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.
Other products from this company:
MAGIC ACX Dante™ WAN Bridge
MAGIC AE1 DAB+ Go Audio Encoder
MAGIC DABMUX Go RF Ensemble Multiplexer
MAGIC DABMUX plus Ensemble Multiplexer
MAGIC THipPro Telephone Hybrid
MAGIC ACip3 Audio Codec
The system supports the G.711, G.722, ISO/MPEG Layer 2, Opus coding algorithms and PCM 16/20/24 Bit in the standard delivery version. Optionally, the Audio Codecs can be upgraded with Enhanced apt-X 16/24 Bit, AAC-LD/AAC-ELD and AAC-LC+V1/V2.
One stereo programme is encoded in the standard version, optionally the system can be upgraded to a second stereo programme by software activation.
Two operating modes are available for the pure IP version: the system can be used for dial-up AoIP connections or IP Leased Line connections. In AoIP mode, the system can register at 5 different SIP servers and automatically accept incoming calls from this SIP server. Audio connections in IP Leased Line and in AoIP dial-up Mode can be established with the Secure Streaming functionality for a highly reliable transmission.
Via the Ember+ protocol 64 inputs and 64 outputs can be programmed, an easy communication between MAGIC ACip3 and e.g. DHD or Lawo mixing consoles is possible.
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MAGIC ACX Dante™ WAN Bridge
The audio connection for the audio inputs/outputs is provided by the integrated 32-channel Dante™ interface, which has redundant GbE interfaces and of course also supports AES67.
The comfortable operating software allows the management of up to 10 systems including graphic surveillance and can be operated from up to 5 workplaces simultaneously.
The front display of the system also shows essential information on the status of the transmission.
The coding is done via PCM16 / PCM24 / PCM32.
Since the limited maximum latency of Dante in wide area networks, high jitter and possible clock differences between sender and receiver prevent a reliable transmission, the MAGIC ACX Dante™ Wan Bridge solves these problems with a jitter buffer including automatic mode and intelligent sample rate adaption (SRA).
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MAGIC AE1 DAB+ Go Audio Encoder
The supplied Windows software allows the configuration of up to 40 systems.
Both analog and digital audio interfaces are available for the input of the audio signal.
In addition to the usual PAD feed via the multiplexer - which supports all standardized services - the PAD services Dynamic Label via FTP or UECP and Slideshow via FTP can also be fed in locally on the device.
In addition, a traffic announcement (TA) can be triggered very easily via UECP or a GPI contact.
Finally, the Audio Encoder allows direct transmission of the program type (PTy) like Rock, Pop etc. via UECP.
The transmission is monitored via the supplied Windows PC software or SNMP. Alternatively, an alarm can also be output via a GPO contact.
Up to three IP addresses can be assigned to the integrated network interface, so that a network separation for different applications is possible. The system also supports VLANs.
Communication between multiplexer and encoder is carried out via the AVTMUX protocol, which enables the control, monitoring and PAD transmission of the encoders from the multiplexer and guarantees secure transmission via Secure Streaming.
With this method, which has also proved its worth in the field of classical audio transmission, all IP packets are transmitted twice - with a delay. Due to the low bit rates with DAB+, the necessary doubling of the data rate is negligible. In addition, different routes can be implemented in the transmission path by using suitable addressing.
At the multiplexer (MAGIC DABMUX Go and MAGIC DABMUX Plus) all IP packets are reassembled correctly in time and duplicate received packets are rejected.
The system can also be used directly with the Open Source Multiplexer ODR DabMux. The necessary interface adaptations are available free of charge on the GitHub platform.
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MAGIC DABMUX Go RF Ensemble Multiplexer
Up to 20 program providers can be connected via external Audio Encoders. An installation of the Encoders directly in the studio avoids effectively an interference in Audio quality because of Codec cascading.
Special value was set on the easy configuration of the Ensemble Multiplexer via web browser, so that even users without DAB expert knowledge are able to set up the system.
The highly compact DAB Ensemble Multiplexers facilitate a very simple Multiplex generation in accordance with standard ETSI EN 300 401. Despite its size, all features such as re-configuration (manually and scheduled), extraction of Sub Channels of other Multiplexers, integration of PAD and NPAD data services, creation of Service Information etc. are integrated.
Audio Services can be supplied via the AVTMUX or the EDI(ETI) protocol from external Multiplexers.
As output signal both Multiplexer variants supply an EDI signal for transmission to the transmitters.
With the RF version, which has an integrated modulator, you can alternatively activate a power amplifier directly. This possibility is particularly of interest if you have only one transmitter site.
The synchronisation is effected via NTP or in case of the RF version via the integrated GPS receiver.
The RF input is intended for future applications.
The configuration, operation and monitoring are effected via a HTML5-compatible web browser.
An external alarm can also be triggered via SNMP.
The system has a GBit Ethernet network interface, which allows the configuration of up to three IP addresses as well as VLANs.
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MAGIC DABMUX plus Ensemble Multiplexer
The device is realized as a 19“ x 1U system with integrated power supply and a redundant 12V power supply. The reliable signal processor-based system has three Gbit Ethernet interfaces which allow the configuration of up to three IPaddresses per interface as well as VLANs. The system also provides 2 x USB 2.0 interfaces and an SD card slot for further applications. Via a module slot, the system can optionally be expanded with an ETI E1/2-Mbit or a dual Ethernet module.
The system has a graphical, coloured display, but a more comfortable control, monitoring and very simple configuration is possible via a HTML5-compatible web browser.
Up to 25 program providers can be connected via external Audio Encoders. An installation of the Encoders directly in the studio avoids effectively an interference in Audio quality because of Codec cascading.
A highlight of the MAGIC DABMUX plus is its automated use in redundant headends. In addition, the system offers dynamic reconfigurations (manual or scheduled), an integrated PAD/NPAD inserter, subchannel extraction via EDI with simplified selection through analysis of the incoming signal, announcement support, service linking support, simplified provision of the program logo via SPI (EPG), audio encoder and multiplexer redundancy, etc. In general, all announcements (except OE announcements) are supported according to ETSI TS 101 756.
As a special feature, the Emergency Warning Break-In upgrade of DAB is optionally supported, which in addition to emergency signalling also enables the replacement of all audio content by an emergency announcement. This ensures that the announcement is audible even with older receivers.
';
MAGIC THipPro Telephone Hybrid
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.
';
Related Products:
Axia Quasar™ SR AoIP Mixing Console
Axia Quasar™ XR Broadcast Console
mc²36 Audio Production Console
Telos Alliance® Axia iQs AES67 Mixing...
RƎLAY Virtual Radio Software
ruby Radio Mixing Console
Power Coreᴿᴾ IP Audio I/O &...
MAGIC THipPro Telephone Hybrid
Artisto
Logitek mixIT-6 Radio Console
Logitek mixIT-12 Radio Console
Axia Quasar™ SR AoIP Mixing Console
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.
Axia Quasar™ XR Broadcast Console
mc²36 Audio Production Console
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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Telos Alliance® Axia iQs AES67 Mixing Console Software
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.
RƎLAY Virtual Radio Software
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.
ruby Radio Mixing Console
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.
Power Coreᴿᴾ IP Audio I/O & DSP Node for Remote Production
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.
System T S500m
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
Type R
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.
GS-CU001B
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:
Express Box MKII
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.
Vita Plus
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.
Vita BB Plus
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.
Paradiso Lite
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.
Paradiso
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.
MinFerno
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.
Inferno
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
Express ip MINI
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.
Express ip MKII
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.
AoIP4O
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.
MAGIC THipPro Telephone Hybrid
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.
Lumo
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.
Artisto
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.
Logitek mixIT-6 Radio Console
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.
