| Audio signal processing, routing and infrastructure |
I/O & Interfaces
I/O & Interfaces
HC Bridge SRC
INTRAPLEX® ASCENT
The scalability of Ascent makes it ideal for applications that require multiple channels of audio encoding and decoding at head-end sites. The high-density solution reduces cost and provides a path for convergence of IT and broadcast infrastructure. The interoperability between Ascent and IP Link codecs provides a complementary solution for remote contribution and distribution use cases.
Product Details
The Ascent platform continues the Intraplex tradition of providing an unprecedented level of reliability for audio over IP application. Built with enhanced network reliability and security in mind, Ascent supports the Dynamic Stream Splicing technology of IP Link codec, which provides “hitless” protection for packet losses using the combination of time and network diversity for packet transmission, and FEC.
To further enhance reliability and security, the platform supports the SRT protocol encapsulation. SRT (Secure Reliable Transport) is an open-source protocol that provides low-latency, reliable and secure streaming of audio and video data. The reliability in SRT is accomplished by a built-in packet re-transmission scheme; the security of streams is handled by the built-in AES-128/256 payload encryption.
In addition to “hitless” protection, like the IP Link codecs, the Ascent also supports automatic failover between different incoming streams or stored audio.
This combination of capabilities gives Intraplex codecs a market-leading position in STL over IP technology and unmatched network transport reliability.
Ascent supports both physical audio interfaces (AES3, Analog) and AES67 (AoIP), with a maximum of 16 full-duplex or a total of 32 channels (combination of encode and decode). Physical audio interfaces can be ordered as 4- or 8-channel cards, supporting both AES3 and analog audio signals. The Ascent product is built on the Commercial-Off-The-Shelf (COTS) x86 architecture to leverage the scalability and cost of the technology. The product is available in a 1RU branded hardware server and as a software-only option.
Main Features of Intraplex® Ascent
- 4, 8, 16 stereo channels. Each channel can be full-duplex, encode-only or decode-only
- Supports AES3, Analog and AES67 audio input and outputs
- Standard Coding: Linear, Opus
- Optional: AAC-HE, AAC-HEv2, AAC- ELD, AAC-xHE, MPEG 2 and MPEG 3 audio coding
- Protocol Encapsulation: Real-time Transport Protocol (RTP), Secure Reliable Transport (SRT)
- Streams: Multicast, unicast, and multi-unicast
- 10 streams per channel with up to 20 destinations per transmit stream; maximum of 100 streams per system
- Three independent IP interfaces for redundant network operation
- Built-in silence sensor with optional stream switchover
- Automatic backup to audio playout from USB drive or external audio source
- Multicoding: can encode the same audio source in multiple formats
- Prioritized stream sources at the output with automatic switch over and switch back between primary and secondary streams and backup sources (streams, USB, external audio source)
- Programmable RTP level Forward Error Correction (FEC) scheme
- Programmable time diversity and Interleaving of streams to combat burst packet losses
- Integrated scheduler for automated scheduled program switching
- Integrated with Intraplex LiveLook (network analytics and monitoring software)
- Web and SNMP for management
- Multiple web account types to restrict access
- GPIOs: up to 32 in, 8 out. Available for stream transport and alarm assignment
- Network reliability
- Dynamic Stream Splicing with network and time diversity for “hitless” packet loss protection
- Programmable RTP level Forward Error Correction (FEC)
- Secure Reliable Transport (SRT): new protocol provides automatic packet retransmission method
- IP Security
- Access control per interface
- Encryption of streams with AES-128
MAGIC ACip3 Audio Codec
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.
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
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).
MAGIC AE1 DAB+ Go Audio Encoder
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.
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
MAGIC DABMUX Go RF Ensemble Multiplexer
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.
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
IQOYA X/LINK-LE
IQOYA X/LINK-LE
Streamlined audio codec
IQOYA X/LINK-LE is a 1U rack streamlined IP audio codec designed for the delivery of a stereo source (or two mono sources) over IP networks for STL and SSL, but also DVB audio, or WEB radio. IQOYA X/LINK-LE benefits from all the major features of X/LINK but at an attractive price. It can be used in legacy analog or AES/EBU audio environments, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a good investment for the migration to IP audio. Like all the IQOYA products, X/LINK-LE is based on Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable connections over inexpensive IP links. Based on a low consumption and fanless powerful hardware platform, IQOYA X/LINK-LE is designed for 24/7/365 use.
key points
Cost effective solution with essential features, and no compromise on reliability
Designed for audio service continuity and failsafe operation
Ongoing product support with flexible options
Invest now in a codec adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure
Easy integration into existing SNMP based supervisors (SET, GET, Traps)
ORBAN OPTIMOD 8700i
ORBAN OPTIMOD 8700i
Xponential Loudness™ brings hyper-compressed music back to life, revealing detail and increasing impact while reducing listening fatigue and distortion.
Built-in streaming server allows you to monitor the 8700i-processed FM and HD/DAB+ audio wherever there is a LAN or Internet connection using free OPUS and MP3 codecs.
Exclusive “Multipath Mitigator” phase corrector reduces multipath distortion without compromising stereo separation.
Dante dual-redundant Audio-Over-IP with AES67 support.
Subharmonic synthesizer adds modern-sounding bass punch to older recordings.
Dual redundant power supplies and safety bypass relays ensure 24/7 operation with no dead-air.
