Matrox DSX Core

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.

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

Matrox DSX LE4 BNC PCIe card offers cost effective 4-channel 3G/HD/SD SDI I/O hardware for ingest, playout, and live production solutions. All SDI inputs and outputs are fully reconfigurable and provide support for embedded audio and ancillary data. The card features full size BNC connectors to simplify wiring.

Key features

 

• Half-length PCI express card


• 4 channel re-configurable SDI IO


• Frame synchronizers


• VANC and HANC support for each input and output


• Analog blackburst reference input (tri-level or bi-level)


• Live zero-frame delay video mixers


• One comprehensive SDK for all our developer cards

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

DSX LE4 FH provides multichannel SDI IO in a half-length PCI express card. This card provides up to 12 reconfigurable IOs, from SD to 4K, along with support for AES/EBU, LTC and GPIO to provide for versatile connectivity.

Matrox DSX LE4 FH is ideal for OEMs who need to create channel-in-a-box systems, video servers, broadcast graphics systems, encoders, transcoders, multiviewers, switchers and other digital media equipment.

Key features Half-length PCI express card Re-configurable IO that can support up to 12 SDI IO Frame synchronizers VANC and HANC support for each input and output Analog blackburst reference input (tri-level or bi-level) Automatic video relay bypass Live zero-frame delay video mixers Up to16 channels of AES/EBU inputs and outputs Up to 8 LTC inputs and outputs RS422 control Single slot all inclusive option

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

DSX LE4 IP offers multi-channel, IP-encapsulated HD and 3G video I/O for SMPTE ST 2022-6, SMPTE ST 2110, and SMPTE ST 2022-7 transport protocols with hardware based packetization, packed into a single half-height, half-length PCIe card.

Key features
Multi-channel IO allowing up to 4 video inputs and 4 video outputs with redundancy in HD
Guaranteed non-bursty packet transmission with zero CPU usage
Support for multiple IP encapsulations
IP and analog blackburst timing reference
IP bypass and fail-safe mechanisms
Feature rich SDK

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

Matrox DSX LE4 LP is a low-profile, half-length PCIe card that offers up to eight reconfigurable SDI I/Os—from SD to 4K. When configured with inputs, the cards are equipped with built-in frame synchronizers to generate clean SDI signals in any environment.

Matrox DSX LE4 LP is ideal for OEMs who need to create compact channel-in-a-box systems, video servers, broadcast graphics systems, encoders, transcoders, multiviewers, switchers and other digital media equipment.

Key features
Low profile, half-length PCIe card
Re-configurable I/O that can support up to 8 SDI inputs or outputs
Frame synchronizers
VANC and HANC support for each input and output
Analog blackburst reference input (tri-level or bi-level)
Automatic video relay bypass (optional)
Live zero-frame delay video mixers

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

The Matrox DSX LE5 12G provides multi-channel reconfigurable 12G/3G SDI I/O for HD and 4K workflows. The DSX LE5 12G allows 4x 12G inputs, 4x 12G outputs and 4x 3G reconfigurable I/O to operate simultaneously on a single half length PCI express card. The DSX LE5 12G is supported by the feature-rich, field proven Matrox DSX SDK.

Key features

• Versatile 12G and quad 3G SDI for 4K connectivity


• Up to 4x 12G inputs and 4x 12G outputs with 4x 3G reconfigurable I/O


• Up to 12x completely reconfigurable 3G I/O


• Frame synchronizers


• Decimation for input preview

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

The Matrox® DSX LE5 Q25 and Matrox DSX LE5 D25 SMPTE ST 2110 network interface controller (NIC) cards provide 25 GbE and 10 GbE interfaces for multi-channel IP video I/O from HD to 4K, with zero CPU usage.

The Matrox DSX LE5 ST 2110 network adapters offer complete ST 2110 offload allowing OEM software solutions to provide reliable and scalable COTS-based 10 GbE and 25 GbE IP solutions. These ST 2110 NIC cards offload all packet processing required for ST 2110-20, ST 2110-30, ST 2110-31, and ST 2110-40, as well as ST 2022-7 redundancy, to free up system resources. With ST 2110-21 packet pacing built into the NIC, software solutions can provide guaranteed narrow senders (Type N) under all conditions independent of CPU load and OS interactions. The DSX LE5 ST 2110 cards also provide line rate capture allowing for wide asynchronous receivers (Type A). Both the permissive wide asynchronous receivers and the, ST 2110-21 packet paced, narrow senders allow software solutions to guarantee universal compatibility and interoperability at scale.

Matrox DSX LE5 ST 2110 NIC cards also offer integrated, hardware-based PTP support allowing for ST 2059-2 accuracy in the nanosecond range eliminating network jitter associated with general-purpose, non-video-aware NIC cards . With built-in support for JT-NM TR-1001-1 -- including NMOS Device Registration and Discovery (IS-04) and NMOS Device Connection Management (IS-05), DSX LE5 cards provide for seamless integration in ST 2110 environments.

With access to standard network functionality, DSX LE5 ST 2110 NIC cards provide a complete comprehensive COTS solution for software applications to seamlessly transition from SDI to ST 2110 with no hassle and guaranteed success.

Key features

• Native SMPTE ST 2110 over 25 GbE support with no CPU usage


• Up to 4 UHDp50/60 inputs and outputs


• Up to 32 HD inputs and outputs


• ST 2110-21 packet pacing in hardware for narrow senders (Type N)


• Wide asynchronous receivers (Type A)


• 32x HD inputs and outputs over 25 GbE


• 256 audio flows from 1 to 64 tracks


• Integrated hardware PTP for ST 2059-2


• Built in NMOS IS-04 and IS-05 support


• 3 year warranty


• One comprehensive SDK for all our developer cards

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

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

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Matrox M264 family of PCI Express cards features hardware-based multi-channel 8- and 10-bit H.264 encoding, decoding and transcoding capabilities. They provide an instantaneous H.264 quality and density boost offering the pristine quality needed for broadcast contribution, production and distribution. With the capability for up to four channels of 4K XAVC encoding/decoding in a single slot card, the M264 family enables OEMs to provide multi-channel 4K production servers in a PC platform. Supporting the H.264-based Sony XAVC and Panasonic AVC-Ultra mezzanine codecs, it makes live 4K production as easy as today’s XDCAM HD workflows in PC platforms.

The M264 family can encode/decode up to four streams of 4Kp60 at 4:2:2 10-bit, 40 streams of HD long GOP at 4:2:2 10-bit or 64 streams of HD at 4:2:0 8-bit. The faster than real-time encoding/decoding ability of the M264 allows multiple blazing fast transcodes for distribution on differing media platforms. With onboard multi-channel, motion-adaptive deinterlacing, and up/down/cross scaling, the M264 family can repurpose content into any resolution before encoding or after decoding, particularly beneficial for OTT workflows.

Specifically designed for high density encoding/decoding (Baseline Profile to High 10 Intra Profile up to Level 5.2), M264 cards provide the highest possible quality at the lowest power per stream (< 1.5 W per HD stream). The M264 S4, S3 and S2 cards are cloud friendly and their resources can be shared across virtual machines.

Matrox M264 cards are the ideal H.264 accelerators for channel-in-a-box systems, video servers, broadcast graphics systems, multi-viewers, and switchers or to create high-density encoders, transcoders and other broadcast media equipment. The Matrox M264 hardware is supported by the Matrox DSX Software Development Kit. Please refer to the Matrox DSX Developer Products datasheet for information on file I/O, software codecs and CPU effects.

Key features PCI Express Cards M264 - Gen 2, ½ length card M264 S2 - Gen 3, ¾ length card M264 S3 - Gen 3, ¾ length card, M264 S4 - Gen 3, full length card. Multi-channel, high-quality H.264 video encoding, decoding and transcoding Up to 4K resolutions supported 4:2:2 10-bit and 4:2:0 8-bit Hardware based deinterlacing and scaling Real-time and faster than real-time operation Dynamic codec control Sony XAVC compliant encoding including 4K XAVC Long and 4K Intra Class 480 Panasonic AVC-Ultra compliant Native MXF file wrapping Native MXF file reading. This includes Trim in and Trim out settings for building playlists File consolidation Time Delay Instant Replay (TDIR) support MXF file truncation and appending during live recording

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » Live and high-density transcoders »

The Matrox® Maevex 6150 quad 4K encoder appliance is the first in the world to deliver quad 4K input capture and encode, supporting at least four concurrent 4K streams and recordings—simultaneously. Delivering the full feature set of the Maevex 6100 quad encoder card, the Maevex 6150 comes in a convenient, rack-ready appliance form factor with added I/O.

Coupled with Maevex PowerStream Plus software, the Maevex 6150 quad 4K enterprise encoder delivers reliable industry-leading technology for flexible management of all Maevex devices and features, ensuring complete reach and control over the entire network.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » Remote production tools »

Featuring an exceptionally low latency of 100 ms glass-to-glass, as well as flexible, future-proof connectivity with 3G, 12G SDI, and ST 2110 over 25 GbE network connections, the Matrox Monarch EDGE encoder and decoder pair delivers best-in-class video transport for today’s most demanding remote production (REMI) workflows. Broadcasters can leverage the 4:2:2 10-bit capable Monarch EDGE encoder and decoder pair to affordably produce live multi-camera events by minimizing on-site expenses and keeping talent in-house, and confidently deliver superlative-quality live programming. The Monarch EDGE encoder and decoder pair delivers high-frame-rate (50 or 60 fps), multi-camera video streams at resolutions up to 4K or quad HD, using a variety of streaming protocols, including SRT, MPEG-2 TS, and RTSP—all packaged in compact, portable, and low power appliances. The 8-bit version of the Monarch EDGE encoder meanwhile, is purpose-built to power next-generation live, synchronized multi-angle productions for cloud-based REMI workflows.

Support » Monitoring, Control & Communications » Test & Measurement » Test and measurement equipment and software »

Matrox VERO is an essential tool for broadcasters, system integrators, and manufacturers of ST 2110 devices and environments. Matrox VERO integrates the components required to allow users to confidently validate ST 2110 devices for compliance before deployment. Matrox VERO combines an adjustable signal generator, a PCAP recorder, and signal diagnostics integrating EBU LIST into a single appliance with an intuitive, web-based GUI.

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

Matrox X.mio3 12G is a half-length PCI Express card that offers two 12G SDI inputs, two 12G SDI outputs, and two reconfigurable 3G SDI I/Os allowing for 4K and HD simulcast for broadcast production. Multi-channel hardware processing accelerates computation-intensive operations including up/down/cross scaling and compositing for resolutions up-to-4K. The card empowers OEMs to provide small-footprint, one-wire SDI solutions for 4K deployments like production trucks. X.mio3 12G also features 4K compositing and mixing to enable graphics workflows.

Matrox X.mio3 12G is ideal for OEMs who need to create advanced channel-in-a-box systems, video servers, broadcast graphics systems, encoders, transcoders, multi-viewers, switchers and other digital media equipment. This card is supported by the Matrox DSX SDK (Software Development Toolkit). Please refer to the Matrox DSX Developer Products datasheet for information on file I/O, software codecs and CPU effects.

Key features
Half-length PCI Express card
Up to two 12G SDI inputs
Up to two 12G SDI outputs
Two re-configurable SD, HD, 3G SDI I/O
Frame synchronizers on each input
VANC and HANC support for each input and output
Analog blackburst reference input (tri-level or bi-level)
Onboard multi-channel 4K scaler
Onboard multi-layer 4K compositor
Live zero-frame delay video and audio mixer
Up to16 channels of AES/EBU inputs and outputs
Up to eight LTC inputs and outputs

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

X.mio3 FH provides multichannel SDI IO with hardware based video processing in a half-length PCI express card. This card provides up to 12 reconfigurable IOs, from SD to 4K, along with support for AES/EBU, LTC and GPIO to provide for versatile connectivity. The multi-channel hardware processing accelerates compute-intensive operations including motion-adaptive de-interlacing, up/down/cross scaling and mixing/compositing for all resolutions, including 4K.

Matrox X.mio3 FH is ideal for OEMs who need to create advanced channel-in-a-box systems, video servers, broadcast graphics systems, encoders, transcoders, multiviewers, switchers and other digital media equipment.

Key features Half-length PCI express card Re-configurable IO that can support up to 12 SDI IO Frame synchronizers VANC and HANC support for each input and output Analog blackburst reference input (tri-level or bi-level) On-board multi-channel MADI (Motion Adaptive De-Interlacer) On-board multi-channel Up/Down/Cross scaler On-board multilayer compositor Automatic video relay bypass Live zero-frame delay video and audio mixers Up to 16 channels of AES/EBU inputs and outputs Up to 8 LTC inputs and outputs RS422 control Single slot all inclusive option

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

X.mio3 IP offers multi-channel, IP-encapsulated HD and 3G video I/O for SMPTE ST 2022-6, SMPTE ST 2110, and SMPTE ST 2022-7 transport protocols with hardware based packetization, plus onboard de-interlacing, scaling and compositing to free up CPU cycles, all packed into a single half-height, half-length PCIe card.

Key features Multi-channel IO allowing up to 4 video inputs and 4 video outputs with redundancy in HD Guaranteed non-bursty packet transmission with zero CPU usage Support for multiple IP encapsulations IP and analog blackburst timing reference IP bypass and fail-safe mechanisms On-board multi-channel MADI (Motion Adaptive Deinterlacer) Onboard multi-channel Up/Down/Cross scaler Onboard multi-layer compositor Feature rich SDK

Publish » Venue Exhibition » Projectors » Professional video projectors »

Matrox DSX LE4 LP is a low-profile, half-length PCIe card that offers up to eight reconfigurable SDI I/Os—from SD to 4K. When configured with inputs, the cards are equipped with built-in frame synchronizers to generate clean SDI signals in any environment.

Matrox DSX LE4 LP is ideal for OEMs who need to create compact channel-in-a-box systems, video servers, broadcast graphics systems, encoders, transcoders, multiviewers, switchers and other digital media equipment.

Key features
Low profile, half-length PCIe card
Re-configurable I/O that can support up to 8 SDI inputs or outputs
Frame synchronizers
VANC and HANC support for each input and output
Analog blackburst reference input (tri-level or bi-level)
Automatic video relay bypass (optional)
Live zero-frame delay video mixers

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

Matrox Xmio3 LP offers an unprecedented level of integration of both multi SDI I/O and hardware processing in the smallest form factor PCIe card.

This low-profile, half-length card offers up to eight reconfigurable SDI I/Os from SD to 4K.
Multi-channel hardware processing accelerates compute-intensive operations including motion-adaptive de-interlacing, up/down/cross scaling and mixing/compositing for all resolutions including 4K.

Matrox X.mio3 LP is ideal for OEMs who need to create advanced yet compact channel-in-a-box systems, video servers, broadcast graphics systems, encoders, transcoders, multiviewers, switchers and other digital media equipment.

Key features
Low profile, half-length PCI-e card
Re-configurable I/O that can support up to 8 SDI inputs or outputs
Frame synchronizers
VANC and HANC support for each input and output
Analog blackburst reference input (tri-level or bi-level)
On-board multi-channel MADI (Motion Adaptive De-Interlacer)
On-board multi-channel Up/Down/Cross scaler
On-board multilayer compositor
Automatic video relay bypass (optional)
Live zero-frame delay video and audio mixer

Publish » Linear Playout » Linear Playout Systems » Channel branding graphics and promotion systems »

X.mio5 Q25 SMPTE ST 2110 network interface controller (NIC) card provides 25GbE and 10GbE interfaces for multi-channel IP video I/O from HD to 4K, with zero CPU usage. The X.mio5 Q25 card features on-board multi-channel HDR conversions along with motion adaptive de-interlacing, pristine scaling, and powerful compositing engines for high-density 4K video processing.

The Matrox X.mio5 Q25 network adapter offers complete ST 2110 offload allowing OEM software solutions to provide reliable and scalable COTS based 10Ge and 25Ge IP solutions. This ST 2110 NIC offloads all packet processing required for ST 2110-20, ST 2110-30, ST 2110-31 and ST 2110-40, as well as ST 2022-7 redundancy, to free up system resources. With ST 2110-21 packet pacing built into the NIC, software solutions can provide guaranteed narrow senders (Type N) under all conditions independent of CPU load and OS interactions. The X.mio5 Q25 ST2110 IO cards also provide line rate capture allowing for wide asynchronous receivers (type A). Both the permissive wide asynchronous receivers and the ST 2110-21 packet paced, narrow senders allow software solutions to guarantee universal compatibility and interoperability at scale.

Key features Native SMPTE ST 2110 support with no CPU usage 4x UHDp60/50 inputs and outputs over 25GbE 32x HD inputs and outputs over 25GbE 256 audio flows from 1 to 64 tracks Multi-channel HDR conversion On-board de-interlacing, scaling, and compositing, Built in NMOS IS-04 and IS-05 support, 3 year warranty

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 »

Professional A/V applications require solutions that can support the ever-increasing Ultra High Bandwidth video of today at the distances they need to reach, and at a price that won't break the bank. Black Box Active Optical Cables (AOCs) do just that, providing an all-in-one, easy-to-install, ultra-reliable solution for Pro AV installs in control rooms, digital signage, healthcare, and any other application where UHD video needs to go the distance.

• Does not compress or alter the video in any way

• No power supplies - installs in small and tight spaces

• Provides optimum video quality for precise visualization

• Customizable cable lengths cover exact distance requirements

• Eliminates signal delays transporting video signals in real-time with no change or color reduction from image

• Transports video signals over fiber, offering same interference immunity for optimum security

• Costs less compared to other AV extenders

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.

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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.

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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

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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
  
  
  
  

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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)

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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

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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

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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

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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

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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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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.

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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.

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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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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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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 »

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.

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

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.

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-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).

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

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).

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

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).

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-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-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).

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

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-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).

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

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

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

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

In normal operation, the unit has PTPv2 time stamping resolution to 8nsec. It uses a combination of a GPS receiver, a PLL (phase lock loop) and a specialist on-board clock device to create the precise, low jitter clock signals required to drive the physical transceiver’s time stamping circuitry, also providing holdover if the GPS signal is lost.

The specialist on board clock is available in three different types: TCXO, OXCO and CSAC (Chip Scale Atomic Clock, Caesium), which vary in both price and accuracy:
AVN-GMCS – TCXO Temperature Compensated Oscillator accurate to 1 part per million (worst case 1 sec gain/loss every 11.5 days). *
AVN-GMCOS – OCXO Oven Controlled Oscillator accurate to 0.1 parts per million (worst case 1 sec gain/loss every 115 days). *
AVN-GMCCS – SAC Quantum Atomic Clock accurate to 0.0005 parts per million (worst case 1 sec gain/loss every 63 years). *

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

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

The unit shows UTC as standard, but can be set to show ‘local time’ on the front panel, by adding a time offset. Daylight saving time changes can be accommodated by entering Spring Forward and Fall Back dates. It has a real time clock so that accurate date and time is available even after the unit is repowered without GPS access.

The built-in webserver, or front panel OLED display, can be used to configure the unit. The webserver is a responsive design meaning that it can be used with small screens on smartphones and tablets.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

-ends-

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.

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

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

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

Applications:

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


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


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


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


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


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

Webserver Software

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

Mix Matrix

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

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

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

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

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

Front Panel Displays, Metering & Controls

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

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

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

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

Detailed Metering Option

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

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

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

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

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

Physical Inputs & Outputs

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

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

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

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

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

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

8 Way Analogue Headphone Distribution System

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

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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 only DAB_ encoder in the marketplce capable of encoding up to 8 stereo channels in DAB+ MPEG4-AAC,

Applications
- single channel DAB+ encoder for studio to Fraunhofer Content Server
- multi channel DAB+ encoder for studio to Fraunhofer Content Server
- multi channel DAB+ encoder for studio to Small Scale multiplexers

Features
- DAB+ audio encoder compresses up to 8 stereo channels in 1RU
- XLR inputs for analog & digital AES audio
- DAB+ audio coding in AAC+ (HE-AACv2 (ETSI TS 102 563)
- optionally: MPEG-1/Layer II for DAB
- compliant to (Small Scale) DAB multiplexes (Open Digital Radio)
- full remote operation via MuxEnc protocol to Fraunhofer DAB+ Content Servers
- PAD support
- Embedded technology means very low power per channel

Connectivity
- XLR inputs for analog & digital AES audio
- 2x RJ-45 ports for streaming and management
- Supported streaming formats: EDI/DCP, MuxEnc
- web and SNMP control, monitoring and alarming

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Seamless integration of networked audio into existing infrastructures for broadcasters as they are progressively moving to RAVENNA and AES67. Encoding up to 64 stereo channels. Support of MADI for legacy systems Input: AES67/Ravenna/MADI Multichannel codec gateway Output streaming formats: ACIP, RTP & UDP By software license field-upgradable up to 64 stereo channels.

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The Q866 Enterprise Internet Radio Transcoder is the perfect
solution to include a large number of streamed webradio stations
into your OTT and cable networks.
You can include virtually any radio station in the world into your
(local) cable network.

It converts up to 48 internet radio stations (Icecast/Shoutcast) into a
DVB compliant MPEG-2 transport stream.
The signals are available via IP (ethernet) or DVB-ASI.

Low power consumption and the compact design in industrynstandard dimensions (19“, 1 U) allow easy integration of the device into your infrastructure.
Our embedded technolody avoids time consuming updates and upgrades.

FEATURES:
- no annual upgrades or maintenance required - „set and forget“
- transcoding of multiple internet radio stations into DVB compliant transport streams
- receives Icecast/Shoutcast streams (MP3 / AAC)
- high security - WAN port (for Icecast) is physically separated from streaming unit and management - no break-in from internet possible
- several compression algorithms (MPEG 1 Layer II, AAC
- compression algorithm can be set individually per radio station
- all bit rates are supported according to the respective standards
- 32kHz, 48kHz sampling rate
- Meta data Tags are converted into UECP data
- 2 years warranty
- By software license field-upgradable up to 48 stereo channels.

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

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

• 4 x balanced analogue inputs on XLR.


• 4 x balanced analogue outputs on XLR.


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


• POE, Link, and Clock LED status indicators.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19" rack-mount form factor.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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