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 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)
Weight
2.50 kg
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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)
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
The Soundweb London BLU-102 offers a fixed configuration of 10 analog inputs and 8 analog outputs, an analog telephone interface, configurable signal processing, AEC processing, and a high bandwidth, fault tolerant digital audio bus.
The RJ-11 port enables the BLU-102 to interface with a standard POTS (aka PSTN or Analog PBX) telephone network.
The BLU-102 has open architecture which is fully configurable through HiQnet™ London Architect. A rich palette of processing and logic objects and a "drag and drop” method of configuration provide a simple and familiar design environment.
The BLU-102 contains dedicated AEC processing for up to 8 independent AEC algorithms. The AEC algorithm can be applied to signals coming from the local analog inputs or from the digital audio bus. 8 individual AEC references (one per algorithm) allow the user to provide a solution for multiple conferencing spaces using a single device.
Automatic Gain Control (AGC) and Noise Cancellation (NC) are also provided per AEC algorithm. AGC ensures that microphone levels remain at an optimum level, and NC removes steady state noise (such as from a projector fan or air conditioning device) from the signal path.
This processor features a low latency, fault tolerant digital audio bus of 48 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.
The BLU-102 is compatible with the entire Soundweb London family and its 48 channel digital audio bus represents channels 1-48 of the larger 256 channel digital audio bus when integrated with the BLU-800, BLU-320, BLU-160, BLU-120, BLU-BIB and BLU-BOB devices.
Analog Inputs provide software configurable gain in 6dB steps up to +48dB per channel and software selectable Phantom Power per channel.
Phantom Power, Signal Present and Clip information per channel is easily accessible, without the requirement for a PC, from clear front panel LED indication. A bi-directional locate function allows devices to be identified both from and within HiQnet London Architect.
12 Control Inputs and 6 Logic Outputs allow the BLU-102 to be integrated with GPIO compatible devices. The Soundweb London Interface Kit, comprehensive documentation which details how Soundweb London systems can be integrated with third party control systems, is included within the installation of HiQnet London Architect.
The BLU-102 and the other members of the Soundweb London family provide the building blocks of the perfectly tailored system solution.
Key Features:
• 10 Analog Inputs (with 48v Phantom Power per Channel)
• 8 Analog Outputs
• RJ-11 Telephone Input/Output Port
• Configurable Signal Processing
• 8 Channels of AEC Processing with Auto Gain Control and Noise Cancellation
• Rich Palette of Processing and Logic Objects
• 48 Channel, Low Latency, Fault Tolerant Digital Audio Bus
• Clear Front Panel LED Indication
• Bi-Directional Locate Functionality
• 12 Control Inputs and 6 Logic Outputs for GPIO Integration
• Soundweb London Interface Kit for Third Party Control System Integration (Documentation)
• HiQnet Device
• Configuration, Control and Monitoring from HiQnet London Architect
mic/line for the BLU-80, BLU-32, BLU-16.
WDT 24 is a wireless transmitter in the operating free band ISM 2.4 GHz, which has been designed to work with the wireless receiver WDR 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 Transmitter 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 ¼\'\' jack inputs.
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
The dbx 560A Compressor/Limiter offers legendary VCA-based compression in the increasingly popular 500 Series format. dbx has a much-heralded history with modular processors dating back to our 900 Series and the 560A continues that tradition.
Featuring the same switch-selectable OverEasy? and Hard Knee compression curves as our industry-standard dbx 160A rackmount processor, the dbx 560A will be your new workhorse compressor! With extremely wide-ranging threshold, ratio and output controls and true RMS level detection providing the most natural-sounding dynamics processing available, the 560A will provide punch to drums, bass, vocals, almost anything! It’s also versatile enough to go from subtle compression to "brick wall" peak limiting.
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