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The MULTIGATE PRO XR4400 provides four frequency-selective expanders/noise gates in a single rack space. Each section comprises an ultra-fast gate, a program dependent expander, a filter section and precision meters indicating both threshold point and gain reduction.
The included side chain parametric filters allow the user to fine-tune control signals so that the gate reacts only when required. The Interactive Attack Control (IAC) circuit analyzes the program material to calculate the attack time of the gate. Separate hold and release controls offer additional flexibility. The XR4400 has a switchable operating mode for the gate and expander. With the MODE function off, the unit works in gate mode using an extremely fast attack time, thus preserving the transient details of the program material. In expander mode, the device analyzes the shape and dynamic contents of the input signal to calculate the control time parameters. The result is that vocals, instruments and complex mix signals can be cleaned up without audible clicks or other detrimental effects.
A couple function on the XR4400 provides a great deal of flexibility to link the expander/gate sections in a master/slave configuration.
Quadruple expander/gateHigh-precision parametric filter for frequency selective operationFully adjustable Ratio control in Expander modeFully adjustable Attenuation control in Gate modeIndependent Hold/Release controls for any envelope shapingFlexLink system for flexible Master/Slave configurationsExtremely low-noise operational amplifiers and high-grade VCA\'sFilter Monitor facility for monitoring the filter sectionHigh-quality detent potentiometers and backlit switchesServo-balanced gold-plated XLR and 1/4" TRS inputs and outputs8-digit LED meters for input level, output level and gain reduction monitoring
| Function |
Four channel expander/gate |
| Number of Channels |
4 |
| Inputs |
4 x XLR, 4 x TRS 1/4" phone |
| Output |
4 x XLR, 4 x TRS 1/4" phone |
| Maximum Input Level |
+21dBu |
| Compression Threshold |
Not applicable |
| Compression Ratio |
Not applicable |
| Attack Time |
Not applicable |
| Release Time |
Not applicable |
| Gate Threshold |
0 to -80dB |
| Gate Rate |
Attack: Program dependant
Hold: 0 to 4 sec, variable
Release: 50ms to 4 sec, variable |
| Maximum Output Level |
+21dBu |
| Make-up Gain |
Not specified by manufacturer |
| Frequency Response |
18HZ to 30kHz |
| Dynamic Range |
Not specified by manufacturer |
| Noise |
>95dBu, unweighted, 22Hz to 22kHz |
| Dimensions |
1U rack mount, 8 1/2" deep |
| Specialties |
Unit will operate at either +4dB or -10 dB |
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The Symphony I/O Chassis from Apogee provides two I/O Module slots for a variety of digital audio conversion options that form the interface side of the Symphony I/O. Elegant and easy to use, the front panel of Symphony I/O Chassis presents the visual feedback needed to monitor levels, sample rate and clock status, along with the controls required to adjust gain, output levels ,and select the Audio Interface Mode (AIM).
The rear panel provides access to the two I/O Module slots that are capable of a combined thirty-two channels of input and output. PCI, USB and Ethernet connectivity ensures complete compatibility with popular workstations, while the panel also includes Wordclock I/O connectors and termination, LoopSynch I/O connectors, and a quiet digitally-controlled fan to keep everything running cool.
The Audio Interface Mode (AIM) allows the system to be optimized for the most popular audio workstation environments, including Symphony PCI, Pro Tools HD PCI, USB audio, or to function as a standalone converter. With a two-step adjustment using a single front panel encoder knob, the interface can be easily switched back and forth between the preferred modes.
The optional five I/O Modules allow the user to choose from standard configurations, or customize the following - channel count, analog to digital input, digital to analog output, microphone preamps (requires an analog card) and digital I/O.
Two input/output module slots Ethernet connector USB 2.0 connector Two Symphony PC-32 ports for direct connection to Symphony 64 PCI card and Pro Tools core card Wordclock I/O and LoopSync I/O Wordclock termination switch Sixteen, 10-segment meters Two multi-function encoder knobs OLED GUI provides function selection and function adjustment and Audio Interface Mode (AIM) selection Two 1/4" headphone outputs 7-segment sample rate display Clock source display Meter source display High-performance, temperature controlled fan Connectors 2 x 1/4" TRS phone Headphone
2 x Symphony I/O PC-32 Ports
1 x Ethernet
1 x USB
2 x BNC Wordclock I/O
2 x BNC LoopSync I/O
Headphones THD+N: -105dB @ 19dBu
Dynamic Range: 119dB, A-weighted
Output Impedance: 30Ω
Power Requirements 90~240VAC, 50/60Hz, 150W
System Requirements Mac OS X 10.5.8 and above
Dimensions (WxDxH) 19 x 11.5 x 3.5" (483 x 292 x 89mm)
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Overview
Dial in YOUR mix exactly as YOU want it, in Real Time.
The PMC16 is a 16 channel personal monitor controller which allows the user to create a custom mix from 16 channels of audio. Designed to be used with the dbx TR1616 or any other BLU link compatible device, the PMC16 allows for monitoring with headphones, in-ear monitors, powered monitors, or traditional wedge monitors. Multiple PMC16s can be daisy chained using CAT5e cable allowing each user to receive 16 channels of high-resolution digital audio and create their own personalized mix to suit their monitoring needs. The PMC16 comprises a 16 channel mixer section with full control of levels, panning, effect send levels, muting, and soloing. Onboard Lexicon© reverb provides the finishing touch. The PMC16ʼs output processing section includes: stereo width control, wedge monitor compensation EQ, high and low master EQ, master level control, and dbx limiting. With a Setup Wizard for ease of configuration, full 16 channel level metering, channel grouping, 16 preset locations for future recall of configurations and mixes, and an intuitive yet powerful user interface, PMC16 gives you the power to dial in YOUR mix exactly as YOU want it, in real time.
Recommended for: Portable, Tour.
Features
- 16 Channel Monitor Mixer with Level, Pan, Mute, Solo, & Reverb Send Control
- 16 Preset Locations for Mix & Configuration Recall
- LCD Display with 16 Channel Metering
- Built-In Lexicon© Reverb
- dbx PeakStopPlus™ Limiting
- Global High & Low EQ
- Global Master Level Control
- Global Stereo Width Control
- BLU Link High Bandwidth, Fault Tolerant Digital Audio Bus
- XLR & 1/4” Stereo or Mono Outputs
- 1/8” & 1/4” Headphone Outputs
- Setup Wizard for Easy Configuration
- Supports 48 kHz & 96 kHz Sampling Rates
- Channel Linking, Grouping, & Multi-Select Capabilities
- Supports up to 60 PMC16s on a Single BLU Link Ring
- Built-In Sturdy Mic Stand Mount
- USB Port for Firmware Updates
| WDR 24 is a wireless receiver in the operating free band ISM 2.4 GHz, which has been designed to work with the transmitter WDT 24, creating a digital audio wireless system.
The transmitter WDT 24 can work with one receiver, or with up to 10 receivers WDR 24, covering a distance of more than 30 metres in open places.
- Wireless Digital Audio Receiver System
- Operating Band ISM 2.4GHz (free band).
- One transmitter WDT 24 paired with one receiver WDR 24, or one transmitter WDT 24 paired with up to ten receivers WDR 24.
- GFSK digital modulation/demodulation.
- External antenna can help to increase the transmission distance.
- Intelligent Frequency Hoping can help to avoid interference effectively and to ensure the channel quality.
- Automatic mute in bad channel quality conditions or high error code rate.
- Transmission distance higher than 30 meters in open places.
- Up to three transmitters WDT 24 in a zone
- Balanced XLR and unbalanced ¼\'\' jack outputs
- Working mode: stereo or mono
| Power Supply |
AC 9V-500 mA 50/60 Hz |
| Frequency |
2404-2476MHz |
| Channel |
25 |
| Modulation/Demodulation |
GFSK |
| Receive Sensitivity (WDR 24) |
-84dBm (min.) |
| Transmitted Power (WDT 24) |
6 dBm |
| Output Level (WDR 24) |
+13dBu (BAL max.) |
| Input Level (WDT 24) |
+5dBu (max.) |
| Bandwidth |
>=1.8 MHz |
| Frequency Response |
20Hz - 20kHz, +3dB / -3dB |
| S/N |
>80 dB |
| Dynamic Range @ 1 kHz |
>90 dB |
| Crosstalk @ 1kHz |
>90 dB |
| THD |
|
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Product Features:
16 MIDAS-designed, fully programmable mic preamps for audiophile sound quality
8 analog, servo-balanced XLR outputs
AES50 network ports featuring KLARK TEKNIKʼs SuperMAC networking capability for ultra-low jitter and latency
Up to 100 m networking capability via CAT-5e cable (not included)
Dual AES50 ports for cascading S16 units—no merger or router required
Precise LED metering plus 7-segment displays for signal control on stage
Phones output assignable to any of the inputs/outputs for on-stage monitoring
Connectivity for BEHRINGER's P-16 Personal Monitoring System for In-Ear applications (not included)
Dual ADAT outputs for use in splitter mode and stand-alone digital multicore applications
MIDI In/Out for bi-directional communication between FOH console and on-stage MIDI devices
USB connector for system updates via PC
"Planet Earth" switching power supply for maximum flexibility (100 - 240 V~), noise-free audio, superior transient response plus low power consumption for energy saving
"Built like a tank" yet compact, all-steel 2U rack-mount chassis
High-quality components and exceptionally rugged construction ensure long life
Conceived and designed by BEHRINGER Germany
KLARK TEKNIK AES50
KLARK TEKNIK SuperMAC (AES50-Compliant) technology gives the S16 its high-capacity digital interconnectivity with ultra-low latency operation, as well as superbly simple deployment and ease of use. KLARK TEKNIK engineers worked hand-in-hand with BEHRINGER product developers to ensure the S16 meets, or exceeds, the requirements of live performance entertainers and their professional audio engineers. Some of the outstanding features are:
48 bidirectional audio channels over CAT5 cable (48 kHz)
24 bidirectional audio channels over CAT5 cable (96 kHz)
Single cable duplex interconnection for audio and sample clocks
Ethernet physical layer audio data transmission
High channel count and ultra-low deterministic latency
Accurate phase-aligned clock distribution.
Comprehensive error detection
and management
Provision for redundant networking
Minimal configuration ñ total ease of deployment and use
Ethernet TCP/IP protocol-compatible auxiliary data channel
Royalty-free implementations available as Xilinx FPGA cores
OEM modules available from developer partners
User's Manual
| The Soundweb London BLU-DAN offers Dante audio and BLU link. This means that the BLU-DAN can be used to ‘add’ Dante to one or more BLU link devices.
The BLU-DAN is configurable through HiQnet Audio Architect. A rich palette of logic objects and a "drag and drop" method of configuration provide a simple and familiar design environment.
The BLU-DAN features Dante audio. The BLU-DAN is capable of simultaneously transmitting and receiving up to 64 channels of Dante audio (64 x 64). Primary and Secondary ports are provided for fault tolerance. Control is through a separate Ethernet port.
The BLU-DAN also features BLU link, a low latency, fault tolerant digital audio bus of 256 channels, which uses standard Category 5e cabling giving a distance of 100m between compatible devices. Fiber media converters can be used to increase the distance between devices to over 40km.
A bi-directional locate function allows devices to be identified both from and within HiQnet Audio Architect.
The BLU-DAN and the other members of the Soundweb London family provide the building blocks of the perfectly tailored system solution.
- AES67 and Dante Audio with support for Dante Domain Manager
- BLU link – 256 Channel, Low Latency, Fault Tolerant Digital Audio Bus
- ‘Add’ Dante / AES67 to one or more BLU link devices
- Clear Front Panel LED Indication
- Bi-Directional Locate Functionality
- HiQnet Device
- Configuration, Control and Monitoring from HiQnet Audio Architect
- 1RU, Half-Rack Device
- AES67 Compatible
- EN 54-16 Compliant for life safety applications
*The price is without VAT
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