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

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

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

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

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

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

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

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

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

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

Publish » Linear Signal Distribution » Distribution & Re-distribution » Encoding and multiplexing for distribution and re-distribution »

IQOYA *SERV/LINK is a high density 1U rack multi-channel IP audio codec designed for the delivery of audio programs in applications such as multiple STL and SSL links; delivery of multiple WEB radios to CDNs; multiple program delivery to DVB/cable operators; multiple intercom and commentary channels; and IP audio transcoding.

IQOYA *SERV/LINK handles  up to 64 stereo (128 mono) input and output channels in 1U, with the possibility to simultaneous encode and stream the audio inputs 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 AES67/RAVENNA and MADI). IQOYA *SERV/LINK can be fully controlled and monitored from its embedded WEB pages and through SNMP.

Key features

• High audio channels density in a 1U rack: 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.


• Two hot swappable redundant power supply units


• Two Ethernet RJ-45 ports (separate control and IP audio traffic, or backup network, or redundant dual streaming)


• Additional RJ-45 ports for AES67


• Optional RS232 ports and GPIOs for auxiliary data tunnelling


• Supported formats: PCM, G711, G722, MPEG L2& L3, Opus, Fraunhofer MPEG4-AAC (AAC-LC, AAC-LD, HE-AACv1, HE-AACv2, AAC-ELD), and optional Qualcomm aptX


• Up to 16 MPX over AES192


• Standard streaming: ACIP RTP, UDP,  Icecast/Shoutcast, MPEG-TS/IP SPTS or MPTS


• Encoding, decoding, and transcoding


• VLANs, QoS, IGMPv3


• WEB and SNMP control, monitoring and alarming

Manage » Manage & Orchestrate » Content & Workflow Management » Storage management »

Digigram’s IQOYA *VIP is unique in simultaneously providing encoding, decoding, and transcoding capacity (for both Windows and Linux applications) and delivering multiple audio content to transmitter sites, web radio CDNs, multiplexers, and other studios in multiple audio formats via multiple protocols. The software service offers the same robustness as hardware-based solutions but is infinitely more cost-effective, scalable, and flexible.

Using IQOYA *VIP to access and leverage Digigram’s renowned AoIP streaming technology as a software service, ecosystem partners can more quickly create their own live streaming, processing, and encoding systems. By integrating seamlessly with third-party applications and simplifying integration of IP streams, IQOYA *VIP brings added value to automation system providers and radio stations.

Typical uses

• Create multiple local programs by inserting various local contents to a national IP audio feed


• Smoothly insert live contributions into programs using IP


• Stream directly from automation systems


• Offer new transcoding and routing services as a service provider

Publish » Linear Signal Distribution » Distribution & Re-distribution » Encoding and multiplexing for distribution and re-distribution »

A complete line of five IP audio codecs dedicated not only to mission critical distribution applications such as STL and SSL links, but also to DVB audio and WEB radio. X/LINK codecs are fully adapted to current legacy audio infrastructure and are also ready to support your future full-IP migrations.

Kindly visit our website to learn more about all the products in this range.

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

Matrox DSX Core is a complete set of 100% software-only modules for media processing in virtualized broadcast infrastructures based on Matrox DSX SDK. Modules include versatile file reading and writing, a large selection of software codecs and video effects, IP streaming and a precise system clock. OEMs can easily build IP- and file-based video applications such as ingest and playout servers, channels in a box, encoders and transcoders for broadcasters and service providers to use in media datacenters.

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

The vm_avp “virtual module” app is a versatile audio and video processing software application that provides routing, processing and glue functionality for the V__matrix eco-system — a complete IP infrastructure solution that also provides a multitude of audio and video processing functions.

In base form, vm_avp provides encapsulation and de-encapsulation of 3G, HD and SD-SDI (ST2022-6/7 only) to IP ST2022-6 and ST2110-20/21/30/31/40, making vm_avp is the logical choice for both gateway and purely IP-based A/V processing, providing up to 160 SDIIP conversions in 3RU. ST2022-7 seamless protection switching is standard, with IP stream format conversion and frame accurate video switching using destination-timed clean and quiet switching (MBB & BBM) with audio V-fade during switching.

To further expand the usefulness of vm_avp, several optional add-on licenses are available:

• +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.

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

The vm_udx “virtual module” is a software app that provides format conversions between SD, HD, 3G and UHD formats in the Lawo V__matrix ecosystem. In addition to up, down and cross conversion the vm_udx also provides audio embedding/de-embedding, proc amp and RGB/YUV color correction and color space conversion (SDR to/from HDR) functionality. Designed from the start for IP networking, the vm_udx app natively supports ST2022-6 and ST2110-20 IP video plus ST2110-30/-31/AES67 and Ravenna IP audio streams. Conversion between IP video and IP audio standards – ST2022 to ST2110, for example – is also possible. To ensure high availability, ST2022-7 hitless protection switching is natively supported.

The vm_udx app provides a format conversion engine capable of processing four SD, HD, 3G or one UHD path for IP and/or SDI signals. Each path provides audio embedding/de-embedding/shuffling functionality. Audio gain, delay and sample rate conversion can be accessed through independent processing blocks, which can be inserted at any point of the processing chain. Eight instances of broadcast quality RGB and YUV color correction and video proc are also available as processing blocks for use by any video source, whether SDI or IP, and available both pre- and post-format conversion.

For even more functionality, add the +HDR option to enable four instances of SDRHDR color space conversion using 3D lookup tables. A large selection of LUTs especially developed for live production are included, and custom LUTs can also be uploaded too. The included LUTs allow for conversion between SDR and HDR in HLG and PQ.

Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »

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.

Produce » Real-Time Production » Audio Production » Audio mixers and consoles – music »

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.

Produce » Real-Time Production » Audio Production » Audio mixers and consoles – music »

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.

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

HC Bridge SRC provides 256 bi-directional channels of sample rate conversion for AoIP networks. HC Bridge SRC facilitates connecting audio between devices running at different sample rates or in different clock domains on Dante (48 kHz and 96 kHz), AES67 or ST 2110-30 networks. The two network connections can be to physically or virtually separate networks, providing control isolation between two sets of equipment where an AoIP discovery and control ‘firewall’ is required. An HC Bridge SRC includes 2 HC cards, each providing 256 Tx and 256 Rx channels at 48 kHz or 96 kHz, with 128 Tx and 128 Rx streams at both sample rates. Sample rate conversion can be disabled if not required. An alternate HC Bridge configuration without sample rate converters is available with higher channel capacity at 48 kHz. An HC Bridge includes 2 HC cards, each provides 512Tx and 512 Rx channels at 48 kHz, 256 Tx and 256 Rx channels at 96 kHz, with 128 Tx and 128 Rx streams at both sample rates. Each card includes a redundant pair of SFP cages for 1GbE RJ45, multi-mode fibre or single-mode fibre Dante/AoIP connectivity.

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

SSL Network I/O & interfaces are a range of devices which facilitate use of MADI, AES, SDI and IP Audio networking using Audinate's Dante technology for broadcast audio production

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

Built on the proven, award-winning Intraplex® IP Link codec technology, GatesAir's Intraplex Ascent platform supports up to 32 audio channels with high-end features.

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.

Product Details

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.

Main Features of Intraplex® Ascent

• 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
  
  
  
  

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

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

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

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

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

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

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

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

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

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

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

Flexible in application: The codec is designed for studio-to-studio, studio-to-transmitter links and cross-media tasks. Even synchronization with microsecond accuracy is possible by an implemented latency control (useful for simultaneous signal output of all decoders of a network or for audio description). In addition, audio streams can be combined to multichannel streams. An optionally available FM Tuner offers further functions for fallback or backup and rebroadcasting. The Easy2connect feature provides operators as well as reporters with an easy to use connection establishment (SIP phonebook) while being in the field.

High compatibility: The IP-4c supports a wide range of protocols for streaming, control and monitoring (e.g. EBU TECH 3326, AES67, Ravenna, Livewire+, Dante, SMPTE ST 2110, PTPv2, RTSP, SAP, SIP, Discovery, Bonjour, SNMP, HTTP, HTTPS, FTP, FTPS or Ember+ and more). Furthermore, the exchange of additional information like GPIO and ancillary data between the audio networks is possible.

Pay as you grow: All soft- and hardware components are individually combinable. The scalability from one to four audio channels using software licenses gives you flexibility in planning your network and reducing your costs.

Multi-format audio coding: Another advantage is the variety of possible algorithms like MPEG1 Layer 2, MPEG2 Layer 3, most AAC profiles including the new xHE-AAC and AAC-ELDv2, OPUS, Ogg Vorbis, PCM, Enhanced aptX, Dolby Digital plus (on request) and more.

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2wcom’s MoIN multi-format, multi-channel multimedia over IP Network Server (hardware, software or cloud): provides professional IP-based audio network technologies for real-time streaming, transcoding, protocol transforming, routing and mixing.

FLEXIBLE IN APPLICATION: designed for studio to studio and studio to transmitter links, as well as cross-media tasks. Multipurpose usage – e.g. audio routing, managing, levelling, loudness, monitoring and mixing between different protocols and environments. Audio streams are combinable to multichannel streams. PTPv2 provides accurate synchronization. The Easy2connect feature is a SIP phonebook that allows for an uncomplicated connection set-up. The number of channels can scale easiliy depending on your needs with permanent or temporary licenses.

HIGH COMPATIBILITY: MoIN supports all currently established standards and protocols like AES67, RAVENNA, Livewire+, Dante, SMPTE ST 2110, SRT, Ember+, SIP, SAP, PTPv2 and RTSP, as well as transcoding from one format into the other.

MULTI-FORMAT AUDIO CODING: The device provides a wide range of codec algorithms – e.g. MPEG Layer 2, MPEG Layer 3, most AAC profiles including the new xHEAAC and AAC-ELDv2, OPUS, Vorbis, Enhanced aptX, AC3 and more.

TRANSMISSION ROBUSTNESS: Dual Streaming and Pro-MPEG FEC ensure rock-solid IP transmission or go beyond with Stream4Sure. The Reliable User Datagram Protocol (RUDP) ensures highest packet recovery with minimum bandwidth and low latency.

SMART MANAGEMENT: configuration set-up via an easy to use web interface for general settings as well as for backup or fallback options. Remote control via HTTP/HTTPS, FTP, Telnet/SSH, NMS, SNMP, JSON, Ember+. Configuration for transcoding in 15 seconds. PTPv2 synchronization and latency control. Control via centralized Network Management System offering individual settings for e.g. adjustable silence detection, IP buffer and jitter check and PLL control. Sophisticated and elaborated alarm concept: forwarding of alarm messages (e.g. via SNMP, Ember+, HTTP/HTTPS…), optional source switching and event logging for documentation.

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The multi-channel PCM system MAGIC ACX Dante™ WAN Bridge enables the transmission of up to 32 uncompressed audio signals via wide area networks (WAN).

The audio connection for the audio inputs/outputs is provided by the integrated 32-channel Dante™ interface, which has redundant GbE interfaces and of course also supports AES67.

The comfortable operating software allows the management of up to 10 systems including graphic surveillance and can be operated from up to 5 workplaces simultaneously.

The front display of the system also shows essential information on the status of the transmission.

The coding is done via PCM16 / PCM24 / PCM32.

Since the limited maximum latency of Dante in wide area networks, high jitter and possible clock differences between sender and receiver prevent a reliable transmission, the MAGIC ACX Dante™ Wan Bridge solves these problems with a jitter buffer including automatic mode and intelligent sample rate adaption (SRA).

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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.

 

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OPTIMOD 8700i LT is a light version of Orban’s flagship audio processor, the OPTIMOD 8700i. It offers the same versatile five-band and two-band audio processing for both analog and digital media (DAB+/HD Radio/Streaming).

The difference to the OPTIMOD 8700i is that the light version has no Dante interface for AoIP, no streaming monitor output and does not come with the Xponential Loudness algorithm.

However, the OPTIMOD 8700i LT Audio Processor includes features such as the Multipath Mitigator phase corrector which reduces multipath distortion without compromising the stereo separation and the Subharmonic Synthesizer which allows you to add modern-sounding bass punch to older recordings. It also includes Orban’s MX limiter technology which lowers distortion, improves transient punch, and minimizes preemphasis-induced high frequency loss. Further outstanding features are monitored and alarmed dual redundant power supplies as well as safety bypass relays for carefree 24/7 operation. Of course, a digital MPX output is also available as well as a 10 MHz clock input.

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The OPTIMOD 6300 is a high-quality, multipurpose stereo audio processor for digital radio (DAB+, HD Radio), digital television, netcasts, STL protection, satellite uplink protection, and digital mastering.

The 6300 features two processing structures: Five-band for a spectrally consistent sound with good loudness control, and Two-band for a transparent sound that preserves the frequency balance of the original program material while also effectively controlling subjective loudness. There are over 60 Factory Presets wich are our "factory recommended settings" for various program formats or types. There are multiple Factory Presets for both radio-oriented and video oriented programming.

Orban's new PreCode™ technology manipulates several aspects of the audio to minimize artifacts caused by low bitrate codecs, ensuring consistent loudness and texture from one source to the next. There are several factory presets tuned specifically for low bitrate codecs.

The OPTIMOD 6300 includes third-generation CBS Loudness Controllers™ for DTV applications. Loudness controllers work with the both Two-Band and Five-Band structures. Material processed by the CBS Loudness Controller has been shown to be well controlled when measured with a long-term loudness meter using the ITU-R BS.1770-2 standard. The 6300 also includes a "BS.1770 Safety Limiter" that follows the CBS Loudness Controller; use the BS.1770 if the BS.1770-2 meter reading must be constrained to a preset value.

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The OPTIMOD 5700i Audio Processor provides stereo enhancement, equalization, AGC, multiband compression, low-IM peak limiting, stereo encoding, and composite limiting - everything that you need to compete in your market. More than 20 excellent sounding, format-specific factory presets get you started. Although the factory presets are fully competent “out of the box,” you can customize them with easy one-knob LESS-MORE control or with more than 60 advanced controls.

Independent Processing for FM and digital radio

The FM and digital media processing paths split after the 5700i's stereo enhancer and AGC. There are two equalizers, multiband compressors, and peak limiters, allowing the analog FM and digital media processing to be optimized separately. The bottom line? Processing that optimizes the sound of your FM channel while punching remarkably crisp, clean, CD-like audio through to your digital channel audience.

The 5700i includes a full-featured RBS/RBDS generator at no additional charge. The generator supports dynamic PS. It can be controlled via the 5700i presets and an ASCII terminal server that can be connected to automation to support displaying title and artist.

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The OPTIMOD 8700i is Orban’s flagship processor. Featuring versatile five-band and two-band processing for both analog FM transmission and digital media, the 8700i provides the industry’s most consistent sound, track-to-track and source-to-source. This consistency allows you to create a sonic signature for your station with the assurance that your signature will stay locked in, uniquely branding your sound

Xponential Loudness™ brings hyper-compressed music back to life, revealing detail and increasing impact while reducing listening fatigue and distortion.

Built-in streaming server allows you to monitor the 8700i-processed FM and HD/DAB+ audio wherever there is a LAN or Internet connection using free OPUS and MP3 codecs.

Exclusive “Multipath Mitigator” phase corrector reduces multipath distortion without compromising stereo separation.

Dante dual-redundant Audio-Over-IP with AES67 support.

Subharmonic synthesizer adds modern-sounding bass punch to older recordings.

Dual redundant power supplies and safety bypass relays ensure 24/7 operation with no dead-air.

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OPTIMOD-FM 5500i puts competitive five- and two-band OPTIMOD processing into a single rack unit package and brings it to you at the most affordable price ever. Quality sound is what 5500i is all about - sound that attracts audiences by providing a polished, professional presentation regardless of format and source material. Exceptional versatility allows you to adjust the processor’s audio texture to brand your audio, knowing that the resulting signature sound will remain consistent, cut-to-cut and source-to-source. Branding builds businesses and no other processors have the consistency to brand your sound like OPTIMODs.

The 5500i can also be used as a superb stand-alone stereo encoder with latency as low as 2 ms and full overshoot limiting in both the left/right and composite baseband domains. When used in this mode, the 5500i must be driven (usually via an STL) by a full-featured FM audio processor (like Orban’s 8700i) that incorporates pre-emphasis-aware HF limiting and peak control. In both modes, the 5500i’s stereo encoder helps deliver a transmitted signal that’s always immaculately clean and perfectly peak limited, with full spectral protection of subcarriers and RDS/RBDS regardless of the amount of composite limiting.

Support » Monitoring, Control & Communications » Audio Monitoring » Audio measurement and loudness control »

Orban's OPTIMOD TV 8685 surround/stereo television loudness controller builds on Orban's 30+ years experience in television audio processing to provide audibly transparent automatic loudness control and dialog intelligibility control for one surround program (up to 7.1) and four stereo programs simultaneously. The stereo processing can operate in dual-mono mode, so it can process four subchannels in stereo or eight subchannels in mono. The OPTIMOD TV 8685 is Orban's second-generation surround/2.0 processor offers standard AES3id and SD, HD-SDI (3G) input/output and audio routing capabilities that include support for optional Dolby-E encoding and decoding. Dual redundant power supplies help ensure maximum uptime where a relay provides hard-wire safety bypass from the SDI input to the SDI output in case of hardware failure. The OPTIMOD TV 8685 is dialnorm-aware. Loudness control is excellent when measured by the ITU BS.1770 standard (as specified in ATSC A/85:2013) or by the OPTIMOD TV 8685's built-in CBS Loudness Meters. When properly installed and set up, the OPTIMOD TV 8685 will automatically make a station compliant with the CALM Act or EBU R128. In stereo mode, the four stereo processors can be made pre-emphasis aware, allowing the OPTIMOD TV 8685 to be purchased for immediate use with analog transmitters with the assurance that it will provide no-compromise processing for digital transmissions when the need arises. A second use for the stereo processors is processing up to four different languages in DTV program streams.

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

MAGIC DABMUX Go RF is the smallest and only DSP-based Ensemble Multiplexers worldwide, thus providing high availability at an excellent performance. With MAGIC DABMUX Go RF RM there is also 19“ versions that can be mounted in a rack.

Up to 20 program providers can be connected via external Audio Encoders. An installation of the Encoders directly in the studio avoids effectively an interference in Audio quality because of Codec cascading.

Special value was set on the easy configuration of the Ensemble Multiplexer via web browser, so that even users without DAB expert knowledge are able to set up the system.

The highly compact DAB Ensemble Multiplexers facilitate a very simple Multiplex generation in accordance with standard ETSI EN 300 401. Despite its size, all features such as re-configuration (manually and scheduled), extraction of Sub Channels of other Multiplexers, integration of PAD and NPAD data services, creation of Service Information etc. are integrated.
Audio Services can be supplied via the AVTMUX or the EDI(ETI) protocol from external Multiplexers.
As output signal both Multiplexer variants supply an EDI signal for transmission to the transmitters.
With the RF version, which has an integrated modulator, you can alternatively activate a power amplifier directly. This possibility is particularly of interest if you have only one transmitter site.
The synchronisation is effected via NTP or in case of the RF version via the integrated GPS receiver.
The RF input is intended for future applications.

The configuration, operation and monitoring are effected via a HTML5-compatible web browser.
An external alarm can also be triggered via SNMP.
The system has a GBit Ethernet network interface, which allows the configuration of up to three IP addresses as well as VLANs.

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

MAGIC ACip3 is designed as 19” system with integrated wide area power supply and provides optionally an external redundant power supply. Three Ethernet interfaces are available which can be used for Audio over IP transmissions, to control the system with the Windows PC software or to integrate it into a network management system via SNMP. The Audio programmes can – flexibly and freely assignable – be fed in or given out, respectively, via an analogue and two digital stereo interfaces.

The system supports the G.711, G.722, ISO/MPEG Layer 2, Opus coding algorithms and PCM 16/20/24 Bit in the standard delivery version. Optionally, the Audio Codecs can be upgraded with Enhanced apt-X 16/24 Bit, AAC-LD/AAC-ELD and AAC-LC+V1/V2.
One stereo programme is encoded in the standard version, optionally the system can be upgraded to a second stereo programme by software activation.

Two operating modes are available for the pure IP version: the system can be used for dial-up AoIP connections or IP Leased Line connections. In AoIP mode, the system can register at 5 different SIP servers and automatically accept incoming calls from this SIP server. Audio connections in IP Leased Line and in AoIP dial-up Mode can be established with the Secure Streaming functionality for a highly reliable transmission.

Via the Ember+ protocol 64 inputs and 64 outputs can be programmed, an easy communication between MAGIC ACip3 and e.g. DHD or Lawo mixing consoles is possible.

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The high quality and professional DSP-based hardware audio encoder MAGIC AE1 DAB+ Go is realized as a fanless ½ x 19‘‘ device and offers excellent audio quality thanks to the licenced Fraunhofer DAB+ algorithm.

The supplied Windows software allows the configuration of up to 40 systems.

Both analog and digital audio interfaces are available for the input of the audio signal.
In addition to the usual PAD feed via the multiplexer - which supports all standardized services - the PAD services Dynamic Label via FTP or UECP and Slideshow via FTP can also be fed in locally on the device.

In addition, a traffic announcement (TA) can be triggered very easily via UECP or a GPI contact.

Finally, the Audio Encoder allows direct transmission of the program type (PTy) like Rock, Pop etc. via UECP.

The transmission is monitored via the supplied Windows PC software or SNMP. Alternatively, an alarm can also be output via a GPO contact.
Up to three IP addresses can be assigned to the integrated network interface, so that a network separation for different applications is possible. The system also supports VLANs.

Communication between multiplexer and encoder is carried out via the AVTMUX protocol, which enables the control, monitoring and PAD transmission of the encoders from the multiplexer and guarantees secure transmission via Secure Streaming.
With this method, which has also proved its worth in the field of classical audio transmission, all IP packets are transmitted twice - with a delay. Due to the low bit rates with DAB+, the necessary doubling of the data rate is negligible. In addition, different routes can be implemented in the transmission path by using suitable addressing.

At the multiplexer (MAGIC DABMUX Go and MAGIC DABMUX Plus) all IP packets are reassembled correctly in time and duplicate received packets are rejected.
The system can also be used directly with the Open Source Multiplexer ODR DabMux. The necessary interface adaptations are available free of charge on the GitHub platform.

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The MAGIC DABMUX plus complements the AVT DAB multiplexer family and offers the best of the Go and the Basic/Professional multiplexers. The MAGIC DABMUX plus can be used for small-scale solutions as well as for high-end systems: MAGIC DABMUX plus combines significantly more computing power with a simple configuration.

The device is realized as a 19“ x 1U system with integrated power supply and a redundant 12V power supply. The reliable signal processor-based system has three Gbit Ethernet interfaces which allow the configuration of up to three IPaddresses per interface as well as VLANs. The system also provides 2 x USB 2.0 interfaces and an SD card slot for further applications. Via a module slot, the system can optionally be expanded with an ETI E1/2-Mbit or a dual Ethernet module.

The system has a graphical, coloured display, but a more comfortable control, monitoring and very simple configuration is possible via a HTML5-compatible web browser.

Up to 25 program providers can be connected via external Audio Encoders. An installation of the Encoders directly in the studio avoids effectively an interference in Audio quality because of Codec cascading.

A highlight of the MAGIC DABMUX plus is its automated use in redundant headends. In addition, the system offers dynamic reconfigurations (manual or scheduled), an integrated PAD/NPAD inserter, subchannel extraction via EDI with simplified selection through analysis of the incoming signal, announcement support, service linking support, simplified provision of the program logo via SPI (EPG), audio encoder and multiplexer redundancy, etc. In general, all announcements (except OE announcements) are supported according to ETSI TS 101 756.

As a special feature, the Emergency Warning Break-In upgrade of DAB is optionally supported, which in addition to emergency signalling also enables the replacement of all audio content by an emergency announcement. This ensures that the announcement is audible even with older receivers.

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This system is the leading digital video and audio delay solution for easily eliminating all unwanted content from your live broadcasts or webcasts.

Key Features Include:

• 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.

Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »

Lumo aims at bringing radio closer to its audience.

It is an all-in-one virtual radio studio including Playout (playlist & jingles players), a Mixing Console with DSP & automixer, a VoIP SIP phone, and an AoIP transmission codec.

Lumo runs on a simple laptop. It is web-native and touch-friendly, you can control your studio with a fingertip from any device (including iPad and Android tablets).

Lumo makes remote operations much easier by reducing to a minimum the amount of gear to deploy and offering intuitive yet powerful user interfaces for technical and non-technical operators.

Licenses can be purchased for temporary use as for yearly contracts, making sure you don't pay for resources that you don't use.

Support » Monitoring, Control & Communications » Audio Monitoring » Audio measurement and loudness control »

Artisto is a software audio engine, providing a complete solution for automating any workflow involving sound. From simple monitoring and detection of levels to advanced workflows for cross-platform media diffusion or virtualised remote studios, Artisto covers every processing and synchronisation need.

Because Artisto is modular, it can be precisely tailored to the specific requirements of any audio application. Artisto can be flexibly configured with an extensive library of processing blocks such as routing, EQs, dynamics, web streaming, AoIP transmission, VoIP phone, recorder, player, loudness levelling and so on. These can be virtually wired together to build a processing pipeline for the desired workflow.

Running on off-the-shelf IT equipment or in the cloud, it eliminates the frustrations inherent in complex hardware infrastructures, solves interoperability issues and dispenses with the need for outdated, insecure control protocols. Artisto responds to any transport requirements from physical or virtual soundcards (AES67, Dante, MADI,…) to low-latency audio-video streams for the cloud (including SIP).

Artisto is fully configurable and controllable via a simple, open web API. Operations can be manual or automated, centralised or distributed, local or remote.

Artisto’s front-end is based on on the most common web technologies, and On-Hertz provides a library with commonly-used components, guaranteeing that any web developer can easily build custom interfaces that fit end-users’ needs.

By design and philosophy, Artisto is scalable and open. It doesn’t lock the customer into one solution. It allows them to choose what part of Artisto they prefer to use or to connect to third-party services.

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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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Ines helps you to manage your audience On Air. Ines is :

A windows app to manage calls on air

A web app (ines light)  to manage your audience cards in the database.

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

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).

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

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).

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

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).

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

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-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-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-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.
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-GMCS to be used for both studio and mobile installations. Moreover this allows for a secondary power source to reduce 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.

The unit can be put into a low-power sleep mode when not in use, with an instant start when power is re-applied. In power off situations, a super capacitor is used to keep the GPS receiver powered in a low power mode for more than 20 hours, enabling the receiver to regain lock immediately rather than having to ‘cold’ start.

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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.

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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.

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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-

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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.

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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.

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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.

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Smart Codecs is the universal remote control of your set of audio codecs. This very robust application has been developed to standardize the use of several models of codecs RTC, ISDN or IP. Smart Codecs can interconnect with your audio / video grid manager.

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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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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 sy