 2:1
    * Centre frequency range = 2.2kHz to 10kHz
    * Q of cut = 3.5
    * Attack time = 0.5ms
    * Release time = 100ms
    * Noise = -94dBu measured with a 20Hz/22KHz bandpass filter

Output Meter
    * Calibrated for 0dBFS = +22dBu and indicates the level after the
      output fader being sent to both the internal AD converter and the
      VoiceMaster Pro XLR output
    * O/L LED = is lit when any section of the unit (including the cans output)
      reaches a level greater than 0dBFS (actual level is +22.4dBu)

The U5 high-voltage DI-preamp combines a unique passive tone selector with a variable
gain preamp and filter. The high input impedance accepts a wide variety of signal levels
and instruments from acoustic guitars to high-output active bass guitars and keyboards.
The U5 direct box is loaded with sonic character and user features professionals demand.

U5 FEATURES
Pure Class A, 100% discrete design
Smooth musical detail and sonic excellence
Variable gain preamp to +30dB.
Dual microphone and line outputs
Hi-Z   3,000,000 ohm input impedance
Very low noise -100dB
High headroom +30dB
Six tone bank selections.
DC coupled Class A output for superior bass
High cut switch eliminates noise
LED active signal indicator
Headphone monitor output jack
Low distortion less than 0.5% THD and IMD
High level 400 watt speaker input
Active to thru selector - ground isolation switch
100% discrete power supplies for audio path
Internal toroidal power supply for low noise
Long lasting, stainless steel hardware
 
SONIC EXCELLENCE
The Avalon U5 Pure Class A instrument DI & preamplifier is the most powerful direct box
available today.  Designed to optimize absolute signal integrity and musical performance,
the U5 includes many features and sonic excellence unequaled by lesser designs. The
U5 is ideal for bass guitar and acoustic instruments, electric guitars, keyboards, synthesizers
and low output pickups. 
Features include state-of-the-art, high input impedance input stage for zero load effect on
sensitive pickups and keyboards, 100% discrete, Pure Class A signal amplifiers, practical
user features and rugged hardware designed to deliver true high performance audio for many
years. 
DESIGNED FOR DEEP & CONTROLLED BASS
Avalon U5 DI-preamps have been in use around the world since 1995. They have found their way
into the world's leading musicians rigs and recording sessions and been featured on thousands
of the best selling (and sounding) albums and live performances.
MINIMUM SIGNAL PATH DESIGN
The U5 includes a variable gain Class A preamplifier to boost low level pickup signals and a high
level speaker input for capturing the live sound of the instruments amplifier. Six passive tone-EQ
curves are also included to enhance a variety of acoustic and electric instruments. A high-cut switch
eliminates unwanted acoustic pickup and high frequency noise, while a  headphone monitor amp
has been included for personal listening. Twin DC coupled Class A output amplifiers drive both low
level microphone preamp inputs and high level +4dB inputs for direct to tape recording or processing.
The active-to-thru switch selects either the instrument input directly or sends the boosted-equalized
signal to the amplifier via the front panel jack. A ground isolation switch eliminates any possible earth
loop or AC buzz problems with complete safety.

U5 SPECIFICATIONS
Circuit Topology:  100% discrete, high-voltage Class A 
Gain:  Switched 3dB steps, maximum gain +30dB 
Input Impedance:  3 ,000,000 ohm unbalanced (single ended) 
Maximum Input Level  +24dBu unbalanced, 400w speaker input 
Output Connectors (2):  XLR type, pin 2 hot electronic balanced 
Maximum Output Level (Class A):  +30dBu DC coupled, balanced 600 ohms (mic and line out) 
Headphone Output:  0.5w into 600 ohms 
Noise 20kHz Unweighted:  -100dBu minimum gain position 
Distortion THD, IMD:  0.1% at +10dB 
Frequency Response, -/+0.5dB:  5Hz to 100kHz 
Frequency Response, -3dB:  1Hz to 500kHz (input band limited) 
High Cut Filter:  -3dB at 8kHz minimum phase design 
Tone Selector:  Six position rotary switch, all passive filters 
AC Power:  Internal toroidal AC supply 
Dimensions:  8.5 x 3.5 x 12 in (216 x 88 x 305mm) 
Weight-Packed:  12lbs (5.4kg) 
Rack mount kit (1) RM-1:  Single U5 19 inch rack mount kit (holds one half rack unit) 
Rack mount kit (2) RM-2:  Dual U5 19 inch rack mount kit (holds two half rack units) 

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)
 

• Road proof housing • High quality components • Professional balanced XLR outputs • Level control The ASC-202 converts consumer level stereo RCA signals to balanced XLR professional line level, and back. One side of the unit, has two XLR inputs and stereo RCA outputs. The other side has stereo RCA inputs and two XLR outputs. Each side has stereo level controls. A possible application of the SC-13 is to use it as a level control in balanced audio lines. The DAP Pro Audio Solutions are a complete line-up of audio tools for the professional musician or audio engineer. It is a series of audio modifying tools like mini-mixers, pre-amps and converters available in active or passive models. All audio solutions are designed to meet the highest stage and audio demands and feature a metal road proof housing with rubber feet and easy accessible controls. The units also have a small footprint which make them ideal to be used and placed in any environment where space is critical. Specifications Frequency Response: 20Hz-20KHz Inputs: XLR/Jack Combo, RCA Outputs: XLR male, RCA Input sensitivity RCA in: -10 dBv Input Impedance RCA in: 10 kOhm Input sensitivity XLR/Jack in: +4 dBu Input Impedance XLR/Jack in: 20 kOhm Output voltage RCA Out: -10 dBV Output Impedance XLR Out: 100 Ohm Output voltage XLR Out: +4 dBu Output Impedance XLR Out: 200 Ohm Power supply: 18V AC (adapter included) Dimensions: 53 x 158 x 135 mm (HxWxD) Weight: 0,65 KgHighly versatile and portable, Sound Devices' MM-1 is a single-channel, portable microphone preamplifier with a powerful headphone monitoring function. Proven in mission-critical applications, the high-performance microphone preamplifier uses a transformer-balanced input and output with a bipolar ±15 volt power supply to produce low-noise, low-distortion gain at all switch positions. It features an "unclippable" dual-stage limiter and two-position high-pass filter. Flexible microphone powering provides phantom power at 48 volts or 12 volts and 12-volt T-power. It is powered from two-AA batteries or external 5-18 VDC. The MM-1 is at home in high-definition television production, corporate/industrial environments, radio/television announce booths, and music production. Its headphone monitor is an advantage in applications where communications channels or mix-minus feeds need to be monitored in headphones. The user can monitor both microphone audio and external audio sources in headphones with level control. The MM-1 is designed to be durable. Its aluminum chassis, oversized end panels, machined knobs, and panel mounted connectors assure reliable mechanical operation.
  • Maximum of 66 dB of gain, in eleven discrete steps
  • Dynamic range exceeding 120 dB
  • 10 Hz to 50 kHz audio bandwidth
  • High immunity to RF interference
  • High current line output driver
  • Transformer-balanced input and output
  • Extended range peak limiter via dual opto-isolators makes preamplifier virtually "unclippable“
  • Selectable 48-volt / 12-volt phantom power or 12-volt "T" power
  • High-pass filter @ 80 or 160 Hz, 6 dB per octave
  • Headphone monitoring of microphone audio, monitor audio, or a mix of both
  • Monitor input accepts mono, balanced or unbalanced line-level inputs on 1/4-inch tip-ring-sleeve connector
  • Separate level control of Monitor gain and overall headphone gain
  • Split ear function via DIP switches
  • Internal battery power (two-AA)
  • Excellent battery life with alkaline and NiMH rechargeable batteries
  • External 5-17 VDC powering (plug included)
  • High-strength aluminum chassis with durable belt clip

 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

Weight5.00 kg5.40 kg1.00 kg0.65 kg3.00 kg2.00 kg
DimensionsN/AN/AN/AN/AN/AN/A
Additional information
Weight 5.00 kg
Weight 5.40 kg
Weight 1.00 kg
Weight 0.65 kg
Weight 3.00 kg
Weight 2.00 kg