RƎLAY Virtual Radio Software

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 »

A__UHD Core is the revolutionary new audio engine for Lawo’s mc² audio production consoles. Designed as a network-based, software-defined IP DSP engine with unparalleled processing density: 1,024 mc²-grade DSP channels, which can either be utilized by a single mc² console, or be shared amongst multiple consoles for effective and space-efficient resource pooling: A__UHD Core’s Pooling license allows users to split the massive DSP power across multiple consoles, powering up to four independent audio consoles with 256 fully-featured mc² DSP channels each. As a result, only one A__UHD Core now does what previously required four individual console cores.

Plenty of reliability features are built-in. A__UHD Core features four independent IP audio streaming engines, each with a redundant pair of 10*/1 GbE network ports (SFP). The device is fully based on open standards and supports Audio-over-IP I/O via ST2110-30/-31, AES67 and RAVENNA. Each network interface supports up to 512 Rx & Tx streams with stream sizes from 1 to 128 audio channels. Redundant hot-swappable PSUs are standard, as well as network interface redundancy for all streaming and management ports, allowing redundant IP network paths using ST2022-7 Seamless Protection Switching and adheres to the Class C specification for deployment within LAN or WAN environments, allowing redundant A__UHD Core units to be housed just about anywhere on the planet—and still take over instantly if the need arises.

Finally, a flexible licensing model makes A__UHD Core ideal for both mobile applications and facility use. For mobile, the scalable DSP performance with temporary licenses is a great way to turn CAPEX into OPEX, whilst in facility applications, the possibility of resource pooling and flexible allocation of DSP resources to multiple physical and GUI-based mixing surfaces maximizes ROI for your audio infrastructure. This flexible software licensing offers fixed and temporary licenses to provide exactly what you need, when you need it. Licenses are stored on USB dongles and can be conveniently downloaded, allowing for seamless transfers from unit to unit.

Best of all, A__UHD Core’s functionally is defined by its software, making it a future-proof investment with a feature-set that is designed to expand.

Support » Monitoring, Control & Communications » System Monitoring & Control » Broadcast system control SW and HW »

 

HOME is a management platform for IP-based media infrastructures. It is designed to connect, manage and secure all aspects and instances of live production environments. HOME provides the tools and centralized services for swift and effective interaction of engineers with their tools.

HOME is cloud-native by design and ready to run anywhere, irrespective of the system’s size. With HOME, the cloud starts on your campus, private and locally. It turns an array of devices, setups, sites, hubs and data centers into a powerful, agile network — quickly and in a perfectly secure way.

Inside HOME, discovery of devices is automatic, while registering and admitting them to the network is only a button press away.

With the adoption of IP well underway, the focus of operators has shifted from whether to adopt IP to how to use its potential with minimal effort and maximum effect. This is where HOME shines: it addresses all pressing issues real-world operators face today and tomorrow. In one place and via a single, platform-agnostic, intuitive user interface.

Lawo’s HOME platform is based on open standards such as ST2110, NMOS, IEEE802.1x and RADIUS and has been designed and built from the ground up using LUX—the Lawo Unified Experience, a framework for conceiving, designing, and building solutions that put you first and defines the standard for user experience and design across the Lawo portfolio.

Discovery and Registration: HOME solves IP complexity with automatic plug & play discovery of IP audio and video devices, which are registered with their name, location, status and type. This applies not only to Lawo products but to third-party solutions as well via NMOS. Discovered devices are managed in a central inventory list, ready for access and configuration.

Device Management: In today’s hectic live broadcast environments, operators rely on speedy, unified device configuration routines, especially when setting generic device parameters or configuring senders and receivers. The ability to save and recall configurations is key to speed up tasks. HOME provides a centralized “mission control” for these processes, providing fast and unified access to device parameters for easy tweaking, irrespective of the end point being controlled.

Operability: With its simple, user-friendly UI, HOME allows users to organize and access processing services. With all required facilities accessible in one place, operators can set up and change stream configurations, and route them across an infrastructure without the need for a separate controller. For large infrastructures HOME works seamlessly with a broadcast controller in the same set-up and helps to speed up configuration and operation. HOME is based on LUX, a UI language common to all Lawo devices and many of their functionalities. Through HOME’s user interface, operators can access and edit device parameters quickly utilizing integral mechanisms that help get the job done efficiently. With HOME, operators quickly get right to what they’re looking for, without distractions and complications, to focus 100% on the task at hand.

Security: The content created by a production crew and transported over a network is any operation’s most valuable asset and deserves strong protection. While a robust security system needs to cover all aspects of media infrastructure and content creation, the key lies in its simplicity. HOME provides a variety of security strategies, first of which is quarantining unknown devices when they come online. Only after being deliberately approved, via an intuitive IEEE802.1X-based routine, can they begin exchanging signals with the HOME network. Secondly, HOME uses an authentication strategy based on a centralized user management system, with dedicated user roles and groups. The LDAP based service allows users to authenticate either locally – within HOME – or via their own corporate IT infrastructure, e.g. Microsoft® Active Directory. Finally comes the arbitration of devices and individual streams based on pinpointed rights management. HOME’s architecture is prepared to manage services such as transport layer security, network segmentation and other IT security mechanisms such as RADIUS.

Scalable Architecture: Home is cloud-native by design, which means that its architecture is built to run detached from hardware constraints. This does not automatically mean that services must be outsourced to an external service provider whose meter is running 24/7; with HOME, the cloud starts on your campus, private and locally, on COTS hardware. The HOME platform is designed as functional blocks that provide microservices, which are self-contained and supply functionality to operators or other services.

HOME can be expanded with additional services at any time to increase its functionality — the platform scales on demand. Should there be a need for a larger RDS, because the installation grows, additional instances of the required resources can be added anytime. One of the core principles of HOME is its focus on the utilization of open standards wherever possible, for broadest compatibility and future-proof integration. With HOME, flexibility and resource utilization in IP media infrastructure maximizes.

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

 

Big performance in a compact size: that’s Lawo’s new second-generation mc²36 audio production console, an incredibly powerful yet compact mixer, now even more capable with a new dual-fader operating bay that allows 48 faders in the same space as a 32-fader board. This makes the new mc²36 even more of an all-rounder for theater, houses of worship, corporate, live and broadcast audio applications – wherever ultra-high audio performance in close quarters is demanded.

With DSP more than doubled from its predecessor, the new mc²36 with built-in A__UHD Core functionality offers 256 processing channels, available at both 48 and 96 kHz, and natively supports ST2110, AES67, RAVENNA, and Ember+. It provides an I/O capacity of 864 channels, with local connections that include 3 redundant IP network interfaces, 16 Lawo-grade mic/line inputs, 16 line outputs, 8 AES3 inputs and outputs, 8 GPIO connections, and an SFP MADI port.

Operating and visualizing features include Button-Glow and touch-sensitive rotary controls, color TFT fader-strip displays, LiveView™ video thumbnails, and super-precise 21.5” full HD touchscreen controls. Its built-in full loudness control is compliant with the ITU 1770 (EBU/R128 or ATSC/A85) standard, featuring peak and loudness metering which can measure individual channels as well as summing busses.

The new mc²36 offers seamless integration of third-party solutions like recording systems, effects engines, and other systems running on external PCs, into its user interface. Applications display right in the console’s screen, while console keyboard, touchpad and touchscreen provide control.

Last but not least: mc²36 offers best-in-class integration of Waves SuperRack SoundGrid, providing operators with access to Waves’ extensive plug-in selection of real-time signal processing alongside the console’s internal processing engine — no additional screens or control devices required.

Moving the mc²36 to the A__UHD Core means that all developments in the future will happen on a single, unified platform, and Lawo continues to provide production file compatibility between all mc² consoles. For instance, a production file from an mc²36 can be used on an mc²96 for post-production — opening the door to a whole new level of performance.

The new mc²36 is based on Lawo’s A__UHD technology and includes Lawo’s innovative new IP Easy functionality that makes IP setup as simple as analog. IP Easy, based on Lawo’s innovative HOME management platform for IP-based media infrastructures, makes connecting IP audio and I/O devices as easy as if they were baseband – the console automatically detects new devices and makes them available at the touch of a button. IP Easy even manages IP addresses, multicast ranges, and VLANs, and includes security features like access control and quarantining of unknown devices to protect your network. HOME is natively built on a cloud-ready microservices architecture, enabling users to connect, manage and secure networked production setups from the ground up. Furthermore, it provides centralized access and control for all Lawo gear within a setup. HOME helps broadcast professionals address some of the most demanding requirements of modern IP infrastructures, including automated discovery and registration of devices, connection management, flow control, software & firmware management, scalability and security.

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

With experience gathered over more than 50 years, German audio innovator Lawo is distinguished by its engineering and manufacturing of the most reliable and most advanced audio mixing consoles available. Originally developed for mobile and studio broadcast environments with zero tolerance for failure, Lawo consoles are also widely chosen for their audio quality in theater, studio and live performance applications.With the mc²96, the mc² platform is pushed beyond limitations, culminating in a pure, unparalleled audio production tool dedicated to those who can tell the difference. Whether it’s intended to be used in broadcast, theater or recording – in each of these applications it stands out with a unique set of features. The mc²96 provides optimized performance within IP video production environments with native ST2110 support. In addition, the revolutionary LiveView™ feature enables thumbnail previews of video streams directly in the fader labeling displays.The mc²96’s unique design combines large screens with a low overbridge height to provide a more ergonomic profile without compromising an engineer’s view. The console’s 21.5” full HD touch-screens work hand-in-hand with the touch-sensitive color-illuminated rotary encoders. An equalizer window will automatically pop-up when touching the equalizer encoders. In multi-user mode, 96 rotary encoders in each 16-fader bay give direct access to all parameters to create an additional central control panel. Color coding of encoders coupled with the high-res color TFTs guarantees a perfect overview. Users can create, save and recall multiple personalized free control layouts directly in their own bay.The mc²96’s extended free control section gives direct access to six parameters in addition to gain control. The result: A perfect balance between decentralized and centralized control workflows. The console allows application of both global and individual parameter assignments for the free controls of each channel strip, and each of the touch-sensitive free controls is color-coded for a comprehensive overview. Additionally, the integrated Remote Desktop function allows seamless integration into the console’s user interface of multiple external PCs running third-party solutions. With the switch of a button, the external PC and its software applications are displayed in the console’s screen while the console’s keyboard, touchpad and touchscreen provide control. The seamless integration of external recording systems, effect engines, or other user interfaces, means less equipment – and the engineer has control over the complete set-up, conveniently from a single, central position.The mc² 96’s comprehensive local I/O features 16 Lawo-grade Mic/Line inputs, 16 Line outputs, 8 AES3 inputs and outputs, 8 GPIOs plus a local MADI port (SFP). The built-in switch provides 4 Ethernet ports, and network connectivity is furnished via two redundant ST2110 / AES67 / RAVENNA ports with ST2022-7 Class C seamless protection switching (SPS). An optimized thermal concept results in an absolutely quiet console, fulfilling a key requirement for concert halls and studio environments. Being an IP-native device based on A__line UHD Core, the mc² 96 includes Lawo’s IP Easy functionality which makes configuring IP setups as simple as analog. The system automatically detects devices and makes them available to the console at the push of a button. Furthermore, the console’s device management includes all needed security features such as access control and quarantining of unknown devices.Selected Features:

• Frames from 24 to 200 faders (with Dual Fader option).


• Remote stand-alone frames of 16 faders.


• 6 banks each with 2 layers.


• TFT metering: mono, stereo or up to 7.1 including bus assignment, gain reduction for dynamics, Audio-follow-Video status, VCA assignment, Mix-Minus, Signal Patching, Meter selection, Automix state.


• GUI page output, e.g. metering, on an external monitor.


• 40-bit floating point signal processing with 1,024 DSP channels (768 inputs and 256 summing buses).


• Up to 768 inputs with A/B input, up to 128 aux buses, up to 96 groups, up to 96 main sums, 32 Automix groups.


• Up to 32 surround channels, 128 VCA groups with metering, 256 GP channels.


• Surround formats: DTS & Dolby ® Digital 5.1, Dolby ® Pro-logic 4.0, DTS ES &  Dolby ® EX 6.1, SDDS 7.1, DTS-HD 7.1, diverse panning characteristics, surround aux bus.


• Loudness Metering according to EBU R128, ATSC A/85 and ARIB, momentary or short term in every channel, integrated measurement on summing channels with display of integrated LUFS value in headline.


• Waves SuperRack SoundGrid ® integration.


• Interfaces include Mic / Line, Line Out, AES3, 3G SDI, HD-SDI, MADI, GPIO, Serial, MIDI, ST2110 / AES67 / RAVENNA / DANTE ® Audio-over-IP.

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.

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 – radio »

ruby, from Lawo, is the mixing console that truly meets the demands of modern radio by blending the hands-on immediacy of physical faders with the context-sensitive efficiency of multi-touch controls. ruby frees your talent to create easily, naturally, effortlessly — in the way that suits them best. ruby is intuitive, uncluttered, powerful; but beneath its svelte surface lie color-programmable visual cues, context-sensitive displays and custom workflow options to help ensure that your hands always find the control you need, first time — every time. The mixing surface has a sleek, modern look that’s both functional and beautiful, designed for fast, accurate operation. From the high-resolution OLEDs that display sources and settings to the programmable function buttons and big, tactile ON/OFF and PFL keys with LED backlighting, ruby looks and feels great. Built in Germany to exacting Lawo standards, ruby features all-metal construction, premium 100 mm faders with built-in dust shield, ergonomic wrist-rest and ultra-reliable switches and rotary controls. There are smart tools like AutoMix, an intelligent algorithm that automatically maintains the balance of multi-mic productions so that creative staff and focus on what they do best — create. Desktop or flush-mount versions in 4, 8, 12 and 16 fader frame sizes, plus standalone 4 and 8 fader and Master Control extenders, allow combining multiple frames to build systems as large as 60 faders.

Today’s board operators interact as much (or more) with screens and displays as they do with the mixing console, so we’ve integrated mixing tools and other console controls into ruby’s multi-touch enabled display (powered by Lawo’s VisTool GUI-building software). No hunting for buried options: multi-touch on-screen controls instantly give what’s needed to control studio devices, apply dynamics, adjust virtual faders, and display true loudness metering. The optional VisTool Unlimited software upgrade even enables you to design context-sensitive screens customized to your station’s unique operating style, using your own custom-built graphics and control layouts.

ruby controls Lawo’s Power Core, perhaps the most powerful mixing engine ever built for radio, with redundant IP networking and redundant power capabilities. At just 1RU, Power Core‘s compact form belies the immense capabilities inside. On the front panel, a high-resolution color OLED gives status and setup information. I/O includes dual AES67 Ethernet ports with SFP, capable of 128 bi-directional AoIP streams, and 4 ports for high-density MADI signals (128 total channels of audio) — perfect for native MADI-to-AES67 AoIP conversion. It complies with the ST2110-30 standard too, ensuring seamless operation in combined radio / TV broadcast plants. ST2022-7 Seamless Protection Switching enables simultaneous dual-redundant network links, if you choose. Clock sync connections and 8 GPI / GPO contact closures complete the scene. Auto-switching dual-redundant power is standard, with Power Core’s internal auto-ranging AC power supply complemented by an inlet which accommodates an external 12VDC backup power supply.

There are also 8 rear-panel expansion slots, which accept a variety of optional IO expansion cards, including MADI, mic/line inputs, line inputs and outputs, AES3 inputs and outputs, and even a Dante interface. And Power Core’s raw audio processing horsepower puts other mix engines to shame, with the ability to handle thousands of simultaneous signals, a routing matrix of up to 1,920 x 1,920 crosspoints, and up to 96 channels of DSP input processing usable for EQ, de-essing, dynamics, etc. tucked into just 1RU. Multiple tiered license options make it easy to choose the combination of price and power that’s just right for you.

Support » Monitoring, Control & Communications » System Monitoring & Control » Data-driven analysis and self-healing »

Lawo’s System Monitoring And Realtime Telemetry Dashboard (aka smartDASH) is a vendor-agnostic enterprise software suite designed to provide full network and media visibility across an all-IP, all-SDI or hybrid WAN/LAN broadcast infrastructure. Based on a LINUX OS, this software-defined networking solution incorporates a powerful and robust database to document and rapidly search any aspect of the operation—from a simple cable ID number to seeking the journey of a multicast across a transnational multi-hop WAN.

By leveraging a vast library of software communication protocols, the smartDASH automatically interrogates live and dormant path connections to create the most intuitive and data rich presentation layers of a COTS-hybrid infrastructure, deriving media network data and documenting the network and supporting infrastructure. With its award-winning network micro-services, smartDASH supports a wide range of network protocols to build up rich graphic representations of the network infrastructure. These automated processes dynamically document the interconnected network design in real-time. SmartDASH users have a zero-footprint installation: SMART can be deployed on-prem, in a private or public cloud, and is accessible from a browser and/or mobile device.

smartDASH supports monitoring and decoding for media formats from low-bit-rate OTT/ABR streams to uncompressed ST2110 studio production flows, in addition to characterizing packet pacing off the delivery network. Unifying network telemetry and mixed media flow brings deep operational visibility into a single glass view. smartDASH even tracks device inventory spares and accounts for CAPEX and OPEX KPIs to manage cost of ownership.

Produce » Real-Time Production » Video Production » Production switchers »

Lawo’s new V__matrix IP broadcast video core infrastructure product will change your idea of what a broadcast facility looks like from legacy to future, quickly transforming any broadcast installation into a flexible, future-proof production facility, addressing a wide range of workflows and supporting your transition to a totally IP-based environment.

V__matrix is the first of its kind. Free from the restrictions of legacy hardware platforms, it offers a completely virtualized real-time routing and processing infrastructure. Instead of connecting single purpose modular products in elaborate production chains, V__matrix employs data center principles of flexibility, fabric computing and COTS economics and makes these available to any live production broadcast environment. Whether an OB truck, a TV studio or a broadcast operation center, V__matrix creates a fully virtualized facility infrastructure. The V__matrix ecosystem is based on generic high capacity FPGA-based processing blades upon which Virtual Modules (VM) are loaded to create the functionality required. Multiple cores are connected through redundant 40GE (or 4x 10GE) Ethernet interfaces to an IP network to form a distributed IP routing and processing matrix that provides frame-accurate, clean switching just like a legacy baseband matrix.

The V__matrix ecosystem scales linearly from tens to thousands of I/O and processing functions which make it ideal for any size live broadcast facility, small or large. Capabilities easily scale as well. An entire production workflow can be remapped in minutes when requirements change from production to production. The functionality of any processing blade can be exchanged, enabling system capabilities to easily be modified or upgraded to address your constantly changing business requirements. The V__matrix pool of generic processing blades provides ultimate flexibility; with software-defined functionality they can be configured and called upon to handle the peaks and troughs of seasonal production demands and with Lawo’s innovative licensing model, Virtual Modules can be assigned to a particular processing blade or be stored in an on-site license server allowing for unprecedented flexibility.

The V__matrix ecosystem can be divided into two parts: physical and virtual. The physical consists of the C100 processing blade and associated hardware which provides the compute and processing capacity of the platform; the more compute power you have the more functions you can run. Since V__matrix is a fully IP-based platform, the C100 processing blade can be placed anywhere there is an IP network. It can be decentralized and spread over one or more facilities or centralized in a core facility or OB truck. A hybrid approach is also possible where some core equipment is kept on-site while a pool of processing power is kept in a remote data center. This decentralized approach allows the technical operation center to be situated in a purpose-built data center outside of town where space, power and cooling is inexpensive, while talent and studios can be in another area.

The virtual world is centered around the software which defines the functionality of the platform. The software packages are called Virtual Modules (VM) and in the V__matrix they allow the function-agnostic core processing hardware to build complex workflows by simply running the appropriate VM. Combining VMs allows the creation of complete production chains fulfilling all broadcast requirements in a fully virtualized environment. As all functions of the V__matrix ecosystem are software-defined it is the ultimate future-proof platform. Changing the functionality of your broadcast plant is as easy as changing the software modules loaded onto the processing blades. By cascading multiple VMs together, the V__matrix scales linearly up to thousands of SDI I/O and audio/video processing functions for unparalleled scalability, flexibility, versatility and cost-efficiency. The current line-up of V__matrix Virtual Modules include:

• vm_streaming – SDI-to-IP Gateway


• vm_udx 4K/HDR Format Converter – IP Up/Down/Cross and Color Space Converter


• vm_mv16-4, vm_mv18-4 & vm_mv24-4 — 4K Multiviewers


• vm_dmv – World’s first expandable distributed 4K IP Multiviewer

V__matrix multiviewer VMs are controlled by Lawo’s groundbreaking touch operated configuration system “theWALL”. This unique HTML5 based GUI makes mosaic configuration with borders, colors, UMDs, tally etc. a simple case of drag and drop.

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.

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

Lawo's V__remote 4 is designed to provide a one-box solution for all the requirements of video and audio signal transport and processing in WAN-based remote productions. It includes everything from Video-over-IP coding to various monitoring and processing tools. All designed for one purpose: to provide a tool that increases the flexibility of any broadcast application, while saving valuable rack-space, set-up time and production costs. With its virtual cabling capability, V__remote 4 leverages IP infrastructure advantages and cost savings to provide an unparalleled degree of flexibility and scalability.

In remote live production, IP is becoming a fundamental requirement, and getting reliable, low latency and high-quality video from venues back to studios at reasonable cost is an absolute necessity. The Lawo V__remote 4 achieves this by combining a bi-directional, four-channel Video-over-IP interface, four local SDI inputs and outputs, and all the processing tools usually needed when contributing video and audio via WAN or LAN to a broadcast production, into a compact 1RU devices.

V__remote 4 provides four 3G / HD/ SD SDI inputs and four 3G / HD / SD SDI outputs for interfacing to external video equipment, and converts these signals into IP streams (and vice-versa). These streams can be transported via standard Layer 3 IP LANs or WANs. V__remote 4 provides parallel encoding in multiple streaming formats, making the same signal available to different applications – e.g. RAW for local production, JPEG2000 for remote sites, MJPEG and H.264 for monitoring applications, and H.264 for internet distribution. The IP-based approach allows easy signal routing via Lawo’s VSM control and monitoring system or other external master control systems. No rewiring is needed as long as all devices are connected to the network. Since the device is based on real network technology with multicast capability, it allows easy transmission of signals to multiple outputs within the network.

The V__remote4’s coding engines are designed to meet the highest demands for video quality and signal transport reliability. The extremely robust J2K codec (with a special protection algorithm for signal transport even on unreliable WAN connections) and ST2022-7 port redundancy ensure signal availability and quality. Lawo’s EPS technology for enhanced protection switching goes even further, enabling system designs with true hardware and network redundancy. Format and quality of the IP video streams can be configured individually to provide the optimum balance of picture quality, latency and bandwidth. Its 6 Ethernet ports are connected to an internal switch allowing “tunneling” of IP traffic such as camera control, RAVENNA streams or even office and Internet IP, through the 10GBit interface.

Support » Monitoring, Control & Communications » Video Monitoring » Multiviewer display processors »

 

vm_dmv is a true revolution in multiviewer technology. Leveraging the power of IP using Lawo’s advanced V__matrix IP video routing & processing platform, vm_dmv lets you build a true 4k UHD multiviewer that can easily grow and adapt to suit your unique requirements. Build a system as small as 24 inputs and 4 control heads, or as large as 768 inputs with 128 heads – and beyond.

The secret lies in the pairing of Lawo’s unique LiveView™ technology and V__matrix architecture. vm_dmv “virtual module” apps run on C100 blades inside a Lawo V__matrix rack unit, which accepts SDI and 40GbE connections. Combining LiveView™ with the ability to “cluster” C100 processing blades keeps bandwidth requirements low, while enabling incredible scalability. Need more inputs or more control heads? Simply load another vm_dmv module to increase system capacity.

vm_dmv takes full advantage of modern network design. It utilizes a distributed architecture, with multiple software modules running on C100 blades within V__matrix frames connected via IP. Individual modules can be hosted in the same frame, or located in different frames within your facility — even in an entirely different location altogether. Which means you can also have multiviewer heads wherever you need them: Master Control, remote production facilities, even OB vans. No other multiviewer system can give you this kind of flexibility.

Traditional multiviewers require massive amounts of bandwidth because they generate control head PiPs using incoming video signals at full-scale. vm_dmv solves the bandwidth problem with cutting-edge LiveView™ MIP-mapping technology. Each incoming signal is optimized and downscaled, then served as four simultaneous streams of differing sizes. These streams become the PiPs used in each multiviewer head; once a source is present, it can be displayed by any (or every) head in the system. LiveView™, coupled with V__matrix’s 40GbE IP backbone, provides true bandwidth efficiency, ultra-low latency, and one-to-many source distribution.

vm_dmv is space-efficient, too. Each V__matrix rack frame accommodates two C100 blades, making it possible to deploy a 48-8 vm_dmv multiviewer system using only 1RU of space. And because the system is built on a true-IP backbone utilizing the ST2110 and ST2022 standards, you can build a distributed architecture, with V__matrix frames placed adjacent to source inputs and connected via Ethernet —instead of running cable bundles to the central rack room.

A groundbreaking multiviewer deserves an equally groundbreaking control and configuration system. vm_DMV is controlled with Lawo’s “theWALL”, a unique HTML5-based GUI that makes mosaic configuration with borders, colors, UMDs, tallies and more a simple case of drag-and-drop. Configure any monitor wall, route signals, change mosaic layouts, save and load user presets. theWALL does all this and more, with a smart, intuitive touch interface that’s a snap to use. Ideal for EICs, line producers and operators, theWALL gives every member of your production team the power to change their monitor wall layouts on-the-fly in seconds flat.

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Support » Monitoring, Control & Communications » System Monitoring & Control » Broadcast system control SW and HW »

A VSM IP control system brings together all the requirements for a safe and flexible broadcast operation. The “everything under one roof” philosophy unites technical devices and operators in a most adaptable way. User panels and interfaces can be configured without limitation to meet the requirements of different workflows and applications, while the whole system is under redundant control.

With hundreds of different protocols implemented and growing by the day, VSM seamlessly integrates with the majority of the most popular broadcast equipment on the market. This includes video routers, video switchers, audio routers, audio consoles, multi-viewers, intercoms, modular equipment and many special third-party devices. By talking native protocols where possible, equipment from different manufacturers can be seamlessly “glued” together, giving unmatched recall and logic control possibilities system-wide. With a modern TCP/IP backbone, VSM utilizes standard IT hardware but enhances reliability and redundancy with sophisticated software redundancy concepts. VSM also provides interfaces to connect serially controlled devices, again freeing you to decide on the best hardware technology, no matter the format of the physical control interface. Real time feedback from every single crosspoint and parameter is available, as well as special applications such as audio metering via the protocol. All VSM user interfaces display the confirmed action from an attached device, guaranteeing maximum transparency for every operator and every control unit. Additionally, a global naming facility for mixers, UMDs, user panels and many other devices provides status information at a glance.

VSM servers are the heart of the control system. Running vsmStudio software, all administration and configuration is both programmed and saved runtime in intuitive and easy to use software. Control interfaces in the form of a wide range of hardware LCD button panels and software panel clients allow simplified operation from highly flexible and custom designed configurable GUIs. Additional VSM hardware includes GPIO interfaces, UMDs for dynamic labelling, and SmartHubs, which convert control signals to and from serial to TCP/IP. SNMP Monitoring capabilities are realized via the vSNMP editor software tool which runs on a separate server.

VSM is the ultimate control system integration solution. Applications include mobile production, in-studio production, multi-studio operation with “boxing” of complete studios for fast switching on the fly, TV and Radio master control rooms, and more. VSM solutions include vsmSOUL Seamless Orchestration & Unification Layer, vsmTally Multi-Studio Tally Management System, vsmSNMP Alarm Management System, and vsmGear control panels and interfaces.

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

Quasar SR is the direct replacement for Axia’s best-selling Fusion console. If you like Fusion, you’ll love Quasar SR—which is comparable to the Fusion console in both price and feature set but also delivers all the power, ergonomics, industrial design, and star appeal of our flagship Quasar XR console.

Quasar SR is not reserved for the most knowledgeable broadcaster but is approachable to any board operator thanks to its streamlined surface design. Quasar SR uses the same frame, power supply, and master module as Quasar XR, but the fader modules are non-motorized, and there are fewer, larger, and easier-to-reach buttons on each channel strip. If the LED meters next to each fader on the Quasar XR console are a bit too much functionality for your surface, then Quasar SR might be the solution for you, featuring Confidence Class Metering only.

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

With the introduction of the Quasar v2.0 Major System Update, the original flagship Quasar console becomes ‘Quasar XR’. Existing Quasar users can upgrade their console to XR by installing this free system-wide update, which adds scalability and modularity to the original console. V2.0 also introduces Quasar Soft and Quasar Cast remote control and monitoring solutions, delivers full integration with Telos Infinity® IP Intercom products, and adds support for new Quasar Accessory Modules, along with numerous other powerful features. This advanced scalability addresses new workflows, like Work from Home, and gives broadcasters more flexibility when it comes to the number of channels they need in their Quasar Engine, starting at 16 channels and scaling up in blocks of 16 channels up to 64 channels.

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

 

Big performance in a compact size: that’s Lawo’s new second-generation mc²36 audio production console, an incredibly powerful yet compact mixer, now even more capable with a new dual-fader operating bay that allows 48 faders in the same space as a 32-fader board. This makes the new mc²36 even more of an all-rounder for theater, houses of worship, corporate, live and broadcast audio applications – wherever ultra-high audio performance in close quarters is demanded.

With DSP more than doubled from its predecessor, the new mc²36 with built-in A__UHD Core functionality offers 256 processing channels, available at both 48 and 96 kHz, and natively supports ST2110, AES67, RAVENNA, and Ember+. It provides an I/O capacity of 864 channels, with local connections that include 3 redundant IP network interfaces, 16 Lawo-grade mic/line inputs, 16 line outputs, 8 AES3 inputs and outputs, 8 GPIO connections, and an SFP MADI port.

Operating and visualizing features include Button-Glow and touch-sensitive rotary controls, color TFT fader-strip displays, LiveView™ video thumbnails, and super-precise 21.5” full HD touchscreen controls. Its built-in full loudness control is compliant with the ITU 1770 (EBU/R128 or ATSC/A85) standard, featuring peak and loudness metering which can measure individual channels as well as summing busses.

The new mc²36 offers seamless integration of third-party solutions like recording systems, effects engines, and other systems running on external PCs, into its user interface. Applications display right in the console’s screen, while console keyboard, touchpad and touchscreen provide control.

Last but not least: mc²36 offers best-in-class integration of Waves SuperRack SoundGrid, providing operators with access to Waves’ extensive plug-in selection of real-time signal processing alongside the console’s internal processing engine — no additional screens or control devices required.

Moving the mc²36 to the A__UHD Core means that all developments in the future will happen on a single, unified platform, and Lawo continues to provide production file compatibility between all mc² consoles. For instance, a production file from an mc²36 can be used on an mc²96 for post-production — opening the door to a whole new level of performance.

The new mc²36 is based on Lawo’s A__UHD technology and includes Lawo’s innovative new IP Easy functionality that makes IP setup as simple as analog. IP Easy, based on Lawo’s innovative HOME management platform for IP-based media infrastructures, makes connecting IP audio and I/O devices as easy as if they were baseband – the console automatically detects new devices and makes them available at the touch of a button. IP Easy even manages IP addresses, multicast ranges, and VLANs, and includes security features like access control and quarantining of unknown devices to protect your network. HOME is natively built on a cloud-ready microservices architecture, enabling users to connect, manage and secure networked production setups from the ground up. Furthermore, it provides centralized access and control for all Lawo gear within a setup. HOME helps broadcast professionals address some of the most demanding requirements of modern IP infrastructures, including automated discovery and registration of devices, connection management, flow control, software & firmware management, scalability and security.

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

Telos Alliance® Axia iQS AES67 Mixing Console Software

Applications

• Console capability without the physical surface


• Distributed / remote workforce


• Temporary studio anywhere (nontraditional studio space)


• Studios lacking physical space


• Multiple users who need concurrent collaboration on a single mix

Telos Alliance® was there for you when you made the leap to IP, and now we’re here to help you on the journey to virtualization, wherever you are on that path. Our goal is to give you broadcasting options that are familiar while creating new ways of working that deliver on virtualization's promise of added scalability, adaptability, cost efficiency, simple deployment, and reliability. And that’s just what we’ve done with our new Axia iQs AES67 Mixing Console Software.

More than a decade ago, Telos Alliance created the Axia iQ AoIP console, now known for its ability to deliver a powerful and flexible mixing experience. Rather than pay for unwanted extras and faders, iQ allowed broadcasters to work smarter, offering the ‘just right’ functionality to mix content how they wanted, without sacrificing features.

Then we took the iQ philosophy and upgraded its underlying technology to create the Axia iQx AES67 console, the first AES67 AoIP console to combine the mix engine and console into one chassis. iQx lets you plug right into your AoIP network —with complete AES67 / SMPTE ST-2110-30 compliance— and enjoy the same capable, easy-to-use experience of the iQ.

Now, the Axia iQ family takes a bold step into the virtual AoIP future with iQs, the software version of iQx that does not require a physical surface. iQs is the first soft console controlled by a full HTML-5 interface, allowing you to not only control a mix from anywhere, but on any device—Mac, Windows, tablet, laptop, even your phone! It’s available in two ways to suit your needs and comfort level.

The Benefits of Virtualization

Non-Proprietary Hardware

You have more options when it comes to designing your system, including commercial off-the-shelf (COTS) options from the IT world.

Customizable

The virtual broadcast studio is customizable, because functionality is not hard-coded to a specific physical attribute on a piece of hardware, like a physical surface.

Scalable

You can run multiple instances of software concurrently, allowing you to scale up for demanding production requirements and scale back down accordingly, only paying for what you need.

Reliable

Virtual software and cloud computing are tested, tried, and true. Broadcasters need not worry about reliability compared with hardware options.

Cost-Efficient

Requires less maintenance, hardware, real estate, conditioned power, HVAC, and associated costs with flexible subscription models to meet OPEX business requirements.

Easier Upgrades

Easier to update software remotely over the Internet or en masse in a centralized data center, eliminating site visits.

2 Ways to Deploy iQs AES67 Virtual Console Mixing Software

iQs + AE-1000 for Easy Virtualization

Telos Alliance understands that virtualizing studio operations can be an overwhelming prospect. That’s why we are eliminating some of the legwork by pre-installing the iQs AES67 Mixing Console Software on a 1RU Telos Alliance AE-1000 Application Engine—our new line of universal servers—to help you ease into the virtual studio and prepare for the future. You can centralize the AE-1000 at the studio, yet give board operators the flexibility to control the iQs mix from a bedroom, a coffee shop, a makeshift studio...anywhere with an Internet connection.

iQs Container for the Data Center

iQs is available as a Docker container. Docker is a set of platform-as-a-service products that uses OS-level virtualization to deliver software in packages called containers. Containers are isolated from one another and bundle their own software, libraries and configuration files; they can communicate through well-defined channels. Docker is an integral part of the IT future.

Because iQs is available as a Docker container, it allows broadcasters to deploy it in a server farm or the cloud, delivering the added benefits of true virtualization, such as easier facility-wide upgrades, concurrent instances, and more, without site visits. Perfect for large installations of iQs console software instances.

iQs in a Docker container lets you realize your all-virtual future now because it’s available as a subscription-based model in addition to singular instances. A subscription allows you to be even more nimble—growing or shrinking the size of your system dynamically as your needs change. You only pay for the functionality you need, and you aren’t locked into any particular console configuration, large or small, for the ultimate in facility flexibility.

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 – radio »

ruby, from Lawo, is the mixing console that truly meets the demands of modern radio by blending the hands-on immediacy of physical faders with the context-sensitive efficiency of multi-touch controls. ruby frees your talent to create easily, naturally, effortlessly — in the way that suits them best. ruby is intuitive, uncluttered, powerful; but beneath its svelte surface lie color-programmable visual cues, context-sensitive displays and custom workflow options to help ensure that your hands always find the control you need, first time — every time. The mixing surface has a sleek, modern look that’s both functional and beautiful, designed for fast, accurate operation. From the high-resolution OLEDs that display sources and settings to the programmable function buttons and big, tactile ON/OFF and PFL keys with LED backlighting, ruby looks and feels great. Built in Germany to exacting Lawo standards, ruby features all-metal construction, premium 100 mm faders with built-in dust shield, ergonomic wrist-rest and ultra-reliable switches and rotary controls. There are smart tools like AutoMix, an intelligent algorithm that automatically maintains the balance of multi-mic productions so that creative staff and focus on what they do best — create. Desktop or flush-mount versions in 4, 8, 12 and 16 fader frame sizes, plus standalone 4 and 8 fader and Master Control extenders, allow combining multiple frames to build systems as large as 60 faders.

Today’s board operators interact as much (or more) with screens and displays as they do with the mixing console, so we’ve integrated mixing tools and other console controls into ruby’s multi-touch enabled display (powered by Lawo’s VisTool GUI-building software). No hunting for buried options: multi-touch on-screen controls instantly give what’s needed to control studio devices, apply dynamics, adjust virtual faders, and display true loudness metering. The optional VisTool Unlimited software upgrade even enables you to design context-sensitive screens customized to your station’s unique operating style, using your own custom-built graphics and control layouts.

ruby controls Lawo’s Power Core, perhaps the most powerful mixing engine ever built for radio, with redundant IP networking and redundant power capabilities. At just 1RU, Power Core‘s compact form belies the immense capabilities inside. On the front panel, a high-resolution color OLED gives status and setup information. I/O includes dual AES67 Ethernet ports with SFP, capable of 128 bi-directional AoIP streams, and 4 ports for high-density MADI signals (128 total channels of audio) — perfect for native MADI-to-AES67 AoIP conversion. It complies with the ST2110-30 standard too, ensuring seamless operation in combined radio / TV broadcast plants. ST2022-7 Seamless Protection Switching enables simultaneous dual-redundant network links, if you choose. Clock sync connections and 8 GPI / GPO contact closures complete the scene. Auto-switching dual-redundant power is standard, with Power Core’s internal auto-ranging AC power supply complemented by an inlet which accommodates an external 12VDC backup power supply.

There are also 8 rear-panel expansion slots, which accept a variety of optional IO expansion cards, including MADI, mic/line inputs, line inputs and outputs, AES3 inputs and outputs, and even a Dante interface. And Power Core’s raw audio processing horsepower puts other mix engines to shame, with the ability to handle thousands of simultaneous signals, a routing matrix of up to 1,920 x 1,920 crosspoints, and up to 96 channels of DSP input processing usable for EQ, de-essing, dynamics, etc. tucked into just 1RU. Multiple tiered license options make it easy to choose the combination of price and power that’s just right for you.

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.

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

Designed for flypack and OB audio applications, the S500m console delivers maximum control in a light weight and reduced form factor, whilst offering the same flagship performance and surface configurability as the S500.

The S500m can be specified as 32 and 48 fader versions with turnkey flight case solutions, or as a larger custom specification surface with up to 96 faders.

The complexities involved in today's major sports and entertainment events mean OB production still requires broadcast consoles with high specification. The S500m provides all the flagship features of the System T S500 console, but is specifically designed to be over 25% lighter to suit OB and flypack applications where weight and portability are key concerns. This allows a complete production system to be easily transported in a purpose built rack case, providing a console stand, meter bridge shipping storage and two 8RU space for Tempest engines and Network I/O.

Robust and optimised for transportation, the S500m's unique modular functionality also lets a console be deployed with or without an extended meter bridge. Meter bridge video can be displayed on console mounted screens or external multi-viewer

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

Type R is a new modular, expandable, IP-based radio system from Calrec Audio, which utilises standard networking technology and combines it with soft panels which can be tailored to operator needs.

Type R’s physical control system consists of just three slimline panels: a fader panel, a large soft panel and a small soft panel. Each is compatible with COTS hardware and powered over ethernet to keep cabling to a minimum.

Type R has a simple 2U core at its heart with integrated I/O resources to get you up and running immediately. A single core can power up to three independent mixing environments, with no sharing of DSP resources. Whether used as independent studio consoles, microphone processors or utility mixing, the ability to use multiple mixing engines combined with the flexibility of the AES67 compatible network provide all the power you will ever need.

The touch-screen soft panels are designed around simple and colourful control elements and can easily be customised as multi-function panels. Soft Panels can be utilised in either landscape or portrait formats, and used to provide adaptable and specific functionality for talent, while ensuring overall control is maintained by the station technical team.

This functionality can be quickly changed from show-to-show using simple memory loads, and can be tailored to fit the needs of the talent. Fader panels are small, sleek and simple, with six faders and immediate access to essential controls. Fader panels can be added or removed as simply as plugging or unplugging an ethernet cable, creating a radio infrastructure which is easily expandable, and making upgrades a breeze.

Broadcast-specific control is clear and concise across the system. Bussing, including the creation of mix-minus feeds, is quick to assign, while EQ and dynamics control is clear and fast. It is a pure radio platform designed for a fast-paced modern environment. And as you would expect from Calrec, Type R is a resilient console system designed for reliable professional use, with all the requisite power, function and scalability to keep you on air for many years to come.

Type R is a thoroughly modern and customer-focused radio broadcast system which adapts to a station’s needs as its requirements evolve, and provides simple customisation across established networks, open control protocols and surface personalisation.

With a native IP Backbone, Type R provides an infrastructure for future expansion. It guarantees stations are not only able to keep pace with changing demands, but provides the facility to ignite their audiences with new and innovative programming.

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

The GS-CU001B MkII is a highly versatile commentary unit. It was designed after close consultation with end users and the result is a stand alone unit which offers a wide range of facilities, and it is equally at home on the terraces, as it is in a studio.

The MkII VersionThe MkII version now adds the following as standard features:

- 48v phantom power

- upgraded input gain pot

- input gain range increased from -20 to +10 dB

- GPOs on Mic and talkback buttons to allow integration with main intercom or talkback system

VERSION 3 (TT): This version has transformer balanced inputs & outputs with high quality Llundhall transformers on the individual mic outputs.

InputsThree front panel inputs are mic/line switchable with selectable 48v phantom power. The front panel also has a small gain adjustment pot. Mic on/off switches can be selected in on/off mode or in cough mode. There is a global low frequency cut that can be selected on or off, and a preset compressor/limiter per input.

Outputs/TalkbackEach input has a direct discrete output on the rear panel along with a mixed output of all of the inputs, all on XLR. The mixed output is also available on an un-balanced 3.5mm jack socket as a local record. The direct output levels can be set in 3 positions:

1: 0dB + limiter: This is the normal operation and it limits peak levels.

2: 0dB: This feeds the output with an un-compressed nominal level of 0dB, for when the peak signal level will be controlled by outboard equipment.

3: -20dB: This feeds the output with a nominal level of -20dB providing extra headroom.

There are 3 common talkback circuits with individual buttons on the two main commentators’ sections (Position B's input cannot be switched to a talkback output). The operation of the talkback buttons features Config+ for configuring in different modes.

MonitoringThere are two main commentator monitoring sections, each with two separate 6.35mm A or B gauge jack sockets for headphones. In this way, the centre position B commentator can choose to share the monitoring of commentator A or C. There are 5 common external sources available for monitoring, plus an additional control which is the sum of the other 2 commentators/inputs. Each of the 6 pots has variable level control and left/both/right switching allowing commentator A and C to achieve their desired mix and balance levels. The sidetone control is located on the rear panel for commentator A and C to adjust the level of their own voice. There is a 7 segment LED PPM meter.

PowerThere is an internal switch mode power supply 100-250v AC, with external power via a 4 pin XLR 9-18v DC.

ModificationsThe GS-CU001 is a complete and versatile base system, and Glensound's most popular commentary unit. It is therefore a perfect starting point for custom requirements. This has resulted in many custom modified units, some listed above, and some not. We are always happy to investigate a particular requirement that you may have. To give you an idea of what is possible, these are some of the modifications, we have designed previously for others:

Added GPIO on a 9 pin D-typeMoved the sidetone control to the top panel on a full size potAdded a talkback channel and monitor between the A and C commentary positions (GS-CU001D and E)Independent monitoring inputs for commentator A and C (GS-CU001E)Additional mic passive outputs (GS-CU001E)Added an additional headphone monitoring input (GS-CU001G)Added two mic inputs for commentator A and C with a simple toggle switch between them (GS-CU001L)

Other Versions are available:

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

The Express Box MKII was designed to provide a quick and easy solution when it is necessary to provide interfacing for two commentators, with simple facilities. This high quality commentary unit is ideal for those looking for cost effective solutions without paying for unwanted features; the Express Box MKII is worthy of investigation.

InputsTwo front panel mic inputs with selectable 48v phantom power.

Outputs/TalkbackEach mic input has its own individual output. There is also a mixed output of the two mic inputs. The two talkback circuits have individual outputs and are common for both commentators.

MonitoringThere are four inputs for external sources, one sidetone pot of their own voice and one other commentator pot for listening to the co-commentator. These are available independently to each commentator on individual pots, so each can adjust the inputs for their own preference of mix level. There are two 6.35mm headphone sockets – 1 for each commentator. A 7 segment LED PPM meter displays level.

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

The VITA PLUS is a very powerful Dante network audio interface, in a traditional desktop style. It has a mic/line input and headphone monitoring connection for a single user, and contains two 4 wire circuits. One of these is typically used as an on air channel, and the other as a talkback channel.

The high quality microphone input, with the Glensound Referee compressor, makes the Vita Plus suitable for on air commentary use, perfect for a small OB or in an off-tube booth .

The flexibility of the talk button configurations, means that the Vita Plus can also be used as a talkback unit as part of a Dante network.

The Vita Plus can connect to any Dante network, or to other audio equipment via Glensound’s own Dante interface units.

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

The VITA BB PLUS is a very powerful Dante network audio interface, in a traditional desktop style.  It has a mic/line input and headphone monitoring connection for a single user, and contains two 4 wire circuits.  One of these is typically used as an on air channel, and the other as a talkback channel.

The high quality microphone input, with the Glensound Referee compressor,  makes the Vita BB Plus suitable for on air commentary use,  perfect for a small OB or in an off-tube booth.

The flexibility of the talk button configurations, means that the Vita BB Plus can also be used as a talkback unit as part of a Dante / AES67 network.

The Vita BB Plus can connect to any Dante or AES67 network, or to other audio equipment via Glensound’s own Dante interface units.

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

Paradiso Lite is a commentators' box for 3 commentators. It can be used by itself to provide the commentary facilities alongside an OB truck for small events or multiple Paradiso Lites can be used at large events that require Worldwide broadcast coverage.

Audio links between the Paradiso and other equipment are by Dante (AES67 compliant) complete with redundant copper links. Analogue I/O and AES3 I/O are also provided directly from the unit for local connections and / or another layer of redundancy.

The Paradiso Lite has been designed to be intuitive & easy to use for Commentators who would rather be talking about the game than working out how the equipment works, and is also built to our exacting rugged & robust standard to make it a reliable piece of broadcast equipment for the busy engineer.

Fully Redundant Copper Network ConnectionsIf an IP link fails you have no broadcast! Our system provides fully redundant glitch free audio & data transport across any of the 2 copper network interfaces.

Automatic Adjustment For High or Low Impedance Headphones Our unique headphone amplifiers automatically provide the correct drive level to high or low impedance headphones.

Instinctive Mic Gain Setting InformationWhen adjusting a mic's input gain, the PPM automatically changes to display just the level of mic input currently being altered.

Multiple Power SourcesFor complete fail-safe redundancy the Paradiso can be powered from any of the following sources: PoE (2 off), Wide Range Mains or external DC.

Local Analogue & AES3 I/OFor 100% reliability local analogue & AES3 I/O is provided as complementary audio connectivity to the network audio.

Referee Compressor/ LimiterGlensound’s World renowned Referee compressor limiter system keeps even the loudest commentators sounding great.

Fully ConfigurableGlenControler remote control/ setup application allows hundreds of different parameters of the Paradiso to configured to meet the users requirements.

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

Paradiso is a commentators box for 3 commentators. It can be used by itself to provide the commentary facilities alongside an OB truck for small events or multiple Paradiso's can be used at large events that require Worldwide broadcast coverage.

Audio links between the Paradiso and other equipment are by Dante (AES67 compliant) complete with redundant copper & fibre links. Analogue I/O and AES3 I/O are also provided directly from the unit for local connections and/ or another layer of redundancy.

The Paradiso has been designed to be intuitive & easy to use for Commentators’ who would rather be talking about the game than working out how the equipment works, and is also built to our exacting rugged & robust standard to make it a reliable piece of broadcast equipment for the busy engineer.ePaper DisplaysFor quick easy intuitive visible Indication of individual headphone source names, levels & pan. Viewable in bright sunlight.

Fully Redundant Copper and Fibre Network ConnectionsIf an IP link fails you have no broadcast! Our system provides fully redundant glitch free audio & data transport across any of the 4 network interfaces.

Automatic Adjustment For High or Low Impedance HeadphonesOur unique headphone amplifiers automatically provide the correct drive level to high or low impedance headphones.

Instinctive Mic Gain Setting InformationWhen adjusting a mic’s input gain, the ePaper display changes to display the gain being applied in dB and the PPM automatically changes to display just the level of mic input currently being altered.

Multiple Power SourcesFor complete failsafe redundancy the Paradiso can be powered from any of the following sources: PoE (2 off), Wide Range Mains or external DC.

Local Analogue & AES3 I/OFor 100% reliability local analogue & AES3 I/O is provided as complimentary audio connectivity to the network audio.

Referee Compressor/ LimiterGlensound’s World renowned Referee compressor limiter system keeps even the loudest commentators sounding great.

Fully ConfigurableGlenController remote control/ setup application allows hundreds of different parameters of the Paradiso to configured to meet the users requirements. 

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

MinFerno is a new, smaller, and even easier to use version of our exceptionally popular Inferno network audio commentary box.

MinFerno is designed for a single commentator/ announcer to use, and like its big brother the Inferno, it provides the very best possible commentary microphone amplifier and compressor limiter circuit for amazing on air sound.

The MinFerno is easy to use for commentators who would rather be talking about the game than figuring out how the equipment works.  It is built to our exacting, rugged & robust standard to make it a reliable piece of broadcast equipment for the busy engineer.

Redundant Powering OptionsThe MinFerno can be powered from any of 3 different sources:1) PoE on the Primary CAT5 Network link2) PoE on the Secondary CAT5 Network link3) External 12V DCFour of rear panel LEDs show the availability of the 3 power sources.

Primary & Secondary SFP SlotsThe SFP (Small Form-Factor Pluggable) fibre slots are standard networking ports that accept standard SFP modules. This means that you decide what type of fibre and connector style you want to use just by the SFP module that you insert. The primary & secondary network circuits allow for glitch free redundancy across both the Fibre & Copper network interfaces.

Primary & Secondary Copper Network Connections with PoETwo CAT6 connections on Neutrik Ethercons (that accept standard networking cables) are provided to allow copper connections to local network switches to carry the Dante/ AES67 audio. Two connections are fitted to allow redundant circuits to be used if required.Both of these connectors can accept a PoE power source for providing the power to the MinFerno.

Input Mode & GainOne pair of push buttons selects the input type of the front panel XLR to be either microphone, line or microphone with 48V phantom power. 3 LEDs indicate which input mode is selected.Two push buttons are used to alter the gain of the input. LEDs indicate if the gain setting is above or below our pre-configured ‘lineup’ levels. The front panel PPM of course provides an accurate indication of the input level.The gain can also be altered remotely by a web browser pointing at the MinFerno's web page.

Up to 4 of Incoming Audio Volume ControlsOn the top panel are up to 5 rotary headphone volume controls. 4 of these are connected to 4 incoming audio circuits from the Dante/ AES67 network. These are normally used for such sources as mixed programme or cue, talkback to director, talkback to producer, talkback to engineer etc.

Sidetone Volume ControlThe 5th front panel rotary headphone volume control is ‘sidetone’. Sidetone is the commentator's own voice in their own ears.

Headphone RoutingEach of the 5 headphone volume controls has an associated left ear, right ear, both routing switch located next to the volume knob. This single push button switch routes the associated source to just the left, just the right or both channels of the stereo headphone amplifier. To enable the commentator to know how they are routing a circuit the first time a routing switch is pressed, a pair of LEDs on the front panel indicate its current routing arrangement. The next time the routing switch is pressed then the next routing option is selected.

Robust Mic On and Talkback ButtonsThere are up to 4 of large bright illuminated buttons (1 for programme and up to 3 for talkback). These switches route the outgoing microphone circuit onto up to 4 different Dante/ AES67 network audio circuits. The operation of these switches (momentary, latching etc) and the interaction of these switches (i.e. pressing a talkback switch mutes the main mic) can be fully configured via the web page.

15 segment LED PPM MeterAudio input level is indicated on the 15 segment PPM meter.

Low Noise Microphone Amplifier With Remote GainWe spent a long time optimising the performance of the THAT corporation microphone amplifier used in the original Inferno and used again on the MinFerno. It features very low noise & distortion circuit that we remote control the gain of in 1dB steps, which allows us to provide the remote webpage gain control as well as the rear panel gain push buttons. We also optimised the circuits to allow correct source impedance switching depending on whether the input has been set as a mic or line input.

Referee Compressor LimiterAs with all our commentary units, the MinFerno features our very popular Referee compressor limiter circuit. This circuit starts to compress the commentator’s voice gradually and slowly increases the compression ratio as the input level becomes overly high, resulting in a very natural sounding and distortion free audio output capable of taming even the loudest of commentators.

High Quality Analogue To Digital Converter (ADC)Dante/ AES67 network audio is a digital circuit and as such the best analogue microphone amplifier would be wasted if we hadn’t paired it up with the best analogue to digital converter. The ADCs job is fairly simple, and if you look at our tech spec you’ll see that we’ve made ours work incredibly well.

Unique Headphone AmplifiersThe commentator’s headphones are a vitally important tool so we take as much care with our headphone amplifiers as with our on air microphones. Our unique headphone amplifier provides the correct output level regardless of the impedance of the attached headphones, meaning that broadcasters can now pick and choose between low impedance ‘cheap’ headphones and high impedance traditional broadcast ones.

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

Links To Dante Compatible Network Or DARK88 Break Out BoxInferno can be used on its own when connected to a Dante audio network. Many audio routers, consoles and intercom systems are Dante compatible, and will accept the Inferno link, enabling Glensound grade commentary into the audio network.

There are primary and secondary Dante network connections

Inferno can also be used outside of a dedicated Dante network when connected to one of our DARK or AoIP44 break out boxes. This connection can be direct or across a structured network. The DARK or AoIP44 units allow all of the audio inputs and outputs to be located in a place separate from the Inferno.

Programmable On A Dante Audio NetworkWhen connected to a Dante network, sources and destinations can be configured from any part of the network using the free Dante controller software. Details are available at www.audinate.com.

Seven Talkback CircuitsThe seven illuminated talkback buttons route the audio input to 7 different destinations and remove the audio from the main programme output.

Eight Input Headphone MixerSeven external sources plus sidetone are presented on eight level controls allowing the user to create their own required monitoring mix of all sources. Each input can be independently selected to be on the user's left ear, right ear or both ears. The headphone connection is available on a 3.5mm and a 6.35mm jack socket.

Single InputThe single input is switchable on the rear panel to be line/mic/48v and has adjustable input gain. The input also features Glensound’s Referee compressor limiter system. This applies a variable rate compressor to the input peaks so that no clipping occurs, without affecting the rest of the audio signal.

Remote ControlThe Inferno contains a web server, allowing remote functionality to an engineer from any PC connected to the network via any web browser.  Enter the IP address of the Inferno in the browser for remote control of the front end mic gain.  It also allows configuration of each push button on the unit to be on/off latching, momentary (push to talk), always on, and off when pushed (cough), or in an intelligent mode where a short press latches or a longer press is just momentary.

15 segment LED PPM Meter Audio input level is indicated on the 15 segment PPM meter.

Power On Dante Link Or LocallyInferno can be powered via the network audio connection if it conforms to the PoE standard. There is also an internal 100v-240v mains supply.

Scalable Commentary~Start with one Inferno into a Dante network~Add Multiple Inferno at any time for 2, 3, 4 ,5 or more commentator system~Use DARK units for a simple point to point networked audio break out box

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

The Express ip Mini is a compact simple to operate, 2 user network audio commentary box, that is ideal for both the occasional & the daily provider of commentary facilities.

Its low cost keeps the accountant happy, its simple facilities makes it very easy to setup for the engineer and the familiar Glensound control surface keeps the commentators satisfied.

Having been involved in designing and manufacturing commentary boxes for over 40 years the Express ip Mini is a culmination of all our years of experience and has been designed to provide just enough of the basic facilities required for everyday commentary in a very robust & easy to use package.

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

The Express ip MKII was designed to provide a quick and easy solution when it is necessary to provide interfacing for two commentators, with simple facilities, into a Dante/ AES67 audio network. This high quality commentary unit is ideal for those looking for cost effective solutions without paying for unwanted features; the Express ip MKII is worthy of investigation.

InputsTwo front panel mic inputs with selectable 48v phantom power.

Outputs/TalkbackEach mic input has its own individual output. The second mic output can be switched on the rear panel to be a mix of both mic inputs.  The two talkback circuits have individual outputs and are common for both commentators.

MonitoringThere are four inputs for external sources, and one sidetone pot of their own voice and another pot for monitoring the other commentator. These are available independently to each commentator on individual pots, so each can adjust the inputs for their own preference of mix level. There are two 6.35mm headphone sockets – 1 for each commentator. A 7 segment LED PPM meter displays level.

NetworkThe network connection is AES67/Dante compatible and is available on the rear panel on an RJ45/CAT5 connection.  It offers 4 input channels and 4 output channels.

PowerThere is an internal switch mode AC power supply, or the Express ip can be powered by PoE if powered by a switch that provides power over Ethernet.

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

The Glensound AoIP4O is packaged in a small & rugged format and designed for outside broadcast, theatre and location applications.

The AoIP4O unit is designed to easily and quickly interface existing analogue equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.

AoIP4O provides 4 balanced analogue audio Outputs from a Dante®/AES67 audio network.

Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.

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

The MAGIC THipPro is the most flexible and functional talkshow system in our portfolio. The 8 lines system is available for POTS, ISDN or VoIP lines, the 16 lines system is available for ISDN or VoIP lines.

A special feature of the MAGIC THipPro is the so-called Mixed Mode in which the system can be operated with POTS or ISDN lines and simultaneously with VoIP lines. For the Mixed Mode, you simply need to upgrade a POTS or ISDN system with VoIP.
MAGIC THipPro provides two analogue and eight digital Audio inputs and outputs plus two handset/headset interfaces. The Audio interfaces can be assigned freely to the installed workplaces and studios. In total, up to 20 workplaces can access the MAGIC THipPro simultaneously. With the Admin Upgrade, six different studios can be configured.

For MAGIC THipPro, two main software versions are available for configuration and control: MAGIC THipPro LAN and MAGIC THipPro Screener. In the standard delivery no PC licence is included so that the user can decide which one is more suitable for his purposes or chose a mixture of both. Additionally, for smaller studios or recording boothes which do not need the ON AIR functionality, the News Desk Client Software could be a cost-effective solution.

MAGIC THipPro can be connected to an MS SQL database Server. Different studios can use the same or individual databases for call screening. Of course, a blacklist function is also available. All features such as e.g. Auto Answer, call forwarding, Voice Disguise, Night Service and caller preselection are supported. As a special feature the answering machine functionality has been implemented – callers can be accepted automatically in Hold, hear a pre-recorded message and are dropped afterwards.
For an easy communication with DHD mixing consoles, MAGIC THipPro allows the configuration of up to 96 DHD SetLogic commands. Via the Ember+ protocol 96 inputs and 96 outputs can be programmed, an easy communication between MAGIC THipPro and e.g. DHD or Lawo mixing consoles is possible.

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Lumo aims at bringing radio closer to its audience.

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

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

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

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

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

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

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

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

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

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

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

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

Powered by the 1 rack-unit high JET67 Audio Engine, Logitek's mixIT-6 delivers these features for only $4800 (USD):

• Auto-configuring mix minus buses available at every fader to provide clean return feeds to telephone hybrids and audio codecs


• Slim tabletop design


• Three utility routers to change feeds to air chains, codecs, or recorders


• EQ and dynamics at every fader


• Any source routes to any fader via simple touchscreen controls


• 2 microphone inputs with phantom power


• 3 stereo analog inputs


• 2 digital inputs


• 4 stereo analog outputs


• 2 digital outputs


• Quick-connect terminal blocks for I/O

The mixIT is a fully networked audio console, featuring AES-67, Ravenna, and Livewire stream discovery via a built-in 4 port Ethernet switch. (2 x 10/100 & 2 x 1 GB connections)

AES-67, Ravenna, and Livewire are available now; an optional Dante module is coming this summer.

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

Powered by the 1 rack-unit high JET67 Audio Engine, Logitek's mixIT 12 delivers the following for only USD $5800:

• Auto-configuring mix minus buses available at every fader to provide clean return feeds to telephone hybrids and audio codecs


• Quick console scene change buttons for different shows


• Three utility routers to change feeds to air chains, codecs, or recorders


• EQ and dynamics at every fader


• Any source routes to any fader via simple touchscreen controls


• 4 microphone inputs with phantom power


• 6 stereo analog inputs


• 2 digital inputs


• 8 stereo analog outputs


• 2 digital outputs


• Quick-connect terminal blocks for I/O

The mixIT is a fully networked audio console, featuring AES-67, Ravenna, and Livewire stream discovery via a built-in 4 port Ethernet switch. (2 x 10/100 & 2 x 1 GB connections)

AES-67, Ravenna, and Livewire are available now; an optional Dante module is coming soon.