Dante-enabled Product Catalog

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.

A pair of PBX series devices allows you to implement transmission over a single fiber:

up to 4 x 12 G /3G/HD-SD SDI, ASI signals

AoIP (4 channels of audio, two inputs and two outputs on each side), Dante / AES 67 protocol , Primary / Secondary ports

The Audio to IP Converter ( AoIP ) is designed to connect microphone or line analog audio signals to local Ethernet transmission networks using the Dante protocol or the AES67 standard , for intercom over IP networks and / or for switching audio signals in a network.

An additional Ethernet port provides connection to a secondary ( Secondary ) audio transmission network, and can also be used for cascading devices (Switched).

There are two versions of the device:

models with transport up to 3 G SDI

models with transport up to 12 G SDI ( index -12 G in device code)

To transmit signals over a single fiber, the principle of CWDM (wavelength division) is used in the range of 1470 nm :1570 nm or 1270nm:1410 nm (index ' H ' in the block code). This allows you to organize both one-way and two-way video transmission.

The optical expansion port UPGRADE (1310 nm ) allows the transmission of additional signals transmitted at wavelengths of 1260nm: 1450 nm .

Thus, by connecting through the UPGRADE expansion port the second body ' P roBox ', operating at wavelengths of 1270nm:1410 nm (index ' L' in the block code), it is possible to transmit 8 3G / HD-SD SDI, ASI signals plus implement two independent AoIP lines through one fiber.

Power is supplied from an external adapter 12 V@1000 mA.

The devices are designed for:

- connection of analog or digital audio signals in AES 3 format to the local audio transmission network ( AoIP ) using AES 67, Dante protocol ;

- connection of microphones to the local audio transmission network ( AoIP ) via AES 67, Dante protocol ;

- audio data exchange between Ethernet Dante , AES 67 and 3 G / HD / SD SDI digital video signals (devices with index ' V '); - creation of a distributed network of audio switches with a common switching field; - transfers

Ethernet ( AES 67, Dante ) over optical links;

- indication of audio signal levels ( PEAI -9090, PEAI -9091).

Ethernet Dante audio interfaces , AES 67 support:

- 8 in, 8 out ( PEAI -9088, PEAI -9090) and 16 in, 16 out ( PEAI -9091) balanced analog audio

or 4 in, 4 out ( PEAI -9088, PEAI -9090) and 8 in, 8 out ( PEAI -9091) AES 3 digital audio channels .

Devices with suffix ' V ' contain a 4-channel audio communication module between Ethernet Dante , AES 67 and 3G / HD / SD SDI digital video signals . The devices have an SDI input and two SDI outputs . Up to 16 audio channels of SDI embedded audio are supported .

Devices with suffix ' R ' contain an AES 3 digital audio input/output module with support for User Bits and Channel Bits (compatible with Riedel intercom equipment ).

The PEAI -9090 and PEAI -9091 devices can be equipped with submodules with microphone inputs (suffix ' M ' in code). This provides a voltage supply of '+48 V ' to the microphones and control of the gain of signals from the microphones.

The front panel of the PEAI -9090 and PEAI -9091 has a display that shows indicators for input and output signals.

The type of audio signals for inputs and outputs, independently of each other, is selected by the User (should be reflected in the device code).

The devices have two electrical Ethernet 100/1000 BaseT ports - primary ( Primary ) and backup ( Secondary ) and one optical (except for PEAI -9091). A redundant electrical Ethernet port provides connection to a redundant audio transmission network or device cascading. The presence of an optical transmission line through a separate SFP module allows you to transmit Ethernet ( AES 67, Dante ) over optical communication lines over long distances

( Ethernet SFP module not included in the device; see section 'SFP modules optical and electrical', section II , pos. 1.1:1.6).

Management and monitoring are carried out over the Ethernet network using the proprietary Dante Controller program and via the WEB interface.

Separate power inputs for main and backup power supply. 'Hot' replacement of power supplies.

Audio connectors DB 25 and RJ 45 (on module ' R ').

The devices are designed for:

- connection of analog or digital audio signals in AES 3 format to the local audio transmission network ( AoIP ) using AES 67, Dante protocol ;

- connection of microphones to the local audio transmission network ( AoIP ) via AES 67, Dante protocol ;

- audio data exchange between Ethernet Dante , AES 67 and 3 G / HD / SD SDI digital video signals (devices with index ' V '); - creation of a distributed network of audio switches with a common switching field; - transfers

Ethernet ( AES 67, Dante ) over optical links;

- indication of audio signal levels ( PEAI -9090, PEAI -9091).

Ethernet Dante audio interfaces , AES 67 support:

- 8 in, 8 out ( PEAI -9088, PEAI -9090) and 16 in, 16 out ( PEAI -9091) balanced analog audio

or 4 in, 4 out ( PEAI -9088, PEAI -9090) and 8 in, 8 out ( PEAI -9091) AES 3 digital audio channels .

Devices with suffix ' V ' contain a 4-channel audio communication module between Ethernet Dante , AES 67 and 3G / HD / SD SDI digital video signals . The devices have an SDI input and two SDI outputs . Up to 16 audio channels of SDI embedded audio are supported .

Devices with suffix ' R ' contain an AES 3 digital audio input/output module with support for User Bits and Channel Bits (compatible with Riedel intercom equipment ).

The PEAI -9090 and PEAI -9091 devices can be equipped with submodules with microphone inputs (suffix ' M ' in code). This provides a voltage supply of '+48 V ' to the microphones and control of the gain of signals from the microphones.

The front panel of the PEAI -9090 and PEAI -9091 has a display that shows indicators for input and output signals.

The type of audio signals for inputs and outputs, independently of each other, is selected by the User (should be reflected in the device code).

The devices have two electrical Ethernet 100/1000 BaseT ports - primary ( Primary ) and backup ( Secondary ) and one optical (except for PEAI -9091). A redundant electrical Ethernet port provides connection to a redundant audio transmission network or device cascading. The presence of an optical transmission line through a separate SFP module allows you to transmit Ethernet ( AES 67, Dante ) over optical communication lines over long distances

( Ethernet SFP module not included in the device; see section 'SFP modules optical and electrical', section II , pos. 1.1:1.6).

Management and monitoring are carried out over the Ethernet network using the proprietary Dante Controller program and via the WEB interface.

Separate power inputs for main and backup power supply. 'Hot' replacement of power supplies.

Audio connectors DB 25 and RJ 45 (on module ' R ').

Press Splitter with Dante input plus analogue audio. Passive distribution and PoE.

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.

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.

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, voting, sign-in 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

Product Features： 1. A connection and distribution function between the host and microphone of the digital conference IoT system 2. Equipped with 4 RJ45 ports, 2-way distribution and 2-way connection; 3. Circular hand-in-hand design, equipped with 2 CAT6 super Category 6 wires; 4. One module has two microphone connections; 5. Any module sets the connection of the chairman microphone.

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 USB audio signal and Dante digital audio signal; expand audio and power through a network cable; fully support Dante software, easy to configure network and audio routing; support USB power supply, Built-in lightning and surge protection; it can be used on any PC, Mac or Linux system without reconfiguring the Dante network. It can record and play dual-channel audio at the same time without any other software.

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.

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).

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. Connect to the previous connection box or directly connect to the conference host through the RJ45 network port interface, and the active expansion connection box; 5. Connect the LINK port or conference unit of the next connection box through the RJ45 network port interface. 6. Support video tracking

Features： 1. A distributor connecting the host and the generator unit; 2. Provide 4 RJ45 wires, 2 of which are used for manual connection between junction boxes, and 2 are used to connect the transmitter unit; 3. Hand-pull connection method, with 2 CAT6 and over Category 6 wires; 4. Each junction box can be connected to 2 transmitter units; 5. The sending unit connected to the unit with chairperson is the chairperson unit, and the others are the representative units.

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 20 outputs (12 analog and 8 digital), using the industry-leading ADI DSP SHARC21489, the highest The working 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. 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.

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. 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.