Dante-enabled Product Catalog

Features： 1. PoE power supply; 2. Soft key-type user switch to avoid mechanical sound when switching the microphone; 3. The entire expandable Dante system has 512 two-way audio channels; 4. Use a network cable to connect to the switch; 5. There are a variety of switch modes to choose from, a variety of gains are available, and a low-cut switch; 6. Equipped with a 3-pin XLRF canon female socket, suitable for quick installation of any XLRM canon male output gooseneck microphone; The sturdy casing cast by die casting and the non-slip silicon foam at the bottom of the microphone holder can reduce the impact of desktop vibration on the microphone.

Features： 1. The wireless conference system eliminates cumbersome wiring, is quick and easy to install, and is simple to operate. It is especially suitable for use in scenarios such as project transformation, mobile conferences, temporary equipment increase, and wiring failure. The system adopts digital and analog fusion design, can realize the camera tracking function, embedded DSP audio processing technology, 4-inch touch screen management and settings, easy to use and intuitive, cooperate with the AOIP network audio transmission module to achieve audio signal interconnection, and the module supports mainstream networks Audio transmission protocol. 2. The wireless radio frequency technology transmits audio and digital control signals, with high-fidelity sound quality, no compression, clear and transparent, and the wireless control signal is reliable and stable; 3. Support camera tracking function, can connect to the camera through RS485, RS232, etc., support VISCA, PELCO-D/P camera control protocol, and can independently connect a camera for camera tracking; 4. It has a video switch RS232 communication port, which can be connected to a high-definition video switcher or a high-definition matrix, and supports up to 8 cameras for camera linkage; 5. Built-in high-speed DSP processor, based on artificial intelligence self-learning algorithm developed adaptive feedback suppression (AFC) and automatic noise reduction technology (ANC), deep learning analysis of the feedback signal that causes howling, and quickly eliminate the cause of howling Repetitive signals, learning and analyzing environmental noise, background noise and current sound, etc., to remove useless noise 6. The host supports a variety of conference modes: such as first-in first-out, chairman priority mode, the number of open microphones supports 1-4, and the system can carry up to 128 microphones; 7. The directivity can be adjusted with the microphone, supporting 5 modes: omnidirectional, cardioid directivity, super cardioid directivity, sharp cardioid directivity, figure 8 directivity, etc.; 8. Extend AOIP network audio transmission to support mainstream. The refined and elegant 4-inch touch screen is used for conference host management and settings, and can display the following information at the same time: a. Speaking mode selection, speaking quantity limit function b. Radio frequency grouping and radio frequency signal strength display c. AFC adaptive feedback suppression is turned on and off, d. ANC automatic noise reduction on and off e. System volume adjustment f. System settings and status view, switch between Chinese and English 6. Network transmission protocol, realize audio signal interconnection and intercommunication

The network audio processor is a network audio processor with 64 digital inputs and 64 digital outputs. It adopts the industry-leading ADI DSP SHARC21489 chip, and the maximum operating frequency can reach 450MHz. Meet the audio processor and transmission needs of various places, such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.

Single 12-inch line array bass (for use with L-205)

Features: The exquisite and minimal structural design highlights the modern atmosphere and conforms to ergonomics. Special electret capacitive dual microphone core design, clear and bright sound quality, high sensitivity; Support mainstream network audio protocols, use network cables to connect to the switch, and transmit audio in the network to realize signal interconnection and intercommunication; Using POE (Power over Ethernet) power supply and analog phantom power supply, digital and analog dual backup, the system is more stable. Provide standard balanced XLR interface to transmit analog audio: Provide standard RJ45 network interface to transmit digital audio: The entire expandable Dante system has 512 two-way audio channels; Original microphone adjustable directivity technology, supports 5 modes: omnidirectional, cardioid directivity, super cardioid directivity, sharp cardioid directivity, figure 8 directivity, etc. Equipped with 5 switch mode options: 1. Power-on mute mode, then press to speak; 2. Power-on speaking mode, and then press mute; 3. Long press the speaking mode to release the mute; 4. Long press the mute mode to release the speech; 5. , Voice control mode; Customizable remote control switch Anti-radio frequency and mobile phone interference; The base is equipped with damping rubber to effectively reduce the noise caused by external vibration to the microphone.

Features： A digital extender, when the conference system host has insufficient power supply or a long cable distance causes the signal to attenuate and cannot work normally, the use of the extender can achieve the function of power supply expansion and communication signal enhancement; Solve the problems of long-line transmission and signal attenuation, enable the system to carry more microphone units, and enhance the stability and reliability of the system.

The network audio processor is a network audio processor with 32 digital inputs and 32 digital outputs. It adopts the industry-leading ADI DSP SHARC21489 chip with a maximum operating frequency of 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.

The LA-AM12D compact line array module with its own power amplifier has a high-energy 12-inch ultra-low frequency unit with continuous directional coverage. Designed for simple system construction, it can provide the best listening experience in the shortest time. At the same time, the entire series is fully integrated with DanteTM. Using the speaker as the system carrier, it fully integrates the functions of signal transmission, processing, power amplification and remote monitoring, abandoning a large number of physical devices such as power amplifiers and peripheral processing equipment, and realizes the system transmission network, processing IT, and control intelligence.

Features： The exquisite and minimal structural design highlights the modern atmosphere and conforms to ergonomics. Special electret capacitive dual microphone core design, clear and bright sound quality, high sensitivity; Support mainstream network audio protocols, use network cables to connect to the switch, and transmit audio in the network to realize signal interconnection and intercommunication; Use POE (Power over Ethernet) for power supply; The entire expandable Dante system has 512 two-way audio channels; Original microphone adjustable directivity technology, supports 5 modes: omnidirectional, cardioid directivity, super cardioid directivity, sharp cardioid directivity, figure 8 directivity, etc. Equipped with 5 switch mode options: 1. Power-on mute mode, then press to speak; 2. Power-on speaking mode, and then press mute; 3. Long press the speaking mode to release the mute; 4. Long press the mute mode to release the speech; 5. , Voice control mode; Customizable remote control switch Anti-radio frequency and mobile phone interference; The base is equipped with damping rubber to effectively reduce the noise caused by external vibration to the microphone.

Features： • Adopt all-digital conference technology, in line with IEC60914 international standard; • Exquisite and minimalist structure design, highlighting the modern atmosphere, accord with 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; • The exquisite and elegant 3.5-inch touch screen is used for the information interaction between the speaker and the conference manager, and the following information can be displayed at the same time: a. Speaker information and time b. Microphone directivity setting c. Sign in and observe the voting status d, microphone gain adjustment e. Information interaction between the speaker and the conference administrator f. System language setting and unit information display • All-digital audio technology, built-in high-performance CPU, faster processing speed, better sound quality; • Digital and analog dual backup function to ensure that the system is stable and reliable, and every important meeting is uninterrupted; • Ring-shaped hand-in-hand connection technology, a microphone failure or replacement 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 representative unit to turn on the microphone; • The chairman unit has two touch buttons, no mechanical button sound, long life, and waterproof function, one is to switch the microphone button, the other is the function button; • 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; • The unit comes with a 1.8-meter dedicated 8-core cable for conferences, and a 1-meter analog backup audio cable (with a Deng Nong head); • Anti-cell phone interference microphone core, stable connection, durable and reliable; • 1 Ø 3.5 mm stereo headphone jack for connecting headphones; • The conference unit has a unique serial number, and the conference system can manually assign an ID to the conference unit.

The network audio processor is a network audio processor with 32 inputs (16 analog and 16 digital) and 32 outputs (16 analog and 16 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. It has high efficiency and high sound energy output, excellent sound quality, perfect protection function and good electromagnetic compatibility characteristics. 2. Array full-frequency sound column with network transmission and pure digital power amplifier. (Equipped with special independent speaker control software), adjust the direction and angle of direct sound (pre-set 0°, ±15°, ±45°, ±60°, bypass) 3. With DSP network active sound column, you can remotely perform EQ, various filter switching, delay, compression, group mixing and other operations on 8 groups of units on the PC side 4. 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). 5. The sound column uses Dante digital audio interfaces (2) and 220V power supply to drive the power amplifier module. It also has a XLR interface with one input and one output, which can be multiple cascaded, and is equipped with two control network ports and a USB interface for The computer adjusts the 8 units of the sound column. 6. The back panel has a display screen and physical buttons, which can switch the sound column input mode, preset effect mode, etc., device TP display mode, lock screen and other operations

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, sign-in, and voting functions.

Features： 1. Adopting all-digital conference technology, in line with IEC60914 international standards; exquisite and minimalist structure design, highlighting the modern atmosphere, in line with ergonomics; special electret capacitive dual microphone core design, clear sound quality, high sensitivity; 2. The self-developed technology of D-net IOT digital conference IOT system realizes five adjustable directional modes of pickup microphones: omnidirectional, cardioid directional, super-cardioid directional, sharp-heart directional, Figure- 8-pointing type; 3. The exquisite and elegant 3.5-inch touch screen is used for the information interaction between the speaker and the conference manager, and can display the following information at the same time: a. Speaker information and time b. Microphone directivity setting c. Sign in and observe the voting status d, microphone gain adjustment e. Information interaction between the speaker and the conference administrator f. System language setting and unit information display 4. Ring-shaped 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; 5. Support line "hot swap", microphone access system can be added at any time 6. With the camera tracking host, after using the computer preset, the camera can be automatically tracked The 7.3.5-inch wall control touch screen 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. 8. Two-color LED indicator light, green when speaking; 9. The unit comes with a 1.8-meter conference dedicated 8-core cable, and a 1-meter analog backup audio cable (with a T-Nong head) 10. Anti-RF microphone core of mobile phone, stable connection, durable and reliable; 11.1 Ø 3.5 mm 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.

The network audio processor is a network audio processor with 32 inputs (16 analog and 16 digital) and 24 outputs (8 analog and 16 digital), using the industry-leading ADI DSP SHARC21489, 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.

A network active speaker is an active full-range speaker that integrates network transmission, pure digital power amplifier and RS-232 control and monitoring. It has high efficiency and high sound energy output, excellent sound quality, perfect protection function and good electromagnetic compatibility characteristics. It has a backup analog input interface and a Dante network interface, which can transmit dual backups. It can be connected to the built-in Dante board through a network cable to realize network audio signal transmission. The IP address is automatically assigned through the router DHCP, and the signal is transmitted with the audio matrix with dante function in the local area network, and a set of all-digital network sound reinforcement system can be easily formed. It is suitable for classrooms in primary and middle schools, small and medium-sized meeting rooms, corridor broadcasting and other places.

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. The chairman unit can initiate sign-in and voting operations. 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.

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

DT22 is a compact 48kHz/96kHz Dante interface enabling convenient placement of 2 mic/line inputs and 2 line outputs wherever you need to get analogue audio in and out of a Dante network. Equipped with standard Phoenix / euroblock connectors, DT22 is designed for fixed installation and can be easily mounted under a meeting table, for example in a breakout room or huddle space, under a podium for local mic/PC and speaker connection, or rack-mounted to interface with analogue equipment such as an AV switcher.

The compact and portable DT02 converts two channels of Dante audio to a pair of XLR line-outputs, enabling simple break-out to analogue equipped hardware for install and live applications.

The compact and portable DT20 converts two analogue inputs, via combo XLR-Jack, to two channels of Dante audio, enabling hassle-free connection of analogue sources to a Dante network for install and live applications.

The Clockaudio Dante Transporter is an advanced professional Dante communication product which combines audio and control transmission across a standard IP network. Based on high performance and pre-configurable hardware platform it guarantees perfect sound quality. The CDT 100 Mk3 has a variety of uses including but not limited to boardrooms, video and conference rooms, or any place else where multiple microphones and control devices are connected back to the AV equipment cabinet.

Features:

• Easy to install with under table mounting

• Dante Audio and control signals transported over standard IP network

• 4 audio inputs, with gain control, high pass filters and Phantom Power

• 2 line-level outputs with high pass filters

• Up to 6 units may be daisy chained

• 4 Touch Switch ports compatible with all logic enabled Clockaudio devices.

• Open API for interfacing with Control Systems

• Compatible with any Dante capable audio DSP

The DD-RN42 is a complete wall-mounted Dante audio network interface. It features two XLR mic or line inputs on the front panels, two balanced line inputs on a rear-panel detachable terminal block and two balanced line outputs on front-panel XLR jacks. The module is configured and adjusted using RDL CONSOLE or other network control software. The DD-RN42 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-RN42 is PoE powered, and is available in multiple finishes with optional customized graphics.

Mic/Line to network section

The two XLR inputs are each converted to a separate Dante network transmit channel. Three gain settings are software-selectable for both the mic and line input ranges to match condenser or dynamic mic levels and standard line levels. P48 phantom voltage may be software enabled when an input is set for mic level.

Line to network section

The two rear-panel balanced or unbalanced inputs are each converted to a separate Dante network transmit channel. Control software sets the gain for an input level of either +4 dBu or -10 dBV. RDL Decora®-style jack plates matching the appearance of the DD-RN42 are optionally available for connection of various audio sources to the rear-panel inputs. An input plate may be located together with the DD-RN42 in a triple gang box or may be mounted remotely in a single box.

Network to line section

Two Dante audio channels are converted to balanced line level feeding front-panel XLR jacks. Each output provides +4 dBu balanced for a network digital audio level of -18 dBFS. Each output is equipped with a software controlled attenuator adjustable in 1 dB steps to a maximum attenuation of 62 dB then OFF. The outputs are connected through a detachable terminal block.

Valid PoE power and synchronization to the Dante network are indicated by green LEDs visible from the front of the unit.

The DD-RN42 is a professional grade product with RDL mic preamplifiers for studio quality fidelity and low noise performance, coupled through XLR jacks housed in a steel chassis with dual powder-coated or stainless steel Decora front plates.

The DD-RN42’s superior performance specifications make it ideally suited to installations requiring network control and 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.

The DD-RN40 is a complete wall-mounted Dante audio network interface. It features four XLR mic or line inputs on the front panels and two line outputs on a rear-panel detachable terminal block. The module is configured and adjusted using RDL CONSOLE or other network control software. The DD-RN40 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-RN40 is PoE powered, and is available in multiple finishes with optional customized graphics.

Mic/Line to network section

The four XLR inputs are each converted to a separate Dante network transmit channel. Three gain settings are software-selectable for both the mic and line input ranges to match condenser or dynamic mic levels and standard line levels. P48 phantom voltage may be software enabled when an input is set for mic level.

Network to line section

Two Dante audio channels are converted to balanced line level. Each output provides +4 dBu balanced for a network digital audio level of -18 dBFS. Each output is equipped with a software controlled attenuator adjustable in 1 dB steps to a maximum attenuation of 62 dB then OFF. The outputs are connected through a detachable terminal block.

The outputs are intended for connection to RDL AMS connectors mounted in Decora®-style plates that match the DD-RN40. AMS audio connectors include RCA, Mini and XLR jacks. The output plate may be located together with the DD-RN40 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-RN40 is a professional grade product with RDL mic preamplifiers for studio quality fidelity and low noise performance, coupled through XLR jacks housed in a steel chassis with dual powder-coated or stainless steel Decora front plates.

The DD-RN40’s superior performance specifications make it ideally suited to installations requiring network control and 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.

The DD-RN31 is a complete wall-mounted Dante audio network interface. It features two XLR mic or line inputs, one stereo Mini-jack line input, left and right RCA jack inputs and one stereo Mini-jack line output on the front panels, plus two line outputs on a rear-panel detachable terminal block. The module is configured and adjusted using RDL CONSOLE or other network control software. The DD-RN31 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-RN31 is PoE powered, and is available in multiple finishes with optional customized graphics.

Mic/Line to network section

The two XLR inputs are each converted to a separate Dante network transmit channel. Gain settings are software-selectable for both the mic and line input ranges to match condenser or dynamic mic levels and standard line levels. P48 phantom voltage may be software enabled when an input is set for mic level.

Line to network section

The Mini-jack and RCA jack left and right inputs each provide a network digital audio level of -18 dBFS for an unbalanced -10 dBV input. The control software allows each of these inputs to feed either or both of two Dante channels. Multiple sources fed to a single Dante output are summed.

Network to line section

Two Dante audio channels are converted to unbalanced line level to feed the front-panel Mini-jack. Each channel provides -10 dBV unbalanced for a network digital audio level of -18 dBFS.

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.

Valid PoE power and synchronization to the Dante network are indicated by green LEDs visible from the front of the unit.

The DD-RN31 is a professional grade product with RDL mic preamplifiers for studio quality fidelity and low noise performance, coupled through XLR jacks housed in a steel chassis with dual powder-coated or stainless steel Decora® front plates.

The DD-RN31’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.

The D SERIES-BTN44 modules are Dante networked, wall-mounted audio interfaces that include bidirectional Bluetooth audio, unbalanced analog stereo or mono audio inputs and outputs, and a balanced or unbalanced line-level output. Stereo audio from a compatible Bluetooth-enabled device feeds two Dante network audio channels. Stereo audio connected to a front-panel Mini-Jack or L and R RCA jacks feeds two other Dante network audio channels.

Stereo audio is available to users through a front-panel Mini-Jack fed from two Dante network channels. A rear-panel terminal block provides either a balanced (+4 dBu) or unbalanced (-10 dBV) output from one Dante network channel. That output may feed a variety of RDL wall panels to provide a specific mic or line level output to users, or it may be used to feed the audio power amplifier in the associated audio zone. One Dante channel may be routed back to a connected device through the Bluetooth interface enabling full duplex communication.

The D SERIES-BTN44 modules are powered from PoE. The installer can customize the module name and programming password through the Bluetooth connection using a smartphone, computer or tablet terminal program. These modules feature studio-quality circuits and analog filters to provide audio performance exceeding expectations from consumer interface technologies. High performance, simplicity of operation and efficiency of installation make RDL the best value in professional Bluetooth interfaces.

The AV-NL4 is a line-level interface that converts four Dante network audio channels to four balanced audio outputs on XLR connectors. Each channel provides +4 dBu balanced for a network digital audio level of -20 dBFS. Each output is equipped with a selector to set the output to line level or attenuate the output by 50 dB to mic level. Special software is not required to configure the module. The AV-NL4 is a professional grade product for studio quality fidelity and low noise performance.

The AV-NL4 may be powered from an external 24 Vdc power supply, available separately. The AV-NL4 is equipped to operate from a PoE enabled network switch. The AV-NL4 will reserve power from the switch even while being powered from an external 24 Vdc supply. If PoE power and an external 24 Vdc supply are both feeding an AV-NL4, the unit will run from the external supply and will seamlessly switch over to PoE power upon loss of the external supply. Valid power and synchronization to the Dante network is indicated by a green LED visible from the front of the unit.

The AV-NL4 is constructed in a durable, professional all-metal enclosure suitable for use on stage or in studio, or mounted to any flat surface. For fixed installations, mounting brackets are included. The chassis is designed to protect the front panel slide switches from accidental damage. This full-featured AV-PAK product is engineered and manufactured in the U.S.A for continuous duty in demanding installations.