Wired, IP and wireless intercoms and talkback
The Beatrice P1 is a one channel handset.
The user of the handset on the P1 can listen to one audio feed from the network and send one audio channel out onto the network. Depending upon how the Dante network has been routed, the incoming audio circuit and outgoing audio circuit can be different locations and the outgoing circuit can be routed to multiple locations.
Audio to / from the handset is automatically turned on when the handset is off the hook and turned off when on the hook.
The Beatrice P1 unit is fully compatible with other manufacturers’ equipment using the Dante protocol.
Other products from this company:
Beatrice B1
Single Channel Network Audio Ultra Compact Beltpack Intercom
Beatrice B2
2 Channel Network Audio Ultra Compact Beltpack Intercom
Beatrice B4+ Single User 4 Channel...
Compact 4ch beltpack using Dante Broadway so can...
Beatrice D8 Plus
8 Channel Network Audio Desktop Intercom with Analogue...
Beatrice LH4
The Glensound Beatrice LH4 provides audio signal present...
Beatrice MIX32
Mixing, Mix Minus and Priority Override For Beatrice...
Signature ADA 1:8G
1 Input, 8 Output Mono Analogue Distribution Amplifier...
Signature ADA 2:6g
1 Input, 6 Output Stereo Distribution Amplifier With...
Signature ADA 2S12MGT
1 Stereo In, 12 Monmo Outputs Transformer Isolated...
Signature ADDA 1
Stereo Analogue To Digital And Digital To Analogue...
Signature SW 8:2+
8 Channel Source Selector/ Switch/ Mixer with PPM
Virgil OB
Dante/ AES67 Headphone Amplifier With Local Analogue Outputs
AoIP22
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
Packaged in a small & rugged format and designed for outside broadcast, theatre and location applications.
The AoIP22 is designed to easily and quickly interface existing analogue equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
EASY TO USEDante® Controller by Audinate is a free application that controls all your Dante® enabled devices on your network.
It will automatically find the AoIP22 and allow you, just by the click of a mouse, to route audio circuits to/ from it.
Dante® network audio has become very popular because it just works and it’s so easy to use. Yet underneath the simple trouble free setup is a very sophisticated Audio over IP (AoIP) protocol working in real time across standard network switches with clever digital clock synchronisation to keep all equipment working in perfect harmony.
PRACTICAL FEATURESBELT CLIP Can be purchased with or without a belt clip.
RUBBER FEET If the AoIP22 is to be sat on a desk or the back of a workbench, then the beltclip can be removed and rubber feet fitted to stop it sliding around.
CABLE TIE HOLES All four front corners of the AoIP22 feature large extremely strong holes that are ideal for threading cable ties through. This makes temporarily installing the unit in out of the way locations very easy.
The AoIP22 is designed to easily and quickly interface existing analogue equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
EASY TO USEDante® Controller by Audinate is a free application that controls all your Dante® enabled devices on your network.
It will automatically find the AoIP22 and allow you, just by the click of a mouse, to route audio circuits to/ from it.
Dante® network audio has become very popular because it just works and it’s so easy to use. Yet underneath the simple trouble free setup is a very sophisticated Audio over IP (AoIP) protocol working in real time across standard network switches with clever digital clock synchronisation to keep all equipment working in perfect harmony.
PRACTICAL FEATURESBELT CLIP Can be purchased with or without a belt clip.
RUBBER FEET If the AoIP22 is to be sat on a desk or the back of a workbench, then the beltclip can be removed and rubber feet fitted to stop it sliding around.
CABLE TIE HOLES All four front corners of the AoIP22 feature large extremely strong holes that are ideal for threading cable ties through. This makes temporarily installing the unit in out of the way locations very easy.
AoIP22 AES3
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
Packaged in a small & rugged format, and designed for outside broadcast, theatre and location applications.
The AoIP22 AES3 is designed to easily and quickly interface existing AES3 equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
The AoIP22 AES3 is designed to easily and quickly interface existing AES3 equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
AoIP44
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
The AoIP44 is an economical subrack designed to interface balanced analogue audio circuits to and from a Network audio system featuring Dante. It provides 4 audio inputs to the network and also 4 audio outputs from the network on balanced 3 pin XLRs. As well as being mains powered with the standard signature backup DC powering option, the AoIP44 can be powered from an external PoE source via its network interface.
The AoIP can be used as a simple low cost audio I/O break out unit on a large Dante audio network where it can be integrated extremely easily using the Dante controller and is fully compatible with any manufacturers Dante equipment. It can also be used in very simple audio over IP scenarios where just 4 bi-directional audio circuits are needed to be distributed across a building's network infrastructure, in which case 2 x AoIP44 units can be used connected together across the network. The AoIP44 is equally suited for high integrity broadcast purposes, intercom, just simple paging facilities or simple distribution of non-critical audio.
Being part of our Signature Range the AoIP44 comes as standard with removable rack ears (to allow front or rear mounting in 19" racks), mounting holes to allow under desk mounting (the holes are equally suited for screwing the unit into odd places!) and an optional external DC power supply for applications requiring redundant power supplies. It is housed in an all anodised aluminium chassis.
The AoIP can be used as a simple low cost audio I/O break out unit on a large Dante audio network where it can be integrated extremely easily using the Dante controller and is fully compatible with any manufacturers Dante equipment. It can also be used in very simple audio over IP scenarios where just 4 bi-directional audio circuits are needed to be distributed across a building's network infrastructure, in which case 2 x AoIP44 units can be used connected together across the network. The AoIP44 is equally suited for high integrity broadcast purposes, intercom, just simple paging facilities or simple distribution of non-critical audio.
Being part of our Signature Range the AoIP44 comes as standard with removable rack ears (to allow front or rear mounting in 19" racks), mounting holes to allow under desk mounting (the holes are equally suited for screwing the unit into odd places!) and an optional external DC power supply for applications requiring redundant power supplies. It is housed in an all anodised aluminium chassis.
AoIP44 AES3
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
Packaged in a small & rugged format, and designed for outside broadcast, theatre and location applications.
The AoIP44 AES3 is designed to easily and quickly interface existing AES3 equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
The AoIP44 AES3 is designed to easily and quickly interface existing AES3 equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
AoIP4I
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
The Glensound AoIP4I is packaged in a small & rugged format and designed for outside broadcast, theatre and location applications.
The AoIP4I unit is designed to easily and quickly interface existing analogue equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
AoIP4I provides 4 balanced analogue audio inputs to a Dante®/AES67 audio network.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
The AoIP4I unit is designed to easily and quickly interface existing analogue equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
AoIP4I provides 4 balanced analogue audio inputs to a Dante®/AES67 audio network.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
AoIP4O
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
The Glensound AoIP4O is packaged in a small & rugged format and designed for outside broadcast, theatre and location applications.
The AoIP4O unit is designed to easily and quickly interface existing analogue equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
AoIP4O provides 4 balanced analogue audio Outputs from a Dante®/AES67 audio network.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
The AoIP4O unit is designed to easily and quickly interface existing analogue equipment to a Dante® / AES67 network audio system. Being powered by PoE means that only one cable needs to be connected to the network to carry both audio and power, providing flexibility and saving time on installation.
AoIP4O provides 4 balanced analogue audio Outputs from a Dante®/AES67 audio network.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
Beatrice B1
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE B1 is an ultra compact, robust, portable single channel beltpack intercom designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard ethernet networks. As such the BEATRICE B1 is also fully compatible with other manufacturers’ equipment using the Dante protocol. It is also AES67 compliant.
This small beltpack was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard ethernet networks. As such the BEATRICE B1 is also fully compatible with other manufacturers’ equipment using the Dante protocol. It is also AES67 compliant.
This small beltpack was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice B2
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE B2 is an ultra compact, robust, portable two channel beltpack intercom designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard ethernet networks. As such the BEATRICE B2 is also fully compatible with other manufacturers’ equipment using the Dante protocol. It is also AES67 compliant.
This small beltpack was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard ethernet networks. As such the BEATRICE B2 is also fully compatible with other manufacturers’ equipment using the Dante protocol. It is also AES67 compliant.
This small beltpack was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice B4
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE B4 is a robust, 4* channel beltpack intercom with crystal clear audio designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. As such the BEATRICE B4 is also fully compatible with other manufacturers’ equipment using the Dante protocol.
The Beatrice B4 is also AES67 compliant.
This small desktop unit was deigned to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. As such the BEATRICE B4 is also fully compatible with other manufacturers’ equipment using the Dante protocol.
The Beatrice B4 is also AES67 compliant.
This small desktop unit was deigned to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice B4+ Single User 4 Channel Belt Pack Dante/AES67 Intercom
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
Compact 4ch beltpack using Dante Broadway chipset so can connect to 4 separate network locations. The 4 channels can also work in IFB/loop mode. Each channels talk button can be configured to be momentary, latching or intelligent.
Four separate level controls for monitoring which can be pushed flush with the front panel to further reduce the unit dimensions.
The Beatrice B4+ has remote control functions using the Windows 10 app GlenController.
Headset connection can be 3 pin mic input only, 4 pin headset or 5 pin headset. 6.35mm headphone jack socket.
It is POE powered via a locking EtherCON connector.
Four separate level controls for monitoring which can be pushed flush with the front panel to further reduce the unit dimensions.
The Beatrice B4+ has remote control functions using the Windows 10 app GlenController.
Headset connection can be 3 pin mic input only, 4 pin headset or 5 pin headset. 6.35mm headphone jack socket.
It is POE powered via a locking EtherCON connector.
Beatrice D16
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE D16 is a versatile and fully featured 16 channel desktop intercom with crystal clear audio designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante® network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE D16 is fully compatible with other manufacturers’ equipment using the Dante® and/ or AES67 protocols.
This sixteen channel desktop intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante® network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE D16 is fully compatible with other manufacturers’ equipment using the Dante® and/ or AES67 protocols.
This sixteen channel desktop intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice D4
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE D4 is a robust, 4* channel Desktop intercom with crystal clear audio designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. As such the BEATRICE D4 is also fully compatible with other manufacturers’ equipment using the Dante protocol.
The Beatrice D4 is also AES67 compliant.
This small desktop unit was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
* From a maximum of 2 network locations.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. As such the BEATRICE D4 is also fully compatible with other manufacturers’ equipment using the Dante protocol.
The Beatrice D4 is also AES67 compliant.
This small desktop unit was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
* From a maximum of 2 network locations.
Beatrice D8
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE D8 is a versatile and fully featured 8 channel desktop intercom with crystal clear audio designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE D8 is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This desktop intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE D8 is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This desktop intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice D8 Plus
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Beatrice D8+ is the fully featured Beatrice R8 in a desktop format. It offers 2 x analogue 4 wires, GPIO, local analogue LS output and expansion capability in addition to the standard features on the smaller Beatrice D8. Because the Beatrice D8+ started life as a 1RU rack, it is quite physically large and we would only recommend it if you specifically need the extra facilities mentioned above. For most customers requiring an 8 key desktop unit, the smaller Beatrice D8 will be sufficient.
The Glensound BEATRICE D8+ is a versatile and fully featured 8 channel desktop intercom with crystal clear audio designed for broadcast,theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks, it is also AES67 compliant. As such the BEATRICE D8+ is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This desktop intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
The Glensound BEATRICE D8+ is a versatile and fully featured 8 channel desktop intercom with crystal clear audio designed for broadcast,theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks, it is also AES67 compliant. As such the BEATRICE D8+ is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This desktop intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice LH4
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound Beatrice LH4 provides audio signal present and call indication via a bright LED ‘lighthouse’ tower, providing 360 degrees of visual signalling.
The Beatrice LH4 is ideal for providing cue/ on air indication to presenters and hosts using the standard Dante/AES67 network. The bright lighthouse LED tower can be set to illuminate on receipt of either an audio signal or a Beatrice 20kHz call signal from any of its 4 network audio inputs. Different colours/ brightness can be set by the user to be associated with each channel
To make it easy to send audible instructions to the users an internal speaker and balanced XLR audio output are also included that through the configuration menu can pass audio from the Dante/AES67 network if required.
The Beatrice LH4 is in a compact unit with a bright lighthouse tower for signalling. There is an onboard configuration screen and it is PoE powered.
The Beatrice LH4 is ideal for providing cue/ on air indication to presenters and hosts using the standard Dante/AES67 network. The bright lighthouse LED tower can be set to illuminate on receipt of either an audio signal or a Beatrice 20kHz call signal from any of its 4 network audio inputs. Different colours/ brightness can be set by the user to be associated with each channel
To make it easy to send audible instructions to the users an internal speaker and balanced XLR audio output are also included that through the configuration menu can pass audio from the Dante/AES67 network if required.
The Beatrice LH4 is in a compact unit with a bright lighthouse tower for signalling. There is an onboard configuration screen and it is PoE powered.
Beatrice MIX32
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE MIX32 is a high density audio mixer for producing fixed ratio audio mixes from/ to Dante and AES67 audio networks. It can be fitted with a maximum of 4 independent mixer cards, each with their own redundant network interface. Each of these mixer cards provides 32 audio inputs and 32 audio outputs from the network, with the 32 outputs being mixes derived from different sets of input channels.
A number of different mix-minus mixes can be set on each of 32 x 32 mixer cards. These make it very practical to use as a central intercom mixing hub.
Audio limiter/ compressor circuits are also provided for each output.
The fixed ratio mixes in the Beatrice MIX32 were originally designed to allow network audio intercom systems to generate multiple listening circuits, however the high quality, low noise, digital mixers can be also be used for many other applications.
A priority mode can be used, and when a signal is present on a priority input, all other inputs are dimmed.
Being designed for 24/7 broadcast operation, the Beatrice MIX32 features both redundant power supplies and also redundant network interfaces.
Dante network audio is a common protocol for distributing high quality linear audio over standard IP networks and it is widely used by many audio equipment manufacturers. The Glensound Beatrice Mix32 Dante audio interface will be compatible with any other manufacturers' Dante audio interface.
A number of different mix-minus mixes can be set on each of 32 x 32 mixer cards. These make it very practical to use as a central intercom mixing hub.
Audio limiter/ compressor circuits are also provided for each output.
The fixed ratio mixes in the Beatrice MIX32 were originally designed to allow network audio intercom systems to generate multiple listening circuits, however the high quality, low noise, digital mixers can be also be used for many other applications.
A priority mode can be used, and when a signal is present on a priority input, all other inputs are dimmed.
Being designed for 24/7 broadcast operation, the Beatrice MIX32 features both redundant power supplies and also redundant network interfaces.
Dante network audio is a common protocol for distributing high quality linear audio over standard IP networks and it is widely used by many audio equipment manufacturers. The Glensound Beatrice Mix32 Dante audio interface will be compatible with any other manufacturers' Dante audio interface.
Beatrice P1
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Beatrice P1 is a one channel handset.
The user of the handset on the P1 can listen to one audio feed from the network and send one audio channel out onto the network. Depending upon how the Dante network has been routed, the incoming audio circuit and outgoing audio circuit can be different locations and the outgoing circuit can be routed to multiple locations.
Audio to / from the handset is automatically turned on when the handset is off the hook and turned off when on the hook.
The Beatrice P1 unit is fully compatible with other manufacturers' equipment using the Dante protocol.
The user of the handset on the P1 can listen to one audio feed from the network and send one audio channel out onto the network. Depending upon how the Dante network has been routed, the incoming audio circuit and outgoing audio circuit can be different locations and the outgoing circuit can be routed to multiple locations.
Audio to / from the handset is automatically turned on when the handset is off the hook and turned off when on the hook.
The Beatrice P1 unit is fully compatible with other manufacturers' equipment using the Dante protocol.
Beatrice P2
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Beatrice P2 is a two channel handset.
The user of the handset on the P2 unit can listen to two audio feeds from the network and send two audio channels out onto the network. Depending upon how the Dante® network has been routed, the incoming audio circuits and outgoing circuits can be different locations and the outgoing circuits can be routed to multiple locations.
Audio to the handset from the two network sources is routed to the telephone's earpiece only when the handset is off the hook. When the handset is picked up (off hook), the handset's microphone is turned on and two switches allow the operater to route the handset's microphone to one or other (or both) of the outgoing network audio channels.
The Beatrice P2 unit is fully compatible with other manufacturers' equipment using the Dante® protocol.
The user of the handset on the P2 unit can listen to two audio feeds from the network and send two audio channels out onto the network. Depending upon how the Dante® network has been routed, the incoming audio circuits and outgoing circuits can be different locations and the outgoing circuits can be routed to multiple locations.
Audio to the handset from the two network sources is routed to the telephone's earpiece only when the handset is off the hook. When the handset is picked up (off hook), the handset's microphone is turned on and two switches allow the operater to route the handset's microphone to one or other (or both) of the outgoing network audio channels.
The Beatrice P2 unit is fully compatible with other manufacturers' equipment using the Dante® protocol.
Beatrice R16
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE R16 is a versatile and fully featured 16 channel rackmount intercom with crystal clear audio designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE R16 is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This 1RU rackmount intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE R16 is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This 1RU rackmount intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice R4
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE R4 is a robust, 4* channel 1RU rackmount intercom with crystal clear audio designed for broadcast, theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. As such the BEATRICE R4 is also fully compatible with other manufacturers’ equipment using the Dante protocol.
The Beatrice R4 is also AES67 compliant.
This small desktop unit was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
* From a maximum of of 2 network locations.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. As such the BEATRICE R4 is also fully compatible with other manufacturers’ equipment using the Dante protocol.
The Beatrice R4 is also AES67 compliant.
This small desktop unit was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
* From a maximum of of 2 network locations.
Beatrice R8
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Glensound BEATRICE R8 is a versatile and fully featured 8 channel rackmount intercom with crystal clear audio designed for broadcast,theatre and professional audio applications.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE R8 is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This 1RU rackmount intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
It is part of our Beatrice intercom system that utilises the reliable and proven Dante network audio transmission protocol to allow real time distribution of uncompressed audio across standard networks. It is also AES67 compliant. As such the BEATRICE R8 is fully compatible with other manufacturers’ equipment using the Dante and/ or AES67 protocols.
This 1RU rackmount intercom was designed to be very easy to use for the operator and simple to set up for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems.
Beatrice W1
Support » Monitoring, Control & Communications » Communications » Wired, IP and wireless intercoms and talkback »
The Beatrice W1 is a single user interface for the Beatrice system, designed to mount flush on a wall or into a desk. it is a useful intercom position allowing two way communication to be possible at standing levels whilst mounted into walls, or at desk level if mounted on a Desk.
Bella 22
Support » Monitoring, Control & Communications » Audio Monitoring » Loudspeakers »
Bella 22 is a 2 buss summing matrix confidence monitor designed for outside broadcast trucks, studios, theatre and professional audio applications. It is perfect for easy and cost effective monitoring of multiple network audio sources, ideal for busy production environments and perfect for monitoring multiple network intercom & programme feeds.
The 2 front panel loudspeakers are driven from a DSP to compensate for their size. The result is surprisingly good, with clear crisp vocals and highly intelligible reproduction of wider band audio sources. They are driven from a class D amplifier and have more than sufficient output level for most environments.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
The 2 front panel loudspeakers are driven from a DSP to compensate for their size. The result is surprisingly good, with clear crisp vocals and highly intelligible reproduction of wider band audio sources. They are driven from a class D amplifier and have more than sufficient output level for most environments.
Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free, whilst the low cost and long asset life will keep the accountant satisfied.
Bella 32
Support » Monitoring, Control & Communications » Audio Monitoring » Loudspeakers »
The Bella 32 is a 32 channel audio monitor for Dante / AES67 networks. 32 push buttons directly select the audio to be monitored and selecting multiple inputs mixes them together. There are two presets for setting up separate mix groups.
Each input can be panned between the left and right speakers, and holding any input will solo it. There are signal present LEDs on each input. There are two local analogue inputs that can also be selected for monitoring.
The currently monitored mix is available as left/right stereo or as a separate mix in analogue and AES3.
Two PPM level meters provide output levels, as well as indication of input level and panning position.
Each input can be panned between the left and right speakers, and holding any input will solo it. There are signal present LEDs on each input. There are two local analogue inputs that can also be selected for monitoring.
The currently monitored mix is available as left/right stereo or as a separate mix in analogue and AES3.
Two PPM level meters provide output levels, as well as indication of input level and panning position.
Bella 4
Support » Monitoring, Control & Communications » Audio Monitoring » Loudspeakers »
The Bella 4 is a 4 loudspeaker confidence monitoring rack. it is a new design for 2019 and features the very latest technology in loudspeaker and amplifier design giving it an extended acoustic frequency range (compared to old legacy equivalent products) and low distortion.
The 4 loudspeakers are each fed from their own* Dante® (AES67 compliant) network audio input. Each Channel has a front panel audio presence indicator and input peak Led and an illuminated Channel mute switch. Front panel volume controls adjust the audio output of the speakers and the headphone circuit.
The 4 loudspeakers are each fed from their own* Dante® (AES67 compliant) network audio input. Each Channel has a front panel audio presence indicator and input peak Led and an illuminated Channel mute switch. Front panel volume controls adjust the audio output of the speakers and the headphone circuit.
DARK1616
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The DARK1616 is a versatile break in/ out box for sending/ receiving both analogue and digital (AES3) audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.
In total there are 16 channels of audio sent from the Dark1616 into the network. The Dark1616 has 8 of AES3 inputs and 16 of analogue inputs, but they cannot be used at the same time. The AES3 inputs take priority over the analogue (if an AES3 input is receiving a valid AES3 signal then it will turn off the equivalent analogue input pair and route its output to the network).
Simultaneously there are 16 channels of audio being received from the network by the Dark1616 and these incoming circuits are provided as outputs from the Dark1616 in both AES3 and analogue.
The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz. The incoming AES3 circuit is always sample rate converted to match the Dante network frequency.
The AES3 outputs are locked to the sample frequency of the Dante network.
For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Being designed for resilient broadcast applications the Dark1616 features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616 lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.
On the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.
In total there are 16 channels of audio sent from the Dark1616 into the network. The Dark1616 has 8 of AES3 inputs and 16 of analogue inputs, but they cannot be used at the same time. The AES3 inputs take priority over the analogue (if an AES3 input is receiving a valid AES3 signal then it will turn off the equivalent analogue input pair and route its output to the network).
Simultaneously there are 16 channels of audio being received from the network by the Dark1616 and these incoming circuits are provided as outputs from the Dark1616 in both AES3 and analogue.
The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz. The incoming AES3 circuit is always sample rate converted to match the Dante network frequency.
The AES3 outputs are locked to the sample frequency of the Dante network.
For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Being designed for resilient broadcast applications the Dark1616 features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616 lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.
On the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.
DARK1616D
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The Glensound DARK1616D is a digital to digital converter designed to connect AES3 audio circuits to/ from a Dante/ AES67 audio network.
Dante network audio is a common protocol for distributing high quality linear audio over standard IP networks and it is widely used by many audio equipment manufacturers. The Glensound Dark1616D Dante audio interface will be compatible with any other manufacturers Dante audio interface.
Being designed for live on-air broadcast applications the Glensound Dark1616D has been designed with multiple redundancy capabilities. It has 2 mains power sources and it also has fully redundant network connections for both Copper & Fibre circuits.
The Dark1616D provides 8 balanced AES3 inputs and 8 balanced AES3 outputs to the Dante network on rear panel D25 connectors wired to AES59 (also known as the Tascam standard).
As per our other Dante equipment 0dBu = -18dBFs
Dante network audio is a common protocol for distributing high quality linear audio over standard IP networks and it is widely used by many audio equipment manufacturers. The Glensound Dark1616D Dante audio interface will be compatible with any other manufacturers Dante audio interface.
Being designed for live on-air broadcast applications the Glensound Dark1616D has been designed with multiple redundancy capabilities. It has 2 mains power sources and it also has fully redundant network connections for both Copper & Fibre circuits.
The Dark1616D provides 8 balanced AES3 inputs and 8 balanced AES3 outputs to the Dante network on rear panel D25 connectors wired to AES59 (also known as the Tascam standard).
As per our other Dante equipment 0dBu = -18dBFs
DARK1616M
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The DARK 1616M is an audio interface that provides connections to a Dante or AES67 IP network.
Inputs & OutputsThere are 16 digital inputs and 16 digital outputs, available on 8 x AES connections. There are also 16 analogue inputs and 16 analogue outputs, available in parallel to the digital. The analogue and digital outputs are always available, and the inputs work either on AES input priority or can be selected via the control app.
The AES3 inputs have sample rate converters and can accept input frequencies up to 192kHz. The incoming AES3 circuit is always sample rate converted to match the Dante network frequency. The AES3 outputs are locked to the sample frequency of the Dante network.
ConnectionsFor ease of cabling, audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Remote Mic AmpsThe analogue inputs are switchable between mic, line and 48v phantom power. These are controllable across the network via the Dark Controller Windows 10 app. This allows remote control by an engineer of the input selection, gain adjust and input on/off control. A meter is also provided for each input on the app, to monitor the input level and help with setting the gain.
Network InterfaceBeing designed for resilient broadcast applications the Dark1616M features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616M lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.
AlarmsOn the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.
Inputs & OutputsThere are 16 digital inputs and 16 digital outputs, available on 8 x AES connections. There are also 16 analogue inputs and 16 analogue outputs, available in parallel to the digital. The analogue and digital outputs are always available, and the inputs work either on AES input priority or can be selected via the control app.
The AES3 inputs have sample rate converters and can accept input frequencies up to 192kHz. The incoming AES3 circuit is always sample rate converted to match the Dante network frequency. The AES3 outputs are locked to the sample frequency of the Dante network.
ConnectionsFor ease of cabling, audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Remote Mic AmpsThe analogue inputs are switchable between mic, line and 48v phantom power. These are controllable across the network via the Dark Controller Windows 10 app. This allows remote control by an engineer of the input selection, gain adjust and input on/off control. A meter is also provided for each input on the app, to monitor the input level and help with setting the gain.
Network InterfaceBeing designed for resilient broadcast applications the Dark1616M features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616M lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.
AlarmsOn the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.
DARK1616S
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The DARK1616S is a versatile break in/ out box for sending/ receiving both analogue and digital (AES3) audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.
In total there are 16 channels of audio sent from the Dark1616S into the network. The Dark1616S has 8 of AES3 inputs and 16 of analogue inputs. Using the GlenController App it is possible to set which inputs (AES3 or Analogue) are being sent to the Dante/ AES67 network, an auto mode is available that sends AES3 (when a valid signal is detected) in preference to the analogue input.
Simultaneously there are 16 channels of audio being received from the network by the Dark1616S and these incoming circuits are provided as outputs from the Dark1616S in both AES3 and analogue.
The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz, the incoming AES3 circuit is always sample rate converted to match the Dante network frequency.
The AES3 outputs have sample rate converters on them and can be locked to the sample frequency of the Word clock input, the first AES3 input or to the Dante network.
A Word clock output is also provided, this is clocked at the same sample rate as the Dante network and can be used for locking external equipment to the network's sample rate.
The unit can be remote controlled using our GlenController Windows 10 App. The App allows such things as the dBFs levels to be set, clock masters and also provides input/ output metering. See the App tab for further details.
For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Being designed for resilient broadcast applications, the Dark1616S features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616S lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616S units can be directly connected with each other with no external network hardware.
In total there are 16 channels of audio sent from the Dark1616S into the network. The Dark1616S has 8 of AES3 inputs and 16 of analogue inputs. Using the GlenController App it is possible to set which inputs (AES3 or Analogue) are being sent to the Dante/ AES67 network, an auto mode is available that sends AES3 (when a valid signal is detected) in preference to the analogue input.
Simultaneously there are 16 channels of audio being received from the network by the Dark1616S and these incoming circuits are provided as outputs from the Dark1616S in both AES3 and analogue.
The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz, the incoming AES3 circuit is always sample rate converted to match the Dante network frequency.
The AES3 outputs have sample rate converters on them and can be locked to the sample frequency of the Word clock input, the first AES3 input or to the Dante network.
A Word clock output is also provided, this is clocked at the same sample rate as the Dante network and can be used for locking external equipment to the network's sample rate.
The unit can be remote controlled using our GlenController Windows 10 App. The App allows such things as the dBFs levels to be set, clock masters and also provides input/ output metering. See the App tab for further details.
For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Being designed for resilient broadcast applications, the Dark1616S features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616S lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616S units can be directly connected with each other with no external network hardware.
DARK16I
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The DARK16I is a versatile box for converting analogue audio inputs to an AoIP system utilising the Dante and AES67 audio over IP (AoIP) protocols.
In total there are 16 audio input channels, all being simultaneously converted to the AoIP network via high quality low noise analogue to digital converters.
For ease of cabling audio inputs are balanced circuits on Neutrik XLRs.
Being designed for resilient broadcast applications, the DARK16I features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16I lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16I units can be directly connected with each other with no external network hardware.
In total there are 16 audio input channels, all being simultaneously converted to the AoIP network via high quality low noise analogue to digital converters.
For ease of cabling audio inputs are balanced circuits on Neutrik XLRs.
Being designed for resilient broadcast applications, the DARK16I features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16I lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16I units can be directly connected with each other with no external network hardware.
DARK16O
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The DARK16O is a versatile break out box for receiving analogue audio from a network utilising the Dante and AES67 audio over IP (AoIP) protocols.
In total there are 16 channels of audio being received from the AoIP network and converted via low noise DACs (Digital to Analogue Converters) to balanced analogue outputs.
For ease of cabling audio outputs are presented as balanced circuits on Neutrik XLRs.
Being designed for resilient broadcast applications the DARK16O features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16O lost its primary link circuit, the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16O units can be directly connected with each other with no external network hardware.
In total there are 16 channels of audio being received from the AoIP network and converted via low noise DACs (Digital to Analogue Converters) to balanced analogue outputs.
For ease of cabling audio outputs are presented as balanced circuits on Neutrik XLRs.
Being designed for resilient broadcast applications the DARK16O features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16O lost its primary link circuit, the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16O units can be directly connected with each other with no external network hardware.
DARK88 MKII
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The DARK88 MKII is a versatile break in/ out box for sending/ receiving analogue audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.
In total there are 8 channels of audio sent from the DARK88 MKII into the network. The DARK88 MKII has 8 of analogue electronically balanced audio inputs on Neutrik XLRs.
Simultaneously there are 8 channels of audio being received from the network by the DARK88 MKII and these incoming circuits are provided as outputs from the DARK88 MKII in analogue.
Network sample rates of up to 192KHz are accommodated seamlessly within the DARK88 MKII.
Being designed for resilient broadcast applications the DARK88 MKII features both redundant power supplies and redundant Dante network links. Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK88 MKII lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK88 MKII units can be directly connected with each other with no external network hardware.
On the front panel, LEDs indicate the status of the 2 power supplies and the 2 network links. GPO status outputs are also provided for external indication of the power supply & network status.
Network connections are placed on the front panel of the DARK88 MKII in order that the network cables (or fibres) match those of a rack mounted professional network switch, making installation and tracing interconnecting cables easy. Fibre connections are via SFP slots, meaning that users can select their own preferred fibre type & connector style by installing their own fibre SFP modules (a selection of modules is available from Glensound if preferred).
The DARK88 MKII features the Brooklyn module from Audinate which is AES67 compliant.
In total there are 8 channels of audio sent from the DARK88 MKII into the network. The DARK88 MKII has 8 of analogue electronically balanced audio inputs on Neutrik XLRs.
Simultaneously there are 8 channels of audio being received from the network by the DARK88 MKII and these incoming circuits are provided as outputs from the DARK88 MKII in analogue.
Network sample rates of up to 192KHz are accommodated seamlessly within the DARK88 MKII.
Being designed for resilient broadcast applications the DARK88 MKII features both redundant power supplies and redundant Dante network links. Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK88 MKII lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK88 MKII units can be directly connected with each other with no external network hardware.
On the front panel, LEDs indicate the status of the 2 power supplies and the 2 network links. GPO status outputs are also provided for external indication of the power supply & network status.
Network connections are placed on the front panel of the DARK88 MKII in order that the network cables (or fibres) match those of a rack mounted professional network switch, making installation and tracing interconnecting cables easy. Fibre connections are via SFP slots, meaning that users can select their own preferred fibre type & connector style by installing their own fibre SFP modules (a selection of modules is available from Glensound if preferred).
The DARK88 MKII features the Brooklyn module from Audinate which is AES67 compliant.
DARK8ADI
Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »
The DARK8ADI (Analogue & Digital Input) is a very powerful Dante®/ AES67 network audio interface, in a robust 1RU 19" subrack It has a 8 of low noise analogue line inputs and 4 of transformer balanced AES3 digital audio inputs. These inputs are transmitted on a fully redundant Dante/ AES67 network interface.
In total 16 audio circuits are sent to the Dante/ AES67 network. Eight of these circuits are always derived from the 4 x AES3 inputs, the other eight are automatically switched between the AES3 and analogue inputs. If the DARK8ADI detects valid AES3 signal on a channel then this will be routed to the network output. If no valid AES3 is detected then the analogue input will be routed to the network output instead of the AES3.
The rugged design, redundant mains powering & redundant network facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.
Eight Analogue Audio InputsThe DARK8ADI has 8 electronically balanced analogue line level audio inputs. Each input is on its own Neutrik 3 pin XLR socket.Quality Analogue To Digital ConvertersTo get the best possible results from your analogue audio inputs, the very best widest range analogue to digital converters (ADCs) currently available are used to make sure the digital audio on your network is as good as it possibly can be.Four AES3 Audio InputsThe DARK8ADI has 4 transformer balanced digital AES3 audio inputs. Each input is on its own Neutrik 3 pin XLR socket. These inputs can accept sample rates up to 192kHz.
Analogue/ AES3 Auto SwitchingEight of the audio outputs are derived from both the 4 x AES3 inputs and the 8 analogue inputs, whereby the AES3 inputs take priority to the analogue circuits. Our input circuitry looks for valid AES3 data streams on the AES3 inputs. If one is detected then the pair of audio outputs (AES3 is two audio channels) associated with that circuit will be routed to the Dante/ AES67 output and if no valid AES3 signal is detected then the analogue input will be routed to that output.
AES3 Network OutputsAs well as the switched outputs, the outputs of the 4 x AES3 input circuits are always routed to 8 audio channels of the AoIP network.
Sample Rate ConvertersThe AES3 audio inputs are fed through sample rate converters so that they match the AoIP networks sample frequency. The AoIP network supports up to 192kHz sampling and the AES3 inputs can be between 32 & 192kHz.
Network InterfaceThere are 4 network interfaces on the DARK8ADI. There are 2 x Neutrik Ethercon (RJ45) connectors and there are also 2 x SFP slots for customers to fit their own preferred fibre interfaces.
Redundant Network InterfaceWhen using the Dante protocol it is possible to set the DARK8ADI to have a fully redundant network interface whereby a completely glitch free automatic redundant audio network link is provided across 2 of the network interfaces.
PowerEach DARK8ADI can be powered from two independent sources to provide a multi-redundant power option.Two wide range switch mode power supplies are fitted as standard to provide redundant mains power supplies.
Alarms & LEDsThe front panel features two LEDs (one indicating OK and the other fault) for both power supplies and both network ports. This data OK/ Fault information is also provided on solid state relay outputs on front panel D connectors.
DARK8MAI MKII
Produce » Real-Time Production » Production Services » Studio and stage facilities »
The DARK8MAI MKII (Microphone Analogue Input) is a very powerful Dante/ AES67 network audio interface, in a robust stage box/ throw down style. It has a 8 of low noise mic/line inputs which are remotely controlled by a Windows 10 app. The outputs of the microphone amplifiers are transmitted on a fully redundant Dante/ AES67 network interface.
Having 8 of low noise high quality microphone inputs makes the DARK8MAI MKII perfect for transporting audio circuits from a stage or a pitch side position to a mixer or other location within an audio network .
The rugged design, PoE powering & remote control facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.
Having 8 of low noise high quality microphone inputs makes the DARK8MAI MKII perfect for transporting audio circuits from a stage or a pitch side position to a mixer or other location within an audio network .
The rugged design, PoE powering & remote control facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.
DARK8MAIR
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
The DARK8MAIR (Microphone Analogue Input Rackmount) is a very powerful Dante/ AES67 network audio interface, in a robust stage box/ throw down style. It has a 8 of low noise mic/line inputs which are remotely controlled by a Windows 10 app. The outputs of the microphone amplifiers are transmitted on a fully redundant Dante/ AES67 network interface.
Having 8 of low noise high quality microphone inputs makes the DARK8MAIR perfect for transporting audio circuits from a stage or a pitch side position to a mixer or other location within an audio network.
The rugged design, PoE/ Mains powering & remote control facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.
Having 8 of low noise high quality microphone inputs makes the DARK8MAIR perfect for transporting audio circuits from a stage or a pitch side position to a mixer or other location within an audio network.
The rugged design, PoE/ Mains powering & remote control facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.
Divine
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
Divine is a new concept in powered network audio monitors. It is housed in a hard-wearing fully sealed die cast enclosure, is powered by Power over Ethernet (PoE) and boasts the very latest low noise high output class D power amplifier.
Internally a Digital Signal Processor (DSP) takes careful care of the audio signals, including state of the art compression and limiting circuits, while a microprocessor provides full setup and control via a small rear panel LCD. Control of setup and day
to day operation of the Divine will also be available on our Windows 10 application GlenController including the ability to group multiple Divines together and control their level simultaneously.
Divine can receive up to four* Dante (AES67 compliant) audio over IP (AoIP) inputs. These inputs can be selected by the user on a large clear front panel select switch. The four audio inputs can also be easily mixed together and their individual levels adjusted.
A priority system is provided to allow one (or more) of the inputs to automatically duck another. This can be very useful if you want to monitor one source but also listen to another when audio is present, such as sending show relay to dressing rooms in a theatre but having the stage manager's call and building fire alarm take precedence when they’re active.
The die cast enclosure has been carefully designed to provide full protection of all control knobs, switches and ports to prevent damage. Uniquely, the housing also features a standard PC screen Vesa mount, meaning that you can purchase any standard Vesa mounting solution to hang/ mount your Divine on, saving you lots of money. Standard microphone stand threads are also provided in the base for an alternative support solution.
Different preset EQs and an LF cut can all be set in the user menu to allow the Divine to be used for a variety of applications.
Divine is so much more than just another powered loudspeaker.
Internally a Digital Signal Processor (DSP) takes careful care of the audio signals, including state of the art compression and limiting circuits, while a microprocessor provides full setup and control via a small rear panel LCD. Control of setup and day
to day operation of the Divine will also be available on our Windows 10 application GlenController including the ability to group multiple Divines together and control their level simultaneously.
Divine can receive up to four* Dante (AES67 compliant) audio over IP (AoIP) inputs. These inputs can be selected by the user on a large clear front panel select switch. The four audio inputs can also be easily mixed together and their individual levels adjusted.
A priority system is provided to allow one (or more) of the inputs to automatically duck another. This can be very useful if you want to monitor one source but also listen to another when audio is present, such as sending show relay to dressing rooms in a theatre but having the stage manager's call and building fire alarm take precedence when they’re active.
The die cast enclosure has been carefully designed to provide full protection of all control knobs, switches and ports to prevent damage. Uniquely, the housing also features a standard PC screen Vesa mount, meaning that you can purchase any standard Vesa mounting solution to hang/ mount your Divine on, saving you lots of money. Standard microphone stand threads are also provided in the base for an alternative support solution.
Different preset EQs and an LF cut can all be set in the user menu to allow the Divine to be used for a variety of applications.
Divine is so much more than just another powered loudspeaker.
Expedition
Connect » Provide Connectivity » Connectivity & Bandwidth » Mobile/cellular connectivity »
The Expedition is a portable 2 channel, digital Broadcasters’ Phone with 7kHz HD Voice audio. It is used for live news and sport reporting or talkback situations. HD Voice is available on compatible 4G LTE and 3G UMTS networks. Narrowband calls are also available on 2G GSM, 3G UMTS, and 4G LTE networks.
Multiple NetworksA 2G GSM, 3G UMTS, and 4G LTE mobile phone with features specifically for broadcast.
High Quality AudioThe HD Voice system allows 7kHz mobile calls on compatible mobile networks using 3G UMTS or 4G LTE networks (see page 3).
Two InputsThere are independent, duplicated controls for both inputs. The illuminated on air button can be configured as latching, momentary or intelligent. There is a small gain control for setting input level.
Two Headphone OutputsThere are two x 6.35mm headphone sockets with a level adjust.
Powering OptionBattery power is available from 6 x AA batteries. There is also an external DC input from 10 - 15V.
Auto AnswerThe Expedition can be set to answer incoming calls automatically.
Output CompressorAn output compressor prevents any peak in the audio being sent onto the HD Voice network. HD Voice is very sensitive to peaks and the output compressor improves the quality of the audio received at the remote end.
Glensound ‘Referee’ Input CompressorGlensound’s Referee compressor has been specially designed and updated over many years, specifically to cope with the excessive peaks often associated with excited sports commentators.
Background Noise SuppressorHD Voice features a background noise suppressor that greatly removes unwanted background noise.
Direct Line OutputA 3 pin XLR plug provides a clean output of the remote end of the call, with it’s own adjustable level control. This makes the Expedition a portable digital hybrid for a mobile telephone.
Diversity AerialsThe Expedition has twin diversity aerials for improved reception and range when on the 4G LTE networks.
Included AccessoriesEach Expedition comes with the following accessories as standard: hard plastic case, plug top power supply, shoulder strap, pair of small SMA aerials, pair of performance SMA aerials.
Multiple NetworksA 2G GSM, 3G UMTS, and 4G LTE mobile phone with features specifically for broadcast.
High Quality AudioThe HD Voice system allows 7kHz mobile calls on compatible mobile networks using 3G UMTS or 4G LTE networks (see page 3).
Two InputsThere are independent, duplicated controls for both inputs. The illuminated on air button can be configured as latching, momentary or intelligent. There is a small gain control for setting input level.
Two Headphone OutputsThere are two x 6.35mm headphone sockets with a level adjust.
Powering OptionBattery power is available from 6 x AA batteries. There is also an external DC input from 10 - 15V.
Auto AnswerThe Expedition can be set to answer incoming calls automatically.
Output CompressorAn output compressor prevents any peak in the audio being sent onto the HD Voice network. HD Voice is very sensitive to peaks and the output compressor improves the quality of the audio received at the remote end.
Glensound ‘Referee’ Input CompressorGlensound’s Referee compressor has been specially designed and updated over many years, specifically to cope with the excessive peaks often associated with excited sports commentators.
Background Noise SuppressorHD Voice features a background noise suppressor that greatly removes unwanted background noise.
Direct Line OutputA 3 pin XLR plug provides a clean output of the remote end of the call, with it’s own adjustable level control. This makes the Expedition a portable digital hybrid for a mobile telephone.
Diversity AerialsThe Expedition has twin diversity aerials for improved reception and range when on the 4G LTE networks.
Included AccessoriesEach Expedition comes with the following accessories as standard: hard plastic case, plug top power supply, shoulder strap, pair of small SMA aerials, pair of performance SMA aerials.
Express Box MKII
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
The Express Box MKII was designed to provide a quick and easy solution when it is necessary to provide interfacing for two commentators, with simple facilities. This high quality commentary unit is ideal for those looking for cost effective solutions without paying for unwanted features; the Express Box MKII is worthy of investigation.
InputsTwo front panel mic inputs with selectable 48v phantom power.
Outputs/TalkbackEach mic input has its own individual output. There is also a mixed output of the two mic inputs. The two talkback circuits have individual outputs and are common for both commentators.
MonitoringThere are four inputs for external sources, one sidetone pot of their own voice and one other commentator pot for listening to the co-commentator. These are available independently to each commentator on individual pots, so each can adjust the inputs for their own preference of mix level. There are two 6.35mm headphone sockets – 1 for each commentator. A 7 segment LED PPM meter displays level.
InputsTwo front panel mic inputs with selectable 48v phantom power.
Outputs/TalkbackEach mic input has its own individual output. There is also a mixed output of the two mic inputs. The two talkback circuits have individual outputs and are common for both commentators.
MonitoringThere are four inputs for external sources, one sidetone pot of their own voice and one other commentator pot for listening to the co-commentator. These are available independently to each commentator on individual pots, so each can adjust the inputs for their own preference of mix level. There are two 6.35mm headphone sockets – 1 for each commentator. A 7 segment LED PPM meter displays level.
Express ip MINI
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
The Express ip Mini is a compact simple to operate, 2 user network audio commentary box, that is ideal for both the occasional & the daily provider of commentary facilities.
Its low cost keeps the accountant happy, its simple facilities makes it very easy to setup for the engineer and the familiar Glensound control surface keeps the commentators satisfied.
Having been involved in designing and manufacturing commentary boxes for over 40 years the Express ip Mini is a culmination of all our years of experience and has been designed to provide just enough of the basic facilities required for everyday commentary in a very robust & easy to use package.
Its low cost keeps the accountant happy, its simple facilities makes it very easy to setup for the engineer and the familiar Glensound control surface keeps the commentators satisfied.
Having been involved in designing and manufacturing commentary boxes for over 40 years the Express ip Mini is a culmination of all our years of experience and has been designed to provide just enough of the basic facilities required for everyday commentary in a very robust & easy to use package.
Express ip MKII
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
The Express ip MKII was designed to provide a quick and easy solution when it is necessary to provide interfacing for two commentators, with simple facilities, into a Dante/ AES67 audio network. This high quality commentary unit is ideal for those looking for cost effective solutions without paying for unwanted features; the Express ip MKII is worthy of investigation.
InputsTwo front panel mic inputs with selectable 48v phantom power.
Outputs/TalkbackEach mic input has its own individual output. The second mic output can be switched on the rear panel to be a mix of both mic inputs. The two talkback circuits have individual outputs and are common for both commentators.
MonitoringThere are four inputs for external sources, and one sidetone pot of their own voice and another pot for monitoring the other commentator. These are available independently to each commentator on individual pots, so each can adjust the inputs for their own preference of mix level. There are two 6.35mm headphone sockets – 1 for each commentator. A 7 segment LED PPM meter displays level.
NetworkThe network connection is AES67/Dante compatible and is available on the rear panel on an RJ45/CAT5 connection. It offers 4 input channels and 4 output channels.
PowerThere is an internal switch mode AC power supply, or the Express ip can be powered by PoE if powered by a switch that provides power over Ethernet.
InputsTwo front panel mic inputs with selectable 48v phantom power.
Outputs/TalkbackEach mic input has its own individual output. The second mic output can be switched on the rear panel to be a mix of both mic inputs. The two talkback circuits have individual outputs and are common for both commentators.
MonitoringThere are four inputs for external sources, and one sidetone pot of their own voice and another pot for monitoring the other commentator. These are available independently to each commentator on individual pots, so each can adjust the inputs for their own preference of mix level. There are two 6.35mm headphone sockets – 1 for each commentator. A 7 segment LED PPM meter displays level.
NetworkThe network connection is AES67/Dante compatible and is available on the rear panel on an RJ45/CAT5 connection. It offers 4 input channels and 4 output channels.
PowerThere is an internal switch mode AC power supply, or the Express ip can be powered by PoE if powered by a switch that provides power over Ethernet.
GS-CU001B
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
The GS-CU001B MkII is a highly versatile commentary unit. It was designed after close consultation with end users and the result is a stand alone unit which offers a wide range of facilities, and it is equally at home on the terraces, as it is in a studio.
The MkII VersionThe MkII version now adds the following as standard features:
- 48v phantom power
- upgraded input gain pot
- input gain range increased from -20 to +10 dB
- GPOs on Mic and talkback buttons to allow integration with main intercom or talkback system
VERSION 3 (TT): This version has transformer balanced inputs & outputs with high quality Llundhall transformers on the individual mic outputs.
InputsThree front panel inputs are mic/line switchable with selectable 48v phantom power. The front panel also has a small gain adjustment pot. Mic on/off switches can be selected in on/off mode or in cough mode. There is a global low frequency cut that can be selected on or off, and a preset compressor/limiter per input.
Outputs/TalkbackEach input has a direct discrete output on the rear panel along with a mixed output of all of the inputs, all on XLR. The mixed output is also available on an un-balanced 3.5mm jack socket as a local record. The direct output levels can be set in 3 positions:
1: 0dB + limiter: This is the normal operation and it limits peak levels.
2: 0dB: This feeds the output with an un-compressed nominal level of 0dB, for when the peak signal level will be controlled by outboard equipment.
3: -20dB: This feeds the output with a nominal level of -20dB providing extra headroom.
There are 3 common talkback circuits with individual buttons on the two main commentators’ sections (Position B's input cannot be switched to a talkback output). The operation of the talkback buttons features Config+ for configuring in different modes.
MonitoringThere are two main commentator monitoring sections, each with two separate 6.35mm A or B gauge jack sockets for headphones. In this way, the centre position B commentator can choose to share the monitoring of commentator A or C. There are 5 common external sources available for monitoring, plus an additional control which is the sum of the other 2 commentators/inputs. Each of the 6 pots has variable level control and left/both/right switching allowing commentator A and C to achieve their desired mix and balance levels. The sidetone control is located on the rear panel for commentator A and C to adjust the level of their own voice. There is a 7 segment LED PPM meter.
PowerThere is an internal switch mode power supply 100-250v AC, with external power via a 4 pin XLR 9-18v DC.
ModificationsThe GS-CU001 is a complete and versatile base system, and Glensound's most popular commentary unit. It is therefore a perfect starting point for custom requirements. This has resulted in many custom modified units, some listed above, and some not. We are always happy to investigate a particular requirement that you may have. To give you an idea of what is possible, these are some of the modifications, we have designed previously for others:
Added GPIO on a 9 pin D-typeMoved the sidetone control to the top panel on a full size potAdded a talkback channel and monitor between the A and C commentary positions (GS-CU001D and E)Independent monitoring inputs for commentator A and C (GS-CU001E)Additional mic passive outputs (GS-CU001E)Added an additional headphone monitoring input (GS-CU001G)Added two mic inputs for commentator A and C with a simple toggle switch between them (GS-CU001L)
Other Versions are available:
The MkII VersionThe MkII version now adds the following as standard features:
- 48v phantom power
- upgraded input gain pot
- input gain range increased from -20 to +10 dB
- GPOs on Mic and talkback buttons to allow integration with main intercom or talkback system
VERSION 3 (TT): This version has transformer balanced inputs & outputs with high quality Llundhall transformers on the individual mic outputs.
InputsThree front panel inputs are mic/line switchable with selectable 48v phantom power. The front panel also has a small gain adjustment pot. Mic on/off switches can be selected in on/off mode or in cough mode. There is a global low frequency cut that can be selected on or off, and a preset compressor/limiter per input.
Outputs/TalkbackEach input has a direct discrete output on the rear panel along with a mixed output of all of the inputs, all on XLR. The mixed output is also available on an un-balanced 3.5mm jack socket as a local record. The direct output levels can be set in 3 positions:
1: 0dB + limiter: This is the normal operation and it limits peak levels.
2: 0dB: This feeds the output with an un-compressed nominal level of 0dB, for when the peak signal level will be controlled by outboard equipment.
3: -20dB: This feeds the output with a nominal level of -20dB providing extra headroom.
There are 3 common talkback circuits with individual buttons on the two main commentators’ sections (Position B's input cannot be switched to a talkback output). The operation of the talkback buttons features Config+ for configuring in different modes.
MonitoringThere are two main commentator monitoring sections, each with two separate 6.35mm A or B gauge jack sockets for headphones. In this way, the centre position B commentator can choose to share the monitoring of commentator A or C. There are 5 common external sources available for monitoring, plus an additional control which is the sum of the other 2 commentators/inputs. Each of the 6 pots has variable level control and left/both/right switching allowing commentator A and C to achieve their desired mix and balance levels. The sidetone control is located on the rear panel for commentator A and C to adjust the level of their own voice. There is a 7 segment LED PPM meter.
PowerThere is an internal switch mode power supply 100-250v AC, with external power via a 4 pin XLR 9-18v DC.
ModificationsThe GS-CU001 is a complete and versatile base system, and Glensound's most popular commentary unit. It is therefore a perfect starting point for custom requirements. This has resulted in many custom modified units, some listed above, and some not. We are always happy to investigate a particular requirement that you may have. To give you an idea of what is possible, these are some of the modifications, we have designed previously for others:
Added GPIO on a 9 pin D-typeMoved the sidetone control to the top panel on a full size potAdded a talkback channel and monitor between the A and C commentary positions (GS-CU001D and E)Independent monitoring inputs for commentator A and C (GS-CU001E)Additional mic passive outputs (GS-CU001E)Added an additional headphone monitoring input (GS-CU001G)Added two mic inputs for commentator A and C with a simple toggle switch between them (GS-CU001L)
Other Versions are available:
GS-CU001B
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
The GS-CU001B MkII is a highly versatile commentary unit. It was designed after close consultation with end users and the result is a stand alone unit which offers a wide range of facilities, and it is equally at home on the terraces, as it is in a studio.
The MkII VersionThe MkII version now adds the following as standard features:
- 48v phantom power
- upgraded input gain pot
- input gain range increased from -20 to +10 dB
- GPOs on Mic and talkback buttons to allow integration with main intercom or talkback system
VERSION 3 (TT): This version has transformer balanced inputs & outputs with high quality Llundhall transformers on the individual mic outputs.
InputsThree front panel inputs are mic/line switchable with selectable 48v phantom power. The front panel also has a small gain adjustment pot. Mic on/off switches can be selected in on/off mode or in cough mode. There is a global low frequency cut that can be selected on or off, and a preset compressor/limiter per input.
Outputs/TalkbackEach input has a direct discrete output on the rear panel along with a mixed output of all of the inputs, all on XLR. The mixed output is also available on an un-balanced 3.5mm jack socket as a local record. The direct output levels can be set in 3 positions:
1: 0dB + limiter: This is the normal operation and it limits peak levels.
2: 0dB: This feeds the output with an un-compressed nominal level of 0dB, for when the peak signal level will be controlled by outboard equipment.
3: -20dB: This feeds the output with a nominal level of -20dB providing extra headroom.
There are 3 common talkback circuits with individual buttons on the two main commentators’ sections (Position B's input cannot be switched to a talkback output). The operation of the talkback buttons features Config+ for configuring in different modes.
MonitoringThere are two main commentator monitoring sections, each with two separate 6.35mm A or B gauge jack sockets for headphones. In this way, the centre position B commentator can choose to share the monitoring of commentator A or C. There are 5 common external sources available for monitoring, plus an additional control which is the sum of the other 2 commentators/inputs. Each of the 6 pots has variable level control and left/both/right switching allowing commentator A and C to achieve their desired mix and balance levels. The sidetone control is located on the rear panel for commentator A and C to adjust the level of their own voice. There is a 7 segment LED PPM meter.
PowerThere is an internal switch mode power supply 100-250v AC, with external power via a 4 pin XLR 9-18v DC.
ModificationsThe GS-CU001 is a complete and versatile base system, and Glensound's most popular commentary unit. It is therefore a perfect starting point for custom requirements. This has resulted in many custom modified units, some listed above, and some not. We are always happy to investigate a particular requirement that you may have. To give you an idea of what is possible, these are some of the modifications, we have designed previously for others:
Added GPIO on a 9 pin D-typeMoved the sidetone control to the top panel on a full size potAdded a talkback channel and monitor between the A and C commentary positions (GS-CU001D and E)Independent monitoring inputs for commentator A and C (GS-CU001E)Additional mic passive outputs (GS-CU001E)Added an additional headphone monitoring input (GS-CU001G)Added two mic inputs for commentator A and C with a simple toggle switch between them (GS-CU001L)
Other Versions are available:
The MkII VersionThe MkII version now adds the following as standard features:
- 48v phantom power
- upgraded input gain pot
- input gain range increased from -20 to +10 dB
- GPOs on Mic and talkback buttons to allow integration with main intercom or talkback system
VERSION 3 (TT): This version has transformer balanced inputs & outputs with high quality Llundhall transformers on the individual mic outputs.
InputsThree front panel inputs are mic/line switchable with selectable 48v phantom power. The front panel also has a small gain adjustment pot. Mic on/off switches can be selected in on/off mode or in cough mode. There is a global low frequency cut that can be selected on or off, and a preset compressor/limiter per input.
Outputs/TalkbackEach input has a direct discrete output on the rear panel along with a mixed output of all of the inputs, all on XLR. The mixed output is also available on an un-balanced 3.5mm jack socket as a local record. The direct output levels can be set in 3 positions:
1: 0dB + limiter: This is the normal operation and it limits peak levels.
2: 0dB: This feeds the output with an un-compressed nominal level of 0dB, for when the peak signal level will be controlled by outboard equipment.
3: -20dB: This feeds the output with a nominal level of -20dB providing extra headroom.
There are 3 common talkback circuits with individual buttons on the two main commentators’ sections (Position B's input cannot be switched to a talkback output). The operation of the talkback buttons features Config+ for configuring in different modes.
MonitoringThere are two main commentator monitoring sections, each with two separate 6.35mm A or B gauge jack sockets for headphones. In this way, the centre position B commentator can choose to share the monitoring of commentator A or C. There are 5 common external sources available for monitoring, plus an additional control which is the sum of the other 2 commentators/inputs. Each of the 6 pots has variable level control and left/both/right switching allowing commentator A and C to achieve their desired mix and balance levels. The sidetone control is located on the rear panel for commentator A and C to adjust the level of their own voice. There is a 7 segment LED PPM meter.
PowerThere is an internal switch mode power supply 100-250v AC, with external power via a 4 pin XLR 9-18v DC.
ModificationsThe GS-CU001 is a complete and versatile base system, and Glensound's most popular commentary unit. It is therefore a perfect starting point for custom requirements. This has resulted in many custom modified units, some listed above, and some not. We are always happy to investigate a particular requirement that you may have. To give you an idea of what is possible, these are some of the modifications, we have designed previously for others:
Added GPIO on a 9 pin D-typeMoved the sidetone control to the top panel on a full size potAdded a talkback channel and monitor between the A and C commentary positions (GS-CU001D and E)Independent monitoring inputs for commentator A and C (GS-CU001E)Additional mic passive outputs (GS-CU001E)Added an additional headphone monitoring input (GS-CU001G)Added two mic inputs for commentator A and C with a simple toggle switch between them (GS-CU001L)
Other Versions are available:
GS-FW012 ip
Create » Content Acquisition and Capture » Audio Aquisition » Audio acquisition accessories »
The GS-FW012 ip is the Dante network audio version of the GS-FW012 4W unit.
There are still four inputs for the top panel loudspeaker or headphone monitoring, each with their own level controls and each being derived from the Dante network. There are still four talkback outputs each being routed to the Dante network.
Both the inputs and outputs are presented via a single rear panel Neutricon RJ45/CAT5 connection that is Dante/AES67 compliant.
The four talkback outputs are identical. Each has a 3 position lever key talk switch that is either off, locked on, or in a sprung push to talk mode. The output of each circuit can be off unless the talkback button is pressed, looping the the associated Dante input input, or outputting the local cue input.
The local cue input is presented on XLR and has it's own level pot for the speaker or headphone monitoring.
A good quality microphone amplifier with adjustable gain followed by a compressor limiter is built into the unit making its talkback outputs to the Dante network crystal clear. Conversion of the audio signals to/ from the Dante network are by low noise, high bandwidth 48K 24bit converters providing superb performance.
Audio I/O can be routed via the digital Dante router. The free Dante Controller software configures all routes and can be downloaded by clicking here.
There is an internal switch mode power supply, or the GS-FW012 ip can be powered via PoE via the CAT5 connection. If both supplies are present then they will act as redundant supplies.
There are still four inputs for the top panel loudspeaker or headphone monitoring, each with their own level controls and each being derived from the Dante network. There are still four talkback outputs each being routed to the Dante network.
Both the inputs and outputs are presented via a single rear panel Neutricon RJ45/CAT5 connection that is Dante/AES67 compliant.
The four talkback outputs are identical. Each has a 3 position lever key talk switch that is either off, locked on, or in a sprung push to talk mode. The output of each circuit can be off unless the talkback button is pressed, looping the the associated Dante input input, or outputting the local cue input.
The local cue input is presented on XLR and has it's own level pot for the speaker or headphone monitoring.
A good quality microphone amplifier with adjustable gain followed by a compressor limiter is built into the unit making its talkback outputs to the Dante network crystal clear. Conversion of the audio signals to/ from the Dante network are by low noise, high bandwidth 48K 24bit converters providing superb performance.
Audio I/O can be routed via the digital Dante router. The free Dante Controller software configures all routes and can be downloaded by clicking here.
There is an internal switch mode power supply, or the GS-FW012 ip can be powered via PoE via the CAT5 connection. If both supplies are present then they will act as redundant supplies.
GS-FW021
Produce » Real-Time Production » Production Services » Outside broadcast and mobile facilities »
- Battery powered from 9volt PP3
- Long battery life (in excess of 100 hours)
- Power on switch
- Power on LED
- Low battery indication
- 3 user selectable modes: 4 wire (for standard connection to other broadcast eqpt) Interrupt loop through (IFB) for monitoring & adding talkback over incoming audio 2 wire conference mode for connecting multiple GS-FW021 together 2 wire mode compatible with prospect C1B
- Headset connection on 3 pole 6.35mm (A/B gauge)
- Push to talk button for mic input
- Limiter circuit for mic input
- Headphone volume control
- Auxiliary audio input (on 3.5mm jack) fed to headphone circuit
- Transformer balanced 4 wire input/ output
- Maximum dimensions just 152 x 42 x 72mm (l x w x d)
GS-HA014
Produce » Real-Time Production » Production Services » Studio and stage facilities »
A very useful headphone amp for studio environments.
This stereo amp mounts underneath desks and allows easy headphone connections for presenters or guest users. A 6.35mm and 3.5mm jack socket is available for the headphones and a full size, covered pot allows level adjust. Audio inputs are on 2 x balanced 6.35mm jack sockets.
The units link power via CAT5. Up to 8 units can be connected from 1 external 9V DC supply. Via an internal jumper, audio can also be routed via the same CAT5 connections as the power. The external power supply is not included and at least 1 must be purchased separately.
The GS-HA014 is suitable for use with a wide range of input levels, a pair of preset gain controls accessed via small holes in the top of the unit allow both broadcast (0dB) and domestic (-10dB) input levels to be accommodated.
This stereo amp mounts underneath desks and allows easy headphone connections for presenters or guest users. A 6.35mm and 3.5mm jack socket is available for the headphones and a full size, covered pot allows level adjust. Audio inputs are on 2 x balanced 6.35mm jack sockets.
The units link power via CAT5. Up to 8 units can be connected from 1 external 9V DC supply. Via an internal jumper, audio can also be routed via the same CAT5 connections as the power. The external power supply is not included and at least 1 must be purchased separately.
The GS-HA014 is suitable for use with a wide range of input levels, a pair of preset gain controls accessed via small holes in the top of the unit allow both broadcast (0dB) and domestic (-10dB) input levels to be accommodated.
GS-MON004
Support » Monitoring, Control & Communications » Audio Monitoring » Loudspeakers »
Modern Loudspeaker Drive Units:The newly designed drive units fitted in the GS-MON004 were originally manufactured as full range drive units in flat screen televisions. They have ultra modern Neodymium magnets with reasonably low tesla magnetic induction properties.
Class D Amplifiers:Each loudspeaker is driven from a new design of Class D amplifier. These amplifiers provide low distortion & low noise outputs. Four watts RMS is provided into each loudspeaker and the peak output is electronically limited to 5 watts.
Inbuilt Loudspeaker Protection:A compressor limiter circuit is provided to prevent overload and damage to the loudspeaker drivers by exceptionally high audio levels. One compressor/ limiter is provided for each loudspeaker and they have been carefully designed not to taint the audio during normal operating levels.
Stereo Inputs For Each Channel:A total of 8 audio inputs are provided for the 4 loudspeakers. Each loudspeaker is driven from 2 audio inputs mixed together, this allows the easy monitoring of stereo signals on the unit. All the audio inputs are electronically balanced on rear panel XLRs.
Channel Mute Switch:Each channel has its own illuminated mute switch. This allows a quick and easy way for an operator to turn the audio off from one (or more ) loudspeakers.
Presence & Peak LEDs:Presence LEDs are provided for each loudspeaker input. These LEDs illuminate when incoming audio is detected and this provides a helpful aid to an operator to know which audio circuit is currently active. Peak LEDs are also provided and these illuminate when the input level reaches a high level.
Headphone Output:A front panel 6.35mm TRS headphone jack is provided. If headphones are plugged in then the audio to the loudspeakers is automatically cut. The headphone output is stereo and it wired such that inputs 1 and 3 feed its left output and 2 and 4 feed the right output channel.
Mains Power Supply:An internal switch mode mains power supply is fitted that is suitable for use Worldwide. The power input is via na industry standard filtered IEC plug.
High Acoustical Output:The modern loudspeaker drive units are capable of providing a high output SPL of 88dB from 1 loudspeaker (at 0.6 metres) or 98dB if all loudspeakers are driven coherently.
Best in Class Frequency Response:Our design provides what we believe is the best in class frequency response from a 4 way 1RU loudspeaker rack. Our acoustical output is within +/-9dB from 22kHz down to 120Hz. The flat mid range and extended LF provide a noticeable improvement in audio clarity
Class D Amplifiers:Each loudspeaker is driven from a new design of Class D amplifier. These amplifiers provide low distortion & low noise outputs. Four watts RMS is provided into each loudspeaker and the peak output is electronically limited to 5 watts.
Inbuilt Loudspeaker Protection:A compressor limiter circuit is provided to prevent overload and damage to the loudspeaker drivers by exceptionally high audio levels. One compressor/ limiter is provided for each loudspeaker and they have been carefully designed not to taint the audio during normal operating levels.
Stereo Inputs For Each Channel:A total of 8 audio inputs are provided for the 4 loudspeakers. Each loudspeaker is driven from 2 audio inputs mixed together, this allows the easy monitoring of stereo signals on the unit. All the audio inputs are electronically balanced on rear panel XLRs.
Channel Mute Switch:Each channel has its own illuminated mute switch. This allows a quick and easy way for an operator to turn the audio off from one (or more ) loudspeakers.
Presence & Peak LEDs:Presence LEDs are provided for each loudspeaker input. These LEDs illuminate when incoming audio is detected and this provides a helpful aid to an operator to know which audio circuit is currently active. Peak LEDs are also provided and these illuminate when the input level reaches a high level.
Headphone Output:A front panel 6.35mm TRS headphone jack is provided. If headphones are plugged in then the audio to the loudspeakers is automatically cut. The headphone output is stereo and it wired such that inputs 1 and 3 feed its left output and 2 and 4 feed the right output channel.
Mains Power Supply:An internal switch mode mains power supply is fitted that is suitable for use Worldwide. The power input is via na industry standard filtered IEC plug.
High Acoustical Output:The modern loudspeaker drive units are capable of providing a high output SPL of 88dB from 1 loudspeaker (at 0.6 metres) or 98dB if all loudspeakers are driven coherently.
Best in Class Frequency Response:Our design provides what we believe is the best in class frequency response from a 4 way 1RU loudspeaker rack. Our acoustical output is within +/-9dB from 22kHz down to 120Hz. The flat mid range and extended LF provide a noticeable improvement in audio clarity
GS-MPI005HD MKII
Connect » Provide Connectivity » Connectivity & Bandwidth » Mobile/cellular connectivity »
Glensound invented the Broadcaster’s mobile phone and our GS-MPI005 and GS- MPI005HD mobile phone racks have been hugely popular with broadcasters and can be found in daily use in outside broadcast trucks and studios around the World.
The completely updated GS-MPI005HD MKII is a mains powered 1RU subrack that can be fitted with up to two mobile phones featuring professional audio interfaces for connecting broadcast audio equipment to cell phone networks.
On the surface the new GS-MPI005HD MKII looks different as it has touch screen colour displays for each phone module fitted. A traditional telephone handset interface is also now provided to allow an operator to communicate off air before handing the call over to broadcast. The handset interface can also dial and answer calls.
No new modern piece of audio broadcast equipment is complete without an AoIP solution so the new GS-MPI005HD MKII has an optional Dante®/ AES67 card.
Of course the crystal clear sound and audio inputs & outputs that are interference free of cell phone noise that made the original units so popular also feature in this new design.
The completely updated GS-MPI005HD MKII is a mains powered 1RU subrack that can be fitted with up to two mobile phones featuring professional audio interfaces for connecting broadcast audio equipment to cell phone networks.
On the surface the new GS-MPI005HD MKII looks different as it has touch screen colour displays for each phone module fitted. A traditional telephone handset interface is also now provided to allow an operator to communicate off air before handing the call over to broadcast. The handset interface can also dial and answer calls.
No new modern piece of audio broadcast equipment is complete without an AoIP solution so the new GS-MPI005HD MKII has an optional Dante®/ AES67 card.
Of course the crystal clear sound and audio inputs & outputs that are interference free of cell phone noise that made the original units so popular also feature in this new design.
Inferno
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
Links To Dante Compatible Network Or DARK88 Break Out BoxInferno can be used on its own when connected to a Dante audio network. Many audio routers, consoles and intercom systems are Dante compatible, and will accept the Inferno link, enabling Glensound grade commentary into the audio network.
There are primary and secondary Dante network connections
Inferno can also be used outside of a dedicated Dante network when connected to one of our DARK or AoIP44 break out boxes. This connection can be direct or across a structured network. The DARK or AoIP44 units allow all of the audio inputs and outputs to be located in a place separate from the Inferno.
Programmable On A Dante Audio NetworkWhen connected to a Dante network, sources and destinations can be configured from any part of the network using the free Dante controller software. Details are available at www.audinate.com.
Seven Talkback CircuitsThe seven illuminated talkback buttons route the audio input to 7 different destinations and remove the audio from the main programme output.
Eight Input Headphone MixerSeven external sources plus sidetone are presented on eight level controls allowing the user to create their own required monitoring mix of all sources. Each input can be independently selected to be on the user's left ear, right ear or both ears. The headphone connection is available on a 3.5mm and a 6.35mm jack socket.
Single InputThe single input is switchable on the rear panel to be line/mic/48v and has adjustable input gain. The input also features Glensound’s Referee compressor limiter system. This applies a variable rate compressor to the input peaks so that no clipping occurs, without affecting the rest of the audio signal.
Remote ControlThe Inferno contains a web server, allowing remote functionality to an engineer from any PC connected to the network via any web browser. Enter the IP address of the Inferno in the browser for remote control of the front end mic gain. It also allows configuration of each push button on the unit to be on/off latching, momentary (push to talk), always on, and off when pushed (cough), or in an intelligent mode where a short press latches or a longer press is just momentary.
15 segment LED PPM Meter Audio input level is indicated on the 15 segment PPM meter.
Power On Dante Link Or LocallyInferno can be powered via the network audio connection if it conforms to the PoE standard. There is also an internal 100v-240v mains supply.
Scalable Commentary~Start with one Inferno into a Dante network~Add Multiple Inferno at any time for 2, 3, 4 ,5 or more commentator system~Use DARK units for a simple point to point networked audio break out box
There are primary and secondary Dante network connections
Inferno can also be used outside of a dedicated Dante network when connected to one of our DARK or AoIP44 break out boxes. This connection can be direct or across a structured network. The DARK or AoIP44 units allow all of the audio inputs and outputs to be located in a place separate from the Inferno.
Programmable On A Dante Audio NetworkWhen connected to a Dante network, sources and destinations can be configured from any part of the network using the free Dante controller software. Details are available at www.audinate.com.
Seven Talkback CircuitsThe seven illuminated talkback buttons route the audio input to 7 different destinations and remove the audio from the main programme output.
Eight Input Headphone MixerSeven external sources plus sidetone are presented on eight level controls allowing the user to create their own required monitoring mix of all sources. Each input can be independently selected to be on the user's left ear, right ear or both ears. The headphone connection is available on a 3.5mm and a 6.35mm jack socket.
Single InputThe single input is switchable on the rear panel to be line/mic/48v and has adjustable input gain. The input also features Glensound’s Referee compressor limiter system. This applies a variable rate compressor to the input peaks so that no clipping occurs, without affecting the rest of the audio signal.
Remote ControlThe Inferno contains a web server, allowing remote functionality to an engineer from any PC connected to the network via any web browser. Enter the IP address of the Inferno in the browser for remote control of the front end mic gain. It also allows configuration of each push button on the unit to be on/off latching, momentary (push to talk), always on, and off when pushed (cough), or in an intelligent mode where a short press latches or a longer press is just momentary.
15 segment LED PPM Meter Audio input level is indicated on the 15 segment PPM meter.
Power On Dante Link Or LocallyInferno can be powered via the network audio connection if it conforms to the PoE standard. There is also an internal 100v-240v mains supply.
Scalable Commentary~Start with one Inferno into a Dante network~Add Multiple Inferno at any time for 2, 3, 4 ,5 or more commentator system~Use DARK units for a simple point to point networked audio break out box
MinFerno
Produce » Real-Time Production » Audio Production » Audio mixers and consoles – radio »
MinFerno is a new, smaller, and even easier to use version of our exceptionally popular Inferno network audio commentary box.
MinFerno is designed for a single commentator/ announcer to use, and like its big brother the Inferno, it provides the very best possible commentary microphone amplifier and compressor limiter circuit for amazing on air sound.
The MinFerno is easy to use for commentators who would rather be talking about the game than figuring out how the equipment works. It is built to our exacting, rugged & robust standard to make it a reliable piece of broadcast equipment for the busy engineer.
Redundant Powering OptionsThe MinFerno can be powered from any of 3 different sources:1) PoE on the Primary CAT5 Network link2) PoE on the Secondary CAT5 Network link3) External 12V DCFour of rear panel LEDs show the availability of the 3 power sources.
Primary & Secondary SFP SlotsThe SFP (Small Form-Factor Pluggable) fibre slots are standard networking ports that accept standard SFP modules. This means that you decide what type of fibre and connector style you want to use just by the SFP module that you insert. The primary & secondary network circuits allow for glitch free redundancy across both the Fibre & Copper network interfaces.
Primary & Secondary Copper Network Connections with PoETwo CAT6 connections on Neutrik Ethercons (that accept standard networking cables) are provided to allow copper connections to local network switches to carry the Dante/ AES67 audio. Two connections are fitted to allow redundant circuits to be used if required.Both of these connectors can accept a PoE power source for providing the power to the MinFerno.
Input Mode & GainOne pair of push buttons selects the input type of the front panel XLR to be either microphone, line or microphone with 48V phantom power. 3 LEDs indicate which input mode is selected.Two push buttons are used to alter the gain of the input. LEDs indicate if the gain setting is above or below our pre-configured ‘lineup’ levels. The front panel PPM of course provides an accurate indication of the input level.The gain can also be altered remotely by a web browser pointing at the MinFerno's web page.
Up to 4 of Incoming Audio Volume ControlsOn the top panel are up to 5 rotary headphone volume controls. 4 of these are connected to 4 incoming audio circuits from the Dante/ AES67 network. These are normally used for such sources as mixed programme or cue, talkback to director, talkback to producer, talkback to engineer etc.
Sidetone Volume ControlThe 5th front panel rotary headphone volume control is ‘sidetone’. Sidetone is the commentator's own voice in their own ears.
Headphone RoutingEach of the 5 headphone volume controls has an associated left ear, right ear, both routing switch located next to the volume knob. This single push button switch routes the associated source to just the left, just the right or both channels of the stereo headphone amplifier. To enable the commentator to know how they are routing a circuit the first time a routing switch is pressed, a pair of LEDs on the front panel indicate its current routing arrangement. The next time the routing switch is pressed then the next routing option is selected.
Robust Mic On and Talkback ButtonsThere are up to 4 of large bright illuminated buttons (1 for programme and up to 3 for talkback). These switches route the outgoing microphone circuit onto up to 4 different Dante/ AES67 network audio circuits. The operation of these switches (momentary, latching etc) and the interaction of these switches (i.e. pressing a talkback switch mutes the main mic) can be fully configured via the web page.
15 segment LED PPM MeterAudio input level is indicated on the 15 segment PPM meter.
Low Noise Microphone Amplifier With Remote GainWe spent a long time optimising the performance of the THAT corporation microphone amplifier used in the original Inferno and used again on the MinFerno. It features very low noise & distortion circuit that we remote control the gain of in 1dB steps, which allows us to provide the remote webpage gain control as well as the rear panel gain push buttons. We also optimised the circuits to allow correct source impedance switching depending on whether the input has been set as a mic or line input.
Referee Compressor LimiterAs with all our commentary units, the MinFerno features our very popular Referee compressor limiter circuit. This circuit starts to compress the commentator’s voice gradually and slowly increases the compression ratio as the input level becomes overly high, resulting in a very natural sounding and distortion free audio output capable of taming even the loudest of commentators.
High Quality Analogue To Digital Converter (ADC)Dante/ AES67 network audio is a digital circuit and as such the best analogue microphone amplifier would be wasted if we hadn’t paired it up with the best analogue to digital converter. The ADCs job is fairly simple, and if you look at our tech spec you’ll see that we’ve made ours work incredibly well.
Unique Headphone AmplifiersThe commentator’s headphones are a vitally important tool so we take as much care with our headphone amplifiers as with our on air microphones. Our unique headphone amplifier provides the correct output level regardless of the impedance of the attached headphones, meaning that broadcasters can now pick and choose between low impedance ‘cheap’ headphones and high impedance traditional broadcast ones.
MinFerno is designed for a single commentator/ announcer to use, and like its big brother the Inferno, it provides the very best possible commentary microphone amplifier and compressor limiter circuit for amazing on air sound.
The MinFerno is easy to use for commentators who would rather be talking about the game than figuring out how the equipment works. It is built to our exacting, rugged & robust standard to make it a reliable piece of broadcast equipment for the busy engineer.
Redundant Powering OptionsThe MinFerno can be powered from any of 3 different sources:1) PoE on the Primary CAT5 Network link2) PoE on the Secondary CAT5 Network link3) External 12V DCFour of rear panel LEDs show the availability of the 3 power sources.
Primary & Secondary SFP SlotsThe SFP (Small Form-Factor Pluggable) fibre slots are standard networking ports that accept standard SFP modules. This means that you decide what type of fibre and connector style you want to use just by the SFP module that you insert. The primary & secondary network circuits allow for glitch free redundancy across both the Fibre & Copper network interfaces.
Primary & Secondary Copper Network Connections with PoETwo CAT6 connections on Neutrik Ethercons (that accept standard networking cables) are provided to allow copper connections to local network switches to carry the Dante/ AES67 audio. Two connections are fitted to allow redundant circuits to be used if required.Both of these connectors can accept a PoE power source for providing the power to the MinFerno.
Input Mode & GainOne pair of push buttons selects the input type of the front panel XLR to be either microphone, line or microphone with 48V phantom power. 3 LEDs indicate which input mode is selected.Two push buttons are used to alter the gain of the input. LEDs indicate if the gain setting is above or below our pre-configured ‘lineup’ levels. The front panel PPM of course provides an accurate indication of the input level.The gain can also be altered remotely by a web browser pointing at the MinFerno's web page.
Up to 4 of Incoming Audio Volume ControlsOn the top panel are up to 5 rotary headphone volume controls. 4 of these are connected to 4 incoming audio circuits from the Dante/ AES67 network. These are normally used for such sources as mixed programme or cue, talkback to director, talkback to producer, talkback to engineer etc.
Sidetone Volume ControlThe 5th front panel rotary headphone volume control is ‘sidetone’. Sidetone is the commentator's own voice in their own ears.
Headphone RoutingEach of the 5 headphone volume controls has an associated left ear, right ear, both routing switch located next to the volume knob. This single push button switch routes the associated source to just the left, just the right or both channels of the stereo headphone amplifier. To enable the commentator to know how they are routing a circuit the first time a routing switch is pressed, a pair of LEDs on the front panel indicate its current routing arrangement. The next time the routing switch is pressed then the next routing option is selected.
Robust Mic On and Talkback ButtonsThere are up to 4 of large bright illuminated buttons (1 for programme and up to 3 for talkback). These switches route the outgoing microphone circuit onto up to 4 different Dante/ AES67 network audio circuits. The operation of these switches (momentary, latching etc) and the interaction of these switches (i.e. pressing a talkback switch mutes the main mic) can be fully configured via the web page.
15 segment LED PPM MeterAudio input level is indicated on the 15 segment PPM meter.
Low Noise Microphone Amplifier With Remote GainWe spent a long time optimising the performance of the THAT corporation microphone amplifier used in the original Inferno and used again on the MinFerno. It features very low noise & distortion circuit that we remote control the gain of in 1dB steps, which allows us to provide the remote webpage gain control as well as the rear panel gain push buttons. We also optimised the circuits to allow correct source impedance switching depending on whether the input has been set as a mic or line input.
Referee Compressor LimiterAs with all our commentary units, the MinFerno features our very popular Referee compressor limiter circuit. This circuit starts to compress the commentator’s voice gradually and slowly increases the compression ratio as the input level becomes overly high, resulting in a very natural sounding and distortion free audio output capable of taming even the loudest of commentators.
High Quality Analogue To Digital Converter (ADC)Dante/ AES67 network audio is a digital circuit and as such the best analogue microphone amplifier would be wasted if we hadn’t paired it up with the best analogue to digital converter. The ADCs job is fairly simple, and if you look at our tech spec you’ll see that we’ve made ours work incredibly well.
Unique Headphone AmplifiersThe commentator’s headphones are a vitally important tool so we take as much care with our headphone amplifiers as with our on air microphones. Our unique headphone amplifier provides the correct output level regardless of the impedance of the attached headphones, meaning that broadcasters can now pick and choose between low impedance ‘cheap’ headphones and high impedance traditional broadcast ones.