Dante-enabled Product Catalog

The QFlex 48 can be used as a standalone self-powered column array device for both spoken word and full-range musical applications in larger scale spaces, where high intelligibility and minimal visual impact are driving concerns. The QFlex 48 can be used as a standalone self-powered column array device for both spoken word and full-range musical applications in larger scale spaces, where high intelligibility and minimal visual impact are driving concerns. QFlex is a range of digitally steerable, multi-channel, array speaker systems for the professional install market. QFlex is designed as a complete solution for applications with difficult acoustics, such as houses of worship, transportation hubs, convention centres, conference facilities, shopping malls, performing arts centres and museums. QFlex is quite revolutionary in that it is able to achieve even coverage and SPL which is maintained across the listening plane. QFlex is able to create an asymmetrical pattern which allows similar SPLs both in the near and far field. It is able to steer the beam away from surfaces that cause reflections to frequencies beyond 12 kHz, making it the first digitally steerable array to maintain music quality over the desired area of coverage, all in a very architecturally pleasing package. The intuitive BeamEngine GUI allows the designer to specify target areas. Based on those target areas a steering algorithm is generated and tailored for that specific area, rather than having to choose from a limited palette of opening and steering angles. New features introduced on QFlex deliver more resolute safety and monitoring facilities, making it fully compliant in any life safety or mass notification environment. In addition QFlex has been weatherised for outdoor or harsh environments without compromising on the product’s sleek aesthetic, making it perfectly suited for large scale transport hub applications such as within airport terminals and major train stations. Among the key improvements on this front are the addition of Pilot Tone Detection capability, Sentinel SM1 System Monitor and certification to IP53 standards.

QFlex 64 is the flagship and largest model available within the QFlex range, comprising of 6 individual connected modules to form a 5.5 metre high array of 32 x 4” LF, 16 x 3” LF, 16 x 1” HF, each with its own discrete channel of amplification and integrated DSP (64 in total).  QFlex 64 is the flagship and largest model available within the QFlex range, comprising of 6 individual connected modules to form a 5.5 metre high array of 32 x 4” LF, 16 x 3” LF, 16 x 1” HF, each with its own discrete channel of amplification and integrated DSP (64 in total). This new model offers greater low frequency beam control (down to 110 Hz), increased effective range and throw (up to 100 metres), and greatly improved headroom and SPL. Typical applications for the QFlex 64 are large public concourses, major transport hubs (such as long railway station platforms) and other long-throw situations, where great intelligibility and LF control over distances are of paramount concern. QFlex is a range of digitally steerable, multi-channel, array speaker systems for the professional install market. QFlex is designed as a complete solution for applications with difficult acoustics, such as houses of worship, transportation hubs, convention centres, conference facilities, shopping malls, performing arts centres and museums. QFlex is quite revolutionary in that it is able to achieve even coverage and SPL which is maintained across the listening plane. QFlex is able to create an asymmetrical pattern which allows similar SPL's both in the near and far field. It is able to steer the beam away from surfaces that cause reflections to frequencies beyond 12 kHz, making it the first digitally steerable array to maintain music quality over the desired area of coverage, all in a very architecturally pleasing package. The intuitive BeamEngine GUI allows the designer to specify target areas. Based on those target areas a steering algorithm is generated and tailored for that specific area, rather than having to choose from a limited palette of opening and steering angles. New features introduced on QFlex deliver more resolute safety and monitoring facilities, making it fully compliant in any life safety or mass notification environment. In addition QFlex has been weatherised for outdoor or harsh environments without compromising on the product’s sleek aesthetic, making it perfectly suited for large scale transport hub applications such as within airport terminals and major train stations. Among the key improvements on this front are the addition of Pilot Tone Detection capability, Sentinel SM1 System Monitor and certification to IP53 standards.

Large scale live production console. L500 helps highly skilled operators deliver outstanding performances in the most demanding applications in Touring, Installation, Houses Of Worship and Theatre. The L500 offers ample power and combines superb audio quality with a comprehensive collection of processing tools, within a superb user interface. Large Scale Production Powerhouse Since its introduction in 2013 the SSL Live L500 console has met widespread praise from many of the world’s leading live engineers. L500 was designed to help highly skilled operators deliver outstanding performances in the most demanding applications in Touring, Installation, Houses Of Worship and Theatre. Like all SSL consoles the secret to the success of the L500 is SSL’s dedication to great sound. There is a sublime transparency, definition and detail to the L500 that is immediately recognisable as the SSL signature. The deceptively simple ergonomic excellence of the control surface enables operators to perform at their very best. A combination of multi-touch screen control with assignable hardware controls, the use of colour coding, the open architecture let engineers configure the console to suit their own way of working. Like all great creative audio technology, SSL Live consoles are instruments for engineers...

Anna™ is the newest introduction to EAW’s flagship line of Adaptive™ Systems, providing all the benefits of Adaptive performance in a high-output mid-sized enclosure. Anna™ is the newest introduction to EAW’s flagship line of Adaptive™ Systems, providing all the benefits of Adaptive performance in a high-output mid-sized enclosure. Weighing just 135 lbs. (61 kg), the extraordinary output-to-weight ratio makes Anna exceptionally versatile, with applications ranging from small clubs and houses of worship to large arenas and touring productions. Anna’s 100° horizontal dispersion further extends the capabilities of Adaptive Systems; columns of Anna can be flown as mains or mixed with columns of Anya™ within the same array for even greater coverage possibilities. Like all Adaptive Systems, Anna is controlled via Resolution™ 2 software over the Dante™ network and utilizes the same standardized power and data infrastructure. Anna modules hang straight, without any vertical splay, and Resolution™ 2 software adapts total system performance to produce custom-tailored coverage that delivers coherent, full-frequency range response across the entire coverage area as defined by the user. It is extremely powerful and immensely scalable, making it suitable for anything from small clubs and theaters, to larger houses of worship and arenas. By carefully crafting the size and spacing of the transducers, Anna is able to create radical coverage patterns (i.e., narrowly focused and directed almost straight down) while providing outstanding fidelity within the coverage area and broadband rejection everywhere else. Each Anna module includes 8x 1-in exit / 35mm voice coil HF compression drivers loaded on a proprietary HF horn that expands to fill nearly the entire face of the enclosure. 4x 5-in MF cone transducers, arranged in two columns of two, use Radial Phase Plugs™ and Concentric Summation Array™ technology to enter the horn and sum coherently with the HF wave front. Dual high-power 10-in LF cone transducers use Offset Aperture™ loading to increase the spacing of the apparent acoustical centers, extending effective horizontal pattern control well into the LF range. The module’s horizontal symmetry ensures coherent summation without anomalies through the crossover regions that result from physically offset acoustic sources. This provides consistent, HF dispersion and broadband pattern control in the horizontal plane. Each Anna module includes a field-replaceable Power Plant with 14 channels of digital signal processing and amplification to drive each of the module’s 14 transducers independently, with all parameters determined by Resolution™ software to provide optimal coverage based to every listening position.

The MA32 series is a multichannel power amplifier platform featuring the worlds highest channel density in small enclosures. Despite it's compact construction in only one rack unit, the amplifier delivers a maximum burst power of about 640W distributed over 32 channels with 20W each. The MA32 series is a multichannel power amplifier platform featuring the worlds highest channel density in small enclosures. Despite it's compact construction in only one rack unit, the amplifier delivers a maximum burst power of about 640W distributed over 32 channels with 20W each. Besides DSP features like IIR-Filters, limiters and delays, it also features 2048 Tap FIR Filters per channel. The integrated webserver and LCD display offers the user an intuitive function overview and easy configuration possibility. To provide high reliability and easy maintenance for project installations, the MA32 series features also a Pilot Tone monitoring for each connected speaker. Audio Inputs can be Dante™ (redundant), MADI / AES10 (1x optical, 2x coaxial redundant) and AES3id which can easily be assigned to any of the desired output channels. With the MA32 series we provide the key for cost effective and space-saving 3D-sound installations, live surround sound applications as well as high channel count research projects.

The MA32 series is a multichannel power amplifier platform featuring the worlds highest channel density in small enclosures. Despite it's compact construction in only 2 rack units, the amplifier delivers a maximum burst power of about 9,000W distributed over 32 channels with 280W each. The MA32 series is a multichannel power amplifier platform featuring the worlds highest channel density in small enclosures. Despite it's compact construction in only 2 rack units, the amplifier delivers a maximum burst power of about 9,000W distributed over 32 channels with 280W each. Besides DSP features like IIR-Filters, limiters and delays, it also features 2048 Tap FIR Filters per channel. The integrated webserver and TFT display offers the user an intuitive function overview and easy configuration possibility. To provide high reliability and easy maintenance for project installations, the MA32 series features also a Pilot Tone monitoring for each connected speaker. Audio Inputs can be Dante™ (redundant), MADI / AES10 (1x optical, 2x coaxial redundant) and AES3id which can easily be assigned to any of the desired output channels. With the MA32 series we provide the key for cost effective and space-saving 3D-sound installations, live surround sound applications as well as high channel count research projects.

The Dante Bridge allows several different classes of audio product to be seamlessly integrated into Dante systems with the minimum of fuss. Analogue outputs allow the Bridge to be used with products which have analogue inputs.  The Dante Bridge allows several different classes of audio product to be seamlessly integrated into Dante systems with the minimum of fuss. Analogue outputs allow the Bridge to be used with products which have analogue inputs. The AES3 output allows the Bridge to be used with products which have an AES3 input. The BvNet connection also allows equipment which has a BvNet control interface to be controlled over an Ethernet network, to which the PodWare application will connect directly.

• Two analog outputs
• AES3 output
• Ethernet control for BvNet products
• Rugged steel enclosure
• Free-standing or rack-mount options
• Power Over Ethernet (PoE)
• Optional power by companion PSU

The control data within Dante is bidirectional, and can flow from the Dante network to a BvNet device and vice-versa. Similarly, digital audio channels within the Dante network can be routed to the BvNet socket, so that the AES3 signal is conveniently available on the BvNet network, where it can be brokenout using a special cable for use as an AES3 feed. The same audio channels are also available on analog outputs for use with any devices with analogue inputs.

The DMM8 U digital automatic microphone mixer is ideal for large meetings and conferences.  The DMM8 U digital automatic microphone mixer is ideal for large meetings and conferences. Its DSP function ensures precise mixing of up to 80 channels by cascading up to 10 DMM8 U mixers. Two additional stereo audio outputs and one stereo USB offer additional routing options. Due to high signal-to-noise ratio, the DMM8 U also keeps noise from multiple open microphones to a minimum. Numerous DSP functions such as filters, compressor/limiter, ducking and routing make the DMM8 U a great fit in a variety of spoken word applications. The DMM8 U (UL, ULD) is made in Germany. Additionally, the UL version offers a LAN interface via Ethernet to enable the control over large distances using, for instance, an AMX control system.  In the ULD version, incorporating a highly sophisticated DANTE chip, all ins are available as direct outs, making it an outstanding tool for broadcast applications, while all outs are also available on the DANTE stream. In total the DMM8 ULD provides 14 free assignable DANTE outs. Additionally, 4 DANTE streams can be used as inputs.

Engineered and made in Germany, the DMM14 U digital automatic microphone mixer is ideal for large meetings and conferences.  Engineered and made in Germany, the DMM14 U digital automatic microphone mixer is ideal for large meetings and conferences. Its DSP function ensures precise mixing of up to 140 channels by cascading up to 10 DMM14 U mixers. Two additional stereo audio outputs and one stereo USB offer additional routing options. Due to high signal-to-noise ratio, the DMM14 U also keeps noise from multiple open microphones to a minimum. Numerous DSP functions such as filters, compressor/limiter, ducking and routing make the DMM14 U a great fit in a variety of spoken word applications. The DMM14 U (UL, ULD) is made in Germany. Additionally, the UL version offers a LAN interface via Ethernet to enable the control over large distances using, for instance, an AMX control system.  In the ULD version, incorporating a highly sophisticated DANTE chip, all ins are available as direct outs, making it an outstanding tool for broadcast applications, while all outs are also available on the DANTE stream. In total the DMM14 ULD provides 22 free assignable DANTE outs. Additionally, 4 DANTE streams can be used as inputs.

The DASR-288 is a digital audio transport system utilizing Audinate’s Dante™ network audio solution. The DASR-288 is a digital audio transport system utilizing Audinate’s Dante™ network audio solution. The DASR-288 connects to your network with a single Ethernet cable. It is a Dante Break Out / ecoder interface in a 2RU form factor designed for all types of commercial AV installations. It offers eight channels of line-level analogue in and 8 line-level analogue outputs. Precision 24-bit A-D and D-A converters are used for superb audio performance with a 119dB dynamic range at sample rates up to 192kHz. This product is used as the audio matrix, each having eight inputs, where you can send uncompressed audio with low latency using standard TCP/ IP Ethernet network. It’s remotely controlled via the network and can be placed anywhere in a facility. Unique to the DASR-288 is its Data tunneling. No other manufacturer offers this feature, which allows you to transport with the Digital audio, 8 contact closures in and 8 contact closures out together combined with either RS-232C serial data. Each of the 8 balanced analog inputs and 8 outputs may be assigned to any Dante flow in the system, either by means of the Audinate’s Dante Controller Windows or OSX application or Inter-M’s own S-Connect Windows Software. The S-Connect allows for system configuration as well as easy channel switching. The S-Connect Software is designed to operate by simple touch screen. Dual redundant Gigabit Dante Network connections are available. Both primary and secondary connections support full Gigabyte ports for full redundancy. Signal Monitoring is available for all 8 inputs and 8 outputs with a front panel mounded dual LEDs, indicating signal presence and clipping. Network status LEDs also monitor network connectivity and transition speed. Brand Interoperability Based on industry standards, Audinate’s Dante, is an uncompressed, multi-channel digital media networking technology, with near-zero latency and synchronization. Dante is the preferred audio networking solution that has been adopted by more pro-audio AV manufacturers than any other networking technology. Interoperability is a reality today. The Inter-M DASR-288 is compatible with hundreds of other Dante-enabled products available from the world’s leading manufacturers, enabling you to mix devices from multiple manufacturers.

The D-3000 is a high power professional amplifier specially designed for the sound re-enforcement market also referred to as SR, which incorporates a DSP, Digital Signal Processor and Audinates’s Dante™ digital audio network. The D-3000 is a high power professional amplifier specially designed for the sound re-enforcement market also referred to as SR, which incorporates a DSP, Digital Signal Processor and Audinates’s Dante™ digital audio network. Dante delivers low latency, robust synchronization, I/O scalability, and installation simplicity via standard IT technology. The D-3000 will deliver up to 3000 watts of power per channel down to 2 ohm loads. Housed in two RU compact frameworks, the D-3000 Class D amplifier offers high power with reduced size and weight. The combination Class D topology amplification circuitry combined with SMPS (Switch Mode Power Supplies) creates a high efficiency amplifier that offers reduced power consumption and low heat dissipation. The D-3000 is ideal for applications where high power and efficiency is required. This may include House of warship, arenas, stadiums, concert venues, touring or any other large-scale facilities.

Next Generation Technology for Large-Scale Broadcast Production System T is the first of a new generation of dedicated broadcast audio production systems from Solid State Logic. For more than 30 years SSL has been at the forefront of innovative broadcast console design, and has incorporated decades of research and customer feedback into the development of System T.

• System T is a fully networked production environment. New ways to combine multiple Control Interfaces, Processor Cores and I/O devices can be specified to create previously impossible system configurations.
• Dante AoIP-based Routing & I/O.
• SSL’s Tempest Engine uses patented Optimal Core Processing for real time CPU based mixing and audio processing.
• SSL’s pristine SuperAnalogue^TM foundations combine with 64-bit floating point processing and mixing to ensure unlimited audio headroom. • System T’s console control surface incorporates the latest multi-gesture touch screen technology and is highly configurable to suit individual preferences.

Orange Box With DiGiCo's compact 2U Orange Box, you can use DMI (DiGiCo Multichannel Interface) cards to create audio paths over whatever interface you desire. Orange Box With DiGiCo's compact 2U Orange Box, you can use DMI (DiGiCo Multichannel Interface) cards to create audio paths over whatever interface you desire. The Orange Box has 2 PSUs for redundancy, and 2 slots to accommodate any of our ten different interfaces that allow you to convert pretty much any format to another. How? We have Slot A and Slot B, let's say you have MADI on your product, but you want to go to Dante. No problem, just buy a DMI with MADI, and a DMI with Dante, connect them up via the two slots, and away you go. It's the same with Hydra 2, Optocore, Aviom, AES, Analogue, and so on. This cool little ‘anything in, anything out' box essentially gives you a choice of Multichannel user interfaces that enable you to send audio wherever you want, in whatever format you choose.

AES to Dante and MADI conversion whenever you need it: Network I/O AES provides 32 Pairs of AES3 audio I/O connectivity to redundant Dante network connections. AES to Dante and MADI Conversion Whenever You Need it Network I/O AES provides 32 Pairs of AES3 audio I/O connectivity to redundant Dante network connections. Sample Rate Convertors allow seamless integration of unlocked AES3 sources. As with all Network I/O products redundant PSUs and Dante connections complete the broadcast specification. Additional coax and dual optical MADI connection, along with internal channel by channel routing provide additional format conversion options.

DCAP-OBiA is the world’s first OBIA processor compliant with DCI/SMPTE Standard and FIPS-140 Level Security. DCAP-OBiA is the world’s first OBIA processor compliant with DCI/SMPTE Standard and FIPS-140 Level Security. It supports a maximum of 256 channels, and can transmit 256 channels in single line, almost zero latency. It can be compatible with all current OBIA formats, and play any OBIA DCP.

DAC-OBiA is designed for converting digital audio signal to analog audio signal up to 32 channels. DAC-OBiA is designed for converting digital audio signal to analog audio signal up to 32 channels. It is based on LEONIS DTA3000, and can be used to work with DCAP-OBiA through Audio over Ethernet technology.

AOIPBox is an AOIP standard terminal, it can support AES67/Dante/Ravenna/Cobranet IP audio protocols. AOIPBox is an AOIP standard terminal that can support AES67/Dante/Ravenna/Cobranet IP audio protocols. AOIPBox is primarily used as a digital audio matrix system in radio stations. AOIPBox can also be used as the network audio recording front terminal or for audio monitoring terminals. AOIPBox has four AES3 inputs with SRC and four AES3 outputs with Bypass function for powerdown. In addition, AOIPBox  has an independent Ethernet interface for audio ppm metering, phase, device status, etc. - very important in the high safety broadcast environment. AOIPBox with 4 AES3 loopout ports Features: 

• Four AES3 IN with SRC and four AES3 out
• Dual AOIP RJ45 interfaces supporting AES67/Dante/Ravenna/Cobranet
• Four AES3 loopout ports
• SRC support from 8K to 192Khz sample frequency
• Dedicate monitoring ethernet port for high-safety broadcast systems
• Dual power supply units
• 1U 19 inch rack mount case
• AC 100~240V @50hz/60hz power supply
• AES3 audio interface is RJ45 interface
• Temperature detection and alarm system
• All aluminum case

The IF-DA64 is TASCAM's option card for TASCAM DA-6400 audio recorder/player. The IF-DA64 is TASCAM's option card for TASCAM DA-6400 audio recorder/player.

The DA-6400 is a cCompact 64-channel digital multitrack recorder/player for Live and Broadcast applictions. TASCAM’s DA-6400 is an all-new 64-track recorder that records to an internal SSD drive caddy for live and event recording. Professional I/O options like MADI and DANTE make it ideal for live recording, event capture, or as a backup for critical DAW sessions. Available with a single or dual/redundant power supply, the DA-6400 is an unbeatable option for professional remote recording. Included with the DA-6400 is an AK-CC25 hot-swappable caddy containing a high-performance 240GB TASCAM solid-state drive. The caddy includes a USB 3 jack for stand-alone use – simply hand the drive to the client to offload the BWAV files into their DAW session. The DA-6400 records 64 tracks at 48kHz/24-bit or 32 tracks at 96kHz/24-bit. Included on the rear panel are SMPTE timecode, word clock, RS-422, parallel and Ethernet connections. I/O options for the DA-6400 include the IF-MA64/EX interface card, with both coaxial and optical MADI connections. This card includes a redundant coaxial connection so that MADI is passed through even if power is lost, making it an ideal backup recorder for DAW sessions. Other I/O options include the lower-cost IF-MA64/BN coaxial MADI card, IF-DA64 Dante card, and IF-AE16 AES/EBU card, with more to come. No dedicated recorder combines the high performance, flexible operation, and complete reliability of TASCAM’s DA-6400 solid-state 64-track recorder. 

This product has reached “end of life”.  Please refer to Grace Design's m108 mic preamp. Possibly the world’s most evolved mic preamplifier, the m802 delivers functionality and performance which simply doesn’t exist in any other mic preamp design. The core is a high fidelity 8 channel microphone preamplifier designed to be extremely transparent and musical. But that’s just the beginning. The m802 is fully remote controllable. With our full featured stand alone hardware RCU (optional), up to 64 channels (a system of 8 m802’s) can be addressed and controlled seamlessly from up to 1000’ away. All preamplifier functions, including gain, phase, 48V phantom are instantly accessible and adjustable. These settings can then all be stored into preset locations and instantly recalled. The m802 can also be controlled directly from Digidesign® ProTools|HD® systems, compatible control surfaces and many other MIDI devices. Working with Digidesign®, we have adapted their control spec to directly address the m802 as an integrated part of a ProTools|HD® system. Additionally, the m802 can be controlled by a wide variety of MIDI mappable devices, making it easily integrated into a wide variety of remote recording situations. Next, consider the optional 8 channel 24bit/192kHz A/D card option. Developed as the perfect complement to the m802 mic preamplifier design, this converter card provides pristine, musical A/D conversion to rival even the finest stand alone units available. Included is our proprietary unique s-Lock™ dual stage PLL for extremely low intrinsic jitter and rock solid digital performance. The m802 now includes ribbon mic mode, which shifts the preamplifier’s entire gain range up 10dB, while deactivating 48V phantom power, optimizing input impedance and bypassing the decoupling capacitors. While the m802 works beautifully as a stand alone mic preamplifier, the optional Remote Control Unit (RCU) provides fantastic ergonomics and enhanced system control. This updated RCU now offers dedicated individual channel select switches and a new highly visible blue LED numeric display for channel gain settings. Both the RCU and the m802 now come with a high contrast, neutral backlight LCD. Whatever your application, we invite you to discover the remarkable sonic performance, functionality and reliability of the new m802. We’re confident that it will help you make the finest recordings of your career.

Like the SD10, the SD5's 37 faders are laid out in three convenient banks of 12 with one master fader with the added benefit of three 15-inch full colour TFT touch screens for even quicker access and multi-user applications. Like the SD10, the SD5's 37 faders are laid out in three convenient banks of 12 with one master fader with the added benefit of three 15-inch full colour TFT touch screens for even quicker access and multi-user applications. In addition, there are two interactive dynamic metering displays (IDMs) as used in its big brother, the SD7. The centrally located master screen has been incorporated into the worksurface meterbridge design to provide a lower profile meter bridge and allow for complete user feedback; this means the engineer never needs to move far from mix position: there is clear visibility of everyone on stage, and all control is less than an arm's length away. To make life even easier, engineers can recall or save presets on the channel strip; and recall or save snapshots using the master screen or or dedicated switches on the console. In addition, special functions can be assigned and accessed instantly via the console's 40 Smart Key Macros (accessible via four layers of 10 RGB backlit keys) at the push of a button. It really is that simple.

A combination of technologies have been seamlessly integrated into the SD7's worksurface to meet the everyday needs of any touring engineer. A combination of technologies have been seamlessly integrated into the SD7's worksurface to meet the everyday needs of any touring engineer. We've managed to pack them into three user-friendly interactive 15-inch high-resolution touch screens, each of which accommodates a bank of 12 faders,  to provide the user with the perfect platform – everything is just 'where it should be'. To make navigation even easier, we've added high-definition interactive meter bridges and multicoloured knobs and switches; and for further functionality and flexibility, 16 assignable faders (one bank of 12, one bank of four) sit neatly below the master screen in the centre of the console, taking the SD7's fader tally to a sizeable 52. To increase the fader count further for multi-engineer applications and shows, an EX-007 Fader Expansion unit is available. Two EX-007s can be connected via a simple Cat5e connector for remote control or simply to increase your fader count to 100.

The SD8 is a particularly versatile console with the same Stealth Digital Processing and floating point Super FPGA technology seen in the flagship SD7. The SD8 is a particularly versatile console with the same Stealth Digital Processing and floating point Super FPGA technology seen in the flagship SD7. The SD8 offers a truly superlative performance at a competitive price, something which will appeal to engineers across the board. There are two versions of the SD8: a 37-channel version, which boasts three banks of 12 motorised faders and one master fader; and the SD8-24, which has all the same power and functionality but in a smaller frame comprising 24 main channel faders and a master fader. At the heart of each console's worksurface is a 15-inch, high resolution TFT LCD touch screen with backlit display. Through this user-friendly interface, any bank of 12 faders can be instantly assigned as input or output channels, allowing all 36 main faders to control inputs if desired; equally, any bank of 12 can also be assigned to the touch screen for fine-tuning. In addition, the SD8 benefits from dedicated multi-function control knobs and electronic labelling.

The SD9 is a super-flexible, complete integrated system powered by Stealth Digital Processing and floating point Super FPGA technology. The SD9 is a super-flexible, complete integrated system powered by Stealth Digital Processing and floating point Super FPGA technology. This lightweight, small-footprint console was designed with multiple applications in mind therefore is equally suited to life on the road or as part of any fixed installation, be it a theatre, conference centre, house of worship or school. The SD9 provides outstanding performance at a very competitive price point. The SD9 features 24 touch-sensitive motorised faders and benefits from quick access function buttons for faster response times. At the heart of the console's worksurface is a 15-inch, high resolution TFT LCD touch screen with backlit display. In addition, the SD9 benefits from dedicated multi-function control knobs and electronic labelling.

SD Ten features three banks of 12 motorised faders and one master fader, each of which benefits from its own high resolution LED bar graph meter. The SD Ten includes many of our flagship SD7's features and benefits, including Stealth Digital Processing and floating-point Super FPGA technology, and provides a fully-loaded feature set that will cater to any front-of-house or monitor engineer's needs.  SD Ten features three banks of 12 motorised faders and one master fader, each of which benefits from its own high resolution LED bar graph meter. In the centre of the worksurface sits the familiar 15-inch backlit colour-keyed TFT touch screen, home of the console's super-intuitive control interface. The sheer intelligence of the SD Ten means even the most complex of tasks become quick and easy: engineers can recall or save presets on the channel strip; and recall or save snapshots using the master screen or hard wired switches on the console. In addition, special functions can be assigned and accessed instantly via the console's 40 Smart Key Macros (accessible via four layers of 10 RGB backlit keys) at the push of a button. It really is that simple.

The SD11 is the smallest by far of all the DiGiCo consoles, though don't let that deceive you: this 19-inch rack mount or table top mixer is a very powerful solution that can be incorporated into pretty much any mixing environment. The SD11 is the smallest by far of all the DiGiCo consoles, though don't let that deceive you: this 19-inch rack mount or table top mixer is a very powerful solution that can be incorporated into pretty much any mixing environment; like its bigger brothers, the SD11 is also powered by Stealth Digital Processing and floating point Super FPGA technology, and transporting it couldn't be easier – just pop it on the back seat of the car. This easy-to-use mixer delivers extremely high performance at a staggeringly competitive price point.    The SD11 is manufactured with a steel chassis and a polycarbonate-overlaid aluminium worksurface which makes it super-lightweight. Its user interface comes courtesy of DiGiCo's trademark 15-inch, high resolution TFT LCD touch screen with backlit display; and its 12 touch sensitive moving faders guarantee a simple and intuitive workflow. 

Suitable for commercial installation, broadcast, live sound, public address and production applications, the BM-A1-64DANTE is an Audio-over-IP (AoIP) audio monitoring unit (AMU) that permits confidence monitoring of any channel present on a Dante(TM) network.  Suitable for commercial installation, broadcast, live sound, public address and production applications, the BM-A1-64DANTE is an Audio-over-IP (AoIP) audio monitoring unit (AMU) that permits confidence monitoring of any channel present on a Dante(TM) network.  Dante, created by Audinate, is a leading Audio-over-IP networking solution, offering virtually perfect synchronization, ultra-low deterministic latency and plug-and-play configuration management.  The BM-A1-64DANTE is designed to provide a simple and effective way of monitoring audio present in Dante networks. Up to 512 channels may be present on a Dante network and the BM-A1-64DANTE can be configured to monitor up to 64 of these (by use of the Dante Controller software application). With up to 64 channels routed to the front panel, it is possible to view each channel's signal strength and also audibly monitor each channel individually. Up to 16 channels may also be combined to create a unique stereo monitor mix, if simulteneous monitoring of multiple channels is required. Two presets allow for different mix combinations to be stored and recalled. The audio present on the speakers is also available via balanced analogue (2 x XLR) and AES/EBU (1 x XLR) outputs. This is useful if connecting the unit to external monitors, or if also using the BM-A1-64DANTE as a simple two channel breakout unit to feed other equipment. Housed in a rugged, compact 1U rack chassis, the BM-A1-64DANTE features a redundant power supply as standard and offers full audio redundancy over a Gigabit Ethernet network.