RUBIDIUM SERIES PTP module C3

The scalability of Ascent makes it ideal for applications that require multiple channels of audio encoding and decoding at head-end sites. The high-density solution reduces cost and provides a path for convergence of IT and broadcast infrastructure. The interoperability between Ascent and IP Link codecs provides a complementary solution for remote contribution and distribution use cases.

The Ascent platform continues the Intraplex tradition of providing an unprecedented level of reliability for audio over IP application. Built with enhanced network reliability and security in mind, Ascent supports the Dynamic Stream Splicing technology of IP Link codec, which provides “hitless” protection for packet losses using the combination of time and network diversity for packet transmission, and FEC.

To further enhance reliability and security, the platform supports the SRT protocol encapsulation. SRT (Secure Reliable Transport) is an open-source protocol that provides low-latency, reliable and secure streaming of audio and video data. The reliability in SRT is accomplished by a built-in packet re-transmission scheme; the security of streams is handled by the built-in AES-128/256 payload encryption.

In addition to “hitless” protection, like the IP Link codecs, the Ascent also supports automatic failover between different incoming streams or stored audio.

This combination of capabilities gives Intraplex codecs a market-leading position in STL over IP technology and unmatched network transport reliability.

Ascent supports both physical audio interfaces (AES3, Analog) and AES67 (AoIP), with a maximum of 16 full-duplex or a total of 32 channels (combination of encode and decode). Physical audio interfaces can be ordered as 4- or 8-channel cards, supporting both AES3 and analog audio signals.    The Ascent product is built on the Commercial-Off-The-Shelf (COTS) x86 architecture to leverage the scalability and cost of the technology. The product is available in a 1RU branded hardware server and as a software-only option.

• 4, 8, 16 stereo channels. Each channel can be full-duplex, encode-only or decode-only


• Supports AES3, Analog and AES67 audio input and outputs


• Standard Coding: Linear, Opus


• Optional: AAC-HE, AAC-HEv2, AAC- ELD, AAC-xHE, MPEG 2 and MPEG 3 audio coding


• Protocol Encapsulation: Real-time Transport Protocol (RTP), Secure Reliable Transport (SRT)


• Streams: Multicast, unicast, and multi-unicast


• 10 streams per channel with up to 20 destinations per transmit stream; maximum of 100 streams per system


• Three independent IP interfaces for redundant network operation


• Built-in silence sensor with optional stream switchover


• Automatic backup to audio playout from USB drive or external audio source


• Multicoding: can encode the same audio source in multiple formats


• 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 scheduler for automated scheduled program switching


• Integrated with Intraplex LiveLook (network analytics and monitoring software)


• Web and SNMP for management


• Multiple web account types to restrict access


• GPIOs: up to 32 in, 8 out. Available for stream transport and alarm assignment


• Network reliability
  
  
  
  • Dynamic Stream Splicing with network and time diversity for “hitless” packet loss protection
  
  
  • Programmable RTP level Forward Error Correction (FEC)
  
  
  • Secure Reliable Transport (SRT): new protocol provides automatic packet retransmission method
  
  
  
  


• IP Security
  
  
  
  • Access control per interface
  
  
  • Encryption of streams with AES-128
  
  
  
  

Lawo’s Power Coreᴿᴾ is a full-featured remote production solution for mc² audio consoles, providing integrated modular I/O, DSP and IP streaming capabilities, all in a compact 1RU device. It accomodates 64 channels of MADI audio via its front-panel port, and I/O count can easily be expanded using the eight rear-panel slots which can be loaded with Mic, Line, AES3 and GPIO cards, in any combination, for a total I/O capacity of up to 128 channels.

Designed for mission-critical applications, Power Coreᴿᴾ includes ST2022-7 compliance for network redundancy and Class C jitter / network latency robustness to eliminate the need for dedicated WAN gateways and minimizing the complexity of remote productions while reducing single points of failure. Its IP interface complies with ST2110-30/-31, AES67 and RAVENNA networking standards to deliver maximum interoperability within your production suite. To accommodate the most sophisticated workflows, Power Coreᴿᴾ features comprehensive audio connectivity via two redundant 1GbE SFP ports for Audio-over-IP, one MADI port (with a 2nd port for interface redundancy), and eight modular I/O slots that can accommodate mix of Mic, Line and AES3 cards. A unique studio card with Mic/Line inputs & outputs plus two headphone amplifiers is available as well.

Power Coreᴿᴾ DSP capabilities include 64 fully-featured processing channels and provide low-latency on-site monitor and IFB mixing. mc2 consoles at home have full control of all relevant channel parameters (gain, fader, mute, EQ, dynamics, Aux Send Level, etc.) of the DSP node at its remote location.

All I/O parameters of the Power Coreᴿᴾ can be remote controlled from an mc² console’s channel strips; in addition, mc² consoles can control the DSP channels inside Power CoreRP to offer control capabilities for up to 64 input channels and 16 stereo Aux busses. Remote channels can be mapped to the host console’s surface just like any other source and offer parameter control for Fader, Mute, EQ+Filters, Dynamics, Delay plus the analog and digital input section. Remote inputs and Auxes of Power Coreᴿᴾ can even be linked to local DSP channels of the host console to ensure continuous linking of parameter values.

In addition to the control integration from an mc² console, Power Coreᴿᴾ features a touch-screen optimized control GUI based on the Lawo VisTool screen design solution. This provides on-site as well as remote access to all parameters of Power Coreᴿᴾ, which is convenient for setting up prior to connecting the host console. Power Coreᴿᴾ ‘s unique remote-control possibilities include continuous and dynamic fader control as well as fader start-type commands, which are essential for latency-critical WAN connections and to avoid speech truncation on air. The GUI’s feature set also includes sophisticated test mode and lineup patterns, as well as controllable DIM levels to the talents’ ear-pieces for talkback from the director. The graphical interface also allows monitoring the control connection to the host console, the on-air status, and Sync. You can also monitor Listen and PFL, as well as the local talkback system.

Lawo A__line nodes are designed to serve as IP audio stageboxes for mc² consoles, audio extensions for the V__matrix ecosystem, or as stand-alone IP audio gateways.

A__line devices provide distributed audio connectivity that makes it easy to scale audio I/O capacity in a networked system — temporarily or permanently. SMPTE ST2110-compliant, A__line offers a granular audio format selection similar to the flexibility found in baseband modular I/O systems, but eliminates system design limitations like maximum channel count per frame, fixed channel capacity per device interconnect, or fixed overall system size. Audio endpoints connected to A__line nodes can be seamlessly shared on LAN or WAN, and thanks to their fully standardized streaming technology, can interconnect to a wide variety of IP-enabled broadcast devices — A__line nodes are brand-agnostic.

A__line devices boast exceptional audio quality. Discrete, Class-A microphone preamplifiers deliver a superb dynamic range of 119dB(A), ultra low noise at all gain levels and a perfectly flat frequency response. Versatile analog I/O accommodates levels as high as +24dBu before clipping. Insertable, precise sample rate conversion is available on each AES3 input. For multichannel baseband interfacing, A__line features bi-directional MADI access via SFP. All devices employ the ST2110-30/31 and AES67 standards to transport uncompressed audio in real-time on Layer-3 IP networks, with ST2022-7 Seamless Protection Switching with dual-redundant network interfaces and ample receive buffer capacity to meet ST2022-7 class C specs for LAN and WAN deployment. Two redundant power inlets complete the package, along with PPM metering for all Analog and AES3 interfaces and PTP/Wordclock sync and conversion.

The A__line family includes:

• A__stage48 – 32x mic/line in, 16x line out, 8x AES3 in, 8x AES3 out, 1x MADI, 8/8 GPI, 3RU


• A__stage64 – 32x mic/line in, 16x line out, 8x AES3 in, 8x AES3 out, 1x MADI, 8/8 GPI, 4RU


• A__stage80 – 32x mic/line in, 32x line out, 8x AES3 in, 8x AES3 out, 1x MADI, 8/8 GPI, 3RU


• A__mic8 – 8x mic/line in, 4x line out, 8/8 GPI, 1RU


• A__digital8 –8x AES3 in, 4x AES3 out, 8/8 GPI, 1RU


• A__digital64 – 32x AES3 in with SRC, 32x AES3 out, 1x MADI, 8/8 GPI, 3RU


• A__madi6 – 6x MADI, 1RU

When controlled with Lawo’s VSM broadcast control system, A__line stageboxes provide an advanced set of networking options; VSM manages A__line’s AoIP connections, I/O settings, its non-blocking internal routing matrix, and GPIO signals for smooth integration into an overarching operational workflow.

RƎLAY virtual radio software from Lawo brings professional radio production to your PC. Mixing, streaming, dynamics processing, routing and even AoIP signal monitoring, using standard RAVENNA / AES67 networking via Lawo’s virtual sound-card software for Windows. Mix and route any audio produced by PC applications, streams from your AES67 network, or any audio signal your PC can ingest. All this power at your fingertips, using touch-screen GUIs so intuitive that talent can learn them in minutes. No exotic hardware needed: just install RƎLAY software on your PC or laptop, and go to work. Perfect for production, live studios, newsrooms — even remotes and OB work.

There are four RƎLAY Virtual Radio apps:

• VRX Virtual Radio Mixer, a powerful mixing console that runs in a virtualized environment. No racks of gear to devour space and money – RƎLAY VRX runs on standard PCs and laptops, with an intuitive, multitouch-enabled interface that’s easy to learn and use. Available in 8-fader and 4-fader versions.


• VPB Virtual Patch Bay, a cross-point patch bay on your PC. RƎLAY VPB gives you unrestricted control over the routing and mixing of audio signals on your computer. Patch inputs to outputs, combine multiple channels, even apply processing using the included Lawo Processing Suite, or your favorite VST plug-ins.


• RƎLAY Virtual Sound Card, an 8×8 AoIP driver (with up to 64 bi-directional channels of stereo I/O) for Windows. RƎLAY VSC turns PC audio into clean, pristine AES67 streams to interface with your studio’s RAVENNA / AES67 network — without the expense of traditional sound cards.


• AoIP Stream Monitor keeps you informed of the state of your critical program streams with real-time confidence metering, LUFS meters, silence sense, audio level alerts, etc., displaying realtime informatics for up to 16 AES67-compatible audio streams. Average and peak bargraph meters for each channel are onscreen constantly, accompanied by a level readout in Loudness Units, and visual alarms for under-level, over-level and error states, plus multiple info tabs with stream health, performance history, and detailed SDP information. Perfect for Master Control displays, stream diagnostics, audio level checks, and much more.

• +audio – adds embedding/de-embedding and shuffling of audio from both IP and baseband I/O with sample rate conversion. 40 TX and RX instances of RAVENNA/AES67/ST2110-30/-31 streaming and an audio crossbar of 512 x 512.


• +audio_matrix – an additional upgrade to the +audio option, the +audio_matrix option provides a total of 88RX and 128TX instances of RAVENNA/AES67/ST2110-30/-31 streaming and an increased audio crossbar of up to 5,312 x 5,312.


• +madi – enables use of V__matrix BNC inputs and outputs to interface with MADI signals (AES10, 48kHz, 64 channels).


• +fs – adds framesync, frame phaser, sample rate conversion and audio/video delay functionality for both IP and baseband video inputs; also provides Dolby-E® auto-alignment functionality.


• +vc2 – adds visually lossless VC2/DiracPro ultra-low latency encoding and decoding; ST2042 low-delay profile.


• +proc_cc – adds YUV & RGB color correction, test pattern generator/inserter, and test tone generator.


• +12g – Adds support for 12 Gbps video standards and provides cross conversion capabilities between UHDTV1 Single-Link and Quad-Link.

Streamlined audio codec

IQOYA X/LINK-LE is a 1U rack streamlined IP audio codec designed for the delivery of a stereo source (or two mono sources) over IP networks for STL and  SSL, but also DVB audio, or WEB radio. IQOYA X/LINK-LE benefits from all the major features of X/LINK but at an attractive price. It can be used in legacy analog or AES/EBU audio environments, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a good investment for the migration to IP audio. Like all the IQOYA products, X/LINK-LE is based on Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable connections over inexpensive IP links. Based on a low consumption and fanless powerful hardware platform, IQOYA X/LINK-LE is designed for 24/7/365 use.

key points

Cost effective solution with essential features, and no compromise on reliability

Designed for audio service continuity and failsafe operation

Ongoing product support with flexible options

Invest now in a codec adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure

The reference audio codec

IQOYA X/LINK is a 1U rack IP audio codec designed for the delivery of a stereo source (or two mono sources) over IP networks for STL and SSL links, but also DVB audio and WEB radio. It can be used in legacy analog or AES/EBU audio infrastructures, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a good investment for the migration to IP audio.  Like all the IQOYA products, X/LINK is based on the Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable IP streaming over inexpensive IP links.  Based on a low consumption and fanless powerful hardware platform, IQOYA X/LINK is designed for 24/7/365 use.

Key points

Adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure

Simultaneous delivery of your audio program to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios, in multiple audio formats

Designed for audio service continuity and failsafe operation

Easy integration into existing SNMP based supervisors (SET, GET, Traps)

Ongoing product support with flexible options

Designed for FULL IP environments

IQOYA X/LINK-AES67 is a 1U rack IP audio codec designed for the delivery of  stereo and/or mono audio sources over IP networks, for STL, SSL, DVB audio, and WEB radio applications. It is dedicated for full-IP audio infrastructures based on  AES67,  Ravenna, or Livewire technologies.  Like all the IQOYA products, X/LINK-AES67 is based on the Fluid IP technology which offers redundant dual streaming, allowing  reliable IP streaming over inexpensive IP links. Based on a low consumption, fanless and  powerful hardware platform, IQOYA X/LINK-AES67 is designed for 24/7/365 use.

Key points

Designed for full IP environments

Supports AES67, RAVENNA, and Livewire technologies

Scalable number of supported IP audio input and ouput channels (from 1 to 8 stereo I/Os)

Simultaneous delivery of input audio programs to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios, in multiple audio formats

Ongoing product support with flexible options

Outstanding possibilities in only 1U rack

IQOYA *SERV/LINK is a extra high density 1U rack multi-channel IP audio codec designed for the delivery of large number of audio programs to transmitter sites (STL), studios (SSL), DVB multiplexers, and WEB radios CDNs. It also serves for the transport of multiple intercom and commentary channels over IP networks. It supports from 4 to 64 stereo input and output channels, with the possibility to simultaneously encode and stream the input channel at multiple formats and protocols, decode IP audio streams to the outputs, and transcode IP audio streams. It features two hot swappable redundant PSU, two Ethernet ports,  and supports different types of audio I/Os (AES/EBUs, or analog, or MADI, or AES67/RAVENNA, or DANTE , or AES67/RAVENNA and MADI). IQOYA *SERV/LINK can be fully controlled and monitored from its embedded WEB pages and through SNMP.

Key points

Maximizes rack space: only 1U to replace up to 64 DANTE stereo channels, up to 16 stereo AES/EBU channels, or 8 analog stereo channels, or 64 MADI stereo channels*, or 64 AES67 stereo channels*, or 64 AES67 and MADI stereo channels*

One single unit simultaneously delivers multiple audio content to transmitter sites, WEB radio CDNs, multiplexers, and other studios, in multiple audio formats

WEB and SNMP control, monitoring and alarming

2 stereo codecs in 1U

IQOYA X/LINK-DUAL is a 1U rack IP audio codec designed for the delivery of two stereo sources (or four mono sources) over IP networks for STL and SSL, DVB audio, WEB radio, intercom and commentary. It can be used in legacy analog and AES/EBU audio infrastructures, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a perfect investment for the migration to in-house IP audio. Like all the IQOYA products, X/LINK-DUAL is based on Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable connections over inexpensive IP links. Based on a low consumption, fanless and powerful hardware platform, IQOYA X/LINK-DUAL is designed for 24/7/365 use.

Key points

Space efficient: two stereo codecs in a 1U rack, with simultaneous delivery to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios

Adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure

Easy integration into existing SNMP based supervisors (SET, GET, Traps)

Ongoing product support with flexible options

The greenMachine HDR Evie+ (Enhanced Video Image Engine), 1 RU half 19” rackmount, is a real-time segmented frame-by-frame broadcast-quality High Dynamic Range (HDR) to Standard Dynamic Range (SDR) converter, with frame sync supporting formats up to 4K UHD (3840x2160). It is the world’s first system that uses the advanced algorithm for sectional dynamic tone mapping, which automatically analyzes different sections of an image in HDR stream and applies optimal corrections on a frame by frame basis in real-time. This unique capability is unlike any other solution today. It is the perfect real-time production tool for sports or any live broadcast event needing high-quality real-time HDR to SDR conversions. HDR EVIE+ fits best in the single native HDR workflow reducing the cost of equipment and manual operations.

The sectional dynamic tone mapping technique is applied to several spatial areas (sectors) in the image. In this type of tone mapping technique, the algorithm adjusts the signal by using several dynamic curves, one per image segment/sector. The algorithm then combines the result with global dynamic curves based on the entire image. While the dynamic curves of the different sectors are perfectly adapted to the respective image content of their segments, the global curve is perfectly adapted to the content of the entire image. Depending on the image content within these sectors, the algorithm reacts dynamically, which allows lights and shadows to be treated independently. Therefore, dark areas can be brightened, and the bright areas can be darkened without getting a flat gradation of the image. “

HDR EVIE+ provides a 1x 4K/UHD processing channel supporting down-conversion from HDR transfer characteristics to SDR through appropriate sectional dynamic tone mapping. It also supports the Wide Color Gamut (WCG) needs of broadcasters and professional AV live events requirements. HDR Evie+ package also includes HDR Static configuration for Static HDR SDR conversions, which performs static tone mapping to realize UP/Down/Cross conversions between HDR and SDR, suited best for the studios or the environments where the light conditions do not change dynamically.

• Eliminate nudity, offensive language, and obscene gestures from live broadcasts while maintaining program continuity.


• Straight-forward to control with two large panic buttons; one for audio (red); and one for video (yellow). Simply press & hold to clean-up your broadcast.


• Allows for dual-user operations, with unique settings to accommodate each operator’s preferences.


• The viewer experience is minimally interrupted with subtle video and audio masking techniques.


• Compensates for the reaction time of human operators to an observed visual or aural event, ensuring every obscenity is concealed in its entirety.


• Solid-state technology equipped with bypass relay circuity to further ensure the integrity of your live television broadcasts.

• 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.

• 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.

• 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.

• 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.

• 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.

• 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.

• 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.

• 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.

• 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.

• 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.

• 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.

C3 Platforms:- C3 module has five different S/W platforms that can be programmed with

• GM = Grandmaster Slave Mode


• SL = Slave Mode


• BC = Boundary Clock


• GF = Grandmaster with fallback


• BG = Boundary Clock with GPS option

C3 Features:-

• 10/100/1000Base-T PTP Ethernet port


• Compatible with SMPTE 2059-2


• Two 10 Mhz outputs come standard, third output is an option


• PPS, time and date data string reference outputs


• Additional unbalanced and balanced PPS outputs


• Leap year/second compatible


• SNMP ready


• Hot swappable


• Failure Relay


• Compatible with TC_link


• Integrated Surge Voltage Protector


• UMID data capable

The PLURA Rubidium Series PTP client module C3 is the most accurate, stable and accurate way to universally create and manage PTP signals flow in the broadcast industry. The module has multiple platforms such as PTP Grandmaster, PTP Slave or PTP boundary clock. The precise acquired reference time and date is transferred in form of a serial protocol and a synchronized seconds pulse. The module comes with two 10 MHz outputs, which are often used for synchronizing a third party SPG system. The PTP slave uses industry leading algorithms to extract time from a PTP input stream and produces stable frequency and time outputs. The module needs to be used in conjunction with a RUBIDIUM Series housing and a power supply. For user specific setups, an initial configuration requires a Windows PC with an USB and/or Ethernet port (only in combination with an IE module). The PTPC module offers a variety of monitoring and control possibilities. Its design allows for the implementation into complex and fail-proof redundant systems. C3 Outputs:- C3 module has the following outputs:

• Time information Serial Interface Output


• Seconds pulse (PPS)


• Four programmable GPIs


• Two 10 MHz-signals (a third output is configurable)


• An Ethernet pass through port

Every module is connected to an internal hot swappable bus, which bilaterally connects all modules within a particular housing.The internal bus can be distributed over several housings by using the RLC port. The RLC plug contains a voltage feed, a failure relay output and aTC_link interface.TC_link is a real time capable proprietary interface, which is based on a customized RS485 interface. On the rear side of every housing, a PC interface (USB) can be found.This serial connection is used for configuration, status control, and also software and firmware updates. Via our IE Ethernet module, browser configuration, status control and SNMP functions are available.

• 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.

• 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.

• 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.