Dante-enabled Product Catalog

Features: The D series converter box is compatible with all Dante interfaces and has all the features of Dante audio network. Each conversion box can provide high-fidelity sound quality restoration, ultra-low latency, and accurate synchronization of sampling rates across the entire network. Compact design, easy to use, plug and play; conversion between analog audio signals and Dante digital audio signals; expand audio and power through a network cable; fully support Dante software, easy to configure the network and audio routing; support PoE power supply, Built-in lightning and surge protection.

The network audio processing center is a network audio processor with 16 inputs (8 analog and 8 digital) and 16 outputs (8 analog and 8 digital), using the industry-leading ADI DSP SHARC21489 chip, The highest operating frequency can reach 450MHz. The input and output modules of the 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 needs of audio processors and transmissions in various places. Such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.

Features： 1. Automatically assign IP address through router DHCP; 2. Each receiving channel can be controlled independently, and it can be applied to any location in the local area network. 3. With backup analog input interface and Dante network interface, transmission double backup; 4. With a link output interface, analog audio and digital audio can be connected hand in hand; 5. Integrate network transmission, pure digital power amplifier and full-range speaker with RS-232 control and monitoring; 6. It has high efficiency and high sound energy output, excellent sound quality, perfect protection function and good electromagnetic compatibility characteristics; 7. It has perfect protection functions (short circuit protection, over current protection, over and under protection, DC protection, power limit, temperature limit, long-term output power limit, clipping limiter).

Features： Anti-cell phone, RF radio frequency interference. The embedded unit is powered by the host computer through the network cable (DC24V); The unit adopts network cable (Category 5 or Category 6) ring-shaped hand-in-hand connection Zinc alloy base is adopted, and the gooseneck is equipped with a green work indicator, which is intuitive and simple to operate and clear instructions; All buttons of the microphone are made of high-quality long-life materials, moisture-proof and dust-proof mute switch, to prevent the influence of noise when starting up; When the microphone is turned on, the switch will automatically emit a green light to indicate that it effectively prevents misoperation. All buttons adopt high-quality, long-life, moisture-proof and dust-proof switches.

1. Exquisite and elegant design, ergonomic, very modern. 2. Original D-net IOT digital conference IoT system technology, in line with IEC60914 international standard 3. With priority button function, you can turn off the representative unit that is speaking. 4. The microphone is equipped with an ultra-quiet electronic touch switch, which can avoid the mechanical sound amplification of the switch, and can control the operation of the microphone pickup. The microphone has a bright luminous halo to indicate the working status. 5. The unique audio processing circuit improves the clarity of the entire system. 7. The chairman unit can use the priority button (Priority) and the approval button (Permit) to control the speaking sequence and control the meeting atmosphere. 8. The unit comes with a 2m Category 6 RJ45 network cable; 10. Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software, so that speakers with different volumes can speak more smoothly; 9.1 Ø 3.5 mm stereo headphone jacks can be connected to headphones. 10. The unique ID number of the microphone unit, the conference system can manually assign the conference unit.

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. Wireless microphone pickup, exquisite pen microphone, automatic frequency linking at boot, simple operation; comes with 6+1 network ports, supports PoE, 100/1000M adaptive, 100 meters transmission distance, supports IEEE802.3af/at standard, single port The maximum power is 60W. Easily meet the high-quality voice needs of local sound reinforcement, recording and broadcasting, and remote interactive education.

The network audio processing center is a network audio processor with 8 inputs (4 analog and 4 digital) and 8 outputs (4 analog and 4 digital), using the industry-leading ADI DSP SHARC21489 chip, The highest operating frequency can reach 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 needs of audio processors and transmission in various places. Such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.

Features： 1. The integration of power over Ethernet and audio data transmission realizes simplified cable layout and conforms to Ethernet engineering specifications; 2. It has high efficiency and high sound energy output, excellent sound quality, perfect protection function and good electromagnetic compatibility characteristics. 3. The built-in DSP network active speaker can remotely monitor and control the EQ of the active speaker, the voltage and current at both ends of the speaker on the PC side. 4. The appearance of the all-metal cabinet is more compact, elegant, and the sound is transparent and bright. 5. Automatically assign IP address through router DHCP; 6. Each receiving channel can be controlled independently, and it can be applied to any location in the local area network 7. It has perfect protection functions (short circuit protection, over current protection, over and under protection, DC protection, power limit, temperature limit, long-term output power limit, clipping limiter). 8. The speaker uses Dante digital audio interface and PoE network power supply to drive the power amplifier module, and only one network cable is needed to realize the simultaneous supply of audio signal and power.

Features： 1. Comes with 2 RJ45 interfaces, adopts digital audio transmission and control technology, and has a good sound quality guarantee; 2. Using a detachable microphone pole, simple operation and convenient product maintenance. 3. Condenser microphone with green ring indicator light, easy to distinguish speaking and waiting status 4. In the entire conference system, the chairman unit has priority to speak, and can mute/close the speech of any representative unit that is speaking, so that the chairman unit can effectively control the order of the meeting place 5. Good anti-GSM interference ability 6. With video tracking

Features： Adopt all-digital conference technology, in line with IEC60914 international standard; The exquisite and minimal structure design highlights the modern atmosphere and conforms to ergonomics; Brand new dual microphone design, longer pickup distance and wider range; Special electret capacitive single microphone core design, clear and bright sound quality, high sensitivity; Original microphone directivity variable technology, users can adjust the microphone directivity according to different environments, locations, meeting methods, and the needs of the speaker, so as to achieve an optimal sound pickup effect, and support 5 modes: omnidirectional, heart -Type directivity, super-cardioid directivity, sharp-heart-type directivity, figure 8 directivity, etc.; Full digital audio technology, built-in high-performance CPU, faster processing speed, better sound quality; Ring hand-in-hand connection technology, the failure or replacement of a microphone will not affect the work of other extensions of the system, and a connection failure between the microphones will not affect the system work, thus making the system more reliable; Support line "hot swap", microphone access system can be added at any time The chairman unit has the priority function of full control of the conference order, and the connection position is not restricted. The chairman unit can short press the function button in the application mode to allow the delegate unit to turn on the microphone. In the conference mode, the chairman unit can short press the function button to turn off all the microphones of the delegate units. If there is a delegate unit waiting to be turned on, it will be turned on immediately. With the camera tracking host, after using the computer preset, the camera can be automatically tracked; There are 4 speaking modes: normal mode, first-in first-out mode, free mode, application mode; Two-color LED indicator, red when speaking; With the sign-in function, the host computer initiates and checks the sign-in situation; Anti-interference from mobile phone microphone core, stable connection, durable and reliable; Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software, so that speakers with different volumes can speak more smoothly; 1 Ø 3.5mm stereo headphone jack can be connected to headphones; The conference unit has a unique serial number, and the conference system can manually assign an ID to the conference unit.

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.

The BL-218SUB subwoofer is composed of two 18-inch high-performance, high-power, long-stroke woofers with 125-core voice coil 253 magnetic steel. The box structure adopts direct-fired design, combined with its own German audio technology, so that its low frequency performance is deep and powerful, plump and soft, thick and flexible, with excellent sound field tolerance, good overall lower limit, pure and natural sound, and full of musical rhythm. It has very good hearing resistance and provides listeners with a steady stream of low frequency experience. Cooperating with full-range speakers will achieve the most ideal effect. Applicable venues: Bars, Disco halls, slow shake bars, ultra-low frequency speakers for entertainment KTV rooms, nightclubs, dance halls, clubs, medium-sized mobile performances, main expansion ultra-low frequencies, conference halls, multi-function halls, main expansion ultra-low frequencies.

Features： 1. The integration of power over Ethernet and audio data transmission realizes simplified cable layout and conforms to Ethernet engineering specifications; 2. It has high efficiency and high sound energy output, excellent sound quality, perfect protection function and good electromagnetic compatibility characteristics. 3. The built-in DSP network active speaker can remotely monitor and control the EQ of the active speaker, the voltage and current at both ends of the speaker on the PC side. 4. The appearance of the all-metal cabinet is more compact, elegant, and the sound is transparent and bright. 5. Automatically assign IP address through router DHCP; 6. Each receiving channel can be controlled independently, and it can be applied to any location in the local area network 7. It has perfect protection functions (short circuit protection, over current protection, over and under protection, DC protection, power limit, temperature limit, long-term output power limit, clipping limiter). 8. The speaker uses Dante digital audio interface and PoE network power supply to drive the power amplifier module, and only one network cable is needed to realize the simultaneous supply of audio signal and power.

Features： 1. Good anti-GSM interference ability; 2. Using a detachable microphone pole, simple operation and convenient product maintenance; 3. Condenser microphone with double color ring indicator light, easy to distinguish speaking and waiting status; 4. Comes with 2 RJ45 interfaces, adopts digital audio transmission and control technology, and has a good sound quality guarantee; 5. It can realize video tracking. When any unit speaks, the video head will automatically turn to the set position. 6. The sound level of this unit can be individually adjusted on each unit to mute the sound. 7. In the entire conference system, the chairman unit has the priority to speak, and can mute/close the speech of any representative unit that is speaking, so that the chairman unit can effectively control the order of the meeting place.

1. Exquisite and elegant design, ergonomic, very modern. 2. Original D-net IOT digital conference IoT system technology, in line with IEC60914 international standard 3. With priority button function, you can turn off the representative unit that is speaking. 4. The microphone is equipped with an ultra-quiet electronic touch switch, which can avoid the mechanical sound amplification of the switch, and can control the operation of the microphone. The microphone has a bright luminous halo to indicate the working status. 5. The unique audio processing circuit improves the clarity of the entire system. 6. Fixed square array microphone pole with variable directivity. 7. The chairman unit can use the priority button (Priority) and the approval button (Permit) to control the speaking sequence and control the meeting atmosphere. 8. The self-developed technology of D-net IOT digital conference IOT system realizes five adjustable modes of directivity of the pickup microphone: omnidirectional, cardioid, super-cardioid, sharp-hearted, Figure- 8-pointing type; 9. Two-color LED indicator light, green when speaking; 10. The unit comes with a 2m Category 6 RJ45 network cable; 11. Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software, so that speakers with different volumes can speak more smoothly; 12.1 Ø 3.5 mm stereo headphone jack can be connected to headphones. 13. The unique ID number of the microphone unit can be manually assigned to the conference unit by the conference system.

The audio processing center is an audio processor audio processing center with 32 inputs (16 analog and 16 digital) and 32 outputs (16 analog and 16 digital). It is an audio processing center with 16 analog inputs and 16-channel analog output audio processor, using industry-leading ADI DSP SHARC21489, the highest operating frequency can reach 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.

BLA-10 line array system, it is a high-quality multifunctional full-range speaker with variable properties suitable for high-end multifunctional conference halls, small and medium-sized theaters and other high-quality sound reinforcement venues. It can be hoisted to form a line array system , You can also do full-frequency sound reinforcement alone.

Features： 1. The integration of power over Ethernet and audio data transmission realizes simplified cable layout and conforms to Ethernet engineering specifications; 2. It has high efficiency and high sound energy output, excellent sound quality, perfect protection function and good electromagnetic compatibility characteristics. 3. The built-in DSP network active speaker can remotely monitor and control the EQ of the active speaker, the voltage and current at both ends of the speaker on the PC side. 4. The sound column is made of wooden box body, sandblasted surface treatment, the sound is loud and powerful. 5. Automatically assign IP address through router DHCP; 6. Each receiving channel can be controlled independently, and it can be applied to any location in the local area network 7. It has perfect protection functions (short circuit protection, over current protection, over and under protection, DC protection, power limit, temperature limit, long-term output power limit, clipping limiter). 8. The speaker uses Dante digital audio interface and PoE network power supply to drive the power amplifier module, and only one network cable is needed to realize the simultaneous supply of audio signal and power.

Features： 1. Equipped with 5 groups of RJ45 loopback signal acquisition interface, "ring hand in hand" connection, the system is more reliable 2. Conventional network cable (Cat5e or Cat6) connection between each unit is convenient and quick 3. Each group of lines can connect 30 units, a single machine can connect 150 units, and an expander can be added to connect more units 4. Support WiFi for all control functions of the central processor 5. Support Dante network transmission, and additionally support three-way LINE/MIC input Dante network, which can be set through software 6. Remote control and remote assistance can be carried out through the TCP/IP network interface 7. Support meeting functions: discussion, video tracking 8. Support system hot swap 9. Support RS232 central control control, and have two-way code sending function, realize seamless connection with intelligent central control system 10. Support external audio input and recording output port 11. Support access port for conference call 12. The system can be set to lock or release the function of the microphone operation, and each unit has an electronic volume adjustment function. 13. Multiple speaking modes: first in first out, last in first out, automatic mode, discussion mode, chairman mode 14. With voice control to turn on the microphone function and speech time limit function (optional 1-999 seconds) 15. Connect to the computer to simulate the scene layout of the on-site meeting, and you can click to turn on or off all the unit microphones at any time 16. The host adopts LCD operation interface (3.5 inches), which can switch various national languages and display perpetual calendar

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

The network audio processor is a network audio processor with 28 inputs (12 analog and 16 digital) and 28 outputs (12 analog and 16 digital). It adopts the industry-leading ADI DSP SHARC21489, the highest The working frequency can reach 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.

Single 18-inch active subwoofer

Features： 1. The audio signal and power transmission of the speaker are realized through the same network cable. 2. The RJ45 interface on the back panel of the box is equipped with XLR firmware, which can be used for waterproofing and strengthen the connection between the box and the wire. 3. The network cable realizes signal transmission and power supply; 4. The IP address is automatically assigned through the router DHCP; 5. Each receiving channel can be controlled independently, and it can be applied to any location in the local area network 1. It has high efficiency and high sound energy output, excellent sound quality, perfect protection function and good electromagnetic compatibility characteristics. 7. The built-in DSP network active speaker can remotely control the EQ and pressure limit of the active speaker on the PC side. 8. It has complete protection functions (short circuit protection, over current protection, over and under protection, DC protection, power limit, temperature limit, long-term output power limit, clipping limiter). 9. The speaker uses Dante digital audio interface and PoE network power supply to drive the power amplifier module, and only one network cable is needed to realize the simultaneous supply of audio signal and power.

Product Description： 1. Use RJ45 network cable to connect in a "hand in hand" way; 2. Each connection box can connect 4 speaking units; 3. Use professional plugs to connect between the unit and the connection box 4. Hand in hand to connect to a box or directly connect to the conference host through the RJ45 network port interface; 5. The power supply is supplied by the host

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) and 220V AC mains dual power supply, digital signal, DSP control, a 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

The audio processing center is an audio processor with 24 inputs (16 analog and 8 digital) and 24 outputs (8 analog and 16 digital), using industry-leading ADI DSP SHARC21489, the highest operating frequency can reach 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.

The coaxial speaker series fully applies the latest sound reinforcement technology in the professional audio industry, adopts a point sound source coaxial structure design, and uses a constant directivity conical waveguide to load the diaphragm compression drive tweeter and low frequency unit, which can provide 80° axisymmetric constant directivity output. The main advantage of the application of coaxial technology is that the radiation of a single-point sound source has excellent phase response, avoiding the weakness of the typical polar lobe interference effect of the traditional horn tweeter and low-frequency speaker unit combination, thereby solving the problem of phase deviation. The wedge-shaped cabinet structure design enables the coaxial series of speakers to have two fixed angles of 30° and 60°, which can meet the requirements of long and short distance stage monitoring. The cabinet is made of high-quality plates, coupled with multi-process, environmentally friendly oil paint, sturdy and durable, integrated, compact cabinet design, perfect frequency response, phase response, first-class sound performance, excellent power output, suitable for FOH (Main sound reinforcement), FILL (auxiliary fill sound) and MONITOR (stage monitor) are used. Application: fixed installation, small mobile rental, theater, gymnasium, multi-function hall, studio, banquet hall, meeting room, academic lecture hall, small and medium-sized mobile performances, high-end entertainment and performing arts venues, etc.