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BARE – TNT-0808D

The audio processing center is an audio processor with 16 inputs (8 analog and 8 digital) and 16 outputs (8 analog and 8 digital). The industry-leading ADI DSP SHARC21489 is used, and the maximum operating frequency is up to 450MHz. The input and output modules of DSP can be customized according to the actual situation. At the same time, it has independent AFC, AEC, ANS, AGC, gain sharing automatic mixing, threshold automatic mixing and other processing modules to meet the audio processor and transmission needs of various places. Such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.
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ASL – Enchorus ENC 1102

The ENCHORUS ENC 1102 Dante™-based audio network module offers six AES3 outputs and six AES3 inputs with a four-band equalizer on each input and output. The ENC 1102 features eight built-in mixing engines and support for sample frequencies up to 192kHz. The device is easy to control and monitor through a web interface. A built-in audio matrix allows flexible routing of all inputs, outputs, and mixer signals. The ENCHORUS Dante™-based audio network was designed specifically for stage applications, offering professional connectors and superior audio quality. All ENCHORUS models include filtering and mixing capabilities and can be configured very rapidly via the integrated web interface. ENCHORUS modules are tough enough on their own but also include rubber bumpers that allow them to be easily and safely stacked on stage.
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  • 6x AES3 outputs
  • 6x AES3 inputs
  • Support of redundant Dante networks or daisy chainable
  • Four-band EQ for each input and output
  • 8x built-in mixing engines (16 inputs, stereo output)
  • Built in audio matrix for flexible routing of all signals
  • Control via web interface
  • Sample rates: 44.1kHz/48kHz/88.2kHz/96kHz/176,4kHz/192kHz
  • Sample rate converter on AES3 inputs from 8kHz to 192kHz
  • 19” rack-mounting kit available
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DCS-100M侧面
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BARE – DCS-100M

Features: 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 that the system can be used efficiently and normally. The system is more reliable and supports line "hot plugging", and microphones can be added to the system at any time 7. With the camera tracking host, after using the computer preset, the camera can automatically track and support the preset panoramic position 8. With D-net IOT digital conference IOT system independently researched and developed to realize five kinds of directional adjustable modes of pickup microphone: omnidirectional, cardioid directional, super cardioid directional, sharp-heart directional, Figure- 8-pointing type; 9. The system host supports dual backup output of analog and digital signals, and supports dual backup power supply of POE power supply and external 24V module; 10. Optional external 3.5-inch wall control touch screen for conference host management and settings, 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 11. The system can be connected to PC control to realize centralized microphone management. Through software, microphone pickup gain adjustment, forced switch, 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
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