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LQ-Series is a family of IP interface devices that connect 2-wire partyline intercom and 4-wire audio and call signalling over IP networks. The LQ-Series are interfaces, available as a compact box or a 1RU panel, that can connect 2-wire, 4-wire and 4-wire with GPIO audio and call signalling over IP networks. LQ Series devices can also connect to Eclipse HX frames via the IVC-32-HX interface card over LAN, WAN and Internet.

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CEDAR's dialogue noise suppression (DNS) technology eliminates traffic noise, air conditioning, wind, rain, babble and general background noise from audio signals. It will also help to compensate for unfavourable acoustic conditions and poor microphone placement, and will even suppress excessive reverberation.

The 2-channel DNS 2 is the first DNS designed specifically for portability and extreme simplicity of use. It hosts a new DNS algorithm that requires nothing more than switching on the Learn function that identifies and adapts to the background noise, and dialing in the amount of noise attenuation required. Retaining the near-zero latency of all its predecessors, it's suitable for use in all situations - location recording, live-to-air broadcasting, live sound in venues such as theatres, concert halls, conferences venues, and places of worship, as well as studio work and post.

What's more, despite being so small and light that you can slip it inside your pocket or equipment bag, the DNS 2 offers analogue line and microphone inputs (the latter with optional 48V phantom powering), an AES3/AES11 digital input, and both analogue and digital outputs. This increases its flexibility even further, and its 12V power input means that you can use it in the middle of a field as easily as in the studio or the edit suite.

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The DNS 8 Live is the first dialogue noise suppressor designed specifically for live use. With eight simultaneous channels of dialogue noise suppression (formatted as AES3 on XLR sockets plus DB25 using the AES59 pin-out) it hosts a new DNS algorithm boasting a simplified '2-knob' approach as well as a detailed editing mode that offers more control than any previous dialogue noise suppressor. Nonetheless, it retains the near-zero latency of all its predecessors, making it suitable for use in all live situations - not just live-to-air broadcasting, but also live sound in venues such as theatres, concert halls, conferences venues, and places of worship.

But that's not the limit of the features offered by the DNS 8 Live. Recognising that many people will want to use it where there's no ready access to mains power, we have added a standard 4-pin 12VDC input.

The DNS 8 Live also offers fail-safe audio bypass implemented in the form of hardware relays that immediately connect the inputs to the outputs in the event of a power failure or other significant disruption. This means that the DNS 8 Live can now be installed with confidence in critical paths within the broadcast chain.

We have anticipated the needs of users who will want to control the DNS 8 Live using modern methods such as laptops, iPads and many other (perhaps as yet unseen) devices, so we gave it the ability to respond to a simple, web-based remote control interface. With its bundled DNS 8 RC software, the DNS 8 Live will work with almost any web browser.

The DNS 8 Live has now been superseded by the DNS 8D with Dante®.

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CEDAR's Dialogue Noise Suppression technology is the standard for removing background noise from dialogue in studios. With near-zero latency, high-resolution processing, and a fast, intuitive user interface, it eliminates traffic noise, air conditioning, wind, rain, babble and general background noise from audio signals. It will also help to compensate for unfavourable acoustic conditions and poor microphone placement, and will even suppress excessive reverberation.

The CEDAR DNS 8D has been designed specifically for broadcast and live sound. With eight channels of noise suppression, it hosts a new algorithm that marries the best of the machine learning and signal processing on which the DNS 8 Live and the DNS 2 dialogue noise suppressors are based. With its slick user-interface that includes standard and detailed modes of operation it offers better control than any previous dialogue noise suppressor, both from the front panel and from almost any web browser that can access its internal remote control software. With its near-zero latency and standard 4-pin 12VDC input in addition to universal mains power, it's suitable for use in all live situations - not just broadcasting, but also live sound in venues such as theatres, concert halls, conferences venues, and places of worship.

Dante® is the de-facto standard for media networking, distributing uncompressed, multi-channel digital audio via standard Ethernet networks with near-zero latency and perfect synchronisation. The DNS 8D is the first DNS with a Dante interface in addition to its AES3 audio connections (DB25 using the AES59 pin-out). With fail-safe AES3 audio bypass implemented in the form of hardware relays that immediately connect the inputs to the outputs in the event of a power failure or other significant disruption plus support for Dante's redundant mode of operation which helps to protect against network failure, it can be installed with confidence in critical paths anywhere within the audio chain.

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STYX (pronounced sticks) is a hugely versatile configurable 48 x 46 mixer/ utility device with 32 x 32 network audio inputs & outputs with the added benefit of 6 x 6 analogue line level circuits, 2 x 2 AES3 circuits, 3(+3) x microphone/line inputs, 2 line up tone inputs & 3 stereo headphone outputs.

The front panel features 24 x assignable rotary encoders and 24 x assignable illuminated switches, plus a mono assignable PPM and LED graph to indicate positioning of the encoders when they are used.

The real power of STYX is its Windows 10 application which in real time can adjust any of the 2208 mixer cross point levels, provides many setup options, and importantly, allows presets to be stored and recalled.

Reliability is key, STYX has multiple redundant power supply sources and redundant network interfaces.

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

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

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

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

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

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

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

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

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FreeSpeak Edge works in the 5GHz bandwidth – beyond what you know about wireless intercom. It extends the entire FreeSpeak™ range, giving you the freedom to think bigger.

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PlayBox Technology has incorporated its most advanced playout engine which is the result of over 15 years of experience in developing playout solutions for the world's

leading broadcasters.

 

PlayBox Technology Channel-in-a-Box provides an integrated playout solution for Broadcasters and OTT service providers in a 1RU or 3RU server, regardless if you are working with IP or SDI.

The CIAB will enable you to playout an IP stream or an SDI in SD, HD or UHD. PlayBox Technology’s CIAB takes care of all your scheduling needs and uses simple, clean and engaging HTML5 for your transmission graphics needs including crawlers and rollers.

 

As you would expect from PlayBox Technology, it offers sophisticated and easy to use automation. We have designed a user friendly ingest which also gives you the ability to FTP your media straight to the Channel In A Box, saving you time.

It is available as a complete turnkey solution for single channel or multichannel HD SDI/SD. Available as both full turnkey solutions and software-only for single and multi-channel operations.

 

Channel in a Box (CIAB) is compatible with a wide range of compression types, including: MPEG-2, DV, DVCPro, DVCPro HD, AVC/H.264, HAVC/H.265, XAVC and more.

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The DNS 4 is the most advanced portable model in the DNS series. With its industry-standard 12V power input and mini-XLRs for many connections, it takes up little more space than the DNS 2 yet doubles the number of channels available.

It's incredibly fast and simple to use; the DNS algorithm requires nothing more than switching on the Learn function that identifies and adapts to the background noise, and then adjusting the amount of noise attenuation required. Notwithstanding this simplicity, it can achieve remarkable results. What's more, despite being so convenient that you can slip it into your pocket or equipment bag, it offers analogue line and ultra low-noise microphone inputs (the latter with optional 48V phantom powering), AES3/AES11 digital inputs, and both analogue and digital outputs. And, to increase its flexibility still further, it runs on 12V power so you can use it in the middle of a field as easily as in the studio or edit suite.

Retaining the near-zero latency of its predecessors, the DNS 4 is suitable for use in all situations - location recording, live-to-air broadcasting and live sound in venues such as theatres, concert halls, conferences venues and places of worship, as well as in the studio.

 

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MediaServicesIQ contains a suite of machine learning (ML) applications. These microservices can be applied to speech-to-text, video intelligence, closed caption conformance, and content classification in media workflows. The microservices in MediaServicesIQ integrate easily with media workflows used by broadcasters and media houses, helping them leverage AI/ML technologies within existing workflows.

Media ServicesIQ provides an easy-to-access portal to cloud-based AI/ML services through custom-developed workflows.

  KEY FEATURES & CAPABILITIES

Metadata Generation

Cloud-based collection of speech-to-text and video intelligence applications automating the generation of speech and video-based metadata Provides off-the-shelf integration with the Avid system Easily interfaces with existing workflows using REST API Video intelligence metadata for face, logo, ad, number, on-screen text, object, and shot change detection Metadata with timecodes for tagging, enabling search and faster content retrieval.

Automatic Closed Caption/Subtitle Generation

Machine-generated, time-synced outputs in various sidecar formats for further processing Faster, scalable, reliable, and secure generation of preliminary outputs Easily ingest sidecar files into workflows for repurposing.

CC Quality Conformance for Multi-Platform publishing

Generate reports on non-conformance parameters within a CC sidecar file Automatically compares against predefined style guides standards from platforms such as Netflix Time coded error reports for easy navigation and correction Ability to ingest multiple sidecar formats.

Auto classify content for better value

Automatically create a classification for content Classify advertisements using various labels to automatically identify and ingest Automatic classification of high-points in a feature-length video Easily integrate and ingest labels into existing workflows.

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A firmware upgrade for Sony’s professional slot-in receiver, DWR-S03D, a part of the DWX wireless series, is now available. Using the dedicated DWA-SLAU1 adaptor, sound engineers can now benefit from the established audio quality of the DWX series using the UniSlot® or SuperSlot® interface. This new compatibility will offer a wideband digital audio solution for location sound recording for movies, dramas and documentaries.