| Audio signal processing, routing and infrastructure |
IQOYA X/LINK
IQOYA X/LINK
The reference audio codec
IQOYA X/LINK is a 1U rack IP audio codec designed for the delivery of a stereo source (or two mono sources) over IP networks for STL and SSL links, but also DVB audio and WEB radio. It can be used in legacy analog or AES/EBU audio infrastructures, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a good investment for the migration to IP audio. Like all the IQOYA products, X/LINK is based on the Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable IP streaming over inexpensive IP links. Based on a low consumption and fanless powerful hardware platform, IQOYA X/LINK is designed for 24/7/365 use.
Key points
Adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure
Simultaneous delivery of your audio program to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios, in multiple audio formats
Designed for audio service continuity and failsafe operation
Easy integration into existing SNMP based supervisors (SET, GET, Traps)
Ongoing product support with flexible options
IQOYA X/LINK-AES67
IQOYA X/LINK-AES67
Designed for FULL IP environments
IQOYA X/LINK-AES67 is a 1U rack IP audio codec designed for the delivery of stereo and/or mono audio sources over IP networks, for STL, SSL, DVB audio, and WEB radio applications. It is dedicated for full-IP audio infrastructures based on AES67, Ravenna, or Livewire technologies. Like all the IQOYA products, X/LINK-AES67 is based on the Fluid IP technology which offers redundant dual streaming, allowing reliable IP streaming over inexpensive IP links. Based on a low consumption, fanless and powerful hardware platform, IQOYA X/LINK-AES67 is designed for 24/7/365 use.
Key points
Designed for full IP environments
Supports AES67, RAVENNA, and Livewire technologies
Scalable number of supported IP audio input and ouput channels (from 1 to 8 stereo I/Os)
Simultaneous delivery of input audio programs to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios, in multiple audio formats
Ongoing product support with flexible options
IQOYA *SERV/LINK
IQOYA *SERV/LINK
Outstanding possibilities in only 1U rack
IQOYA *SERV/LINK is a extra high density 1U rack multi-channel IP audio codec designed for the delivery of large number of audio programs to transmitter sites (STL), studios (SSL), DVB multiplexers, and WEB radios CDNs. It also serves for the transport of multiple intercom and commentary channels over IP networks. It supports from 4 to 64 stereo input and output channels, with the possibility to simultaneously encode and stream the input channel at multiple formats and protocols, decode IP audio streams to the outputs, and transcode IP audio streams. It features two hot swappable redundant PSU, two Ethernet ports, and supports different types of audio I/Os (AES/EBUs, or analog, or MADI, or AES67/RAVENNA, or DANTE , or AES67/RAVENNA and MADI). IQOYA *SERV/LINK can be fully controlled and monitored from its embedded WEB pages and through SNMP.
Key points
Maximizes rack space: only 1U to replace up to 64 DANTE stereo channels, up to 16 stereo AES/EBU channels, or 8 analog stereo channels, or 64 MADI stereo channels*, or 64 AES67 stereo channels*, or 64 AES67 and MADI stereo channels*
One single unit simultaneously delivers multiple audio content to transmitter sites, WEB radio CDNs, multiplexers, and other studios, in multiple audio formats
WEB and SNMP control, monitoring and alarming
Lumo
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
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.
IQOYA X/LINK-DUAL
IQOYA X/LINK-DUAL
2 stereo codecs in 1U
IQOYA X/LINK-DUAL is a 1U rack IP audio codec designed for the delivery of two stereo sources (or four mono sources) over IP networks for STL and SSL, DVB audio, WEB radio, intercom and commentary. It can be used in legacy analog and AES/EBU audio infrastructures, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a perfect investment for the migration to in-house IP audio. Like all the IQOYA products, X/LINK-DUAL is based on Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable connections over inexpensive IP links. Based on a low consumption, fanless and powerful hardware platform, IQOYA X/LINK-DUAL is designed for 24/7/365 use.
Key points
Space efficient: two stereo codecs in a 1U rack, with simultaneous delivery to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios
Adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure
Easy integration into existing SNMP based supervisors (SET, GET, Traps)
Ongoing product support with flexible options
IP-4c
IP-4c
High compatibility: The IP-4c supports a wide range of protocols for streaming, control and monitoring (e.g. EBU TECH 3326, AES67, Ravenna, Livewire+, Dante, SMPTE ST 2110, PTPv2, RTSP, SAP, SIP, Discovery, Bonjour, SNMP, HTTP, HTTPS, FTP, FTPS or Ember+ and more). Furthermore, the exchange of additional information like GPIO and ancillary data between the audio networks is possible.
Pay as you grow: All soft- and hardware components are individually combinable. The scalability from one to four audio channels using software licenses gives you flexibility in planning your network and reducing your costs.
Multi-format audio coding: Another advantage is the variety of possible algorithms like MPEG1 Layer 2, MPEG2 Layer 3, most AAC profiles including the new xHE-AAC and AAC-ELDv2, OPUS, Ogg Vorbis, PCM, Enhanced aptX, Dolby Digital plus (on request) and more.
MoIN
MoIN
FLEXIBLE IN APPLICATION: designed for studio to studio and studio to transmitter links, as well as cross-media tasks. Multipurpose usage – e.g. audio routing, managing, levelling, loudness, monitoring and mixing between different protocols and environments. Audio streams are combinable to multichannel streams. PTPv2 provides accurate synchronization. The Easy2connect feature is a SIP phonebook that allows for an uncomplicated connection set-up. The number of channels can scale easiliy depending on your needs with permanent or temporary licenses.
HIGH COMPATIBILITY: MoIN supports all currently established standards and protocols like AES67, RAVENNA, Livewire+, Dante, SMPTE ST 2110, SRT, Ember+, SIP, SAP, PTPv2 and RTSP, as well as transcoding from one format into the other.
MULTI-FORMAT AUDIO CODING: The device provides a wide range of codec algorithms – e.g. MPEG Layer 2, MPEG Layer 3, most AAC profiles including the new xHEAAC and AAC-ELDv2, OPUS, Vorbis, Enhanced aptX, AC3 and more.
TRANSMISSION ROBUSTNESS: Dual Streaming and Pro-MPEG FEC ensure rock-solid IP transmission or go beyond with Stream4Sure. The Reliable User Datagram Protocol (RUDP) ensures highest packet recovery with minimum bandwidth and low latency.
SMART MANAGEMENT: configuration set-up via an easy to use web interface for general settings as well as for backup or fallback options. Remote control via HTTP/HTTPS, FTP, Telnet/SSH, NMS, SNMP, JSON, Ember+. Configuration for transcoding in 15 seconds. PTPv2 synchronization and latency control. Control via centralized Network Management System offering individual settings for e.g. adjustable silence detection, IP buffer and jitter check and PLL control. Sophisticated and elaborated alarm concept: forwarding of alarm messages (e.g. via SNMP, Ember+, HTTP/HTTPS…), optional source switching and event logging for documentation.
MAGIC ACX Dante™ WAN Bridge
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).
greenMachine HDR Evie+
greenMachine HDR Evie+
The greenMachine HDR Evie+ (Enhanced Video Image Engine), 1 RU half 19” rackmount, is a real-time segmented frame-by-frame broadcast-quality High Dynamic Range (HDR) to Standard Dynamic Range (SDR) converter, with frame sync supporting formats up to 4K UHD (3840x2160). It is the world’s first system that uses the advanced algorithm for sectional dynamic tone mapping, which automatically analyzes different sections of an image in HDR stream and applies optimal corrections on a frame by frame basis in real-time. This unique capability is unlike any other solution today. It is the perfect real-time production tool for sports or any live broadcast event needing high-quality real-time HDR to SDR conversions. HDR EVIE+ fits best in the single native HDR workflow reducing the cost of equipment and manual operations.
The sectional dynamic tone mapping technique is applied to several spatial areas (sectors) in the image. In this type of tone mapping technique, the algorithm adjusts the signal by using several dynamic curves, one per image segment/sector. The algorithm then combines the result with global dynamic curves based on the entire image. While the dynamic curves of the different sectors are perfectly adapted to the respective image content of their segments, the global curve is perfectly adapted to the content of the entire image. Depending on the image content within these sectors, the algorithm reacts dynamically, which allows lights and shadows to be treated independently. Therefore, dark areas can be brightened, and the bright areas can be darkened without getting a flat gradation of the image. “
HDR EVIE+ provides a 1x 4K/UHD processing channel supporting down-conversion from HDR transfer characteristics to SDR through appropriate sectional dynamic tone mapping. It also supports the Wide Color Gamut (WCG) needs of broadcasters and professional AV live events requirements. HDR Evie+ package also includes HDR Static configuration for Static HDR SDR conversions, which performs static tone mapping to realize UP/Down/Cross conversions between HDR and SDR, suited best for the studios or the environments where the light conditions do not change dynamically.
ORBAN OPTIMOD 8700i LT
ORBAN OPTIMOD 8700i LT
The difference to the OPTIMOD 8700i is that the light version has no Dante interface for AoIP, no streaming monitor output and does not come with the Xponential Loudness algorithm.
However, the OPTIMOD 8700i LT Audio Processor includes features such as the Multipath Mitigator phase corrector which reduces multipath distortion without compromising the stereo separation and the Subharmonic Synthesizer which allows you to add modern-sounding bass punch to older recordings. It also includes Orban’s MX limiter technology which lowers distortion, improves transient punch, and minimizes preemphasis-induced high frequency loss. Further outstanding features are monitored and alarmed dual redundant power supplies as well as safety bypass relays for carefree 24/7 operation. Of course, a digital MPX output is also available as well as a 10 MHz clock input.
ORBAN OPTIMOD 6300
ORBAN OPTIMOD 6300
The 6300 features two processing structures: Five-band for a spectrally consistent sound with good loudness control, and Two-band for a transparent sound that preserves the frequency balance of the original program material while also effectively controlling subjective loudness. There are over 60 Factory Presets wich are our "factory recommended settings" for various program formats or types. There are multiple Factory Presets for both radio-oriented and video oriented programming.
Orban's new PreCode™ technology manipulates several aspects of the audio to minimize artifacts caused by low bitrate codecs, ensuring consistent loudness and texture from one source to the next. There are several factory presets tuned specifically for low bitrate codecs.
The OPTIMOD 6300 includes third-generation CBS Loudness Controllers™ for DTV applications. Loudness controllers work with the both Two-Band and Five-Band structures. Material processed by the CBS Loudness Controller has been shown to be well controlled when measured with a long-term loudness meter using the ITU-R BS.1770-2 standard. The 6300 also includes a "BS.1770 Safety Limiter" that follows the CBS Loudness Controller; use the BS.1770 if the BS.1770-2 meter reading must be constrained to a preset value.
