| Audio signal processing, routing and infrastructure |
Analog and digital audio interfaces, routing, signal processing and conversion devices.
I/O & Interfaces
A comprehensive range of interfaces for Dante, AES67 and SMPTE 2110 audio networks.
I/O & Interfaces
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
SSL Network I/O & interfaces are a range of devices which facilitate use of MADI, AES, SDI and IP Audio networking using Audinate's Dante technology for broadcast audio production
HC Bridge SRC
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
HC Bridge SRC provides 256 bi-directional channels of sample rate conversion for AoIP networks. HC Bridge SRC facilitates connecting audio between devices running at different sample rates or in different clock domains on Dante (48 kHz and 96 kHz), AES67 or ST 2110-30 networks. The two network connections can be to physically or virtually separate networks, providing control isolation between two sets of equipment where an AoIP discovery and control ‘firewall’ is required. An HC Bridge SRC includes 2 HC cards, each providing 256 Tx and 256 Rx channels at 48 kHz or 96 kHz, with 128 Tx and 128 Rx streams at both sample rates. Sample rate conversion can be disabled if not required. An alternate HC Bridge configuration without sample rate converters is available with higher channel capacity at 48 kHz. An HC Bridge includes 2 HC cards, each provides 512Tx and 512 Rx channels at 48 kHz, 256 Tx and 256 Rx channels at 96 kHz, with 128 Tx and 128 Rx streams at both sample rates. Each card includes a redundant pair of SFP cages for 1GbE RJ45, multi-mode fibre or single-mode fibre Dante/AoIP connectivity.
INTRAPLEX® ASCENT
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
Built on the proven, award-winning Intraplex® IP Link codec technology, GatesAir's Intraplex Ascent platform supports up to 32 audio channels with high-end features.
Product Details
Main Features of 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 is a pure IP Audio Code for high-quality Audio transmission.
MAGIC ACip3 Audio Codec
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
MAGIC ACip3 is designed as 19” system with integrated wide area power supply and provides optionally an external redundant power supply. Three Ethernet interfaces are available which can be used for Audio over IP transmissions, to control the system with the Windows PC software or to integrate it into a network management system via SNMP. The Audio programmes can – flexibly and freely assignable – be fed in or given out, respectively, via an analogue and two digital stereo interfaces.
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.
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.
AVN-PA8, 8 Stereo Analogue Line Inputs & 8 Stereo Analogue Line Outputs, AES...
AVN-PA8, 1 of 3 powerful audio mix engine interfaces for audio routing, metering and equalisation.
AVN-PA8, 8 Stereo Analogue Line Inputs & 8 Stereo Analogue Line Outputs, AES67 Portal
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
The AVN-Portal series is a group of three powerful audio mix engine interfaces (AVN-PA8, AVN-PD8 & AVN-PM8) capable of audio routing, metering and equalisation. They have both physical audio I/O and AES67 stream I/O and can mix and route freely from any input to any output, useful for a multitude of applications.
Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.
Applications:
Webserver Software
A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.
Mix Matrix
The key to the success of the AVN-PA8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.
The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.
Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.
DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points.
The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.
Front Panel Displays, Metering & Controls
The AVN-PA8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and D-types for rear panel connectivity.
Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.
Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.
A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.
Detailed Metering Option
The ‘D’ version of the portal (e.g. AVN-PA8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.
The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.
Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.
Phase metering can be displayed per stereo channel and channel idents can be shown either above or below the metering to identify each input/output.
On devices without a meter display, a smaller set of monochrome meters are shown on the main OLED display.
Physical Inputs & Outputs
For the analogue audio, the AVN-PA8 uses D-type sockets with AES59 analogue pinout, paralleled with 8 x RJ45 connectors using StudioHub® pinout.
The ‘T’ version (e.g. AVN-PA8T) uses rear panel terminal block connectors for all physical inputs and outputs.
The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.
There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.
A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.
For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.
8 Way Analogue Headphone Distribution System
The AVN-PA8 can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.
Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.
Applications:
- 8 output analogue zone mixer, with individual output gain control.
- 8 channel clean-feed generator, with input mixing and gain control on inputs and outputs.
- Distribute 8 stereo channels of audio over an SFP fibre connection.
- IFB generator to send 64 x AES67 streams to individual belt-packs.
- 8 output headphone distribution system, with separate input mix for each headphone output and individual gain control.
- Input mixer with input/output metering and AES67 stream generation.
Webserver Software
A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.
Mix Matrix
The key to the success of the AVN-PA8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.
The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.
Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.
DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points.
The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.
Front Panel Displays, Metering & Controls
The AVN-PA8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and D-types for rear panel connectivity.
Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.
Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.
A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.
Detailed Metering Option
The ‘D’ version of the portal (e.g. AVN-PA8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.
The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.
Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.
Phase metering can be displayed per stereo channel and channel idents can be shown either above or below the metering to identify each input/output.
On devices without a meter display, a smaller set of monochrome meters are shown on the main OLED display.
Physical Inputs & Outputs
For the analogue audio, the AVN-PA8 uses D-type sockets with AES59 analogue pinout, paralleled with 8 x RJ45 connectors using StudioHub® pinout.
The ‘T’ version (e.g. AVN-PA8T) uses rear panel terminal block connectors for all physical inputs and outputs.
The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.
There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.
A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.
For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.
8 Way Analogue Headphone Distribution System
The AVN-PA8 can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.
IQOYA X/LINK-AES67
IQOYA X/LINK-AES67 is a stereo to multi stereo FULL IP audio codec dedicated to distribution.
IQOYA X/LINK-AES67
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
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
The MAGIC ACX Dante™ WAN Bridge enables the connection of Dante™ Networks over wide area networks.
MAGIC ACX Dante™ WAN Bridge
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
The multi-channel PCM system MAGIC ACX Dante™ WAN Bridge enables the transmission of up to 32 uncompressed audio signals via wide area networks (WAN).
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).
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).
AVN-AO16R 16 Output Dual Dante®Interface, PoE
Audio converter & interface converts up to 16 analogue outputs from the Dante AoIP network standard.
AVN-AO16R 16 Output Dual Dante®Interface, PoE
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
The AVN-AO16R audio converter and interface converts up to sixteen analogue outputs from the Dante Audio-over-IP networking standard. Dual Ethernet ports allow the unit to operate in redundant mode, ensuring audio routing is maintained in the event of loss of link on either of the network connections.
This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).
All analogue outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.
(The AVN-AO16 is available with a single Ethernet port).
This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up, except line-up levels, done via the standard Dante Controller software and power via PoE (Power Over Ethernet).
All analogue outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock. Global 0dBFS line-up can be set to +12dBu, +18dBu or +24dBu to meet your specific requirement via the front panel recessed toggle switch.
- 16 x balanced analogue outputs on XLR.
- Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
- 2 x RJ45 Dante connectors (1Gb/s Ethernet Port) allowing the unit to operate in redundant or switched modes.
- PoE and Link LED status indicators for each Ethernet port.
- Clock LED status indicator.
- Configuration using Dante Controller.
- AES67 operation & Dante Domain Manager compliant.
- Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.
- 1U 19" rack-mount form factor.
(The AVN-AO16 is available with a single Ethernet port).
MAGIC AE1 DAB+ Go Audio Encoder
MAGIC AE1 DAB+ Go is a high quality and professional DSP-based hardware Audio Encoder.
MAGIC AE1 DAB+ Go Audio Encoder
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The high quality and professional DSP-based hardware audio encoder MAGIC AE1 DAB+ Go is realized as a fanless ½ x 19‘‘ device and offers excellent audio quality thanks to the licenced Fraunhofer DAB+ algorithm.
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.
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.
AVN-PD8, 8 Stereo AES3 Digital Inputs & 8 Stereo AES3 Digital Outputs, AES67...
AVN-PD8, 1 of 3 powerful audio mix engine interfaces for audio routing, metering and equalisation.
AVN-PD8, 8 Stereo AES3 Digital Inputs & 8 Stereo AES3 Digital Outputs, AES67 Portal
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
The AVN-Portal series is a group of three powerful audio mix engine interfaces (AVN-PA8, AVN-PD8 & AVN-PM8) capable of audio routing, metering and equalisation. They have both physical audio I/O and AES67 stream I/O and can mix and route freely from any input to any output, useful for a multitude of applications.
Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.
Applications:
Webserver Software
A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.
Mix Matrix
The key to the success of the AVN-PA8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.
The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.
Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.
DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points.
The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.
Front Panel Displays, Metering & Controls
The AVN-PD8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and D-types for rear panel connectivity.
Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.
Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.
A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.
Detailed Metering Option
The ‘D’ version of the portal (e.g. AVN-PD8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.
The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.
Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.
Physical Inputs & Outputs
For the digital audio, the AVN-PD8 uses D-type sockets with AES59 digital pinout, paralleled with 8 x RJ45 connectors using StudioHub® pinout. There is individual sample rate conversion on each input.
The ‘T’ version (e.g. AVN-PD8T) uses rear panel terminal block connectors for all physical inputs and outputs.
The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.
There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.
A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.
For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.
8 Way AES3 Headphone Distribution System
The AVN-PD8 can be combined with multiple AVN-HD1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.
Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.
Applications:
- 8 channel digital mixer,
- 64 channel AES67 stream distribution amplifier from digital sources,
- Distribute 8 stereo channels of audio over an SFP fibre connection.
- IFB generator to send 64 x AES67 streams to individual belt-packs.
- 8 output headphone distribution system, with separate input mix for each headphone output and individual gain control.
- Input mixer with input/output metering and AES67 stream generation.
Webserver Software
A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.
Mix Matrix
The key to the success of the AVN-PA8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.
The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.
Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.
DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points.
The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.
Front Panel Displays, Metering & Controls
The AVN-PD8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and D-types for rear panel connectivity.
Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.
Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.
A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.
Detailed Metering Option
The ‘D’ version of the portal (e.g. AVN-PD8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.
The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.
Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.
Physical Inputs & Outputs
For the digital audio, the AVN-PD8 uses D-type sockets with AES59 digital pinout, paralleled with 8 x RJ45 connectors using StudioHub® pinout. There is individual sample rate conversion on each input.
The ‘T’ version (e.g. AVN-PD8T) uses rear panel terminal block connectors for all physical inputs and outputs.
The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.
There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.
A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.
For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.
8 Way AES3 Headphone Distribution System
The AVN-PD8 can be combined with multiple AVN-HD1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.
IQOYA X/LINK-DUAL
IQOYA X/LINK is a DUAL stereo IP audio codec dedicated to distribution applications.
IQOYA X/LINK-DUAL
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
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
AVN-TB6 6 Button Talkback Intercom
AVN-TB6, a 6 channel talkback intercom control unit from the Sonifex AVN range of IP based products.
AVN-TB6 6 Button Talkback Intercom
Connect » Real-Time Signal Transfer » Audio Interfacing & Conversion » Audio signal processing, routing and infrastructure »
The AVN-TB6 is a 6 channel talkback intercom control unit from the Sonifex AVN range of IP based products. The AVN-TB6 is a freestanding version which can be rackmounted with the AVN-TB6RK 1U rack kit. There is also a desktop version, the AVN-TB6D which can be fitted flush into a work surface if required. Both models offers similar functionality.
This unit provides broadcast quality audio communication between studios, offices and different areas in a facility or building complex, using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion. RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. The AVN range use RAVENNA as the communication method providing compatibility with other AES67 systems.
Each of the 6 channels on the AVN-TB6 can be configured to provide communications with other remote networked units, and an independently configurable ‘page’ function can contact selected units with priority over standard intercom calls if required.
There is a monitor channel that can route the audio from an AoIP source to the headphones and speaker. This could be used to take an IFB feed or an off-air transmission signal or simply to listen to any audio source.
A user configurable GPIO system, with 10 physical ports and 10 virtual ports, can be used to control operational functions on local or networked units, or drive outputs as selected states change, and a voltage free relay contact can be used to operate external equipment.
A built-in web server provides complete configuration control of the units and also allows for firmware updates and configuration backup. An Ember+ interface also gives access to the configuration options as well as providing remote control and monitoring of the GPIO and virtual GPIO ports.Webserver Software.
The AVN-TB6 has a built-in webserver for setup and configuration. The webserver is responsive, and resizes depending on the size of your screen, meaning that it can be used on large monitors or small handheld devices such as smart-phones. Help information is shown on the right hand side of the screen so it’s a good place to go to find out how the unit operates.
Category: AES67/Dante AoIP Products.Product Function: Provides broadcast quality audio communication using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion.Typical Applications: Ideal for broadcast quality audio communication between studios, offices and different areas in a facility or building complex.
Features:
This unit provides broadcast quality audio communication between studios, offices and different areas in a facility or building complex, using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion. RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. The AVN range use RAVENNA as the communication method providing compatibility with other AES67 systems.
Each of the 6 channels on the AVN-TB6 can be configured to provide communications with other remote networked units, and an independently configurable ‘page’ function can contact selected units with priority over standard intercom calls if required.
There is a monitor channel that can route the audio from an AoIP source to the headphones and speaker. This could be used to take an IFB feed or an off-air transmission signal or simply to listen to any audio source.
A user configurable GPIO system, with 10 physical ports and 10 virtual ports, can be used to control operational functions on local or networked units, or drive outputs as selected states change, and a voltage free relay contact can be used to operate external equipment.
A built-in web server provides complete configuration control of the units and also allows for firmware updates and configuration backup. An Ember+ interface also gives access to the configuration options as well as providing remote control and monitoring of the GPIO and virtual GPIO ports.Webserver Software.
The AVN-TB6 has a built-in webserver for setup and configuration. The webserver is responsive, and resizes depending on the size of your screen, meaning that it can be used on large monitors or small handheld devices such as smart-phones. Help information is shown on the right hand side of the screen so it’s a good place to go to find out how the unit operates.
Category: AES67/Dante AoIP Products.Product Function: Provides broadcast quality audio communication using RAVENNA/AES67 as the transport mechanism, allowing simple CAT 5 cabling and expansion.Typical Applications: Ideal for broadcast quality audio communication between studios, offices and different areas in a facility or building complex.
Features:
- 6 illuminated key-cap Talk buttons plus Listen & Page buttons.
- Dual 1Gb Ethernet & 1Gb SFP port.
- Mic & headset inputs, headphone & speaker outputs with volume control.
- Loudspeaker & Mic Mute buttons.
- Dual AC & DC power supply inputs.
- Advanced echo cancellation & mic AGC to prevent acoustic feedback.
- 10 user assignable GPIO ports.
- Responsive design Ethernet webserver.
- AVN-TB6RK 19” rack kit available.
