Dante-enabled Product Catalog

MondeF 4-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 4 Dante® outputs, 8 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 4-zone network audio server (Dante®)

4 Dante® outputs

8 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 64-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 64 Dante® outputs, 32 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 64-zone network audio server (Dante®)

64 Dante® outputs

32 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 8-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 8 Dante® outputs, 8 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 8-zone network audio server (Dante®)

8 Dante® outputs

8 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 16-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 16 Dante® outputs, 16 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 16-zone network audio server (Dante®)

16 Dante® outputs

16 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

A 1U 19inch rack mounted AoIP monitoring and mixing unit.

4 models are available the RL64, RL128, RL192 and RL256 and each unit can be configured at time of purchase to be Dante or Ravenna

RL Series units allows two modes of operation a mixing mode which can mix up to 256 channels at 48kHz (128 channels at 96kHz) plus microphone input (via headphone connector) and AES input to create a 259 into 2 mixer. Alternatively, it can be set into a simple Listen mode where touching a channel instantly routes to the LR output and cancels any previous selection.

The outputs of the LR bus are presented to the user via the inbuilt 4mm x 9mm full range speakers or via the 3.5mm headphone jack. The speakers are automatically muted when a headphone jack is inserted. Simultaneous outputs are also available in the form of the stereo balanced line outputs and AES output accessed via XLR connectors on the rear of the RL Series Unit. The mix output is also routed to the AoIP output.

The RL Series units are operated and controlled via the large letterbox 6.6” diagonal (168mm) 1440 x 240 24-bit colour LCD and capacitive touch panel.

A simple and elegant menu system has been designed to be intuitive to navigate allowing the user to access all the features of the RL Series unit very quickly.

The high-resolution screen is easy to read presenting comprehensive metering, routing and status information to the user.

Firmware updates of the RL Series unit are achieved via a dedicated USB micro AB connector on the front panel. There is also a dedicated 3.5mm jack socket for the headphone output

Stacks Image 30

On the rear of the RL Series unit there are 4x Primary and Secondary ports for up to 4x AoIP inputs. 2x balanced line outputs on 3 pin male XLRs and AES output also on 3 pin male XLR. These outputs can be configured to follow the main volume settings of the speakers/headphones or they can also be set to follow the main Mute control.

Additionally, there is an AES input via 3 pin female XLR connector

The RL Series unit can be powered via either of the two 2.5mm DC input connectors requiring 24V @ 2A. These inputs also have a screw lock to prevent accidental unplugging. Using both inputs provides redundancy.

A 1U 19inch rack mounted AoIP monitoring and mixing unit.

4 models are available the RL64, RL128, RL192 and RL256 and each unit can be configured at time of purchase to be Dante or Ravenna

RL Series units allows two modes of operation a mixing mode which can mix up to 256 channels at 48kHz (128 channels at 96kHz) plus microphone input (via headphone connector) and AES input to create a 259 into 2 mixer. Alternatively, it can be set into a simple Listen mode where touching a channel instantly routes to the LR output and cancels any previous selection.

The outputs of the LR bus are presented to the user via the inbuilt 4mm x 9mm full range speakers or via the 3.5mm headphone jack. The speakers are automatically muted when a headphone jack is inserted. Simultaneous outputs are also available in the form of the stereo balanced line outputs and AES output accessed via XLR connectors on the rear of the RL Series Unit. The mix output is also routed to the AoIP output.

The RL Series units are operated and controlled via the large letterbox 6.6” diagonal (168mm) 1440 x 240 24-bit colour LCD and capacitive touch panel.

A simple and elegant menu system has been designed to be intuitive to navigate allowing the user to access all the features of the RL Series unit very quickly.

The high-resolution screen is easy to read presenting comprehensive metering, routing and status information to the user.

Firmware updates of the RL Series unit are achieved via a dedicated USB micro AB connector on the front panel. There is also a dedicated 3.5mm jack socket for the headphone output

Stacks Image 30

On the rear of the RL Series unit there are 4x Primary and Secondary ports for up to 4x AoIP inputs. 2x balanced line outputs on 3 pin male XLRs and AES output also on 3 pin male XLR. These outputs can be configured to follow the main volume settings of the speakers/headphones or they can also be set to follow the main Mute control.

Additionally, there is an AES input via 3 pin female XLR connector

The RL Series unit can be powered via either of the two 2.5mm DC input connectors requiring 24V @ 2A. These inputs also have a screw lock to prevent accidental unplugging. Using both inputs provides redundancy.

PENTA 721s-DNT is a Dante to Dante converter with built-in Samplerate Converter, MADI and AES, which provides a versatile and flexible solution for signal distribution of Dante audio between two different networks as well as routing via AES/EBU and MADI and optical fibre networks.

It is a compact 1U digital audio interface with a dual set of Dante IP Audio network ports, Pro|Mon and dual power supply, providing routing and selection of sample rate between two separated Dante networks and individual domains. One of the domains has a capacity of 128 Dante channels and 128 audio flows, and the other domain has a capacity of 64 Dante channels and 32 audio flows.

All Dante channels can be routed to and from any of the digital interfaces of the unit. It comes with eight channel AES/EBU input/output channels, MADI and two IP Audio Ethernet in/outs, as well as an optional mini-module slot with dual SFP MADI optical in/out connection.

PENTA 721s-SDI is a Dual SDI embedder/de-embedder with onboard Samplerate Converter, MADI, AES and Dante, which provides a versatile and flexible solution for signal distribution of embedded audio via AES/EBU, MADI, as well as routing via IP Gigabit Ethernet and optical fibre networks.

Penta 721s-SDI is a compact 1U digital audio interface with a dual set of 3G SDI input/output ports, Pro|Mon and dual power supply. It provides embedding and de-embedding of 16 audio channels per port where samplerate conversion can be enabled if desired.

All embedded channel can be routed to and from any of the digital interfaces on the unit. It comes with eight channel AES/EBU input/output channels, MADI and Dante IP Audio Ethernet in/outs, as well as an optional mini-module slot with dual SFP MADI optical in/out connection.

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.

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

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

Quantum MC is a 1U rack mount IP audio codec extending the Quantum product family.

Quantum MC is a Multi-Channel IP codec for bridging phase-locked 5.1 audio signals over Internet between two such devices. (This is 6 phase-locked mono audios).

Alternatively this product might be used for an efficient low delay bidirectional bridge of 6 voices/audios between two production sites.

Quantum MC fully supports IP (TCP and UDP). This enables remote monitoring/configuring and data/audio transportation over data communications links (LAN, WAN, 3G/4G, Internet…)

This unit can be controlled from its built-in web graphical interface, its front panel display menu, or from the ProdysControlPlus application.

Quantum2 XL is a complete Commentary Unit including 5 mic/line inputs and 4 headphone outputs. Additional sources to its embedded mixers are USB Audio, AES/EBU or AoIP/AES67 (DANTE or Ravenna optional) inputs.

Quantum2 XL is an all-in-one box solution for remote commentaries linking two bidirectional audio streams with four independent voices, sounds or mixed audios to the studio or Host Broadcast Center.

Perfect sounds are encoded using a choice of advanced and low delay compression algorithms for sport and music.

Highly reliable transmissions are achieved using Diversity and Bonded streams over a group of network interfaces (2x Ethernet, 2x LTE and Wi-Fi).

Additionally, a video downlink stream can be decoded internally and shown on a general purpose monitor, enriching the information supplied to the talents.

A video video uplink is also possible for the transmission of images from the commentators to the studio.

It also allows to insert audio into the received video and to forward a single synchronised video and audio stream to the studio.

Quantum2 W is a complete Commentary Unit including 3 mic/line inputs and 3 headphone outputs. Additional sources to its embedded mixers are USB Audio, AES/EBU or AoIP/AES67 (DANTE or Ravenna optional) inputs.

Quantum2 W is an all-in-one box solution for remote commentaries linking two bidirectional audio streams with four independent voices, sounds or mixed audios to the studio or Host Broadcast Center.

Perfect sounds are encoded using a choice of advanced and low delay compression algorithms for sport and music.

Highly reliable transmissions are achieved using Diversity and Bonded streams over a group of network interfaces (2x Ethernet, 2x LTE and Wi-Fi).

Additionally, a video downlink stream can be decoded internally and shown on a general purpose monitor, enriching the information supplied to the talents.

A video video uplink is also possible for the transmission of images from the commentators to the studio.

It also allows to insert audio into the received video and to forward a single synchronised video and audio stream to the studio.

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 ').

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.

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.

Pi is a multipurpose series of power amps for touring and installation applications, based on the legendary QuantaPulse™ switching mode power supply with an innovative class H 3 steps topology.

This series includes a completely renewed PMS™ which incorporates a set of protection systems in power supply and audio modules which works in real time continuously maintaining all variables of the amp within safe working threshold always. So, Pi amps are appropriate machines for the most exigent applications with the best reliability.

Pi amps have been designed with a non-symmetrical class H topology which allows working with very high voltages given incredible headroom and a great punch.

All these characteristics make Pi amplifier an interesting device to work with asymmetric loads to squeeze every last drop of power in each way of the sound system.

Pi series has an extra-large 4.3″ IPS display with capacitive touch panel whereby it is possible to control and manage every parameter of the amp and select all System Presets, Uses, User EQ and Snapshots created.

Also a powerful FIR DSP has been implemented in these amps with a software specifically developed for this series.

Pi is a multipurpose series of power amps for touring and installation applications, based on the legendary QuantaPulse™ switching mode power supply with an innovative class H 3 steps topology.

This series includes a completely renewed PMS™ which incorporates a set of protection systems in power supply and audio modules which works in real time continuously maintaining all variables of the amp within safe working threshold always. So, Pi amps are appropriate machines for the most exigent applications with the best reliability.

Pi amps have been designed with a non-symmetrical class H topology which allows working with very high voltages given incredible headroom and a great punch.

All these characteristics make Pi amplifier an interesting device to work with asymmetric loads to squeeze every last drop of power in each way of the sound system.

Pi series has an extra-large 4.3″ IPS display with capacitive touch panel whereby it is possible to control and manage every parameter of the amp and select all System Presets, Uses, User EQ and Snapshots created.

Also a powerful FIR DSP has been implemented in these amps with a software specifically developed for this series.

Pi is a multipurpose series of power amps for touring and installation applications, based on the legendary QuantaPulse™ switching mode power supply with an innovative class H 3 steps topology.

This series includes a completely renewed PMS™ which incorporates a set of protection systems in power supply and audio modules which works in real time continuously maintaining all variables of the amp within safe working threshold always. So, Pi amps are appropriate machines for the most exigent applications with the best reliability.

Pi amps have been designed with a non-symmetrical class H topology which allows working with very high voltages given incredible headroom and a great punch.

All these characteristics make Pi amplifier an interesting device to work with asymmetric loads to squeeze every last drop of power in each way of the sound system.

Pi series has an extra-large 4.3″ IPS display with capacitive touch panel whereby it is possible to control and manage every parameter of the amp and select all System Presets, Uses, User EQ and Snapshots created.

Also a powerful FIR DSP has been implemented in these amps with a software specifically developed for this series.

Pi is a multipurpose series of power amps for touring and installation applications, based on the legendary QuantaPulse™ switching mode power supply with an innovative class H 3 steps topology.

This series includes a completely renewed PMS™ which incorporates a set of protection systems in power supply and audio modules which works in real time continuously maintaining all variables of the amp within safe working threshold always. So, Pi amps are appropriate machines for the most exigent applications with the best reliability.

Pi amps have been designed with a non-symmetrical class H topology which allows working with very high voltages given incredible headroom and a great punch.

All these characteristics make Pi amplifier an interesting device to work with asymmetric loads to squeeze every last drop of power in each way of the sound system.

Pi series has an extra-large 4.3″ IPS display with capacitive touch panel whereby it is possible to control and manage every parameter of the amp and select all System Presets, Uses, User EQ and Snapshots created.

Also a powerful FIR DSP has been implemented in these amps with a software specifically developed for this series.

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)

In 2010, TAIDEN developed the first paperless multimedia conference system in the world. Paperless conference, video service, conference services, etc. were successfully introduced into conference system, and promoted conference system technology to a new stage. Now, TAIDEN paperless multimedia conference system has been successfully applied to the United Nations Headquarters, the World Bank Headquarters, Lombardy district government in Italy, the People's Great Hall of Zhejiang Province, Wuhan International Expo Center and other high-end meeting places. The concept of paperless multimedia conference system

has also been widely recognized and followed by global industry of conference system, and has become the representative of the fourth generation of conference system products.

In 2014, TAIDEN continued to lead the development of conference system technology, and introduced a new generation of paperless multimedia conference system. The new generation paperless multimedia terminals are equipped with 10" with 1280X800 high resolution LCD touch screen, and the new experience of capacitive touch screen which supporting multi-touch, make paperless experience and operation more convenient and efficient, and can achieve a variety of HD video service, and a built-in 5 megapixel camera, and video intercom function, and E-ink nameplate with the latest electronic ink technology, at the same time, it can also realize interactive conference control and management (discussion, vote, 64 channel simultaneous interpretation x 2), conference services, etc.

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.

AK series Dante network amplifier, with analog /digital automatic backup switching function, the analog signal will switch automatically when there is a problem with the digital signal, ensuring that the signal is not terminated due to device failure. The product is divided into two channel and four channel two specifications, using Dante, AES digital audio input, 10 channel signal input can be matrix selection and mutual thermal backup, intelligent network monitoring, and center control. PFC switch power supply, high performance DSP sound control with FIR balance, AGC automatic overload protection, square wave self-load impedance suppression. Intelligent temperature control with overheat protection, constant resistance and constant pressure.

※Optional functions: remote switch, center control shutdown, remote maintenance, sound channel backup, speaker intelligent control and others.

Applications: Acoustic systems in large government projects, large stadiums, cultural centers, etc.