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Overview: The Liquid Channel is a revolutionary professional channel strip that can emulate any classic mic-pre and compressor. Combining radical new analogue preamp technology with Dynamic Convolution? techniques, The Liquid Channel fuses cutting-edge analogue design with lightning fast SHARC DSP. Augmented by fully digital controls and optional remote software, The Liquid Channel provides the ultimate fluid vintage collection. Liquid technology Rather than creating a similar sound to vintage units, as with modelling devices, The Liquid Channel samples their sonic behaviour. This is achieved through Dynamic Convolution?; the application of a nique, level-dependant set of responses to an audio signal. These measured responses, sampled at numerous levels and with every possible setting combination, are applied to the input stream on a sample-by-sample basis for accurate emulation. Liquid preamplification Mic-pre emulation can?t be achieved with software alone. Hardware is required in addition to account for the physical interaction with the microphone. As a result, Focusrite invested vast amounts of time and energy into designing and building the ultimate ?liquid? preamplifier, able to match the input impedance and signal path (transformer or electronic) of the device being emulated. Not all vintage units are born equal Second order harmonic distortion is a beneficial artefact of analogue circuits (especially tube and transformer- coupled designs) providing the much loved ?warmth?. However, often two units of the same type will vary in the amount of second order distortion produced, so an additional control for modifying this vintage property is provided. This allows for precise matching of the Liquid Channel to your own beloved mic-pre. The best of both worlds - analogue and digital The Liquid Channel combines a highly complex, massively flexible analogue front-end with Dynamic Convolution? processing, which utilises lightening fast SHARC chips and runs at sample rates of up to 192kHz. The front panel controls are digital, with tactile rotary encoders; all parameters can be stored in one of 99 program memories. At the press of button, you can reload all mic-pre, compressor and EQ settings for an individual session. If using The Liquid Channel in conjunction with a recording platform (e.g. Pro Tools), both the session files and The Liquid Channel?s program memory can be sent via standard data transfer methods, providing a completely mobile recording session. Perfectly emulated vintage classics with the power and ease of use of the digital domain. Infinite expansion and remote control The USB port on the rear panel allows remote control of The Liquid Channel, leaving the processor safely racked away. But that?s not all. The software application also serves as an archiving system for additional emulations and program memories, and permits downloads of additional classic units from www.ffliquid.com. So, The Liquid Channel is infinitely expandable. Specifications Converter performance Sample rate: 44.1, 48, 88.2, 96, 176.4 and 192kHz Bit depth: 24-bit ADC SNR: 120dB measured with 20Hz/22kHz bandpass A-weighted filter Frequency response: ±0.05dB between 20Hz ? 22 kHz Maximum input level: +22dBu THD+N: 0.00035% (-109dB) DAC Dynamic range: 116dB measured with 20Hz/22kHz bandpass A-weighted filter Frequency response: ±0.05dB between 20Hz ? 22kHz Maximum output level: +22dBu THD+N: 0.0007% (-103dB) Jitter Internal clock: AES digital output: External clock: Analogue and digital path Mic pre Gain range: +6dB to +80dB, switched in 1dB steps Frequency response: variable, set by pre-amp chosen THD+N at analogue out: 0.001% measured with a +4dBu 1kHz input signal with 20Hz/22kHz bandpass filter THD+N at AES digital out: 0.0005% measured with a +4dBu 1kHz input signal with 20Hz/22kHz bandpass filter Mic noise: EIN = -126dB measured at 80dB of gain with 150 Ohm source impedance and 20Hz/22kHz bandpass filter Noise at analogue out: -92dBu measured at +6dB gain with 20Hz/22kHz bandpass A-weighted filter Noise at AES digital out: -119dBFS measured at + 6dB gain with 20Hz/22kHz bandpass A-weighted filter Maximum input level: +16dBu Input impedance: variable, set by preamp chosen CMRR: Transformer: 123dB @ 60dB of gain Electronic: 102dB @ 60dB of gain Line input Gain range: ?10dB to +10dB, switched in 1dB steps Frequency response: 0dB ±0.1dB between 20Hz and 20kHz THD+N at analogue out: 0.001% measured with a +18dBu 1kHz input signal with 20Hz/22kHz bandpass filter THD+N at AES digital out: 0.0004% measured with a +18dBu 1kHz input signal with 20Hz/22kHz bandpass filter Noise at analogue out: -92dBu measured at 0dB gain with 20Hz/22kHz bandpass A-weighted filter Noise at AES digital out: -120dBfs measured at 0dB gain with 20Hz/22kHz bandpass A-weighted filter Maximum input level: +22dBu High pass filter Roll off frequency: switchable between 75Hz and 120Hz, frequency measured at ?6dB down point, 12dB per octave roll-off Harmonics Distortion range: 0 to 15 where 15 (maximum) = 10% of 2nd-, 20% of 3rd- and 10% of 5th-order at 0dBFS (level-dependent distortion) Compressor In ?As Original ? mode the parameter ranges will be the same as on the original unit being emulated. In ?Free? mode the parameter ranges are as follows: Threshold range: -40dB to 20dB switched in 1dB steps Ratio range: 1:1 to limit Attack range: 0.1ms to 2.5s Release range: 0.1ms to 2.5s Make-up gain: -20dB to +20dB switched in 0.5dB steps EQ High Shelf Frequency range: 200Hz to 20kHz Gain: +/-18dB Mid Band Frequency range: 100Hz to 10kHz Gain: +/-18dB Q: variable between 0.8 and 2.5 Low Shelf Frequency range: 10Hz to 1kHz Gain: +/-18dB Weight 8.6kg Dimensions 484mm (W) x 85mm (H) x 270mm (D) 2U rackmount
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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
Input A
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(2) ¼” TRS balanced 10 Ohms
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Input B
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(2) ¼” TRS balanced 10 Ohms
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External input
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TRS stereo unbalanced 10 Ohms
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Headphone amplifier output power
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150 mW RMS (60 Ohms) per channel
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Audio outputs
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¼” TRS stereo unbalanced per channel
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Master controls
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Talk button, Talkback volume, Input A, Input B
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Channel controls
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Headphone level, Mix A/B control, External input level, Mono, Mute
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External talkback
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XLR with TS external control jack
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Power
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IEC 110V-240V internal switching
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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
| ISA OneClassic Single-channel Microphone Pre-amplifier with Independent D.I.
Key Features
ISA Series Transformer-based Pre-amplifier
Encased in a rugged and portable chassis, ISA One offers the classic Focusrite microphone pre-amplifier at its lowest cost to date.
Flexible Independent D.I. Channel
ISA One is ideal for both engineers and demanding performers alike, featuring independent gain control, an output for routing to an amp, an independent XLR output on the rear and routing to the optional ADC.
Switchable Impedance
Choose one of four carefully selected input impedances, including the original ISA 110 setting, to suit any microphone.
Optional Stereo 192kHz ADC
This optional card embodies cutting-edge conversion technology, incorporating Focusrite analogue circuitry to deliver the best A-D performance in its class (Dynamic range of 119dB).
Rugged Custom Flight Case
Built to protect your ISA One from the rigors of the road. Rugged plywood construction with a solid polypropylene exterior and reinforced corners.
Headphone Output with Volume Control
ISA One can feed either a sum of the two inputs to the headphones output, or an external stereo cue mix (such as a stereo mix feed from an interface) via two TRS Jack inputs on the rear of the unit.
Dedicated Insert Point
Allows you to place extra processing between the pre-amplifi er or D.I. and the optional converter, such as an EQ or compressor.
Specifications
Analogue Channel Inputs
- XLR Mic input
- One XLR and one TRS line input
- XLR Instrument input TRS Jack
- External ADC input TRS Jack
- TRS jack return
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Analogue Channel Outputs
- TRS jack send
- XLR balanced line output
- XLR DI output
- TS jack DI through
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Analogue Channel Additional I/O
- TRS Jack cue mix left input
- TRS Jack cue mix right input
- 1/4” TRS Jack headphones output
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Mic Input Response
Gain range |
0dB to 60dB in 10dB steps + 20 dB of variable gain |
Input Impedance |
Switched Impedance setting Equivalent Input Impedance at 1kHz |
Low = 600 Ohms
ISA110 = 1400 Ohms
Med = 2400 Ohms
High = 6800 Ohms |
EIN (Equivalent Input Noise) |
-126dB measured at 60dB of gain with 150 Ohm terminating impedance and 22Hz/22kHz band-pass filter |
Noise |
Noise at output with unity gain (0 dB) and 22 Hz-22 kHz band pass filter |
-97 dBu |
Signal-to-Noise Ratio |
106 dB relative to max headroom (9dBu) |
Total Harmonic Distortion + Noise |
Measured at medium gain (30dB) with a 1kHz -20dBu input signal and with a 22Hz/22kHz band-pass filter |
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Frequency Response |
At minimum gain (0 dB) |
-0.5dB down at 10Hz and -3dB down at 125kHz |
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At maximum gain (60dB) |
-3dB down at 16Hz and -3dB down 118kHz |
CMRR (Common Mode Rejection Ratio) |
98dB (Channel 1, 1kHz, maximum gain with +24 dBu input) |
Crosstalk Channel to Channel |
With 10dB@1kHz input to chA, chB output =104dBrA. With 10dB@10kHz input to chA, chB output = 84dBrA |
Line Input Response
Gain range |
-20dB to +10dB in 10dB steps + 20 dB of variable gain |
Input Impedance |
10 kΩ from 10 Hz to 200 kHz |
Noise |
Noise at main output with gain at unity (0 dB) measured with 50 Ω source impedance and a 22Hz - 22 kHz band pass filter |
-96 dBu |
Signal-to-Noise Ratio |
Measured with a 22 Hz-22 kHz band pass filter |
120 dB relative to max headroom (24 dBu)
118dB relative to 0dBFS (+22dBu) |
Total Harmonic Distortion + Noise |
Measured with a 0 dBu input signal, and a 22 Hz-22 kHz band pass filter |
0.0001% |
Frequency Response |
At unity gain (0 dB) |
-0.3dB down at 10Hz and -3dB down at 200kHz |
Instrument Input Response
Gain range |
10dB to 40dB continuously variable |
Input Impedance |
High = greater than 1M
Low =greater than 300k |
Noise |
Measured with 22 Hz-22 kHz band pass filter |
Minimum gain (+10 dB): -92 dBu
Maximum gain (+40dB): -62dBu |
THD |
At minimum gain (+10) |
Frequency Response |
At minimum gain (+10 dB) |
-10dB input: 10Hz-100kHz +/- 0.6dB |
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At maximum gain (+40 dB) |
-40dB input: -2.5dB down at 10Hz and 0dB at 100kHz |
Meters
Moving Coil (MC) Meter |
factory calibrated to 0VU = +4dBu with 1kHz sinewave. With the VU Cal button pressed the meter can be adjusted on the rear panel to allow 0VU to equal +10dBu to +26dBu with the centre detent being equal to +22dBu. |
Peak LED Meters |
Calibrated in the detent position for 0dBFS = +22dBu, calibration is adjustable on the rear panel to allow 0dBFS to equal +10dBu to +26dBu |
Routing for MC and Peak1 meter is after the HPF, pre insert send or switched post insert return. Peak2 is always pre ADC channel 2, which can be fed by external input or Instrument input. |
LED Levels |
As follows, when peak calibration is set to center detent on the rear panel. (This is when using the internal ADC). |
0 = +22dBu
-2 = +20dBu
-6 = +16dBu
-12 = +10dBu
-18 = +4dBu
-42 = -20dBu |
Frequency Response |
At minimum gain (+10 dB) |
-10dB input: 10Hz-100kHz +/- 0.6dB |
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At maximum gain (+40 dB) |
-40dB input: -2.5dB down at 10Hz and 0dB at 100kHz |
High-Pass Filter
Roll-Off |
18 dB per octave (3 pole filter) |
Frequency |
Fixed 75Hz measured at the 3dB down point |
Weight and Dimensions
W x D x H |
220mm (W) x 104mm (H) x 254 - 290mm (D - top to bottom) |
8.66" (W) x 4.1" (H) x 10" - 11.4" (D - top to bottom) |
Weight |
3.9 kg |
8.6 lbs |
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Designed for high-definition field production environments, the studio-quality MP-1 is an ideal line-driver for critical radio, television, and film applications.
The MP-1 is extremely durable and easy to use. With rugged mechanical and electrical construction and high-quality components it will provide years of superb audio performance under the most punishing field conditions.
MP-1 Key Features - Superb audio specifications
- 66 dB of gain, in eleven discrete steps
- Transformer-balanced input and output
- Phantom power - 48 volt or 12 volt
- High pass filter @ 80 Hz or 160 Hz, 6 dB/octave
- Limiter makes unit virtually "unclippable“
- High current line output drive
- Excellent immunity to RF interference.
- Battery power (two x AA)
- Durable mechanical construction
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