Sonifex AVN-GMCS IEEE1588 PTP Grandmaster Clock with GPS Receiver
Sonifex AVN-GMCS IEEE1588 PTP Grandmaster Clock with GPS Receiver
The AVN-GMCS is a PTPv2 grandmaster clock for use with AoIP applications. IEEE1588-2008 PTPv2 (precision time protocol) is used to synchronise all the nodes within a network. To achieve this one of the nodes must become the master clock and distribute time packets to the others. The AVN-GMCS is designed to perform this role simply and accurately, enabling sub micro second synchronisation between all nodes.
RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. For this to be possible, each of the nodes needs to be time synchronised with one another. RAVENNA uses PTP time stamping to achieve this, which distributes the network time but also works out the latency involved in the delivery and adjusts the time at each node accordingly.
Unit configuration is achieved easily either with the front panel controls or the webserver, including the setup of the PTP profiles.
The AVN-GMCS supports the Default (RAVENNA), Media (AES67) and AES-R16-2016 (SMPTE-ST 2059-2 & AES67 compatible) profiles and has a ‘Custom’ profile page for you to define your own.
In normal operation, the unit has PTPv2 time stamping resolution to 8nsec. It uses a combination of a GPS receiver, a PLL (phase lock loop) and a specialist on-board clock device to create the precise, low jitter clock signals required to drive the physical transceiver’s time stamping circuitry, also providing holdover if the GPS signal is lost.
The specialist on board clock is available in three different types: TCXO, OXCO and CSAC (Chip Scale Atomic Clock, Caesium), which vary in both price and accuracy:
AVN-GMCS – TCXO Temperature Compensated Oscillator accurate to 1 part per million (worst case 1 sec gain/loss every 11.5 days). *
AVN-GMCOS – OCXO Oven Controlled Oscillator accurate to 0.1 parts per million (worst case 1 sec gain/loss every 115 days). *
AVN-GMCCS – SAC Quantum Atomic Clock accurate to 0.0005 parts per million (worst case 1 sec gain/loss every 63 years). *
GPS presence and the number of satellites received is shown on the front panel, together with status information on output sample rates, sync type and profile type. The unit also has a screen-saver option which shows the current time.
Although designed as a grandmaster clock, a separate clock input can act as an alternative reference source to GPS which the unit can ‘slave’ to. Clock outputs, driven from the physical transceiver, can be used to provide media clocks for external equipment local to the AVN-GMC when it is in both ‘master’ and ‘slave’ states. The clock outputs are available as a single AES-3id output and two outputs which can be selected as either word clock or variable PPS. The wordclock can operate at 32, 44.1, 48, 96, 176.4 and 192kHz. When set as a variable PPS output, the unit can act as a clock master to distribute a reference frequency to test and measurement equipment.
The unit shows UTC as standard, but can be set to show ‘local time’ on the front panel, by adding a time offset. Daylight saving time changes can be accommodated by entering Spring Forward and Fall Back dates. It has a real time clock so that accurate date and time is available even after the unit is repowered without GPS access.
The built-in webserver, or front panel OLED display, can be used to configure the unit. The webserver is a responsive design meaning that it can be used with small screens on smartphones and tablets.
Front panel LEDs show the synchronisation status, GPS lock and the status of the AC and DC power supply inputs.
The brightness of the OLED display and LED indicators can be adjusted for low or high lighting conditions 4 general purpose outputs indicate critical states for the unit using a 9 way D-type connector mounted on the rear panel. Pull down when active pins are supplied for GPS lock status, external sync present, AC power present and DC power present.
The unit has a front panel power button and dual power connectors – an IEC mains input and a 12V DC input, which allows the AVN-GMCS to be used for both studio and mobile installations. Moreover this allows for a secondary power source to reduce the effect of power down events. In any case, the unit monitors the status of both power sources and displays this on the front panel.
The unit can be put into a low-power sleep mode when not in use, with an instant start when power is re-applied. In power off situations, a super capacitor is used to keep the GPS receiver powered in a low power mode for more than 20 hours, enabling the receiver to regain lock immediately rather than having to ‘cold’ start.
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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.
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Main Features of Intraplex® Ascent
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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. “
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- Configured specifically to address the needs of different applications, skillets and workflows.
- Engineers and supervisors can restrict sources, modes and front panel functions to streamline operation and reduce user error.
- A feature set that can change with your requirements, including optional MADI support or Loudness Monitoring.
The latest release includes the brand-new audio phase metering mode included as standard, plus additional functions that can be purchased separately and installed as a license.
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- MADI License - customers will be able to Mix, Monitor and Measure up to 128 MADI sources on the SAM-Q-SDI.
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Network Switches
Drawing on Yamaha expertise in signal processing as innovators in electronic instruments, we swiftly transitioned those skills towards advanced network technologies. For over 25 years, we've played a vital role in the development of the internet, which has become a lifeline for modern society.
Moving forward, we remain committed to providing even better services and support to make your connections smoother and more convenient than ever before. We continue to innovate and enhance the way you connect.
10-Gigabit/Multi-Gigabit switches for flexible deployment in high-capacity networks
Enhancing backbone networks has become crucial for businesses, concert halls, and other installations due to the rising demand for high-quality video and audio with low latency. By introducing switches that support 10-Gigabit/Multi-Gigabit Ethernet, our new lineup creates an ideal environment for backbone networks requiring high speeds, capacities, and capabilities. These new L3 intelligent switches support systems handling increased media traffic across multiple segments, while our L2 switches are perfect for transmitting single-segment audio and video media. Like our other existing products, this expanded lineup is also supported by Yamaha LAN Monitor and includes a range of features optimized for Dante and AVoIP systems.
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R Series Dante Audio Interfaces
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Due to their compact yet flexible design, DM7 series mixers can be used in a wide range of audio applications. With their comprehensive functionality, expansion options, and outstanding ease of use, DM7 consoles excel in a truly impressive array of use cases.
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RIVAGE PM Series Mixing Consoles
The RIVAGE PM series includes five consoles, two DSP engines, two I/O rack units, and two network protocols. Any of these components can be combined to create systems that match a wide range of budget requirements in broadcast applications, such as in fixed studios or OB vehicles
All five consoles feature a similar interface with up to three bays of 12 faders each, touch-sensitive display panels, and the Yamaha Selected Channel Section. Mix engineers who are familiar with one RIVAGE PM console can easily switch to any other and be up and running in a matter of minutes.
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Abekas AirCleaner
Key Features Include:
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All analogue inputs and 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.
- 8 x balanced analogue inputs on XLR.
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(The AVN-AIO8 is available with a single Ethernet port).
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Smart Codecs
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INES
A windows app to manage calls on air
A web app (ines light) to manage your audience cards in the database.
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ORBAN OPTIMOD 8700i LT
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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.
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AVN-AO16R 16 Output Dual Dante®Interface, PoE
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).
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AVN-AI16R 16 Input Dual Dante® Interface, PoE
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 inputs 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 inputs on XLR.
- Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
- 2 x RJ45 Dante connector (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-AI16 is available with a single Ethernet port).
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AVN-AI16 16 Input Dante® Interface, PoE
All analogue inputs 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 inputs on XLR.
- Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
- 1 x RJ45 Dante connector (1 Gb/s Ethernet Port).
- PoE, Link, and Clock LED status indicators.
- Configuration using Dante Controller.
- AES67 operation & Dante Domain Manager compliant.
- Powered by PoE.
- 1U 19" rack-mount form factor.
(The AVN-AI16R is available with dual redundant Ethernet ports).
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AVN-AESIO8R 8 AES3 Input, 8 AES3 Output Dual Dante® Interface, PoE
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 done via the standard Dante Controller software and power via PoE (Power Over Ethernet).
All digital AES3 inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link and Clock.
- 8 x balanced digital stereo AES3 inputs on XLR, supporting input rates of 32kHz – 192kHz.
- Sample rate conversion on physical inputs to Dante system sample rate.
- 8 x balanced digital stereo AES3 outputs on XLR, output rate matches Dante system sample rate.
- 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.
- AES3 Lock LED status indicators for each input.
- 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-AESIO8 is available with a single Ethernet port).
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AVN-AESIO8 8 AES3 Input, 8 AES3 Output Dante® Interface, PoE
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 done via the standard Dante Controller software and power via PoE (Power Over Ethernet).
All digital AES3 inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link and Clock.
- 8 x balanced digital stereo AES3 inputs on XLR, supporting input rates of 32kHz – 192kHz.
- Sample rate conversion on physical inputs to Dante system sample rate.
- 8 x balanced digital stereo AES3 outputs on XLR, output rate matches Dante system sample rate.
- 1 x RJ45 Dante connector (1Gb/s Ethernet Port).
- PoE, Link, and Clock LED status indicators.
- AES3 Lock LED status indicators for each input.
- Configuration using Dante Controller.
- AES67 operation & Dante Domain Manager compliant.
- Powered by PoE.
- 1U 19” rack-mount form factor.
(The AVN-AESIO8R is available with dual redundant Ethernet ports).
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AVN-AO16 16 Output Dante® Interface, PoE
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).
- 1 x RJ45 Dante connector (1 Gb/s Ethernet Port).
- PoE, Link, and Clock LED status indicators.
- Configuration using Dante Controller.
- AES67 operation & Dante Domain Manager compliant.
- Powered by PoE.
- 1U 19" rack-mount form factor.
(The AVN-AO16R is available with dual redundant Ethernet ports).
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AVN-TB6D 6 Button Talkback Intercom
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-TB6D 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 SoftwareThe AVN-TB6D 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.
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AVN-AIO8 8 Input, 8 Output Dante® Interface, PoE
All analogue inputs and 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.
- 8 x balanced analogue inputs on XLR.
- 8 x balanced analogue outputs on XLR.
- Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
- 1 x RJ45 Dante connector (1Gb/s Ethernet Port).
- PoE, Link, and Clock LED status indicators.
- Configuration using Dante Controller.
- AES67 operation & Dante Domain Manager compliant.
- Powered by PoE.
- 1U 19” rack-mount form factor.
(The AVN-AIO8R is available with dual redundant Ethernet ports).
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ORBAN OPTIMOD TV 8685
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ORBAN OPTIMOD 5700i
Independent Processing for FM and digital radio
The FM and digital media processing paths split after the 5700i's stereo enhancer and AGC. There are two equalizers, multiband compressors, and peak limiters, allowing the analog FM and digital media processing to be optimized separately. The bottom line? Processing that optimizes the sound of your FM channel while punching remarkably crisp, clean, CD-like audio through to your digital channel audience.
The 5700i includes a full-featured RBS/RBDS generator at no additional charge. The generator supports dynamic PS. It can be controlled via the 5700i presets and an ASCII terminal server that can be connected to automation to support displaying title and artist.
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ORBAN OPTIMOD FM 5500i
The 5500i can also be used as a superb stand-alone stereo encoder with latency as low as 2 ms and full overshoot limiting in both the left/right and composite baseband domains. When used in this mode, the 5500i must be driven (usually via an STL) by a full-featured FM audio processor (like Orban’s 8700i) that incorporates pre-emphasis-aware HF limiting and peak control. In both modes, the 5500i’s stereo encoder helps deliver a transmitted signal that’s always immaculately clean and perfectly peak limited, with full spectral protection of subcarriers and RDS/RBDS regardless of the amount of composite limiting.
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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.
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Matrox DSX Core
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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 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 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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Sonifex AVN-GMCS IEEE1588 PTP Grandmaster Clock with GPS Receiver
RAVENNA (of which AES67 is a subset) allows for the distribution of audio across a network. For this to be possible, each of the nodes needs to be time synchronised with one another. RAVENNA uses PTP time stamping to achieve this, which distributes the network time but also works out the latency involved in the delivery and adjusts the time at each node accordingly.
Unit configuration is achieved easily either with the front panel controls or the webserver, including the setup of the PTP profiles.
The AVN-GMCS supports the Default (RAVENNA), Media (AES67) and AES-R16-2016 (SMPTE-ST 2059-2 & AES67 compatible) profiles and has a ‘Custom’ profile page for you to define your own.
In normal operation, the unit has PTPv2 time stamping resolution to 8nsec. It uses a combination of a GPS receiver, a PLL (phase lock loop) and a specialist on-board clock device to create the precise, low jitter clock signals required to drive the physical transceiver’s time stamping circuitry, also providing holdover if the GPS signal is lost.
The specialist on board clock is available in three different types: TCXO, OXCO and CSAC (Chip Scale Atomic Clock, Caesium), which vary in both price and accuracy:
AVN-GMCS – TCXO Temperature Compensated Oscillator accurate to 1 part per million (worst case 1 sec gain/loss every 11.5 days). *
AVN-GMCOS – OCXO Oven Controlled Oscillator accurate to 0.1 parts per million (worst case 1 sec gain/loss every 115 days). *
AVN-GMCCS – SAC Quantum Atomic Clock accurate to 0.0005 parts per million (worst case 1 sec gain/loss every 63 years). *
GPS presence and the number of satellites received is shown on the front panel, together with status information on output sample rates, sync type and profile type. The unit also has a screen-saver option which shows the current time.
Although designed as a grandmaster clock, a separate clock input can act as an alternative reference source to GPS which the unit can ‘slave’ to. Clock outputs, driven from the physical transceiver, can be used to provide media clocks for external equipment local to the AVN-GMC when it is in both ‘master’ and ‘slave’ states. The clock outputs are available as a single AES-3id output and two outputs which can be selected as either word clock or variable PPS. The wordclock can operate at 32, 44.1, 48, 96, 176.4 and 192kHz. When set as a variable PPS output, the unit can act as a clock master to distribute a reference frequency to test and measurement equipment.
The unit shows UTC as standard, but can be set to show ‘local time’ on the front panel, by adding a time offset. Daylight saving time changes can be accommodated by entering Spring Forward and Fall Back dates. It has a real time clock so that accurate date and time is available even after the unit is repowered without GPS access.
The built-in webserver, or front panel OLED display, can be used to configure the unit. The webserver is a responsive design meaning that it can be used with small screens on smartphones and tablets.
Front panel LEDs show the synchronisation status, GPS lock and the status of the AC and DC power supply inputs.
The brightness of the OLED display and LED indicators can be adjusted for low or high lighting conditions 4 general purpose outputs indicate critical states for the unit using a 9 way D-type connector mounted on the rear panel. Pull down when active pins are supplied for GPS lock status, external sync present, AC power present and DC power present.
The unit has a front panel power button and dual power connectors - an IEC mains input and a 12V DC input, which allows the AVN-GMCS to be used for both studio and mobile installations. Moreover this allows for a secondary power source to reduce the effect of power down events. In any case, the unit monitors the status of both power sources and displays this on the front panel.
The unit can be put into a low-power sleep mode when not in use, with an instant start when power is re-applied. In power off situations, a super capacitor is used to keep the GPS receiver powered in a low power mode for more than 20 hours, enabling the receiver to regain lock immediately rather than having to ‘cold’ start.
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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
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AVN-DIO10 Dante® to 3G/HD/SD-SDI Embedder/De-Embedder
The AVN-DIO10 takes any SDI feed, de-embeds the 16 audio channels and places them on channels 1-16 of the Dante network, mapped using Dante Controller. It simultaneously takes the 16 input channels mapped to the device on Dante Controller and re-embeds them onto the SDI output, with an embed enable switch for each channel pair.
It has a single 3G/HD/SD-SDI input and a reclocked output, with dual redundant Primary and Secondary Dante network ports, using Neutrik EtherCon® Ethernet connectors and is powered by PoE. It is a fully Dante compliant and AES67 compatible device that uses Dante Controller for configuration and supports the full range of Dante sample rates.
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AVN-AIO4 4 Input, 4 Output Dante® Interface, PoE
All analogue inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for PoE, Link, and Clock.
- 4 x balanced analogue inputs on XLR.
- 4 x balanced analogue outputs on XLR.
- 1 x RJ45 Dante connector (100Mb/s Ethernet Port).
- POE, Link, and Clock LED status indicators.
- Configuration using Dante Controller.
- AES67 operation & Dante Domain Manager compliant.
- Powered by PoE.
- 1U 19" rack-mount form factor.
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AVN-PA8, 8 Stereo Analogue Line Inputs & 8 Stereo Analogue Line Outputs, AES67 Portal
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.
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AVN-PD8, 8 Stereo AES3 Digital Inputs & 8 Stereo AES3 Digital Outputs, AES67 Portal
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.
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AVN-PM8 8 Mic/Line Inputs, 8 Stereo Analogue Line Outputs, AES67 Portal
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 microphone input mixer, with individual output gain control, input/output metering and AES67 stream generation.
- 8 channel clean-feed generator, with input mixing and gain control on inputs and outputs.
- Distribute 8 microphone 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 stream AES67 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-PM8 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. There is an adjustable input and output gain/trim and an additional mic pre-amp gain adjustment for each mic input.
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-PM8 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 RJ45s 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-PM8D) 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 output 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 microphone audio, the AVN-PM8 uses 8 x mic/line XLR sockets for the inputs and 8 x RJ45 connectors using StudioHub® pinout for the stereo analogue line outputs. +48V phantom power is available for each microphone input with a red LED presence indication.
The ‘T’ version (e.g. AVN-PM8T) 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-PM8 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.
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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)
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Q568 MULTI CHANNEL DAB+ AUDIO ENCODER
Applications
- single channel DAB+ encoder for studio to Fraunhofer Content Server
- multi channel DAB+ encoder for studio to Fraunhofer Content Server
- multi channel DAB+ encoder for studio to Small Scale multiplexers
Features
- DAB+ audio encoder compresses up to 8 stereo channels in 1RU
- XLR inputs for analog & digital AES audio
- DAB+ audio coding in AAC+ (HE-AACv2 (ETSI TS 102 563)
- optionally: MPEG-1/Layer II for DAB
- compliant to (Small Scale) DAB multiplexes (Open Digital Radio)
- full remote operation via MuxEnc protocol to Fraunhofer DAB+ Content Servers
- PAD support
- Embedded technology means very low power per channel
Connectivity
- XLR inputs for analog & digital AES audio
- 2x RJ-45 ports for streaming and management
- Supported streaming formats: EDI/DCP, MuxEnc
- web and SNMP control, monitoring and alarming
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Q880 64 STEREO CHANNEL RAVENNA/AES67 IP AUDIO GATEWAY CODEC
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Q866 48 CHANNEL ENTERPRISE INTERNET RADIO TRANSCODER
solution to include a large number of streamed webradio stations
into your OTT and cable networks.
You can include virtually any radio station in the world into your
(local) cable network.
It converts up to 48 internet radio stations (Icecast/Shoutcast) into a
DVB compliant MPEG-2 transport stream.
The signals are available via IP (ethernet) or DVB-ASI.
Low power consumption and the compact design in industrynstandard dimensions (19“, 1 U) allow easy integration of the device into your infrastructure.
Our embedded technolody avoids time consuming updates and upgrades.
FEATURES:
- no annual upgrades or maintenance required - „set and forget“
- transcoding of multiple internet radio stations into DVB compliant transport streams
- receives Icecast/Shoutcast streams (MP3 / AAC)
- high security - WAN port (for Icecast) is physically separated from streaming unit and management - no break-in from internet possible
- several compression algorithms (MPEG 1 Layer II, AAC
- compression algorithm can be set individually per radio station
- all bit rates are supported according to the respective standards
- 32kHz, 48kHz sampling rate
- Meta data Tags are converted into UECP data
- 2 years warranty
- By software license field-upgradable up to 48 stereo channels.
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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
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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
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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.
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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
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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.
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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.
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RUBIDIUM SERIES PTP module C3
C3 Platforms:- C3 module has five different S/W platforms that can be programmed with
- GM = Grandmaster Slave Mode
- SL = Slave Mode
- BC = Boundary Clock
- GF = Grandmaster with fallback
- BG = Boundary Clock with GPS option
C3 Features:-
- 10/100/1000Base-T PTP Ethernet port
- Compatible with SMPTE 2059-2
- Two 10 Mhz outputs come standard, third output is an option
- PPS, time and date data string reference outputs
- Additional unbalanced and balanced PPS outputs
- Leap year/second compatible
- SNMP ready
- Hot swappable
- Failure Relay
- Compatible with TC_link
- Integrated Surge Voltage Protector
- UMID data capable
The PLURA Rubidium Series PTP client module C3 is the most accurate, stable and accurate way to universally create and manage PTP signals flow in the broadcast industry. The module has multiple platforms such as PTP Grandmaster, PTP Slave or PTP boundary clock. The precise acquired reference time and date is transferred in form of a serial protocol and a synchronized seconds pulse. The module comes with two 10 MHz outputs, which are often used for synchronizing a third party SPG system. The PTP slave uses industry leading algorithms to extract time from a PTP input stream and produces stable frequency and time outputs. The module needs to be used in conjunction with a RUBIDIUM Series housing and a power supply. For user specific setups, an initial configuration requires a Windows PC with an USB and/or Ethernet port (only in combination with an IE module). The PTPC module offers a variety of monitoring and control possibilities. Its design allows for the implementation into complex and fail-proof redundant systems. C3 Outputs:- C3 module has the following outputs:
- Time information Serial Interface Output
- Seconds pulse (PPS)
- Four programmable GPIs
- Two 10 MHz-signals (a third output is configurable)
- An Ethernet pass through port
Every module is connected to an internal hot swappable bus, which bilaterally connects all modules within a particular housing.The internal bus can be distributed over several housings by using the RLC port. The RLC plug contains a voltage feed, a failure relay output and aTC_link interface.TC_link is a real time capable proprietary interface, which is based on a customized RS485 interface. On the rear side of every housing, a PC interface (USB) can be found.This serial connection is used for configuration, status control, and also software and firmware updates. Via our IE Ethernet module, browser configuration, status control and SNMP functions are available.
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AVN-CU4-DANTE Commentator Unit, 4 Commentators
It is a dual version of the AVN-CU2-DANTE providing four mic/line inputs with a wide, adjustable gain range and four stereo headphone outputs with lockable jack sockets, suitable for operation by three or four commentators.
The feature set is as per the AVN-CU2-DANTE, with the following differences. There are:4 x On-air buttons.4 x Page buttons, 2 for each half of the display.4 x Cough buttons.8 x Talkback buttons, up to 4 for each user and3 x User buttons.
As for the AVN-CU2-DANTE, the illuminated ‘Sonifex’ logo acts as a power indication and illuminated LEDs indicate network clock status, AoIP Primary and AoIP Secondary link status, PoE Primary, PoE Secondary and AC power active.
The front panel houses 4 x locking mic/line inputs with +48V phantom power indication and 4 x headphone outputs on locking 6.35mm jack sockets.
There’s an abundance of 4 wire connections on the rear panel: 4 x analogue line inputs on XLR sockets with latching locks, 6 x analogue line outputs on XLR plugs and an RJ45 AES3 stereo input & output.
The unit has dual redundant network ports on both RJ45 (PoE+ using 2 x Neutrik EtherCON® connectors) and SFP cages. There is an AC mains input on an IEC inlet, with a universal supply.
The 10 configurable GPIO and single switched changeover output use a 15 way D-type connector.
-end-
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AVN-CU2-DANTE Commentator Unit, 2 Commentators
The AVN-CU2-DANTE provides two mic/line inputs with a wide, adjustable gain range and has two stereo headphone outputs with lockable jack sockets, suitable for operation by two commentators.
It’s powered using Power over Ethernet (PoE), using Neutrik EtherCON connectors, with primary and secondary ports for power and data redundancy. There’s an additional 4 pin XLR 12V DC input. The unit supports up to 16 input and output AoIP channels and up to 16 simultaneous input and output AoIP streams.
The 6 x push-button rotary encoders and 12 x key-cap buttons are fully configurable, to control input & output levels and panning.
Each rotary encoder has a separate colour-coded meter section showing the channel name, detailed level metering, left/right panning and a limiter indication, on a bright daylight reading display. Colours can be programmed per encoder to quickly identify particular source groups, so headphone source selection becomes intuitive.
Metering is available per input/output, with output metering configurable as pre or post level adjustment. The top of the display shows output metering, a limiter indication and the name of the output, a pre-defined logo or nothing. A limiter is available on every output.
The unit has 2 x locking mic/line inputs with +48V phantom power indication and 2 x headphone outputs on locking 6.35mm jack sockets.
Four wire I/O on rear panel RJ45 connectors provide an AES3 or analogue input and output that can be assigned as mic outputs (line level), talkback outputs, programme inputs or talkback inputs as desired. In addition, the AES/analogue connections can be used as an insert or exit point into/out from the AoIP network.
The unit has dual redundant network ports on both RJ45 (PoE using 2 x Neutrik EtherCON® connectors) and SFP cages.
There are 10 x configurable GPIO on a 15 way D-type connector with 1 x switched changeover output.
All of the buttons have key-cap text and can be configured. There are some standard operations available:
2 x On-Air buttons can be used to connect mic audio to the main output, either over AoIP or via the AES digital audio connection. The On-Air buttons can be locked if required.
A Menu button can be used to access limited setup options on the TFT display.
2 x Page buttons change the display and encoders to monitor an additional set of sources, mix points or outputs. Up to 4 pages can be pre-programmed, e.g. one for talkback inputs, one for outputs, one to monitor other sources.
2 x Cough buttons take the commentator off-air while pressed.
A User button can be programmed to perform any function using the web server.
4 x T/B (talkback) buttons can be configured to initiate talkback over AoIP or AES digital audio connection, using 4 x talkback busses. The talkback buttons operate with lazy talkback, taking the commentator off-air when invoked.
The illuminated ‘Sonifex’ logo acts as a power indication and there are illuminated LEDs to indicate network clock status, AoIP Primary and AoIP Secondary link status, PoE Primary, PoE Secondary and DC power active.
Stream setup to and from the unit is initially via Dante® Controller with more detailed configuration performed by using the built-in web GUI.
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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.
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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.
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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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AVN-TB6 6 Button Talkback Intercom
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.
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