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DCS-101主机

I/O Interfaces

BARE – DCS-101DMX

1.
Original D-net IOT digital conference IoT system technology, in line with
IEC60914 international standard
2. Support 41kHz/48kHz audio sampling rate, long-distance transmission will
not have any impact on sound quality
3. The host computer and the computer can be connected through TC/IP or
network access WiFi to realize wireless control mode, and the host can
realize IP-based address allocation to realize distributed application
management
4. D-net IOT technology can realize the flexible completion of networking
or independent partitioning of multiple conference systems, and a variety of
conference fields can be used for expansion
5. Dante network audio interface, support to match any Dante audio
equipment, and realize computer host computer software control and I Pad
software remote control to set various parameter values through the use of
TCP/IP connection;
6. Support Ethernet (POE) power supply, digital signal, DSP control, one
cable digital link system unit adopts loop loop connection (1 output, 1
backup input) to ensure the efficient and normal use of the system.
7. The system supports line “hot plugging”, and microphones can
be added to the system at any time
8. With the camera tracking host, after using the computer preset, the
camera can automatically track and support the preset panoramic
position
9. The self-developed technology of D-net IOT digital conference IOT system
realizes five adjustable modes of directional microphones: omnidirectional,
cardioid directional, super-cardioid directional, sharp-heart directional,
Figure- 8-pointing type;
10. The system host supports dual-backup output of analog and digital
signals, and supports dual-backup power supply for POE power supply and
external 24V module;
11. Optional external 4-inch wall control touch screen for conference host
management and settings, which is convenient and efficient, and can display
the following information at the same time:
a. Time-limited mode: The speech time limit can be set to 1-60 minutes, and
the speech will be automatically closed when the time is up. After reaching
the power-on quantity, when another representative unit wants to speak, it
will automatically enter the request-to-speak registration queue.
b. First-in-first-out mode: First-in-first-out speaking mode (factory
preset position), after reaching the number of power-on, the speaking unit
opened first is closed by the subsequent speaking unit.
c. Application for speaking mode: the speaking of the representative unit
needs to be approved by the chairman unit or the operator. In this mode, the
chairman unit always keeps the normally open state.
d. First-in-first-out mode; you can choose 1/2/4/6 microphones to speak at
the same time, and the chairman microphone and VIP microphone can be preset
without restriction.
e. Volume adjustment of main output
12. The system can be connected to PC control to achieve centralized
microphone management. Through software, microphone pickup gain adjustment,
forced switching, speaker information registration, can initiate sign-in
function, detect microphone line conditions, software set VIP unit, VIP unit
The switch is not limited by the number of conference units that are open at
the same time and the speaking mode
13. Network-based collaborative management, the system host can set the IP
address, and use advanced TCP/IP network connection between the control PC
and the host, such as the network access WIFI, which can realize the wireless
control and management of the PC and the host, and realize the distributed
Application and management.
14. Input and output interface:
4-channel balanced microphoneline input, using phoenix plug
interface;
4-channel balanced audio output, using Phoenix plug interface;
4-channel network DANTE audio input, using RJ45 network interface;
4-channel network DANTE audio output, using RJ45 network interface;
Support up to 48K/96K, 24BIT uncompressed CD-level sound quality;
15. Provide two wireless UHF ultra-high frequency handheld microphones for
auxiliary pickup.
16. Built-in feedback suppressor, noise cancellation, echo cancellation and
other main algorithms;
17. A variety of control methods, which can be managed by computer, central
control, touch panel, etc.;
18. Provide 6 100/1000M adaptive RJ45 ports, single port maximum power 60W,
automatic detection and identification of standard PoE power receiving
equipment;
19. Provide 1 100/1000M adaptive RJ45 port as uplink port;
20. Provide 1 Gigabit SFP optical fiber module expansion slot;
21. With a variety of audio processing functions such as; 31-band
equalizer, 13-parameter equalizer, compressor, limiter, expander, crossover,
crossover, mixer, filter, noise gate, dumb Sounder, automatic gain, signal
indicator and signal generator, etc.;
22. Multiple conference hosts can be used as independent conference
systems, or can be easily expanded to form a large-scale conference system to
achieve flexible configuration of multiple conference rooms
23. Dante interface, transmit audio signals through the network, low delay,
high efficiency, realize real-time audio interconnection in large conferences
or multiple conference rooms, and support mainstream network audio protocols

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