ProCo – AoDDP Dante Wall Plates


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  • BARE

BARE – DCS-200M

Features: 1. This product uses a high-performance microcomputer processor as the core design of a new generation of products, equipped with a high-definition and sensitive 3.5-inch OLED touch screen. Adopting all-digital conference technology, conforming to IEC60914 international standard, professional, novel, simple appearance design, and comprehensively adopting the perfect combination of digital control technology, advanced audio processing technology and network technology, intelligent control, reaching the requirements of high-definition CD sound quality, which can be widely used Used in various occasions: large and small meeting rooms or multi-function halls, small and medium-sized performance occasions, hotels, conferences, public security, courts and other government departments. 2. 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 ID allocation address to realize distributed application management 3. With the camera tracking host, after using the computer preset, the camera can automatically track and support the preset panoramic position 4. 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; 5. The 3.5-inch wall-control touch screen or even a PC is used 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 f. Centralized microphone management, such as microphone pickup gain adjustment, forced switch, and speaker information registration g, can initiate voting and sign-in functions h. Interconnection with speaking microphones, such as tea, pen and paper, etc. i. Check the microphone line j. VIP unit can be set through software, and the switch of VIP unit is not limited by the number of conference units that can be opened at the same time and speaking mode 6. Built-in self-learning algorithm based on artificial intelligence has developed adaptive feedback suppression (AFC) and automatic noise reduction technology (ANC), deep learning analysis of feedback signals that cause howling, and elimination of ring-shaped hand-in-hand connection technology that causes howling. The failure or replacement of a microphone will not affect the work of other extensions in the system, support the line "hot swap", you can add microphones 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. Built-in real-time clock, perpetual calendar function, can display the current time on the conference host, conference terminal, convenient for participants to view 9.2 channels of audio external line input and 2 channels of audio line output, 1 channel of balanced output 10.4 channels of 8-core aviation connector can form 2 groups of hand-in-hand loop wiring to ensure a more stable system
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Core_X10
  • Q-SYS

Server Core X10

Network I/O or paired with distributed processing. Features & Benefits
  • 256 x 256 networked audio channels (Q-LAN / AES67)
  • 64x AEC channels @ 200 ms
  • Includes 8 x 8 Software-based Dante channels (licensable up to 128 x 128)
  • Supports up to twelve (12) Q-SYS NM-T1 tabletop microphones
  • Up to 32 VoIP softphone instances
  • 64 x 64 Media/WAN channel capacity
  • 16 multitrack playback channels (up to 128 with optional stackable feature license)
  • 4 multitrack record channels
  • Two (2) 2.5 Gbps Ethernet ports for redundant networked audio
  • Two (2) 2.5 Gbps independent, auxiliary Ethernet ports for segregation of network services such as VoIP, SNMP, LLDP, LDAP and WAN Media streaming
  • Onboard 256 GB media drive
  • 1 RU form factor
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  • Dante Ch: 8x8
  • 48kHz
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  • RDL

RDL – DDS-BN2M

The DD-BN2M is a complete wall-mounted Dante audio network interface. It features two XLR mic or line inputs on the front panels, plus two line outputs on a rear-panel detachable terminal block. The DD-BN2M is a complete wall-mounted Dante audio network interface. It features two XLR mic or line inputs on the front panels, plus two line outputs on a rear-panel detachable terminal block. Special software is not required to configure the DD-BN2M. Each XLR input provides three switches that may be set from the front of the unit when the cover plate is not installed. One switch enables or disables P48 phantom; the second switch selects the mic or line gain range; the third switch sets the gain. Each rear-panel output provides a switch to set the output to balanced professional or unbalanced consumer level. The DD-BN2M fits a standard US dual-gang electrical box or an RDL WB-2 back box for installations in thinner European or equivalent walls. The DD-BN2M is PoE powered, and is available in multiple finishes with optional customized graphics. The two XLR inputs are each converted to a separate Dante network transmit channel. Three gain settings are switch-selectable for both the mic and line input ranges to match condenser or dynamic mic levels and standard line levels. Two Dante audio channels are converted to balanced line level on the rear-panel detachable terminal block. Each output provides +4 dBu balanced for a network digital audio level of -18 dBFS. Each output is equipped with a selector to unbalance the audio and attenuate the level to -10 dBV. These switches are located on the top of the chassis and are set by the installer prior to mounting the unit. The rear-panel outputs are intended for connection to RDL AMS connectors mounted in Decora-style plates that match the DD-BN2M. AMS audio connectors include RCA, Mini and XLR jacks. The output plate may be located together with the DD-BN2M in a triple gang box or may be mounted remotely in a single box. Valid PoE power and synchronization to the Dante network are indicated by green LEDs visible from the front of the unit. The DD-BN2M is a professional grade product with discrete mic preamplifiers for studio quality fidelity and low noise performance, coupled through metal XLR jacks housed in a stainless steel chassis with dual powder-coated or stainless steel Decora front plates. The DD-BN2M’s superior performance specifications make it ideally suited to the most demanding installations, and an exceptional value in commercial networked audio systems. This full-featured product is engineered and manufactured in the U.S.A for continuous duty in demanding installations. Designed to outperform. Built to last.
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