Dante-enabled Product Catalog

CAPITOL IP PLUS is a new digital audio broadcast mixing console, specifically designed for ON AIR control but adapts to a diversity of workflows and functionalities. It incorporates all the basic features required for such operations : automatic monitor muting, cough muting, fader start, control signalling, control signals for automating external equipment, external communications management, talkback or intercom functions, and many more.

The AEQ SOLARIS is a compact, scalable and virtualisable multichannel audio codec with IP connectivity. It is designed for a variety of broadcast applications, including multiple STL (Studio-to-Transmitter Link) connections and remote contributions, allowing users to start with eight bidirectional stereo channels and expand up to 64 channels in the same 1U 19-inch rack unit. Its front panel features a full-colour, high-resolution display used to show general status of the unit as well as the configuration of the network interfaces and software versions, while the rear has five network ports to allow for traffic control, encoded audio (WAN) and local audio I/O separation through AES67/Dante. It boasts two AC power supplies and features local audio inputs and outputs that can connect with other audio devices using AES67/Dante. It can also be installed as an application on a commercial server, or operate in a virtualised environment and controlled through a web interface. In order to achieve unified control with other AEQ audio codecs, it can also be controlled from AEQ’s ControlPhoenix software, while an API is available to integrate it with orchestrators and third-party applications. Apart from the recommended OPUS algorithm and the uncompressed PCM format, it incorporates the G711 and G722 algorithms to communicate with VoIP devices. For special uses, C and apt-X encoding algorithms are available.

The NETBOX DSP is able to receive audio from the Dante network and then return it, mixed and processed, to be used in another device or subsystem. It can also generate test signals and vumeters. Effectively, it is an audio matrix for AoIP networks. The NETBOX DSP is able to receive audio from the Dante network and then return it, mixed and processed, to be used in another device or subsystem. It can also generate test signals and vumeters. Effectively, it is audio matrix for AoIP networks. NETBOX DSP can be used in combination with any AEQ or third-party device from more than 350 manufacturers which are compatible with Dante AoIP protocol. NETBOX DSP also incorporates 16 GPI and 16 GPO (each GPIO connector includes a power supply pin in order to feed external circuitry). GPIO can also be transported through the IP network between compatible devices. This way, a GPIO can drive the GPOs in different piece of equipment.

Standalone commentary unit (CU), or AoIP connected with 8 channels via Dante protocol. Scalable architecture: simple routing to DanteTM IP devices; integrated in IP intercom system, or connected to IP commentary system matrix. Standalone commentary unit (CU), or AoIP connected with 8 channels via DanteTM protocol. Scalable architecture: simple routing to DanteTM IP devices; integrated in IP intercom system, or connected to IP commentary system matrix.  Standalone mono or stereo sound mixer with mixing, routing, tone and dynamics control. 3 commentator inputs and a dual-mono or stereo line level input. Listening of 8 remote and 2 local sources.

• Operates as an intercom panel at the same time as a Commentary Unit.
• Configurable as interpreter desk up to three languages. 
• 3 1Gigabit IP ports per unit for redundancy, daisy chain and auxiliary data or video transport.
• Dual power supply: 48VDC via PoE or external local power supply.
• Software Configuration and remote control.
• Rugged and ergonomic mechanics, suitable for indoors and outdoors locations.

AEQ VENUS is a dual IP stereo, full duplex audiocodec system. With a single unit it is possible to establish two bi-directional stereo or four mono (to one or two different locations) AEQ VENUS is a dual IP stereo, full duplex audiocodec system. With a single unit it is possible to establish two bi-directional stereo or four mono (to one or two different locations). It was designed to meet the international recommendation N/ACIP EBU Tech3326 and includes the option of an adaptive buffer in order to absorb network jitter, DHCP allowing the automatic configuration of the IP Network parameters, FEC and automatic reference clock adjustment in order to synchronize both extremes of the communications. The unit has balanced analogue Audio Inputs and Outputs through XLR connectors as well as digital audio AES/EBU I/O’s.

Series 8000 user intercom panels have been designed to provide broadcast quality audio and to be used with the digital technology of Conexia and CrossNET matrixes. Audio is captured and processed at 48kHz, with 24 bits/sample, providing full audio bandwidth with negligible noise and distortion levels. Series 8000 user intercom panels have been designed to provide broadcast quality audio and to be used with the digital technology of Conexia and CrossNET matrixes. Audio is captured and processed at 48kHz, with 24 bits/sample, providing full audio bandwidth with negligible noise and distortion levels. Ease of installation has also been taken into account, featuring a redundant IP port that handles high-quality DANTE™ -format audio and also compatible with AES 67 standard. Audio is digitally processed to cancel acoustic echo and automatically compensate for voice level and particular speech habits of each operator. Acoustics has been seriously engineered to reach the highest possible intelligibility and sound clarity.

CrossNet 168 has 128-ports Dante IP interface, 8 Kroma digital intercom ports, 12 balanced, broadcast quality analogue ports, and 20 compressed audio IP ports.  CrossNet 168 has 128-ports Dante IP interface, 8 Kroma digital intercom ports, 12 balanced, broadcast quality analogue ports, and 20 compressed audio IP ports. CrossNET is a compact intercom matrix, built into a 1U-height 19” enclosure, featuring basic IP connectivity based on Dante™ technology, that is compatible with the recent AES-67 standard, allowing for the transportation of broadcast quality audio. It incorporates high-quality analog balanced inputs and outputs and supports former Kroma IP panels and interfaces with reduced bandwidth compressed audio. It also supports Kroma panels with digital audio link. The proper matrix features a 4-key intercom user interface on its front panel. 

CrossNet 104 has 64-ports Dante IP interface, 8 Kroma digital intercom ports, 12 balanced, broadcast quality analogue ports, and 20 compressed audio IP ports. CrossNet 104 has 64-ports Dante IP interface, 8 Kroma digital intercom ports, 12 balanced, broadcast quality analogue ports, and 20 compressed audio IP ports. CrossNET is a compact intercom matrix, built into a 1U-height 19” enclosure, featuring basic IP connectivity based on Dante™ technology, that is compatible with the recent AES-67 standard, allowing for the transportation of broadcast quality audio. It incorporates high-quality analog balanced inputs and outputs and supports former Kroma IP panels and interfaces with reduced bandwidth compressed audio. It also supports Kroma panels with digital audio link. The proper matrix features a 4-key intercom user interface on its front panel. 

CrossNet 72 has 32-ports Dante IP interface, 8 Kroma digital intercom ports, 12 balanced, broadcase quality analogue ports, and 20 compressed audio IP ports. CrossNet 72 has 32-ports Dante IP interface, 8 Kroma digital intercom ports, 12 balanced, broadcase quality analogue ports, and 20 compressed audio IP ports. CrossNET is a compact intercom matrix, built into a 1U-height 19” enclosure, featuring basic IP connectivity based on Dante™ technology, that is compatible with the recent AES-67 standard, allowing for the transportation of broadcast quality audio. It incorporates high-quality analog balanced inputs and outputs and supports former Kroma IP panels and interfaces with reduced bandwidth compressed audio. It also supports Kroma panels with digital audio link. The proper matrix features a 4-key intercom user interface on its front panel. 

The FORUM is a self-contained digital audio mixing console specifically designed to meet the most rigorous demands of the radio and television broadcast industry. Its flexible design allows it to readily be adapted to any ON-AIR application. FORUM adapts perfectly to any environment: auto-control, studio control, television production, sound production, mixed configurations, integration in large installations, etc. The control surface can be configured with a minimum of 4 faders, and easily expanded to 8, 12 or even 20 channels with the FRGRAND chassis.

Dante card for FORUM console. Dante card for FORUM console.

Dante card for Capitol console. Dante card for Capitol console.

Dante card for BC-2000D router or ARENA digital mixing console. Dante card for BC-2000D router or ARENA digital mixing console.

CAPITOL IP is an 8 channel digital mixing console based on the AEQ CAPITOL console and the experience gained with this model. The control surface has been redesigned with new softer buttons, more programmable keys and a new arrangement of the monitoring section controls. AoIP multi-channel networking connectivity in CAPITOL IP console is implemented through a single module with 16 input and 16 output channels incorporated in its core.

The AEQ ARENA is a digital mixing console for radio and television broadcast applications. It is specifically designed for controlling on-air transmissions but can easily be adapted to any working environment. The ARENA incorporates all the features needed for any working situation. The core of the ARENA is the BC-2000D Routing system from AEQ.

The backbone of the BC-2000 D Router Digital Audio system is the double TDM bus, housed in the BC-2000 D frame, with the capacity to route, mix or distribute more than a thousand channels simultaneously. The TDM bus is divided into an input bus and an output bus, with the possibility of inserting DSP cards, achieving a higher DSP processing capacity. The system may be decentralized through the use of DANTE AoIP or MADI links as the interconnection between the different sub-frames. The BC-2000 D Routing system is also the heart and engine of AEQ’s commentary systems OLYMPIA.

The Olympia commentary system uses the audio platform TITAN - BC 2000D router. This system platform has an overall capacity of over 5000 x 5000 audio channels and complies with the requisites demanded at the largest broadcasting events in the World. Nonetheless, Olympia can be configured to service very small events as well as on-board systems for OB-vans. The Olympia counts on automatic system planning and engineering with generation of equipment listing, interconnection procedures through simple definition of needs.

NETBOX 8 AD inferface features 8 inputs and 8 outputs, organized in 4 mono analog and 2 digital stereo channels. Stereo digital ones can be configured as AES/EBU or SPDIF standards. The second digital stereo channel can also be switched to a USB connector to ease the connection to an audio workstation. It also provides 4 GPI and 4 GPO. The GPO port includes a power supply pin to feed the output circuitry. NETBOX 8 AD inferface features 8 inputs and 8 outputs, organized in 4 mono analog and 2 digital stereo channels. Stereo digital ones can be configured as AES/EBU or SPDIF standards. The second digital stereo channel can also be switched to a USB connector to ease the connection to an audio workstation. It also provides 4 GPI and 4 GPO. The GPO port includes a power supply pin to feed the output circuitry. Due to its small footprint, it can be useful to give IP access to analog or digital consoles that are not ready for this type of connectivity from factory, for recording rooms, talk-rooms or any other auxiliary location.

The NETBOX 4MH interface has been designed to cover the need for high quality microphone inputs and headphone outputs availability outputs, for example in radio and TV studios, stage or multimedia installations. The multichannel IP connectivity (Audinate DANTE™ protocol) makes the Inputs of the unit available at any point of the network. The NETBOX 4MH interface has been designed to cover the need for high quality microphone inputs and headphone outputs availability outputs, for example in radio and TV studios, stage or multimedia installations. The multichannel IP connectivity (Audinate DANTE™ protocol) makes the Inputs of the unit available at any point of the network. NETBOX 4MH interface also includes 4 General Purpose inputs and outputs (GPIO) as interfaces for signaling transport between the different devices in the IP network. As a complementary feature, the unit is soporting the same inputs and outputs as analogue balanced line Inputs and Outputs. Further, the new interface has additional GPIOs that can be used in for example a studio installation and as an interface to signaling terminals such as the AEQ SRC 03. This greatly facilitates the implementation of services such as remote CUE and Cough-cut, User keys, ON AIR and READY lights, etc. Technical specifications

• Channels: 4 microphone-line input, 4 line outputs, in parallel with 4 stereo headphone outputs. Mute and gain control between +12 and -30 dB, individually or in stereo pairs.
• 48V Phantom Power supply and electronic balancing for all microphone inputs.
• Data format: DANTE Audio-over-IP technology.
• Scalable and flexible network topology supporting a wide range of audio transmitters and receivers.
• Supports 100Mbps, 1 Gbps and 10 Gbps network segments.
• 2 x RJ45, 100 BASE-T Ethernet ports with galvanic isolation, usable as “daisy chain” connections.

NETBOX 32 AD interface features 32 input and 32 output channels organized in 16 mono analog + 8 stereo digital channels. The stereo digital audio channels can be configured as AES/EBU or SPDIF standard. It also incorporates 16 GPI and 16 GPO (each GPO connector includes a power supply pin to feed the output circuitry). NETBOX 32 AD interface features 32 input and 32 output channels organized in 16 mono analog + 8 stereo digital channels. The stereo digital audio channels can be configured as AES/EBU or SPDIF standard. It also incorporates 16 GPI and 16 GPO (each GPO connector includes a power supply pin to feed the output circuitry). Due to its high input and output capacity, it is especially suitable for central controls and link dispatches and also to increase or distribute the capacity of TDM BUS matrixes such as the AEQ BC 2000D.

The FORUM is a self-contained digital audio mixing console specifically designed to meet the most rigorous demands of the radio and television broadcast industry. Its flexible design allows it to readily be adapted to any ON-AIR application. FORUM adapts perfectly to any environment: auto-control, studio control, television production, sound production, mixed configurations, integration in large installations, etc. The control surface can be configured with a minimum of 4 faders, and easily expanded to 8, 12 or even 20 channels with the GRAND FORUM chassis.