Dante-enabled Product Catalog

Dante network Digital audio Adaptor for conversion of 2 Dante signals into 2 line level analogue signals and vice versa.

DX32R is your versatile digital audio Swiss knife - an interface and matrix designed for multiple audio applications such as multi-room recording studios, post-production facilities and audio distribution installations. It allows you to route all digital input and output and manage them on a mono channel basis within a single unit or a combination of units.

With near zero latency, the DX32R can be used to expand I/O channel count and interface digital components, providing a high quality bridge between AES/EBU, MADI and the Dante IP audio network.

DX32R include a 384x384 multiplexer, enabling all digital inputs and outputs to be routed in any combination; this enables the unit to also split signals for advanced signal distribution.

The DX32R IP Audio protocol is based on the robust tried-and-tested Dante™ digital audio network technology, and will interoperate with products of other brands that comply with Dante.

The protocol is compliant with the AES67 standard for AoIP. The Dante and NTP IP Audio formats provide fast, flexible, and economical audio routing via IP and are compatible with the AX32 AD/DA converter, and other Dante devices. The IP Audio routing provides low latency, tightly synchronized, transport of uncompressed audio over Gigabit IP Ethernet Layer 3 networks using off the shelf switches, and routers for audio routing via one or more sub-nets.

A total of 512 channels can be routed on a 1 Gigabit network, and more if the network capacity is higher.

DX32R is TCP/IP controlled via one of the two Ethernet ports vi the DADman Control software available for both Windows and OSX.

The routing of the IP audio via Dante is done by using the Audinate Dante controller software.

The world's first paperless multimedia congress system, developed by TAIDEN as early as in 2010, has ushered in a new era for the conferencing industry and has been used at many high-profile international conferences and meeting venues, testifying in return to its excellence and superiority.

Not stopping there, TAIDEN later launched its third generation paperless multimedia congress system in 2016. The new system boasts of superb HD paperless

experience, greater system reliability, budget-conscious integration of functions and convenient management with its ultra-thin design, larger capacitive touch screen and higher megapixel camera. Either tabletop or flush-mounted model provides its users with faster, more convenient and efficient operation experience. Another innovative and brand-new member — the dual microphone paperless multimedia congress terminal with an extra backup microphone is also added to the growing product line.

The world's first paperless multimedia congress system, developed by TAIDEN as early as in 2010, has ushered in a new era for the conferencing industry and has been used at many high-profile international conferences and meeting venues, testifying in return to its excellence and superiority.

Not stopping there, TAIDEN later launched its third generation paperless multimedia congress system in 2016. The new system boasts of superb HD paperless

experience, greater system reliability, budget-conscious integration of functions and convenient management with its ultra-thin design, larger capacitive touch screen and higher megapixel camera. Either tabletop or flush-mounted model provides its users with faster, more convenient and efficient operation experience. Another innovative and brand-new member — the dual microphone paperless multimedia congress terminal with an extra backup microphone is also added to the growing product line.

Features

• Two Mic/Line Terminal Block Inputs on Rear Panel.

• Converts Mic or Line Audio Signals to 2 Channels Dante Network.

• Switchable Mic/Line/48V Phantom Power for Each Input.

• Terminal Block for Two Audio Outputs on Rear Panel

• Two Audio Signal Outputs from Dante Network.

• Converts 2 Channels Dante Signals to Two Line Terminal Block Outputs.

• Balanced or Unbalanced Signals for Inputs and Outputs.

• Adjustable Gain Control for Output Level on Rear Panel.

• LED indicators for Signal Level and Clip of Each Input Channel on Front Panel.

• Adjustable Gain from -20 dB to +20dB for Each Input Channel on Front Panel.

• LED indicators for Signal Level of Each Output Channel on Front Panel.

• Power over Ethernet (PoE) or 24VDC power supply

• LED Indicators Show the Status of Power and Network.

• High Resolution 24 Bit for DAC and ADC

• Provides High Quality of Preamplifier for Microphones.

• Dimension: 143mm x 45mm x 93mm (WHD)

Blocks are designed for:

connecting analog or digital AES3 audio signals to sound studios and audio mixers using the AES67 or Dante protocol;

transmission of audio signals over Ethernet 100/1000 BaseT;

performing the function of an 8x8 audio switch and creating a distributed network of switches with a common switching field, limited network bandwidth (one audio channel takes about 1 Mbps).

Supports up to 16 channels of balanced analog audio (8 inputs and 8 outputs) or up to 8 channels of AES 3

digital audio (4 inputs and 4 outputs). The type of audio signals (' xxxx ') for inputs and outputs is selected by the user in the device code independently of each other. The first pair of ' xx ' defines the input signals and the second pair of ' xx ' the output signals (' A ' indicates 2 analog stereo signals; ' E ' indicates 2 AES digital signals ). An additional Ethernet port provides connection to a redundant ( Redundant )) audio transmission networks or device cascading. Management and monitoring is carried out over an Ethernet network using a proprietary program Dante Controller . HDB44 audio connectors.

The Model 392 is designed for a variety of applications that require a visual status indicator. An array of red/green/blue (RGB) LEDs provide backlighting for a distinctively shaped polycarbonate (plastic) lens assembly. The Model 392 can serve as a broadcast "on-air" light, a room occupied display, or an intercom call signal indicator. In addition, the unit can serve as an audio level display, with a green, yellow, and red indication of audio level. Full operation requires only a Power-over-Ethernet (PoE) 100 Mb/s data connection. It's compatible with the popular Dante® audio-over-Ethernet protocol but offers support for a range of additional applications. Configuration is performed using the Studio Technologies' STcontroller software application. Selected configuration values are stored within the Model 392's non-volatile memory. A variety of methods are provided to allow control of the Model 392's LED array. These include using STcontroller's "virtual" button, receiving network-transported UDP commands, and responding to an audio signal level associated with a Dante digital audio connection.

Key Features

Dante audio-over-Ethernet technology

General-purpose visual indicator with RGB LEDs

On, flashing, and audio level displays

Responds to high-frequency (18-23 kHz) call, UDP commands, and PCM audio data

Mounts in a US-standard 2-gang electrical box

Single Decora-style lens opening dimensions

Configured using STcontroller application

Power-over-Ethernet (PoE) powered

The AES16O-D is a 16 Channel Dante to AES output converter.

Housed in a 1U unit with dual redundant Internal power supplies as standard.

8 x XLR outputs for 16 Channels of AES

Primary and Secondary Dante connections

44.1/48/88.2/96K Sample Rates.

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

Features: ●The high-quality pre-op amplifier guarantees the output sound quality and controls the output volume of each channel ●Advanced digital reverberation modulation function ●Twelve microphone input sockets. Independent phantom power supply for each channel, independent input volume adjustment, independent switch indicator ●The device itself provides a mixing channel, mixing input twelve microphones (each microphone can adjust the volume of the output to the mixing channel), while controlling the final output mixing volume ●Each input channel and mixing channel have an independent output, and the input channel and mixing channel can be monitored locally ●The equipment allows nearby installation, and the network transmits the audio of independent channels and mixed channels to the remote power amplifier or mixer ●With device superimposition function (when there are more than twelve microphones, two sets of this device can be superimposed and used) ●Twelve channels of microphones can be adjusted individually, which can also use various types and indicators of microphones to ensure that the microphone volume input and output in the system are balanced ●Only one audio equipment of the power amplifier, with this equipment can drive multiple, even dozens of microphones to work ●The panel displays the network status, which is convenient for fault diagnosis ●RS232/RS485 remote control interface reserved ●Only one network port, daisy chain connection is not allowed ●The network IP of the device can be set independently ●Microphone connection matching: phantom or non-phantom powered microphone ●Microphone connection: 12 XLR sockets

The network audio processing center is a network audio processor with 20 inputs (12 analog and 8 digital) and 20 outputs (12 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.

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.