Overview
The DM84 is an 8-in, 4-out digital matrix processor with the capability to route any combination of inputs to any combination of outputs with extremely low throughput latency. Automatic microphone mixing is accomplished via a patented Adaptive Proportional Gain algorithm. Sophisticated filtering is offered on each input and output. A compressor is available on each input, a compressor and limiter on each output. In addition, each input and output can be configured with a delay setting of up to 100 ms for inputs, 250 ms for outputs. The DM84 runs on the latest-generation DSP architecture and can be integrated as part of a larger system by connecting other DM units (like the DMTH4) via the Digital Audio Network Interface (DANI) bus.
The DM84 has enhanced expandability - up to a limit of three, combining multiple DM84 units expands not only the number of the mic inputs, but also the number of distinct outputs/mix busses in the system. Stacking 2 units creates a 16x8 mixing matrix; stacking 3 units creates a 24x12 mixing matrix. While individual outputs/mix buses are normally treated as independent NOM buses for microphone mixing, the DM84's "phantom" mix mode capability allows multiple mix busses to be combined into a single NOM bus for mix-minus sound reinforcement applications.
The DM84 offers eight mic/line inputs with four mic/line outputs; switchable phantom power to handle condenser microphones; programmable front-panel control knobs for easy input and output gain adjustment; front panel indicators for input and output channel activity; and auto mic mixing with Proportional Gain and AutoSkew. The unit is AMX® and CRESTRON® compatible with USB and RS-232 interfaces for setup and control. Also, a programmable I/O interface allows control using contact closures or voltage levels as well as system status monitoring.
Features
- 8-in/4-out digital matrix architecture
- Stackable with enlarged matrix up to 24-in/12-out
- Front panel level controls with activity LEDs
- 6 filters plus compressor on each input
- 9 filters plus compressor/limiter on each output
- USB and RS-232 interfaces for setup and control
- Fully balanced audio signal flow through entire system
- Digital I/O ports for connection to other LecNet 2 devices
- Proportional gain auto mixing algorithm with AutoSkew™ - US Patent 5,414,776
- Simultaneous LecNet2 and Third-Party control
Specifications
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Audio inputs
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Gain:
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-10 dB to +60 dB, programmable in 1 dB steps |
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Input impedance:
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2.5 k Ohm |
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Phantom voltage:
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15 V, programmable |
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Connector:
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5-pin Phoenix |
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Audio outputs:
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Nominal level:
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Floating balanced, either side can be grounded 0 dBu all outputs, -40 dBu selectable on outputs 9 through 12 |
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Output impedance:
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• 450 Ohm differential programmable outputs at line level
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• 5 Ohm differential programmable outputs at microphone level
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Input Dynamic Range:
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96 dB at -50 dBu input level 102 dB at all other levels (unweighted 20 - 20 kHz)
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Output Dynamic Range:
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105 dB (unweighted 20 - 20 kHz)
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Audio Performance:
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IMD + noise:
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0.1% max.
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0.02% nominal input level
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THD + noise:
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0.1% (worst case)
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0.02% nominal input level
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EIN:
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-126 dBu
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Connectors:
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Audio I/O:
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5-pin Phoenix
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Expansion:
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RJ45
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Logic I/O:
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DB25
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Serial:
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Standard USB and mini TRS
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| Digital Audio Network Interface (DANI) | |
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Physical level:
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LVDS (Low Voltage DIfferential Signal) high speed |
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Connector:
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Four RJ-45 |
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Cable quality:
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Shielded CAT-5 |
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Transmission speed:
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50 Mbits/s |
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Programmable control inputs
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Number of inputs:
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11 |
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Analog voltage range:
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0-5V |
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Logic input:
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TTL, LVTTL, CMOS, LVCMOS |
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Programmable control outputs
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Number of logic outputs:
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8 |
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Logic control:
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Active low |
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Max sink current:
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100 mA |
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Max supply voltage:
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40 V |
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Supply voltage for control I/O:
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5 V |
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Max current:
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750 mA |
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Power requirements:
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100-240 VAC, 47-63 Hz |
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Power consumption:
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15 Watts |
