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The AVN-AESIO8 audio converter and interface converts up to eight digital stereo AES3 inputs and eight digital stereo AES3 outputs to and from the Dante Audio-over-IP networking standard. Each input can accept sample rates from 32kHz to 192kHz, which will be sample rate converted to the Dante system sample rate. A valid input signal is confirmed by front panel AES3 lock LEDs for each input. All outputs follow the Dante system sample rate.

This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All digital AES3 inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link and Clock.

• 8 x balanced digital stereo AES3 inputs on XLR, supporting input rates of 32kHz – 192kHz.


• Sample rate conversion on physical inputs to Dante system sample rate.


• 8 x balanced digital stereo AES3 outputs on XLR, output rate matches Dante system sample rate.


• 1 x RJ45 Dante connector (1Gb/s Ethernet Port).


• PoE, Link, and Clock LED status indicators.


• AES3 Lock LED status indicators for each input.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19” rack-mount form factor.

(The AVN-AESIO8R is available with dual redundant Ethernet ports).

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The AVN-AESIO8R audio converter and interface converts up to eight digital stereo AES3 inputs and eight digital stereo AES3 outputs to and from the Dante Audio-over-IP networking standard. Each input can accept sample rates from 32kHz to 192kHz, which will be sample rate converted to the Dante system sample rate. A valid input signal is confirmed by front panel AES3 lock LEDs for each input. All outputs follow the Dante system sample rate.

Dual Ethernet ports allow the unit to operate in redundant mode, ensuring audio routing is maintained in the event of loss of link on either of the network connections. This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All digital AES3 inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link and Clock.

• 8 x balanced digital stereo AES3 inputs on XLR, supporting input rates of 32kHz – 192kHz.


• Sample rate conversion on physical inputs to Dante system sample rate.


• 8 x balanced digital stereo AES3 outputs on XLR, output rate matches Dante system sample rate.


• 2 x RJ45 Dante connectors (1Gb/s Ethernet Port) allowing the unit to operate in redundant or switched modes.


• PoE and Link LED status indicators for each Ethernet port.


• Clock LED status indicator.


• AES3 Lock LED status indicators for each input.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.


• 1U 19” rack-mount form factor.

(The AVN-AESIO8 is available with a single Ethernet port).

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The AVN-AI16 audio converter and interface converts up to sixteen analogue inputs to the Dante Audio-over-IP networking standard. This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue inputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.

• 16 x balanced analogue inputs on XLR.


• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).


• 1 x RJ45 Dante connector (1 Gb/s Ethernet Port).


• PoE, Link, and Clock LED status indicators.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19" rack-mount form factor.

(The AVN-AI16R is available with dual redundant Ethernet ports).

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The AVN-AI16R audio converter and interface converts up to sixteen analogue inputs to the Dante Audio-over-IP networking standard. Dual Ethernet ports allow the unit to operate in redundant mode, ensuring audio routing is maintained in the event of loss of link on either of the network connections.

This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue inputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.

• 16 x balanced analogue inputs on XLR.


• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).


• 2 x RJ45 Dante connector (1Gb/s Ethernet Port) allowing the unit to operate in redundant or switched modes.


• PoE and Link LED status indicators for each Ethernet port.


• Clock LED status indicator.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.


• 1U 19" rack-mount form factor.

(The AVN-AI16 is available with a single Ethernet port).

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The AVN-AIO4 audio converter and interface converts up to four analogue inputs and four analogue outputs to and from the Dante Audio-over-IP networking standard. This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up done via the standard Dante Controller software and power via POE (Power Over Ethernet).

All analogue inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock.

• 4 x balanced analogue inputs on XLR.


• 4 x balanced analogue outputs on XLR.


• 1 x RJ45 Dante connector (100Mb/s Ethernet Port).


• POE, Link, and Clock LED status indicators.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19" rack-mount form factor.

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The AVN-AIO8 audio converter and interface converts up to eight analogue inputs and eight analogue outputs to and from the Dante Audio-over-IP networking standard. This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.

• 8 x balanced analogue inputs on XLR.


• 8 x balanced analogue outputs on XLR.


• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).


• 1 x RJ45 Dante connector (1Gb/s Ethernet Port).


• PoE, Link, and Clock LED status indicators.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19” rack-mount form factor.

(The AVN-AIO8R is available with dual redundant Ethernet ports).

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The AVN-AIO8R audio converter and interface converts up to eight analogue inputs and eight analogue outputs to and from the Dante Audio-over-IP networking standard. Dual Ethernet ports allow the unit to operate in redundant mode, ensuring audio routing is maintained in the event of loss of link on either of the network connections.

This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.

• 8 x balanced analogue inputs on XLR.


• 8 x balanced analogue outputs on XLR.


• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).


• 2 x RJ45 Dante connectors (1Gb/s Ethernet Port) allowing the unit to operate in redundant or switched modes.


• PoE and Link LED status indicators for each Ethernet port.


• Clock LED status indicator.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.


• 1U 19” rack-mount form factor.

(The AVN-AIO8 is available with a single Ethernet port).

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The AVN-AO16 audio converter and interface converts up to sixteen analogue outputs from the Dante Audio-over-IP networking standard. This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.

• 16 x balanced analogue outputs on XLR.


• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).


• 1 x RJ45 Dante connector (1 Gb/s Ethernet Port).


• PoE, Link, and Clock LED status indicators.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19" rack-mount form factor.

(The AVN-AO16R is available with dual redundant Ethernet ports).

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The AVN-AO16R audio converter and interface converts up to sixteen analogue outputs from the Dante Audio-over-IP networking standard. Dual Ethernet ports allow the unit to operate in redundant mode, ensuring audio routing is maintained in the event of loss of link on either of the network connections.

This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.

• 16 x balanced analogue outputs on XLR.


• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).


• 2 x RJ45 Dante connectors (1Gb/s Ethernet Port) allowing the unit to operate in redundant or switched modes.


• PoE and Link LED status indicators for each Ethernet port.


• Clock LED status indicator.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.


• 1U 19" rack-mount form factor.

(The AVN-AO16 is available with a single Ethernet port).

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The AVN-CU2-DANTE is a portable commentator unit using Dante® AoIP.

The AVN-CU2-DANTE provides two mic/line inputs with a wide, adjustable gain range and has two stereo headphone outputs with lockable jack sockets, suitable for operation by two commentators.

It’s powered using Power over Ethernet (PoE), using Neutrik EtherCON connectors, with primary and secondary ports for power and data redundancy. There’s an additional 4 pin XLR 12V DC input. The unit supports up to 16 input and output AoIP channels and up to 16 simultaneous input and output AoIP streams.

The 6 x push-button rotary encoders and 12 x key-cap buttons are fully configurable, to control input & output levels and panning.

Each rotary encoder has a separate colour-coded meter section showing the channel name, detailed level metering, left/right panning and a limiter indication, on a bright daylight reading display. Colours can be programmed per encoder to quickly identify particular source groups, so headphone source selection becomes intuitive.

Metering is available per input/output, with output metering configurable as pre or post level adjustment. The top of the display shows output metering, a limiter indication and the name of the output, a pre-defined logo or nothing. A limiter is available on every output.

The unit has 2 x locking mic/line inputs with +48V phantom power indication and 2 x headphone outputs on locking 6.35mm jack sockets.

Four wire I/O on rear panel RJ45 connectors provide an AES3 or analogue input and output that can be assigned as mic outputs (line level), talkback outputs, programme inputs or talkback inputs as desired.  In addition, the AES/analogue connections can be used as an insert or exit point into/out from the AoIP network.

The unit has dual redundant network ports on both RJ45 (PoE using 2 x Neutrik EtherCON® connectors) and SFP cages.

There are 10 x configurable GPIO on a 15 way D-type connector with 1 x switched changeover output.

All of the buttons have key-cap text and can be configured. There are some standard operations available:

2 x On-Air buttons can be used to connect mic audio to the main output, either over AoIP or via the AES digital audio connection. The On-Air buttons can be locked if required.

A Menu button can be used to access limited setup options on the TFT display.

2 x Page buttons change the display and encoders to monitor an additional set of sources, mix points or outputs. Up to 4 pages can be pre-programmed, e.g. one for talkback inputs, one for outputs, one to monitor other sources.

2 x Cough buttons take the commentator off-air while pressed.

A User button can be programmed to perform any function using the web server.

4 x T/B (talkback) buttons can be configured to initiate talkback over AoIP or AES digital audio connection, using 4 x talkback busses. The talkback buttons operate with lazy talkback, taking the commentator off-air when invoked.

The illuminated ‘Sonifex’ logo acts as a power indication and there are illuminated LEDs to indicate network clock status, AoIP Primary and AoIP Secondary link status, PoE Primary, PoE Secondary and DC power active.

Stream setup to and from the unit is initially via Dante® Controller with more detailed configuration performed by using the built-in web GUI.

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The AVN-CU4-DANTE is a portable commentator unit using Dante® AoIP.

It is a dual version of the AVN-CU2-DANTE providing four mic/line inputs with a wide, adjustable gain range and four stereo headphone outputs with lockable jack sockets, suitable for operation by three or four commentators.

The feature set is as per the AVN-CU2-DANTE, with the following differences. There are:4 x On-air buttons.4 x Page buttons, 2 for each half of the display.4 x Cough buttons.8 x Talkback buttons, up to 4 for each user and3 x User buttons.

As for the AVN-CU2-DANTE, the illuminated ‘Sonifex’ logo acts as a power indication and illuminated LEDs indicate network clock status, AoIP Primary and AoIP Secondary link status, PoE Primary, PoE Secondary and AC power active.

The front panel houses 4 x locking mic/line inputs with +48V phantom power indication and 4 x headphone outputs on locking 6.35mm jack sockets.

There’s an abundance of 4 wire connections on the rear panel: 4 x analogue line inputs on XLR sockets with latching locks, 6 x analogue line outputs on XLR plugs and an RJ45 AES3 stereo input & output.

The unit has dual redundant network ports on both RJ45 (PoE+ using 2 x Neutrik EtherCON® connectors) and SFP cages. There is an AC mains input on an IEC inlet, with a universal supply.

The 10 configurable GPIO and single switched changeover output use a 15 way D-type connector.

-end-

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The AVN-DIO01 is a Dante AoIP network to analogue XLR stereo output converter in the Sonifex DIO range of Dante input/output devices. It features two balanced analogue XLR outputs and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

The A/D and D/A circuitry offers 120dB of dynamic range - ten times better than similar competing products. All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

• 2 x balanced XLR analogue outputs.


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range D/A conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO02 is an analogue XLR stereo input to Dante AoIP network converter in the Sonifex DIO range of Dante input/output devices. It features two balanced analogue XLR inputs and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

The A/D and D/A circuitry offers 120dB of dynamic range - ten times better than similar competing products. All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

• 2 x balanced XLR analogue inputs.


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range A/D conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO03 is Dante AoIP network-to-headphones convertor in the Sonifex DIO range of Dante input/output devices. It features a single stereo headphone output available on two connections for 1/4" and 3.5mm jacks and a volume control for headphones level. A limiter with threshold setting and LED indication can be switched in to prevent over-shooting and protect hearing.

The front panel potentiometer adjusts headphone volume from mute (fully anticlockwise) to +6dB of gain when fully clockwise. This is useful if the Dante stream level is low and requires boosting.

A simple headphone limiter is included to prevent hearing damage by limiting the audio level sent to the headphones. The limit level can be set using a trimmer adjustment tool, or small flat blade screwdriver, between approximately -12dBu (fully anticlockwise) and 0dBu (fully clockwise).

When limiting, the blue limit LED illuminates. This should prompt you to turn the headphone volume down until the blue LED extinguishes, as audio quality will be reduced whilst the limiter is active.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

The A/D and D/A circuitry offers 120dB of dynamic range - ten times better than similar competing products. All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

* Note: Because the headphone output of this device is unbalanced, performance is 6dB lower than for balanced AVN-DIO products.

• 1/4-inch and 3.5mm jack analogue headphone outputs.


• Headphones volume control.


• Limiter on/off, threshold control and LED indicator.


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES 67 compatible.


• Ultra-high quality, wide dynamic range D/A conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO04 is a Dante to analogue phono stereo input & output convertor in the Sonifex DIO range of Dante input/output devices. It features stereo analogue input and output phono connections and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

Note: Depending on the nature of your PoE supply, it may be necessary to earth the unit using the rear panel earth tag to maximise audio performance.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

The A/D and D/A circuitry offers 114dB(*) of dynamic range - ten times better than similar competing products. All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

* Note: Because the inputs & outputs of this device are unbalanced, performance is 6dB lower than for balanced AVN-DIO products.

• 2 x analogue phono-type inputs.


• 2 x analogue phono-type outputs.


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range D/A and A/D conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO05 is a Dante to analogue terminal block input and output convertor in the Sonifex DIO range of Dante input/output devices. It features balanced stereo analogue inputs and outputs on a terminal block connector and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

The A/D and D/A circuitry offers 120dB of dynamic range - ten times better than similar competing products. All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

• 12 x terminal block connections (balanced stereo inputs and outputs).


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES 67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range D/A and A/D conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO06 is a Dante to AES3 digital input and output audio convertor in the Sonifex DIO range of Dante input/output devices. It features stereo AES3 digital audio inputs and outputs on Neutrik XLR connectors, and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

• 1 x stereo AES3 XLR input.


• 1 x stereo AES3 XLR output.


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO07 is a Dante to AES-3id digital input and output audio convertor in the Sonifex DIO range of Dante input/output devices. It features stereo AES-3id digital audio inputs and outputs on BNC connectors, and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

• 1 x stereo AES-3id BNC input.


• 1 x stereo AES-3id BNC output.


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO08 is a Dante to AES3 digital input and output audio convertor in the Sonifex DIO range of Dante input/output devices. It features stereo AES3 digital audio inputs and outputs on terminal block connectors, and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

• 6 x terminal block connections (balanced stereo inputs and outputs).


• Neutrik EtherCon® Ethernet connection.


• Fully Dante compliant device.


• AES67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO09 is a Microphone input to Dante converter in the Sonifex DIO range of Dante input/output devices with A/D circuitry offering a world-class E.I.N. of 129dB. It features a single high quality mic preamp with balanced XLR input, coarse and fine gain controls, high pass filter, phantom power, tri-color level LED  and one Neutrik EtherCon® connector for direct connection to a Dante AoIP network.

All Sonifex DIO interfaces provide a simple, convenient, and elegant plug and play method of connecting legacy analogue and digital audio equipment to the Dante AoIP audio network.

All DIO products use Dante Controller for configuration and are powered by PoE (Power over Ethernet). They use rugged aluminium boxes with side slots for screw-mounting and contain superior audio circuitry for optimal audio performance. All feature rugged Neutrik EtherCon® connectors and Neutrik lockable audio connectors for ultra-reliable connectivity.

The AVN-DIO09 has coarse and fine mic gain with the coarse gain set using a toggle switch, providing 20dB/50dB of gain, and the fine gain set using a trimmer adjustment tool, or small flat blade screwdriver, adding between 0dB and 36dB of additional gain. An on/off toggle switch turns the high pass filter on or off and when enabled, it acts on frequencies below 125Hz at a rolloff of 6dB/octave.

Phantom power is enabled/disabled via a toggle switch on the front panel and when enabled, a 48V DC supply is provided to power an appropriate microphone. A red LED illuminates to show when phantom is enabled.

A front panel audio level LED helps to set the gain and shows the audio level being sent to the Dante network.

Note: If using a phantom powered microphone, it may be necessary to earth the unit using the rear panel earth tag to eliminate mains hum, depending on the nature of your PoE supply.

• 1 x balanced microphone input on XLR socket with latch lock.


• Neutrik EtherCon® Ethernet connection.


• Single turn pot setting fine mic gain (0dB – 36dB).


• Coarse mic gain switch (+20db/+50dB).


• High pass filter on/off switch.


• Phantom power on/off switch.


• Phantom power LED indicator.


• Level LED indicator.


• Fully Dante compliant device.


• AES67 compatible.


• Dante Domain Manager compliant.


• Ultra-high quality, wide dynamic range A/D conversion.


• Powered via PoE (Power over Ethernet).

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The AVN-DIO10 is a 3G/HD/SD-SDI to Dante® Embedder/De-Embedder in a single product solution. This is a simple plug and play audio/video interface that provides a convenient and elegant method of connecting legacy 3G/HD/SD-SDI equipment to the Dante® AoIP audio network.

The AVN-DIO10 takes any SDI feed, de-embeds the 16 audio channels and places them on channels 1-16 of the Dante network, mapped using Dante Controller. It simultaneously takes the 16 input channels mapped to the device on Dante Controller and re-embeds them onto the SDI output, with an embed enable switch for each channel pair.

It has a single 3G/HD/SD-SDI input and a reclocked output, with dual redundant Primary and Secondary Dante network ports, using Neutrik EtherCon® Ethernet connectors and is powered by PoE. It is a fully Dante compliant and AES67 compatible device that uses Dante Controller for configuration and supports the full range of Dante sample rates.

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The AVN-PA8/D and the AVN-PM8/D Portals can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers and the unit can be mounted using the CM-MNT1 desk mount panel.

The AVN-PA8/D and AVN-PM8/D can be combined with multiple Sonifex AVN-HA1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.

• Front panel 6.35mm (1/4”) headphone socket and volume control knob.


• Front panel Mute/GPO push button.


• Analogue audio input on RJ45 (the connector provides power to the unit and a GPO back to the portal).


• Loop through audio output on RJ45 (power and GPO signal are not connected).


• Locking DC power connector (if a portal is not being used to supply the unit with power).

Note: The AVN-HA1 is an analogue input product taking an analogue audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate analogue input.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The AVN-PD8/D Portal can be combined with multiple AVN-HD1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers and the unit can be mounted using the CM-MNT1 desk mount panel.

The AVN-PD8/D can be combined with multiple Sonifex AVN-HD1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.

• Front panel 6.35mm (1/4”) headphone socket and volume control knob.


• Front panel Mute/GPO push button.


• AES3 digital input on RJ45 (the connector provides power to the unit and a GPO back to the portal).


• AES3 digital output on RJ45 (power and GPO signal are not connected).


• Locking DC power connector if a portal is not being used to supply the unit with power.

Note: The AVN-HD1 is an AES3 digital input product taking an AES3 audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate AES3 audio input.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-Portal series is a group of three powerful audio mix engine interfaces (AVN-PA8, AVN-PD8 & AVN-PM8) capable of audio routing, metering and equalisation. They have both physical audio I/O and AES67 stream I/O and can mix and route freely from any input to any output, useful for a multitude of applications.

Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.

Applications:

• 8 output analogue zone mixer, with individual output gain control.


• 8 channel clean-feed generator, with input mixing and gain control on inputs and outputs.


• Distribute 8 stereo channels of audio over an SFP fibre connection.


• IFB generator to send 64 x AES67 streams to individual belt-packs.


• 8 output headphone distribution system, with separate input mix for each headphone output and individual gain control.


• Input mixer with input/output metering and AES67 stream generation.

Webserver Software

A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.

Mix Matrix

The key to the success of the AVN-PA8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.

The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.

Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.

DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points.

The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.

Front Panel Displays, Metering & Controls

The AVN-PA8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and D-types for rear panel connectivity.

Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.

Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.

A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.

Detailed Metering Option

The ‘D’ version of the portal (e.g. AVN-PA8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.

The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.

Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.

Phase metering can be displayed per stereo channel and channel idents can be shown either above or below the metering to identify each input/output.

On devices without a meter display, a smaller set of monochrome meters are shown on the main OLED display.

Physical Inputs & Outputs

For the analogue audio, the AVN-PA8 uses D-type sockets with AES59 analogue pinout, paralleled with 8 x RJ45 connectors using StudioHub® pinout.

The ‘T’ version (e.g. AVN-PA8T) uses rear panel terminal block connectors for all physical inputs and outputs.

The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.

There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.

A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.

For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.

8 Way Analogue Headphone Distribution System

The AVN-PA8 can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-Portal series is a group of three powerful audio mix engine interfaces (AVN-PA8, AVN-PD8 & AVN-PM8) capable of audio routing, metering and equalisation. They have both physical audio I/O and AES67 stream I/O and can mix and route freely from any input to any output, useful for a multitude of applications.

Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.

Applications:

• 8 channel digital mixer,


• 64 channel AES67 stream distribution amplifier from digital sources,


• Distribute 8 stereo channels of audio over an SFP fibre connection.


• IFB generator to send 64 x AES67 streams to individual belt-packs.


• 8 output headphone distribution system, with separate input mix for each headphone output and individual gain control.


• Input mixer with input/output metering and AES67 stream generation.

Webserver Software

A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.

Mix Matrix

The key to the success of the AVN-PA8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.

The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.

Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.

DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points.

The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.

Front Panel Displays, Metering & Controls

The AVN-PD8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and D-types for rear panel connectivity.

Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.

Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.

A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.

Detailed Metering Option

The ‘D’ version of the portal (e.g. AVN-PD8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.

The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.

Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.

Physical Inputs & Outputs

For the digital audio, the AVN-PD8 uses D-type sockets with AES59 digital pinout, paralleled with 8 x RJ45 connectors using StudioHub® pinout. There is individual sample rate conversion on each input.

The ‘T’ version (e.g. AVN-PD8T) uses rear panel terminal block connectors for all physical inputs and outputs.

The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.

There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.

A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.

For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.

8 Way AES3 Headphone Distribution System

The AVN-PD8 can be combined with multiple AVN-HD1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-Portal series is a group of three powerful audio mix engine interfaces (AVN-PA8, AVN-PD8 & AVN-PM8) capable of audio routing, metering and equalisation. They have both physical audio I/O and AES67 stream I/O and can mix and route freely from any input to any output, useful for a multitude of applications.

Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.

Applications:

• 8 channel microphone input mixer, with individual output gain control, input/output metering and AES67 stream generation.


• 8 channel clean-feed generator, with input mixing and gain control on inputs and outputs.


• Distribute 8 microphone channels of audio over an SFP fibre connection.


• IFB generator to send 64 x AES67 streams to individual belt-packs.


• 8 output headphone distribution system, with separate input mix for each headphone output and individual gain control.


• Input mixer with input/output metering and stream AES67 generation.

Webserver Software

A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.

Mix Matrix

The key to the success of the AVN-PM8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.

The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.

Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.

DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points. There is an adjustable input and output gain/trim and an additional mic pre-amp gain adjustment for each mic input.

The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.

Front Panel Displays, Metering & Controls

The AVN-PM8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and RJ45s for rear panel connectivity.

Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.

Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.

A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.

Detailed Metering Option

The ‘D’ version of the portal (e.g. AVN-PM8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.

The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.

Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.

Phase metering can be displayed per stereo output channel and channel idents can be shown either above or below the metering to identify each input/output.

On devices without a meter display, a smaller set of monochrome meters are shown on the main OLED display.

Physical Inputs & Outputs

For the microphone audio, the AVN-PM8 uses 8 x mic/line XLR sockets for the inputs and 8 x RJ45 connectors using StudioHub® pinout for the stereo analogue line outputs. +48V phantom power is available for each microphone input with a red LED presence indication.

The ‘T’ version (e.g. AVN-PM8T) uses rear panel terminal block connectors for all physical inputs and outputs.

The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.

There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.

A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.

For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.

8 Way Analogue Headphone Distribution System

The AVN-PM8 can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-TB20AD is a desktop version of the rackmount AVN-TB20AR intercom with a smaller form factor and an elegant sloped front. It has the same feature set and connectivity (except for an additional headset connector on the rear of the AVN-TB20AR).
Features:
• 20 illuminated key-cap Talk buttons plus Listen & Page buttons.
• Phone button for remote dialling and control of an external telephone hybrid.
• Page button and Group Talk facilities.
• Callback button with callback source display.
• Three user definable buttons.
• Speaker & microphone mute buttons.
• Mic & headset inputs, headphone & speaker outputs.
• Front panel volume control which operates on speaker/headphone outputs and incoming source levels.
• +48V phantom power for the mic inputs.
• Ethernet webserver and front panel control & configuration.
• Front panel display providing source & destination information.
• Sources from AoIP, 1 x balanced, 2 x unbalanced or S/PDIF digital inputs.
• Destinations to AoIP or rear panel balanced & unbalanced outputs.
• Advanced echo cancellation & mic AGC to prevent acoustic feedback.
• Dual 1Gb lan ports & 1Gb SFP fibre port.
• 10 user assignable GPIO ports.
• GPIO button for triggering external events, via physical GPIO or network commands.
• Front panel LEDs for network audio presence, Talk activity, AGC activity, clock sync and power supply activity.
• Two front panel monitor buttons for routing audio directly to the speaker e.g. to take an IFB feed or off-air transmission signal.
• Ducking or mixing of inputs to speaker/headphones.
• Dual AC & DC power supply inputs.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-TB6 is a 6 channel talkback intercom control unit from the Sonifex AVN range of IP based products. The AVN-TB6 is a freestanding version which can be rackmounted with the AVN-TB6RK 1U rack kit. There is also a desktop version, the AVN-TB6D which can be fitted flush into a work surface if required. Both models offers similar functionality.

This unit provides broadcast quality audio communication between studios, offices and different areas in a facility or building complex, using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion. RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. The AVN range use RAVENNA as the communication method providing compatibility with other AES67 systems.

Each of the 6 channels on the AVN-TB6 can be configured to provide communications with other remote networked units, and an independently configurable ‘page’ function can contact selected units with priority over standard intercom calls if required.

There is a monitor channel that can route the audio from an AoIP source to the headphones and speaker. This could be used to take an IFB feed or an off-air transmission signal or simply to listen to any audio source.

A user configurable GPIO system, with 10 physical ports and 10 virtual ports, can be used to control operational functions on local or networked units, or drive outputs as selected states change, and a voltage free relay contact can be used to operate external equipment.

A built-in web server provides complete configuration control of the units and also allows for firmware updates and configuration backup. An Ember+ interface also gives access to the configuration options as well as providing remote control and monitoring of the GPIO and virtual GPIO ports.Webserver Software.

The AVN-TB6 has a built-in webserver for setup and configuration. The webserver is responsive, and resizes depending on the size of your screen, meaning that it can be used on large monitors or small handheld devices such as smart-phones. Help information is shown on the right hand side of the screen so it’s a good place to go to find out how the unit operates.

Category: AES67/Dante AoIP Products.Product Function: Provides broadcast quality audio communication using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion.Typical Applications: Ideal for broadcast quality audio communication between studios, offices and different areas in a facility or building complex.

Features:

• 6 illuminated key-cap Talk buttons plus Listen & Page buttons.


• Dual 1Gb Ethernet & 1Gb SFP port.


• Mic & headset inputs, headphone & speaker outputs with volume control.


• Loudspeaker & Mic Mute buttons.


• Dual AC & DC power supply inputs.


• Advanced echo cancellation & mic AGC to prevent acoustic feedback.


• 10 user assignable GPIO ports.


• Responsive design Ethernet webserver.


• AVN-TB6RK 19” rack kit available.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-TB6D is a 6 channel talkback intercom control unit from the Sonifex AVN range of IP based products. The AVN-TB6D is a desktop version which can be fitted flush into a work surface if required. There is also a freestanding version which can be rackmounted with the AVN-TB6RK 1U rack kit. Both models offers similar functionality.

This unit provides broadcast quality audio communication between studios, offices and different areas in a facility or building complex, using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion. RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. The AVN range use RAVENNA as the communication method providing compatibility with other AES67 systems.

Each of the 6 channels on the AVN-TB6D can be configured to provide communications with other remote networked units, and an independently configurable ‘page’ function can contact selected units with priority over standard intercom calls if required.

There is a monitor channel that can route the audio from an AoIP source to the headphones and speaker. This could be used to take an IFB feed or an off-air transmission signal or simply to listen to any audio source.

A user configurable GPIO system, with 10 physical ports and 10 virtual ports, can be used to control operational functions on local or networked units, or drive outputs as selected states change, and a voltage free relay contact can be used to operate external equipment.

A built-in web server provides complete configuration control of the units and also allows for firmware updates and configuration backup. An Ember+ interface also gives access to the configuration options as well as providing remote control and monitoring of the GPIO and virtual GPIO ports.

Webserver SoftwareThe AVN-TB6D has a built-in webserver for setup and configuration. The webserver is responsive, and resizes depending on the size of your screen, meaning that it can be used on large monitors or small handheld devices such as smart-phones. Help information is shown on the right hand side of the screen so it’s a good place to go to find out how the unit operates.

Category: AES67/Dante AoIP Products.Product Function: Provides broadcast quality audio communication using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion.Typical Applications: Ideal for broadcast quality audio communication between studios, offices and different areas in a facility or building complex.

Features:

• 6 illuminated key-cap Talk buttons plus Listen & Page buttons.


• Dual 1Gb Ethernet & 1Gb SFP port.


• Mic & headset inputs, headphone & speaker outputs with volume control.


• Loudspeaker & Mic Mute buttons.


• Dual AC & DC power supply inputs.


• Advanced echo cancellation & mic AGC to prevent acoustic feedback.


• 10 user assignable GPIO ports.


• Responsive design Ethernet webserver.


• AVN-TB6RK 19” rack kit available.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-GMCS is a PTPv2 grandmaster clock for use with AoIP applications. IEEE1588-2008 PTPv2 (precision time protocol) is used to synchronise all the nodes within a network. To achieve this one of the nodes must become the master clock and distribute time packets to the others. The AVN-GMCS is designed to perform this role simply and accurately, enabling sub micro second synchronisation between all nodes.

RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. For this to be possible, each of the nodes needs to be time synchronised with one another. RAVENNA uses PTP time stamping to achieve this, which distributes the network time but also works out the latency involved in the delivery and adjusts the time at each node accordingly.

Unit configuration is achieved easily either with the front panel controls or the webserver, including the setup of the PTP profiles.
The AVN-GMCS supports the Default (RAVENNA), Media (AES67) and AES-R16-2016 (SMPTE-ST 2059-2 & AES67 compatible) profiles and has a ‘Custom’ profile page for you to define your own.

In normal operation, the unit has PTPv2 time stamping resolution to 8nsec. It uses a combination of a GPS receiver, a PLL (phase lock loop) and a specialist on-board clock device to create the precise, low jitter clock signals required to drive the physical transceiver’s time stamping circuitry, also providing holdover if the GPS signal is lost.
The specialist on board clock is available in three different types: TCXO, OXCO and CSAC (Chip Scale Atomic Clock, Caesium), which vary in both price and accuracy:
AVN-GMCS – TCXO Temperature Compensated Oscillator accurate to 1 part per million (worst case 1 sec gain/loss every 11.5 days). *
AVN-GMCOS – OCXO Oven Controlled Oscillator accurate to 0.1 parts per million (worst case 1 sec gain/loss every 115 days). *
AVN-GMCCS – SAC Quantum Atomic Clock accurate to 0.0005 parts per million (worst case 1 sec gain/loss every 63 years). *

GPS presence and the number of satellites received is shown on the front panel, together with status information on output sample rates, sync type and profile type. The unit also has a screen-saver option which shows the current time.

Although designed as a grandmaster clock, a separate clock input can act as an alternative reference source to GPS which the unit can ‘slave’ to. Clock outputs, driven from the physical transceiver, can be used to provide media clocks for external equipment local to the AVN-GMC when it is in both ‘master’ and ‘slave’ states. The clock outputs are available as a single AES-3id output and two outputs which can be selected as either word clock or variable PPS. The wordclock can operate at 32, 44.1, 48, 96, 176.4 and 192kHz. When set as a variable PPS output, the unit can act as a clock master to distribute a reference frequency to test and measurement equipment.

The unit shows UTC as standard, but can be set to show ‘local time’ on the front panel, by adding a time offset. Daylight saving time changes can be accommodated by entering Spring Forward and Fall Back dates. It has a real time clock so that accurate date and time is available even after the unit is repowered without GPS access.
The built-in webserver, or front panel OLED display, can be used to configure the unit. The webserver is a responsive design meaning that it can be used with small screens on smartphones and tablets.

Front panel LEDs show the synchronisation status, GPS lock and the status of the AC and DC power supply inputs.

The brightness of the OLED display and LED indicators can be adjusted for low or high lighting conditions 4 general purpose outputs indicate critical states for the unit using a 9 way D-type connector mounted on the rear panel. Pull down when active pins are supplied for GPS lock status, external sync present, AC power present and DC power present.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The Sonifex AVN-PXH12 12 x 2 Channel Mix Monitor is a world first – it can simply and effectively simultaneously monitor and mix up to 24 AES67 AoIP sources in a compact 1U rack to a speaker/headphone combination in a quick and intuitive way so it is ideal for use in production galleries, OB trucks and engineering racks rooms. The product was winner of the AMI \'Best of Show Award\' for IBC2017, the BaM Awards™ April 2018 and has been nominated for a 2018 ‘AV Award’.

The 24 audio sources can be selected from 4 discrete stereo analogue audio inputs (1 x front panel 3.5mm jack socket, 2 x rear panel 3.5mm jack sockets and 1 x rear panel stereo XLR input pair) or from any RAVENNA, AES67 or AES67-enabled Dante® AoIP connected streams.

These stereo signals are routed to the 12 x control channels on the front panel, each of which have a ‘Normal’ and an ‘Alternate’ input selection. Each channel has three buttons: one for input selection, another to Mute the channel and the third to select whether the channel input is routed to the left, right or stereo output legs. The knob for each channel controls the level of the input routed to the output and the knob also illuminates either green, amber or red to show input level. Pressing the knob ‘Solos’ the channel input to the output.

The front panel has 3 outputs: paralleled stereo headphones on 6.35mm (¼”) jack and 3.5mm jack sockets, each with their own individual attenuation settings, and a mono-mix speaker output. There are discrete volume controls for the headphones and the speaker, and the latter also has a mute button.

The rear panel has an additional 3 line level XLR-3 audio outputs, which can be designated as mono mix or left or right channel outputs of the mixed audio content (similar to the speaker and headphone outputs respectively), or any of the physical inputs or AoIP input sources.

The unit also sends to the network, as AoIP AES67 streams, the 8 channels of the 4 physical stereo inputs, together with a stereo mix of the speaker output.

Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product. There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection.

A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software controlled functionality. A voltage free relay contact can be used to operate external equipment.

A built-in web server provides complete configuration control of the unit including source assignment to each channel and also allows for firmware updates and configuration backup. The unit can be controlled by suitable Ember+ commands.

-ends-

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-TB10AR is a 10 button intercom meaning that 10 other ‘stations’ can be defined, one per button, for communication. Comms can be made as a Talk action, a Listen action or a duplex Talk/Listen action to/from each station. Coloured LEDs in the buttons help to show which action is being used and there is also a Callback button for when you’re unavailable to receive a call.

The stations can be from anywhere on the AoIP network and the use of Bonjour Device Discovery means that other stations can be found quickly and sometimes automatically.

The Page button is used to speak to all stations (or a defined list of stations) and Group Talk functions can be enabled to page particular groups of stations.

Two monitor buttons allow for routing audio directly to the speaker e.g. to take an IFB feed or an off-air transmission signal. Signals can be ducked or mixed when a talkback input is received to the speakers or headphones.

Three user defined buttons can be programmed for different functions, such as for Group Talk.

The speaker mutes automatically when headphones are inserted and the volume level of headphones, speaker and incoming sources can all be controlled with one front panel rotary encoder volume control knob, which shows the level using RGB LEDs around the outside of the knob.

Advanced acoustic echo cancellation & built-in microphone AGC (automatic gain control) ensure that there’s no acoustic feedback between microphone and speaker.

Buttons are available for microphone mute (cough) and speaker mute actions and these can be controlled remotely by GPI or network commands.

Each unit has a built-in webserver which is where the majority of settings and configurations are made. The front panel OLED display can also be used to configure the unit, although more functionality is available by using the webserver. The webserver is a responsive design meaning that it can be used with small screens on smartphones and tablets.

The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.

Front panel LEDs show the AoIP network status, synchronisation status, whether AGC is being used and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be adjusted for low or high lighting conditions.

The unit has a front panel power button and dual power connectors - an IEC mains input and a 12V DC input, which allows the AVN-TB10AR to be used for both studio and mobile installations. Also, a secondary power source reduces the effect of power down events. In any case, the unit monitors the status of both power sources and displays this on the front panel.

10 GPIOs (general purpose inputs/outputs) and a programmable relay output can be configured to indicate critical states for the unit, via the 15 way D-type connector, for example, to show loss of DC power, or to show a button press action.

Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »

The AVN-TB20AR is a 2U rackmount 20 button version of the AVN-TB10AR with the same specification, but more station buttons allowing greater communication for larger facilities. It also has ‘GPI/O’ and ‘Phone’ buttons for remote control of external equipment and Sonifex DHY-04 telephone hybrids, useful in an OB truck or production gallery.

Features:
• 20 illuminated key-cap Talk buttons plus Listen & Page buttons.
• Phone button for remote dialling and control of an external telephone hybrid.
• Page button and Group Talk facilities.
• Callback button with callback source display.
• Three user definable buttons.
• Speaker & microphone mute buttons.
• Mic & headset inputs (front & rear panel headset connection), headphone & speaker outputs.
• Front panel volume control which operates on speaker/headphone outputs and incoming source levels.
• +48V phantom power for the mic inputs.
• Ethernet webserver and front panel control & configuration.
• Front panel display providing source & destination information.
• Sources from AoIP, 1 x balanced, 2 x unbalanced or S/PDIF digital inputs.
• Destinations to AoIP or rear panel balanced & unbalanced outputs.
• Advanced echo cancellation & mic AGC to prevent acoustic feedback.
• Dual 1Gb lan ports & 1Gb SFP fibre port.
• 10 user assignable GPIO ports.
• GPI/O button for triggering external events, via physical GPIO or network commands.
• Front panel LEDs for network audio presence, Talk activity, AGC activity, clock sync and power supply activity.
• Two front panel monitor buttons for routing audio directly to the speaker e.g. to take an IFB feed or off-air transmission signal.
• Ducking or mixing of inputs to speaker/headphones.
• Dual AC & DC power supply inputs.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

With its swiss-army-knife approach, the Nimbra 400 family is the very definition of versatility. Engineered for the harsh realities of the internet, the Nimbra 400 family supports a wide range of video and transport technologies. The combination of native support for both AVC and HEVC encoding/decoding, compressed video over IP and ASI and an open and flexible approach to IP networking in a super-efficient 1 RU form factor makes the Nimbra 400 family a perfect companion for space constrained remote sites and flight kits. With native support for RIST, SRT, Zixi and our own Edge Connect software means you can get your content anywhere. The integrated encryption and firewall paired with transport of any remote management traffic and comms also minimizes the need for third party equipment on site.

Paired with its standalone software counterpart, the Virtual Nimbra, means there are solutions ranging from highly redundant appliances for flight-kit and racks, to flexible software for datacenters and public cloud providers.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Combining native video, audio and high quality ethernet transport with switching, the Nimbra 300 provides a high-quality multiservice solution for demanding media applications. Multi-service capabilities allow you to transport live and file-based content across the same network, and with native support for DTT (ATSC 3.0, DVB-T2, ISDB-T) transport, Broadcast Radio distribution and video contribution the Nimbra 300 series gives you all the flexibility you need.

With support for service admission control and guaranteed bandwidth, performance and characteristics different services can safely be mixed no matter the network load. With hitless 1+1 protection of media and data services you can be sure that the expected service quality will always be delivered.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Next generation WAN transport-platform designed for all-IP services and all-IP networks, guaranteeing quality for the most demanding live media events while providing agility through automated real-time provisioning. Scaling both switching and link capacities to levels unprecedented in the media WAN space, enabling event sites, stadiums, studios, and production sites to be interconnected using a total of 1,2 Tbps of guaranteed capacity in just a compact 3RU modular form factor.Future proof building practice with fully flexible cooling and power configurations allow deployment in any modern service provider, broadcaster or data center environment. Designed with a virtualization architecture, using one common hardware platform for all media and network functions, enabling flexible deployment of functions anywhere in the network. Modular and redundant, with hot swappable and protected components guarantees uptime and reliability.

Connect » Provide Connectivity » Connectivity & Bandwidth » Fiber and private IP connectivity »

The LTN® Network is a broadcast-quality IP multicast network with 99.999% reliability and ultra-low latency. The Network connects 3,000+ locations with global audiences and powers LTN solutions to drive content value.

With proprietary protocols for dynamic multicarrier routing and rapid error recovery, the LTN Network is engineered to outsmart congestion and preempt failure, ensuring consistent live video delivery.

Backed by the LTN Network, major media producers and distributors expand their reach, increase their revenue, and deliver transformative video experiences.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Whether it’s for contribution or distribution, full-time or occasional use, LTN Leaf is a cost-effective, fully managed solution for transmitting or receiving broadcast-quality video.

The Leaf appliance requires minimal on-site setup, and features a compact, integrated design that harnesses the flexibility of multiple encode and decode configurations. Leaf also enables seamless access to the LTN Network, with real-time monitoring and always-on support.

The Leaf encode mode supports interactive interviews and live shots with lower end-to-end latency than satellite. Leaf also features broadcast-quality encoding for fast-motion sports and complex tape playout content at bitrates up to 20 Mbps, with high-quality multi-codec decoding of up to four simultaneous feeds.

Create the future of video with LTN Leaf.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Next generation of media transport designed for ultimate reliability, flexibility and versatility. Nimbra Edge is a fully cloud-based media transport solution designed for private or public cloud environments.Nimbra Edge augments existing Nimbra devices by drastically simplifying the network architecture through offloading complexity such as redundancy and scalability to its cloud-based backend. With its open approach to networking, Nimbra Edge connects to any media device supporting RIST, SRT, Zixi or Net Insights Edge connect software. The latter designed to cloud-enable any third-party device or workflow.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

LTN Leaf Flypack™ is a portable connection to the LTN Network that enables professional broadcasters to deliver flawless live events — easily, at scale, and from any location. Packing the power of a satellite truck in a cost-efficient carry-on, Flypack provides broadcast quality, ultra-low latency, and global reach, backed by our fully managed and monitored multicast IP network.

A complete IP solution for any live event, Flypack features plug-and-play setup, offers encryption capabilities, and encodes and decodes at up to 20 mbps with SD and HD support.

By connecting Flypack to the LTN Network, you can manage and control video from origin to destination, monitoring every signal with LTN Portal and always-on NOC support.

Power the future of video with LTN Leaf Flypack.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

CatonNet Video Platform (CVP), powered by Caton Transport Protocols, provides broadcast grade media transmission services. CVP is designed as a highly available, low latency and secure service ensuring your video can be transmitted and received without compromising on quality, security or speed. With a point of presence in over 60 countries worldwide, some of the largest content creators, broadcasters, satellite operators and service providers are using this fully managed end-to-end service with up to 99.999% stream availability to securely deliver superior video to their customers locally, regionally and globally. All while enjoying significant cost savings compared to traditional network services.

Support » Monitoring, Control & Communications » System Monitoring & Control » Broadcast system control SW and HW »

 

HOME is a management platform for IP-based media infrastructures. It is designed to connect, manage and secure all aspects and instances of live production environments. HOME provides the tools and centralized services for swift and effective interaction of engineers with their tools.

HOME is cloud-native by design and ready to run anywhere, irrespective of the system’s size. With HOME, the cloud starts on your campus, private and locally. It turns an array of devices, setups, sites, hubs and data centers into a powerful, agile network — quickly and in a perfectly secure way.

Inside HOME, discovery of devices is automatic, while registering and admitting them to the network is only a button press away.

With the adoption of IP well underway, the focus of operators has shifted from whether to adopt IP to how to use its potential with minimal effort and maximum effect. This is where HOME shines: it addresses all pressing issues real-world operators face today and tomorrow. In one place and via a single, platform-agnostic, intuitive user interface.

Lawo’s HOME platform is based on open standards such as ST2110, NMOS, IEEE802.1x and RADIUS and has been designed and built from the ground up using LUX—the Lawo Unified Experience, a framework for conceiving, designing, and building solutions that put you first and defines the standard for user experience and design across the Lawo portfolio.

Discovery and Registration: HOME solves IP complexity with automatic plug & play discovery of IP audio and video devices, which are registered with their name, location, status and type. This applies not only to Lawo products but to third-party solutions as well via NMOS. Discovered devices are managed in a central inventory list, ready for access and configuration.

Device Management: In today’s hectic live broadcast environments, operators rely on speedy, unified device configuration routines, especially when setting generic device parameters or configuring senders and receivers. The ability to save and recall configurations is key to speed up tasks. HOME provides a centralized “mission control” for these processes, providing fast and unified access to device parameters for easy tweaking, irrespective of the end point being controlled.

Operability: With its simple, user-friendly UI, HOME allows users to organize and access processing services. With all required facilities accessible in one place, operators can set up and change stream configurations, and route them across an infrastructure without the need for a separate controller. For large infrastructures HOME works seamlessly with a broadcast controller in the same set-up and helps to speed up configuration and operation. HOME is based on LUX, a UI language common to all Lawo devices and many of their functionalities. Through HOME’s user interface, operators can access and edit device parameters quickly utilizing integral mechanisms that help get the job done efficiently. With HOME, operators quickly get right to what they’re looking for, without distractions and complications, to focus 100% on the task at hand.

Security: The content created by a production crew and transported over a network is any operation’s most valuable asset and deserves strong protection. While a robust security system needs to cover all aspects of media infrastructure and content creation, the key lies in its simplicity. HOME provides a variety of security strategies, first of which is quarantining unknown devices when they come online. Only after being deliberately approved, via an intuitive IEEE802.1X-based routine, can they begin exchanging signals with the HOME network. Secondly, HOME uses an authentication strategy based on a centralized user management system, with dedicated user roles and groups. The LDAP based service allows users to authenticate either locally – within HOME – or via their own corporate IT infrastructure, e.g. Microsoft® Active Directory. Finally comes the arbitration of devices and individual streams based on pinpointed rights management. HOME’s architecture is prepared to manage services such as transport layer security, network segmentation and other IT security mechanisms such as RADIUS.

Scalable Architecture: Home is cloud-native by design, which means that its architecture is built to run detached from hardware constraints. This does not automatically mean that services must be outsourced to an external service provider whose meter is running 24/7; with HOME, the cloud starts on your campus, private and locally, on COTS hardware. The HOME platform is designed as functional blocks that provide microservices, which are self-contained and supply functionality to operators or other services.

HOME can be expanded with additional services at any time to increase its functionality — the platform scales on demand. Should there be a need for a larger RDS, because the installation grows, additional instances of the required resources can be added anytime. One of the core principles of HOME is its focus on the utilization of open standards wherever possible, for broadest compatibility and future-proof integration. With HOME, flexibility and resource utilization in IP media infrastructure maximizes.

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Connect » Provide Connectivity » Access Connectivity »

The next generation of digital STL
Our challenge: to design a studio-to-transmitter link (STL) for your most demanding 950 MHz applications — one as reliable and robust as Intraplex® T1 and IP audio links. Digital STL manages much more than audio, so installation and configuration need to be straightforward, not a science project. Data should not be optional, requiring additional boxes and complexity. Your STL should be ready, out of the box, for AM, FM and HD Radio™, as well as future multimedia applications.

Earlier digital STLs were not designed for IP data transport. Adding IP data to them required optional modules and external add-ons, and many engineers working on HD Radio installations reported spending much time and money trying to eliminate glitches.

Our solution: HD Link™ is designed to manage all HD Radio™ transport scenarios, regardless of where you place your importer and exporter. Its two prioritized Ethernet paths give preference to HD Radio™ data over control and other LAN/WAN data. It supports both UDP and TCP, and even handles the switching of TCP return packets over asymmetric IP paths with plug-and-play simplicity.

HD Link offers RF power to spare, an integrated IP gateway with sophisticated data handling capabilities, and multiple channels of audio. The intuitive front panel and remote interfaces tap into the most complete feature set of any microwave STL, yet take less time to configure. It can even operate on both RF and IP simultaneously, and allows automatic backup of all services from one to the other.

Intraplex is broadcasting’s first choice for rock-solid, full-time operation of T1 STLs. With HD Link, you can now count on the same dependable performance, superior support and long-term value for your microwave links.

Connect »

The Intraplex® IP Link family of IP audio codecs provides high-end features at an affordable price. Offering an array of audio coding options along with IPConnect technology for data tunneling, the IP Link codecs are suitable for use in Studio-to-Transmitter Links (STLs) as well as audio contribution and distribution networks. Support for IP multicast and multiple unicast streams enables one encoder to feed multiple decoders.

By incorporating three IP Interfaces that can be used for streaming and management, the IP Link systems can provide a level of reliability not seen in comparably-priced codecs.

As the latest additions to the Intraplex line of data transport products, the IP Link family of audio codecs bring legendary Intraplex reliability to the IP codec market.

IP Link 100/100p: Single bidirectional stereo audio channel IP Link 200: Two bidirectional stereo audio channels IP Link 200A: Two bidirectional stereo audio channels with one channel being AES67 Standard: Linear, AAC-LC, Opus and G.722 audio coding Optional: AAC-HE, AAC-HEv2, AACELD, MPEG2, MPEG3 and Enhanced aptX audio coding Optional: Automatic audio loudness leveling and metering compliant with EBU R-128 and ITU-R Optional: IPConnect capability to reliably transport external IP packets Other transport modes: Transparent AES up to 192 kbps to support composite FM multiplex signal over AES Protocol Encapsulation: RTP, SHOUTcast/Icecast, MPEG-TS Three independent IP interfaces for redundant network operation Optional redundant power supply: 12VDC or 48VDC Built-in silence sensor with optional stream switch over Automatic backup to audio playout from USB drive or external audio source Multicoding can encode the same audio source in multiple formats for STL, backup, and web streaming Optional Dynamic Stream Splicing providing “hitless” operation and T1/ E1 circuit like performance on less predictable IP networks Prioritized stream sources at the output with automatic switch over and switch back between primary and secondary streams and backup sources (streams, USB, external audio source) Programmable RTP level Forward Error Correction (FEC) scheme Programmable time diversity and interleaving of streams to combat burst packet losses Integrated with Intraplex IP Link Scheduler for automated scheduled program switching Integrated with Intraplex LiveLook (network analytics and monitoring software) N+1 redundancy with integrated control of external switching equipment IP Link 200/200A/100p: SynchroCast™ option provides dynamically managed precision delay for Single Frequency Network (SFN) broadcasting and simulcasting Support for IP multicast and multi-unicast Web browser user interface and SNMP network management Eight multipurpose contact closure inputs and outputs provide: Transport of logic signals with time-alignment to audio Stream control Alarm notification

Connect »

Intraplex® IPConnect is an IP networking gateway solution for fortification of mission-critical communication across wide area networks
IPConnect solves the problem of reliable transport across complex networks that traditionally suffer data loss through dropped packets. IPConnect provides an added layer of reliability for many types of IP traffic such as HD Radio, DAB/DAB+, or DRM by employing a combination of packet protection schemes with network/time diversity and packet-level forward error correction. Additionally, IPConnect can bridge local area network (LAN) segments across wide-area networks with seamless tunneling to enhance reliability of program and signal transport across large geographical regions with multiple receiving and transmitting sites.

With its three network interfaces, IPConnect’s data gateway provides extra protection through packet encapsulation, which encloses external IP data packets in a GatesAir protocol wrapper as it moves across IP networks. IPConnect provides this protection for studio-generated data, as well as IP data from external sources. While the former might include RDS data or SNMP control signals to trigger a command at the transmitter site, the latter may incorporate program or control data coming from a network operations center, satellite feed or advertising service.

As the next-generation member in GatesAir Intraplex family of IP codecs, the award-winning IPConnect leverages the strengths of previous IP Link codecs—including GatesAir’s patented Dynamic Stream Splicing (DSS) technology to mitigate IP packet loss, eliminate off-air time, and optimize stream reliability.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK1616S is a versatile break in/ out box for sending/ receiving both analogue and digital (AES3) audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.

In total there are 16 channels of audio sent from the Dark1616S into the network. The Dark1616S has 8 of AES3 inputs and 16 of analogue inputs. Using the GlenController App it is possible to set which inputs (AES3 or Analogue) are being sent to the Dante/ AES67 network, an auto mode is available that sends AES3 (when a valid signal is detected) in preference to the analogue input.

Simultaneously there are 16 channels of audio being received from the network by the Dark1616S and these incoming circuits are provided as outputs from the Dark1616S in both AES3 and analogue.

The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz, the incoming AES3 circuit is always sample rate converted to match the Dante network frequency.

The AES3 outputs have sample rate converters on them and can be locked to the sample frequency of the Word clock input, the first AES3 input or to the Dante network.

A Word clock output is also provided, this is clocked at the same sample rate as the Dante network and can be used for locking external equipment to the network's sample rate.

The unit can be remote controlled using our GlenController Windows 10 App. The App allows such things as the dBFs levels to be set, clock masters and also provides input/ output metering. See the App tab for further details.

For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.

Being designed for resilient broadcast applications, the Dark1616S features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616S lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616S units can be directly connected with each other with no external network hardware.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The Glensound DARK1616D is a digital to digital converter designed to connect AES3 audio circuits to/ from a Dante/ AES67 audio network.

Dante network audio is a common protocol for distributing high quality linear audio over standard IP networks and it is widely used by many audio equipment manufacturers. The Glensound Dark1616D Dante audio interface will be compatible with any other manufacturers Dante audio interface.

Being designed for live on-air broadcast applications the Glensound Dark1616D has been designed with multiple redundancy capabilities. It has 2 mains power sources and it also has fully redundant network connections for both Copper & Fibre circuits.

The Dark1616D provides 8 balanced AES3 inputs and 8 balanced AES3 outputs to the Dante network on rear panel D25 connectors wired to AES59 (also known as the Tascam standard).

As per our other Dante equipment 0dBu = -18dBFs

Create » Content Acquisition and Capture » Audio Aquisition » Audio acquisition accessories »

The GS-FW012 ip is the Dante network audio version of the GS-FW012 4W unit.

There are still four inputs for the top panel loudspeaker or headphone monitoring, each with their own level controls and each being derived from the Dante network.  There are still four talkback outputs each being routed to the Dante network.

Both the inputs and outputs are presented via a single rear panel Neutricon RJ45/CAT5 connection that is Dante/AES67 compliant.

The four talkback outputs are identical.  Each has a 3 position lever key talk switch that is either off, locked on, or in a sprung push to talk mode.  The output of each circuit can be off unless the talkback button is pressed, looping the the associated Dante input input, or outputting the local cue input.

The local cue input is presented on XLR and has it's own level pot for the speaker or headphone monitoring.

A good quality microphone amplifier with adjustable gain followed by a compressor limiter is built into the unit making its talkback outputs to the Dante network crystal clear. Conversion of the audio signals to/ from the Dante network are by low noise, high bandwidth 48K 24bit converters providing superb performance.

Audio I/O can be routed via the digital Dante router.  The free Dante Controller software configures all routes and can be downloaded by clicking here.

There is an internal switch mode power supply, or the GS-FW012 ip can be powered via PoE via the CAT5 connection. If both supplies are present then they will act as redundant supplies.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK16O is a versatile break out box for receiving analogue audio from a network utilising the Dante and AES67 audio over IP (AoIP) protocols.

In total there are 16 channels of audio being received from the AoIP network and converted via low noise DACs (Digital to Analogue Converters) to balanced analogue outputs.

For ease of cabling audio outputs are presented as balanced circuits on Neutrik XLRs.

Being designed for resilient broadcast applications the DARK16O features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16O lost its primary link circuit, the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16O units can be directly connected with each other with no external network hardware.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK16I is a versatile box for converting analogue audio inputs to an AoIP system utilising the Dante and AES67 audio over IP (AoIP) protocols.

In total there are 16 audio input channels, all being simultaneously converted to the AoIP network via high quality low noise analogue to digital converters.

For ease of cabling audio inputs are balanced circuits on Neutrik XLRs.

Being designed for resilient broadcast applications, the DARK16I features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16I lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16I units can be directly connected with each other with no external network hardware.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK88 MKII is a versatile break in/ out box for sending/ receiving analogue audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.

In total there are 8 channels of audio sent from the DARK88 MKII into the network. The DARK88 MKII has 8 of analogue electronically balanced audio inputs on Neutrik XLRs.

Simultaneously there are 8 channels of audio being received from the network by the DARK88 MKII and these incoming circuits are provided as outputs from the DARK88 MKII in analogue.

Network sample rates of up to 192KHz are accommodated seamlessly within the DARK88 MKII.

Being designed for resilient broadcast applications the DARK88 MKII features both redundant power supplies and redundant Dante network links. Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK88 MKII lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK88 MKII units can be directly connected with each other with no external network hardware.

On the front panel, LEDs indicate the status of the 2 power supplies and the 2 network links. GPO status outputs are also provided for external indication of the power supply & network status.

Network connections are placed on the front panel of the DARK88 MKII in order that the network cables (or fibres) match those of a rack mounted professional network switch, making installation and tracing interconnecting cables easy. Fibre connections are via SFP slots, meaning that users can select their own preferred fibre type & connector style by installing their own fibre SFP modules (a selection of modules is available from Glensound if preferred).

The DARK88 MKII features the Brooklyn module from Audinate which is AES67 compliant.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK8ADI (Analogue & Digital Input) is a very powerful Dante®/ AES67 network audio interface, in a robust 1RU 19" subrack It has a 8 of low noise analogue line inputs and 4 of transformer balanced AES3 digital audio inputs. These inputs are transmitted on a fully redundant Dante/ AES67 network interface.

In total 16 audio circuits are sent to the Dante/ AES67 network. Eight of these circuits are always derived from the 4 x AES3 inputs, the other eight are automatically switched between the AES3 and analogue inputs. If the DARK8ADI detects valid AES3 signal on a channel then this will be routed to the network output.  If no valid AES3 is detected then the analogue input will be routed to the network output instead of the AES3.

The rugged design, redundant mains powering & redundant network facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.

Eight Analogue Audio InputsThe DARK8ADI has 8 electronically balanced analogue line level audio inputs. Each input is on its own Neutrik 3 pin XLR socket.Quality Analogue To Digital ConvertersTo get the best possible results from your analogue audio inputs, the very best widest range analogue to digital converters (ADCs) currently available are used to make sure the digital audio on your network is as good as it possibly can be.Four AES3 Audio InputsThe DARK8ADI has 4 transformer balanced digital AES3 audio inputs. Each input is on its own Neutrik 3 pin XLR socket. These inputs can accept sample rates up to 192kHz.

Analogue/ AES3 Auto SwitchingEight of the audio outputs are derived from both the 4 x AES3 inputs and the 8 analogue inputs, whereby the AES3 inputs take priority to the analogue circuits. Our input circuitry looks for valid AES3 data streams on the AES3 inputs.  If one is detected then the pair of audio outputs (AES3 is two audio channels) associated with that circuit will be routed to the Dante/ AES67 output and if no valid AES3 signal is detected then the analogue input will be routed to that output.

AES3 Network OutputsAs well as the switched outputs, the outputs of the 4 x AES3 input circuits are always routed to 8 audio channels of the AoIP network.

Sample Rate ConvertersThe AES3 audio inputs are fed through sample rate converters so that they match the AoIP networks sample frequency. The AoIP network supports up to 192kHz sampling and the AES3 inputs can be between 32 & 192kHz.

Network InterfaceThere are 4 network interfaces on the DARK8ADI. There are 2 x Neutrik Ethercon (RJ45) connectors and there are also 2 x SFP slots for customers to fit their own preferred fibre interfaces.

Redundant Network InterfaceWhen using the Dante protocol it is possible to set the DARK8ADI to have a fully redundant network interface whereby a completely glitch free automatic redundant audio network link is provided across 2 of the network interfaces.

PowerEach DARK8ADI can be powered from two independent sources to provide a multi-redundant power option.Two wide range switch mode power supplies are fitted as standard to provide redundant mains power supplies.

Alarms & LEDsThe front panel features two LEDs (one indicating OK and the other fault) for both power supplies and both network ports. This data OK/ Fault information is also provided on solid state relay outputs on front panel D connectors.

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The DARK 1616M is an audio interface that provides connections to a Dante or AES67 IP network.

Inputs & OutputsThere are 16 digital inputs and 16 digital outputs, available on 8 x AES connections.  There are also 16 analogue inputs and 16 analogue outputs, available in parallel to the digital.  The analogue and digital outputs are always available, and the inputs work either on AES input priority or can be selected via the control app.

The AES3 inputs have sample rate converters and can accept input frequencies up to 192kHz.  The incoming AES3 circuit is always sample rate converted to match the Dante network frequency.  The AES3 outputs are locked to the sample frequency of the Dante network.

ConnectionsFor ease of cabling, audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.

Remote Mic AmpsThe analogue inputs are switchable between mic, line and 48v phantom power.  These are controllable across the network via the Dark Controller Windows 10 app.  This allows remote control by an engineer of the input selection, gain adjust and input on/off control.  A meter is also provided for each input on the app, to monitor the input level and help with setting the gain.

Network InterfaceBeing designed for resilient broadcast applications the Dark1616M features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616M lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.

AlarmsOn the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.

Publish » Non-Linear Publishing » Non-Linear Publishing Systems » Cable/IPTV VOD servers and systems »

Software solution for sophisticated transport stream splicing. Runs on generic enterprise grade Windows server hardware. Scalable and flexible configuration. Low propagation delay and avoids full steam decoding when operating in frame accurate mode,

Applications include Ad Insertion, Slate insertion (covering content) or live content replacement.

Support » Design, Build, Run » Implementation Services » Consulting services »

Edge, is available as a lightweight functional block on a VM, container in-line with other software or on bare metal which is specified according to the required throughput.

It can either be used standalone through a web interface or in most cases connects over websockets to our control and monitoring platform called CORE (deployed on prem or in the cloud).

Typical use-cases are secure and reliable Internet transit, a monitoring probe, 2022-7 protection, Input protection and (De)Scrambling.

 

Functions that set itself apart from a native SRT implementation are:

•    SMPTE 2022-7 seamless input with per path statistics

•    SRT RTP header passthrough allowing 1+1 paths followed by downstream 2022-7 seamless protection

•    BISS-2 scrambling and descrambling with seamless odd/even key transition

•    Thumbnailing (based on ffmpeg library)

•    ETR290 Priority 1, Priority 2 & Priority 3 (fully written in house)

•    UDP and RTP output with packet pacing and launch delay offset

•    Threshold based input redundancy with support passive or active backup sources

•    CORE (CORE is the name of our control platform) - Centralised control and monitoring capable of controlling and monitoring through firewalls and presenting a northbound REST control and monitoring API

•    CORE - High availability horizontally scalable cluster based control and monitoring capable of managing tens of thousands of EDGEs.

•    CORE has a REST API with existing integrations with ATOS BNCS and Skyline Dataminer.

•    CORE supports local, LDAP, SAML and Azure AD authentication.

•    CORE - Grafana native driver for correlated monitoring of all software components

•    Floating centralised license model with both OPEX and CAPEX options

•    EDGE Debugging tools - top talkers & PCAP

•    HLS input and output

Publish » Internet Distribution » Internet Distribution » Content delivery networks (CDNs) »

Network 1 is a reliable and cost-effective internet-based platform, for the primary distribution of linear content. It offers content owners 24/7/365, point to multi-point channel distribution to global audiences. Network 1 empowers customers to focus on the content, not the distribution.

From public service broadcasters, to global media giants, the UK's oldest commercial network and premier league football teams, Network 1 customers are increasingly utilising global connectivity to transport linear video content.

Network 1 offers significantly quicker implementation than traditional fibre and satellite distribution methods, with set-up normally measured in hours rather than months. Network 1 is multi-cloud compatible, protocol agnostic and delivers over managed or unmanaged networks. Infinite scalability ensures on-boarding new channels is easy and if no hardware is required, customers can deliver to new affiliates or broadcast platform operators straight away.

Network 1 is a fully managed service, with network management and monitoring capabilities, offering complete reliability.

Publish » Internet Distribution » Internet Distribution » Internet distribution platforms (incl. social media) »

Livelink is a robust IP delivery solution for transporting live linear and OTT content from point to single-point or multi-point. It allows users to self-manage their live content for delivery to any destination worldwide by utilizing cloud environments available in every region. Livelink is cloud agnostic and compatible with any transport protocol including Zixi, RIST, SRT, HLS and RTMP.

Livelink’s expand on demand infrastructure ensures that teams can quickly scale their requirements within minutes. This flexible approach enables stakeholders managing live broadcasts and events to adapt to changing circumstances. Livelink does not rely on traditional infrastructure which makes it extremely cost-effective. Its intuitive self-service interface enables users to easily transport, manage and extract value from live content. The platform integrates rights management, automatic booking updates and customisable monitoring and alerts so that customers retain control of all aspects of delivery.

Livelink offers broadcasters the ability to deploy and deliver live feeds with minimal on-site presence. The platform offers global reach and is the ideal solution to reduce live event overheads and also supports current social distancing measures which the broadcasting industry has implemented. The self-management interface monitors availability, routing and scheduling for all feeds, as well as tracking cloud environment costs. Cerberus Tech can fulfil hardware requirements, direct to the end user if required. However, in many instances no additional hardware is required for set-up, just an internet connection.

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The DARK1616 is a versatile break in/ out box for sending/ receiving both analogue and digital (AES3) audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.

In total there are 16 channels of audio sent from the Dark1616 into the network. The Dark1616 has 8 of AES3 inputs and 16 of analogue inputs, but they cannot be used at the same time.   The AES3 inputs take priority over the analogue (if an AES3 input is receiving a valid AES3 signal then it will turn off the equivalent analogue input pair and route its output to the network).

Simultaneously there are 16 channels of audio being received from the network by the Dark1616 and these incoming circuits are provided as outputs from the Dark1616 in both AES3 and analogue.

The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz.  The incoming AES3 circuit is always sample rate converted to match the Dante network frequency.

The AES3 outputs are locked to the sample frequency of the Dante network.

For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.

Being designed for resilient broadcast applications the Dark1616 features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616 lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.

On the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Scalable multiple channel transport stream switching running on generic enterprise grade Windows servers. Flexible control options and auto redundant configurations

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

L-GRID tackles the problem of speed mismatches at the source and avoids an extra investment in large buffer core switches.

L-GRID will configure the network card to only allow bursts of the configured bandwidth. Data flows to the preconfigured destinations are sent out of the network card perfectly shaped to the given bandwidth. As the L-GRID eliminates bursts before they enter the network, no exhaustive buffer space is required from the connected switches. The traffic is controlled/managed at the source. The cause for congestion is remediated before it can happen.

L-GRID enables the user to configure specific bandwidths for specific/configured clients. L-GRID runs as a user-space app on a Linux server and is configured via a config file or a REST interface. A certified network card is required

L-GRID will thereby improve the efficiency of your installed network.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

GRID encompasses a technology and a methodology to enable guaranteed data flows with effectively-enforced QoS in any IP network: a studio network, a post-production network, a data center network, etc.

GRID’s software transforms the IP network and the data flows to be fully predictable, non-blocking and 100% reliable. As a result, your network will be used much more cost-efficiently. GRID also provides several wizards to make configuration and management of the network easy and intuitive. The software enables a 100% service guarantee with a specific bandwidth, low latency and no packet-loss per service or flow.

GRID is compatible with the hardware of multiple vendors.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The AVN-PA8/D and the AVN-PM8/D Portals can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers and the unit can be mounted using the CM-MNT1 desk mount panel.

The AVN-PA8/D and AVN-PM8/D can be combined with multiple Sonifex AVN-HA1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.

• Front panel 6.35mm (1/4”) headphone socket and volume control knob.


• Front panel Mute/GPO push button.


• Analogue audio input on RJ45 (the connector provides power to the unit and a GPO back to the portal).


• Loop through audio output on RJ45 (power and GPO signal are not connected).


• Locking DC power connector (if a portal is not being used to supply the unit with power).

Note: The AVN-HA1 is an analogue input product taking an analogue audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate analogue input.

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The AVN-PD8/D Portal can be combined with multiple AVN-HD1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers and the unit can be mounted using the CM-MNT1 desk mount panel.

The AVN-PD8/D can be combined with multiple Sonifex AVN-HD1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.

• Front panel 6.35mm (1/4”) headphone socket and volume control knob.


• Front panel Mute/GPO push button.


• AES3 digital input on RJ45 (the connector provides power to the unit and a GPO back to the portal).


• AES3 digital output on RJ45 (power and GPO signal are not connected).


• Locking DC power connector if a portal is not being used to supply the unit with power.

Note: The AVN-HD1 is an AES3 digital input product taking an AES3 audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate AES3 audio input.