iTrans DB22


More Details

Similar Products

DCS-101主机
  • BARE

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
    I/O Interfaces
Dante官网logo
  • BARE

BARE – TX-D

A network media matrix mainly used for local meetings, education and remote interactive education. It is equipped with excellent digital audio processing technology, eight-channel audio processing, including full-band adaptive echo cancellation, automatic noise suppression, and feedback suppression algorithms. Long-distance pickup microphone, easy to install, simple to operate, comes with 6+1 network ports, supports PoE, 100/1000M adaptive, 100 meters transmission distance, supports IEEE802.3af/at standard, and the maximum power of a single port is 60W. Easily meet the high-quality voice needs of local sound reinforcement, recording and broadcasting, and remote interactive education.
    I/O Interfaces
directout-prodigy-mp
  • DirectOut

PRODIGY.MP

PRODIGY.MP is a multifunction audio processor that is designed to address numerous applications in pro audio, broadcast, installation and studio applications. PRODIGY.MP provides flexible I/O, dual networked audio, DSP (EQ, Delay, Limiter, Matrix/Mixing, Inserts…), Sample Rate Conversion, multiple sync generation/handling/distribution and powerful hard- and software.

Housed in a 2U rack frame, it supports networking selectable from Dante, RAVENNA and SoundGrid, while remote options allow access to hardware settings, DSP, SRC, network and a channel-based routing matrix. Fully modular, the hardware can be configured to specific requirements, giving budget savings as well as the flexibility to address changes in the global media environment through future upgrades.

Modular Approach, Maximum Flexibility

I/O includes four local slots for analog line-level/mic input and AES3 modules. With eight channels per module and interface direction, PRODIGY.MP provides up to 32 local inputs and outputs.

The hardware additionally supports two MADI and two network audio options. MADI slots can be equipped with BNC, SC optical and SFP modules, while network audio boards add Dante, RAVENNA/AES67 or SoundGrid, and increase channel capacity to 416 inputs/420 outputs.

DSP, SR & Sync management

With FPGA-based processing, user-definable processing paths and pick-off points, analog and digital inserts, outputs and summing matrices, PRODIGY.MP boasts very low latency DSP for EQ (FIR and IIR filters), Time Adjust/Delays and Limiters. The internal DSP reference can be set to Word Clock or Video input, any network/digital input, or internally generated clock. Sample Rate Converters on AES inputs and free-running MADI and network I/Os are available, allowing simultaneous Media/Format/Sample Rate and Sync conversion, DSP and full routing at up to 192 kHz.

Control Is Key, Multiple Access Locally & Remotely

globcon remote control, a browser-based GUI, and a backlight front panel IPS touch display ensure ultimate usability. Where globcon software offers integrated control of multiple pieces of equipment in Mac/PC/Linux-based systems, an integrated web server supports remote access via HTML and JavaScript-based user interfaces.

The browser-based GUI allows immediate remote access without the need to install software. The management network port is independent of the audio network, and can employ a built-in switch for in-band management of the device.

Third-party remote control protocols can extend system integration further. Automated surveillance and red light control is possible through GPIOs, while standard MIDI input and outputs give communication and active/passive control via MIDI. USB ports on the front and rear panels for connection and control of legacy devices.

Networked Audio & Seamless Interoperability

PRODIGY.MP’s network audio modules accommodate audio formats including DirectOut’s AES67-compliant RAVENNA stack that supports SMPTE ST2110-30/31 and ST2022-7 for redundant streaming. With the 64-channel Dante option, the hardware can be a front-end for audio consoles, or a standalone converter, DSP, SRC and routing system.

Waves Audio’s SoundGrid offers extremely low latency, and 128 channels of audio processing for live sound, studio and broadcast. Waves servers can be connected via SoundGrid for plug-in inserts. Virtual soundcards for macOS and Windows can directly interface with any DAW for multitrack recording and virtual soundchecking.

Secure Operation, Monitoring

Two front panel headphone outputs provide monitoring of any I/O at the device. Two phase-redundant power supplies with separate IEC inlets and AC power switches support hardware redundancy. An internal switch on the network module also ensures redundancy for the network domain. While the Dante module supports Dante redundancy, the RAVENNA module offers stream redundancy in accordance with ST2022-7. EARS (Enhanced Automatic Redundancy Switching) is available for several combinations of inputs.

    I/O Interfaces
  • Dante Ch: 64x64
  • 24-bit