AVN-DIO09 Microphone Input to Dante®

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

LTN Wave is an IP multicast transport solution that gives you end-to-end visibility and control. The smarter alternative to legacy distribution, Wave unites the quality and availability of satellite with the flexibility and revenue-generating potential of IP.

LTN enables highly reliable, secure delivery via the industry’s only fully managed and monitored global network. Through the LTN Network, Wave integrates with the LTN Ecosystem of universal media technology services, from monitoring and control to ad enablement, for a solution targeted to your needs and objectives.

We accommodate any support requirement, from operations enablement for customer teams to complete LTN service management. And our expert network operations center is always on.

Harness reliability and efficiency at scale with a new wave of broadcast-quality transmission.

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

Matrox® Extio 3 is the world's first IP KVM extender capable of supporting 4Kp60 or quad 1080p60 4:4:4 video, keyboard, mouse, USB 2.0, and audio at unprecedented low-bitrates over a standard Gigabit Ethernet network. With exceptional remote performance, smooth video playback, and pristine image quality. Designed to ease integration and provide operational flexibility for information sharing, fast decision making, and intuitive collaboration, Matrox Extio 3 IP KVM is ideal for a wide range of secure extension and switching applications in environments such as process control rooms and dispatch, as well as network operation centers in healthcare, military, transportation, government, broadcasting, and industrial automation and control.

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.

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

Vena, our smart broadcast network, is set to transform your business by intelligently connecting your media supply chain. From contribution to distribution, Vena gives you control and visibility. Any-time changes, real-time tracking, unrivalled availability, and 24/7 access, all making it quicker and easier to deliver your content.

Get ready to be ...

More versatile

You can select, configure and book temporary on-demand and permanent services quickly. Adding and removing services as you need them for greater control.

More efficient

Enjoy complete, end-to-end control at your fingertips, automate your workflows, and create role-based rules.

More in control

You can see your network usage and capacity across real-world maps in order to schedule bookings and make business decisions in real-time.

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

 

The Magellan™ SDN Orchestrator (SDNO) is a proven software control solution deployed in multiple installations worldwide. It delivers full support for SMPTE ST 2110 and ST 2022-6, plus the most comprehensive list of third-party IP endpoints via native APIs and open protocols, making it ideal to orchestrate complete live production and playout operations.

Magellan Live Manager is an integrated management environment across various Imagine products. Operating hand in hand with Magellan SDNO, LM consists of "LM-Apps" including PathView and AlarmView, which provide high-value user interaction and information

 

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

 

Selenio Network Processor (SNP) is the industry’s first pure-IP media processing platform, supporting uncompressed UHD based on SMPTE ST 2110. This high-density, scalable platform enables media companies to seamlessly transition from SDI to all-IP networks, while adhering to the SMPTE ST 2022-6 and ST 2110 standards.

The standards-based SNP interoperates with Imagine’s other UHD- and IP-ready solutions, as well as with commercial off-the-shelf (COTS) IP switches, providing media companies the flexibility to work with a variety of technology suppliers for a seamless end-to-end production workflow.





 

Publish » Content Protection » Content Protection & Encryption » CAS, encryption, DRM »

The XC Platform is the modular head-end platform for IPTV and broadcast, supporting content acquisition, compression, stream processing including scrambling and multiplexing, with distribution options for cable, satellite and terrestrial networks.

Publish » Content Protection » Content Protection & Encryption » CAS, encryption, DRM »

The X Platform is a high-capacity, versatile, ultra-low latency platform for high-speed video networking, enhanced IP security and advanced compression for the remote production, contribution and distribution markets.

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

Our TV Outside Broadcast service has the largest fibre footprint at venues around the UK, including large capacity sports stadiums, entertainment venues and exhibition centres. With more than 60 years’ experience, combined with our high specification fibre, satellite and microwave connectivity, we can help you broadcast from anywhere, anytime.

Our resilient network can deliver your content in UHD/4K, HDR or HD, compressed or uncompressed and with minimal latency. We can support multiple JPEG2000 or MPEG4 visions at once, or send your feeds over digital microwave, satellite or as a data feed. This means we can get your pictures seamlessly from the venue and send them back in high quality to a designated point. You can feel safe in the knowledge that your content is secure on a network that is safe from piracy with our market leading BISS-CA encryption.

Our record speaks for itself, having successfully transmitted footage for the IAAF Athletic Championships in London, UEFA Champions League Finals, Olympics, Commonwealth Games and Royal Wedding. By continually innovating, we are helping our customers in their shift to remote production by delivering multiple HD visions across our high capacity contribution network, and of course, we helped to make ‘Project Restart’ a reality. For lower tier events, we’ve launched our Flightcase solution. Using our portfolio, we can help you customise the right solution for your needs, bringing your audience the uninterrupted, high-quality live programmes they have come to expect and love.

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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 » 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.

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Our network makes it easy to distribute your content across the Globe. We link major media hubs in Amsterdam to Sydney and on every continent in between. Through our pre-connected gateways, you can start sending video and large media files straight away. Hundreds of potential customers are within reach, with broadcasters, platforms operators, sports venue studios and satellite teleports joining our connected family. We’ve delivered high profile events across the globe, from the World Cup and the Olympics to the UEFA Champions League Final and IAAF World Championships.

We constantly monitor our Global Media Network, providing you with an industry-leading 99.999% availability. You’ll be protected from high latency, jitter and downtime and if there’s any trouble, you’ll instantly switch to a second video path to ensure your content is always delivered. When you become part of our large connected broadcast community, you can enjoy new markets for your video channels as well as sending UHD video.

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

As audience participation media experts, we help the UK media industry create an active TV experience.  We use Europe’s largest in-network call termination system to answer up to a quarter of a million televote calls a minute. Our mass call technology and tailored programmes (capped entries, loyalty points, etc) enable media companies and content owners to generate more revenue through spontaneous campaigns and exciting ‘paid for’ interactions with their audience.

Supporting your campaigns

Whether it’s by mobile voice short code, SMS or MMS, our Agilemedia service provides the platform and APIs you need to support your mobile campaigns.

Organising your voting

Our self-service online tools help you to plan, manage and launch your competition and voting events in a secure and compliant way.

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

FacilityLine Plus is a high-capacity, multi-format video exchange service that lets you share standard definition, high definition and now ultra-high definition video, directly with your collaborators and customers. Our London-based, uncompressed video network connects broadcasters with a global community at BT Tower – it's perfect if you operate within a 35km radius of the tower. We can connect you directly with over 100 media partners via our digital switch infrastructure, and over 600 media production and distribution customers. As a dedicated service that bypasses the open internet, there’s no chance of video stream degradation due to a congested network so you can deliver the highest quality video.

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

TICO-XS is a disruptive, innovative, visually lossless, light-weight and line-based compression technology, specifically designed to be used instead of uncompressed video in many applications. It is currently under standardization as JPEG XS.

TICO-XS is a smart upgrade path to manage higher resolutions (4K, 8K, …) and frame rates while assuring visual quality, keeping power and bandwidth at a reasonable budget and significantly reducing the complexity and cost of a system, a chip or an infrastructure.

 

Applications:

From 3G-SDI to IP in 4K

The mapping of the TICO-XS mezzanine compression to carry UHDTV 4K 60P over a single 3G-SDI cable has been specified in the SMPTE RDD35. intoPIX provides the needed building blocks to create a 4K over 3G-SDI solutions. Have a look at the intoPIX TICO-XS IP-cores and peripheral IP to map the compression over SDI.

 

4K with TICO compression over SMPTE2022-6

As SMPTE ST2022-6, defines the transport of HD-SDI or 3G-SDI over IP, using TICO-XS, it is easy to leverage an existing SMPTE 2022-6 system to carry up to three UHDTV 4K 60P streams over a single 10 GbE port using the SMPTE 2022-6 standard.

 

SMPTE 2110 and TICO compression

SMPTE ST2110-22 specifies the transport of compressed video over IP (including JPEG-XS), enabling you to transport any video formats over IP with full flexibility and all the benefits of TICO-XS compression.

 

8K over a single 10GbE

It has already been proven. TICO-XS can enable you to carry with lossless quality a 8K 60fps video stream over a single 10GbE link using TICO RTP payload and a SMPTE 2110-22 transport.

 

8K over a single 12G-SDI

It has already been proven. TICO-XS can enable you to carry a 8K 60fps video stream over a single 12G-SDI link with lossless quality.

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

JPEG XS compression ​for HD, 4K & 8K ​At the speed of light!

Available for CPU or GPU, implementing the newJPEG-XS standardand many more features, the intoPIX FastTICO-XS SDKs provide the highest quality, lowest latency, visually lossless codec for the optimization of your application. FastTICO-XS SDKs are perfect for all professionals looking to deploy ultra-low latency, lossless encoding over their whole infrastructure and workflows.

Feature overview

Quality & latency

• Best-in-class visually lossless video codec , compliant with JPEG XS


• Adjustable compression ratio (on frame by frame) with constant bitrate throughput (CBR)


• From Lossless down to near-lossless / visually lossless


• Ultra-low latency capability (down to only few lines in CPU) with an optimal API


• Optimized for real-time operations

Processors (CPU / GPU)

• Highly parallelized processing in encode/decode


• x86-64 Intel / AMD CPU   (SSE 4.1, AVX2 or newer) in CPU


• Nvidia GPU (CUDA)

Image formats & frame rates

• Supports monochrome (4:0:0, Alpha) up to 16bit,


• Support 4:2:2  and 4:4:4 8/10/12 bit


• Validated up 8K UHDTV-1 (8192)


• Frame rate depends on CPU/GPU performance & SDK packages

Compression

• Compliant with JPEG XS standard & profiles


• Additional advanced features by intoPIX improving performances & quality.

Rate control

• Adjustable compression rate


• Constant (predictable) bitrate (CBR)

Operating systems

• Windows & Linux (32-bit/64-bit) + macOS (64 bit)

Add-ons

• Embedded Downscaler Add-On: Proxy x1 & x2 decoding (ie. decode directly HD resolution or 4K resolution from  8K XS-compressed video stream)  and reduce your CPU or GPU load. 


• FFmpeg Add-On : connect directly the intoPIX SDK to FFmpeg


• Insertion/Extraction of Color Specification box / Video Support box from JPEG-XS part 3


• SDI Mapping Add-On: enables users to map 4K/8K effciently over existing 3G/HD-SDI cables. This unique feature reformats the XS codestream to fully map the SDI active area avoiding the forbidden EAV/SAV codes, offering a 20% gain of compression/quality performances.

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

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).

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

Artel’s SMART Media Delivery Platform™ is the conduit to simplifying the convergence of IT and broadcast ecosystems. As a base platform supporting native IP media delivery, the SMART platform’s architecture provides feature flexibility and future functional expandability via software download. By using the latest networking and processor technology the SMART Media Delivery Platform has achieved a high level of integration and the agility to navigate media through next-generation networks.

•Integrated Switching Capability - A non-blocking layer 2/3 switching and routing engine has been incorporated into the SMART platform to interface with the network eliminating the need for an external switch or router. The integrated switching capability allows for deep packet inspection, traffic management, flow control, protection switching, and link aggregation that can be easily configured through an SNMP interface.

•Control and Management - Simplifying the migration to an IP infrastructure, the SMART platform’s control and management interface allows the ease of integration, operation, problem detection and isolation.

•Designed for Service Provider and Broadcast Networks - The SMART platform is designed to support transport and data services, and broadcast network video, audio, and IT network applications. With a single module, multiple ports of media may be aggregated and injected into the network without the need for different physical elements. The SMART supports SMPTE ST 2022-7 hitless switching, VLAN tagging, QoS, and traffic management. These functions along with greater port density drastically reduces power consumption, size and cost.

•Software Defined Functionality - The migration to IP has increase the need for more agile solutions. The SMART platform is designed to update functionality and add new applications via software download providing a solution to manage multi-platform content delivery.

The SMART Media Delivery Platform’s small footprint, true data networking and transport versatility, software-defined functionality, and high density opens up new opportunities for media transport and broadcast applications and allows for a seamless transition as IP networks evolve.

A finalist for the 2018 IABM BaM Awards in the Connect category, the SMART Media Delivery Platform™ is a carrier-grade, software-defined platform with integrated nonblocking Layer 2/3 switching and routing capabilities. Designed to attach seamlessly to the IP network without the need for external network elements, the SMART platform supports SMPTE ST 2022: 1, 2, 5, 6, and 7 hitless switching; QoS; VLAN tagging; and traffic management. The platform features four video ports for transport of video, audio, and ancillary data and four GigE data ports bridged to a 10G interface. The SMART solution is software-enabled, providing an easy and efficient platform for adding or upgrading functionality via software download.

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ANEMAN Free Version established the idea of an audio management tool that could be multi-platform and multi-vendor. There is a clear need for an intuitive and informative method of configuring the network, monitoring the performance and to store and recall settings. The Enterprise version is building on these basics but with the idea to control much larger networks that might require a degree of customisation and ongoing support. Enterprise is a web service operating 24/7 on your network, logging events and alerting errors. Extensive metrics allow the user to track and fix problems very easily. It is designed for multi-user operation with the option to add levels of security management. In fact the system administrator can segment the network into zones with individual access rights.
Merging Technologies can offer network and integration design, custom developments for specific widgets and workflows and custom development of ANEMAN to include additional devices, company personalisation and discovery protocols. The company is very open to suggestions and specific user requests.

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

IP Transit allows you to create a global network. Optimize connectivity with access to a blend of the best ISPs in every region and more than 15 Tbps of bandwidth spanning long-haul routes and intra-regional connections. You can improve digital user experience on six continents, including emerging markets like China, India, Brazil, Indonesia, and many more with Zenlayer’s global network.

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Cloud Networking allows you to create a network on demand with instant connections to data centers and clouds

• Performance


• Reduce latency & jitter by up to 40% during peak periods (vs. public Internet)


• Zero packet loss


• High availability

Connectivity

• Quickly build hybrid-or multi-cloud environments to meet business needs


• Directly connect to public clouds and SaaS

Elasticity

• Scale up or down depending on demand


• Choose from 1 Mbps to 10 Gbps

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

GTT’s top ranked, Tier 1 IP network spans over 600 PoPs, which are located in the world’s leading data centers, ensuring seamless internet service on a global basis. Our network maintains extensive peering relationships with other major global carriers and access providers to ensure traffic reaches its intended destination quickly and efficiently.

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

GTT’s Ethernet service enables clients to design a network environment best suited to their needs, with point-to-point and point-to-multipoint topology options, flexible bandwidth ranging from 10Mbps to 10Gbps, run over GTT’s fully resilient and self-healing global MPLS network. Ethernet traffic gets premium treatment across GTT’s backbone network to support our clients’ high performance needs for branch-office connectivity to the Cloud or DC-to-DC connectivity use cases like data replication, disaster recovery and big data.

Monetize » Analyze Data » Data Analytics » Consumer data analysis »

EdgeFunctions simplifies development and reduces time to production for streaming video and content delivery use cases.

Limelight EdgeFunctions is a serverless compute platform that automatically makes your code available to many edge locations around the world, executes at the network edge close to your users to ensure the lowest latency, and scales to meet demand. EdgeFunctions is ideal for customers needing to implement streaming video and content delivery use cases such as personalized streaming, access control, dynamic ad insertion, A/B testing, image manipulation, and more. It is tightly integrated with one of the world’s highest-performing video and Content Delivery Networks (CDNs). EdgeFunctions gives developers tools to boost performance, customize workflows, and scale rapidly while only paying for resources used.

With EdgeFunctions, developers are able to run their code without having to worry about the underlying infrastructure, simplifying development, and reducing time to production. It enables developers to concentrate on what they are good at: coding!

Support » Compute » Cloud Compute »

The edge is the new cloud. Analysts project that more and more computing will move to the network edge, offering the potential to reduce latency, transit costs, and centralized server load. For many media and entertainment use cases, these advantages translate to improved user experience and enable new applications.

Connect » Provide Connectivity » Access Connectivity » Bonded cellular devices »

Our media network combines fibre, IP-based and satellite teleport transfer to ensure your audiences never compromise on quality content regardless of platform, device or geography.