Dante-enabled Product Catalog

The MONISMS MNS-AMP260AC Dante Audio Amplifier is a compact power amplifier module that drives small 4-8Ω speakers (Ceiling Speaker, In-wall Speaker, Pendant Speaker Column Speaker etc.)  The MONISMS MNS-AMP260AC Dante Audio Amplifier is a compact power amplifier module that drives small 4-8Ω speakers (Ceiling Speaker, In-wall Speaker, Pendant Speaker Column Speaker etc.) The MONISMS Dante Amplifier has two outputs,  each with 30W of power into an 4-8Ω load, This makes the module ideal for powering small areas consisting of only a few low power speakers. The module's audio input is via an Ethernet audio network, allowing the use of the DANTE/ AES67 audio network protocol. MNS-AMP260AC MAX power 2*30W,  Which build-in AC 100-240V power module. Dante Audio Amplifier can be mounted on a DIN rail or screwed onto a panel or wall for easy installation in a remote closet near the speaker area.

The POE DANTE/AES67 Audio Amplifier module MNS-AMPMC230POE, Built-in to your 4-8 ohm speaker, that instantly Dante-enables your speakers. The POE DANTE/AES67 Audio Amplifier module MNS-AMPMC230POE, Built-in to your 4-8 ohm speaker, that instantly Dante-enables your speakers. The MONISMS Dante Audio Amplifier is a compact power amplifier module that drives small 4-8Ω speakers. The MONISMS Dante Amplifier has two outputs, each with 15W of power into an 4-8Ω load, This makes the module ideal for powering small areas consisting of only a few low power speakers. The module's audio input is via an Ethernet audio network, allowing the use of the DANTE/ AES67 audio network protocol. MNS-AMP230POE is powered by standard POE+ and has a MAX power of 30 watts [2*15W]. If there is not  POE+  used,  It can used the external 24VDC power supply.

Scorpio is a premium, portable, 32 channel, 36 track mixer-recorder and the most powerful product ever designed by Sound Devices.  Scorpio is a premium, portable, 32 channel, 36 track mixer-recorder and the most powerful product ever designed by Sound Devices. With 16 mic/line preamplifiers, 32 channels of Dante in and out, AES in and out, 12 analog outputs and multiple headphone outputs, Scorpio is well-suited for any production scenario. A fully-customizable routing matrix enables sound professionals to send any input to any channel, bus, or output. Up to 12 buses may be individually mixed. Due to its compact form factor, the Scorpio is equally at home over-the-shoulder or in a mobile rig. Scorpio incorporates Sound Devices’ most cutting-edge technology. An ultra-powerful engine comprised of three FPGA circuits and six ARM processors deliver the horsepower needed for the most complex tasks. FPGA-based audio processing with 64-bit data paths ensures the highest sound quality and reliability. The Scorpio also features Sound Devices’ latest and best in analog microphone preamplifier design. These preamps have the smoothest sound and lowest noise of any preamp in the company’s 20-year history and include built-in analog limiters, high pass filters, delay, 3-band EQ and phantom power. The Scorpio has an internal 256GB SSD and can simultaneously record to two SD cards for redundancy. For additional flexibility, sound professionals can send different files to their choice of media. The companion SD-Remote Android application allows for control of the Scorpio via a large display. Great attention to detail has been paid to every aspect of the Scorpio’s design. Most common menus are accessible with only one or two button presses, and many menu shortcuts can be achieved with only one hand. Scorpio features a built-in dual L-Mount battery charger and may be powered with L-Mount batteries or via the TA4 DC inputs using smart batteries, NP-1 batteries, or in-line power supplies. The ultra-accurate, fully-featured timecode generator contains its own battery to hold timecode for up to four hours after power off.

The D10-1 digital amplifier is a 10 Amp single output class ‘D’ driver; meaning a 60%+ increase in energy efficiency over existing solutions. The D10-1 digital amplifier is a 10 Amp single output class ‘D’ driver; meaning a 60%+ increase in energy efficiency over existing solutions. The driver features capacitive touch front panels with intuitive menus, built in test signals, and are fully networkable with a Wi-Fi accessible standard browser based control panel for remote set-up, monitoring and email alerts. The D10-1 features a 10 Amp loop output plus ample voltage headroom, making it the most flexible solution on the market, suitable for a huge range of applications. The added power of the D10-1 provides a solution for installations in environments containing very high levels of metal, previously not possible without the use of combiners. In another first for high-power Class D Induction Loop drivers, installation can be performed with total confidence, as unique multi-stage filtering ensures compatibility with both other system equipment and global EMC regulations. Features

• Dante Input for Additional Network Audio
• Drives 1 output channels at 10ARMS
• Networkable with remote browser interface
• Digital display & intuitive ‘touch’ menu system
• 3 menu modes; Main, Status & Quick
• Highly energy efficient Class-D amplifier with low heat dissipation
• Up to 1,450m2 Perimeter Loops area coverage
• Compact 1U rack mount unit with internal transformer for simple rack installation
• Optimised for speech frequencies with unmatched intelligibility & capable of high quality musical reproduction
• AGC & Dual Slope MLC
• Active status monitoring & remote fault reporting via email
• Data compliant with: IEC 62489-1 Standard

Applications include

• Lecture Theatres & Conference facilities
• Stadia, Sports Halls, Cinemas & Theatres
• Courts Rooms, Airports & Railway Stations

Evo-NG is Steffens Systems’ Next Generation of self-powered, modular, electronically steerable column array loudspeaker. Evo-NG is Steffens Systems’ Next Generation of self-powered, modular, electronically steerable column array loudspeaker. The Evo-NG brings improved acoustic characteristics and advanced FIR beamforming algorithms for ultimate speech intelligibility while its appealing, slim design blends seamlessly into any architectural environment. Evo-NG modules can be combined to build column arrays consisting of up to six modules. The Evo-NG system consists of 2 models; a master and a slave. While the 8 individual 50W class D amplifier channels are common to both modules, only the Master module contains a power supply and 16 channel DSP board, allowing the Master module to function standalone, or to become the core of a larger array, potentially with multiple masters and slaves. This approach provides a convenient, flexible and cost-effective solution, while minimizing the number of SKUs required by the integrator. Features:

• 8 individual channels per module, each driven by a 50W class D amplifier
• 8x 4” neodymium full-range speakers
• 16x 1.3” neodymium soft dome tweeters per module
• Columns up to 5.6m tall from only 2 SKUs (3x master + 3x slave modules)
• Up to 5 separately controllable beams per array, each with individual user-selectable acoustic centre
• User-definable 8 band parametric equalizer
• Look-ahead limiter
• User-defined delay

Interfaces:

• Power (100 to 240 VAC, 50 to 60 Hz)
• Balanced analogue line in (THD+N < 0,01%)
• AES in
• AES out
• RS485
• Optional DANTE / network control

Dimensions: 930mm x 132mm x 154mm per module RAL 9010 (custom RAL colour optional)

Evo-PBS is a self-powered, modular, electronically steerable column array speaker with excellent speech intelligibility and sound quality even in difficult acoustic environments. Evo-PBS is a self-powered, modular, electronically steerable column array speaker with excellent speech intelligibility and sound quality even in difficult acoustic environments. Evo PBS modules can be combined to build column arrays consisting of up to six modules. Each of the eight individual audio channels per module is driven by a 100W class D amplifier. The advanced beamsteering algorithm enables configuration of up to six individually controllable beams per array, each with a freely selectable acoustic centre. Additionally, side lobes can be suppressed by up to 20dB, maximizing speech intelligibility even in the most challenging acoustic environments. Each module contains:

• 8x 4¼” neodymium full-range speakers and
• 16x 1.1” neodymium tweeters

Additional features:

• FIR-based beamforming algorithm
• User-definable 8-Band parametric equalizer
• Look-ahead RMS and peak Limiter
• User-definable delay.
• Side Lobe Suppression (SLS)

Interfaces:

• Power (100 to 240 VAC, 50 to 60 Hz)
• Balanced analogue line in (THD+N < 0,01%)
• AES in
• AES out
• RS485
• Network (for configuration)
• DANTE (optional)

Dimensions: 930mm x 156mm x 170mm per module RAL 9010 (custom RAL colour optional)

Mastertone is a modular digital audio mixing matrix with a wide range of signal processing features. Mastertone is a modular digital audio mixing matrix with a wide range of signal processing features. The standard configuration offers 8 analogue inputs and 8 analogue outputs and can be extended by up to 8 additional inputs and 8 outputs (analogue or AES) plus 16 DANTE inputs and 16 DANTE outputs. As the core component of an audio system, Mastertone delivers a comprehensive featureset including delays, parametric equalizers, compressors, channel grouping, phantom power and more. Additionally, the sophisticated Open Microphone Controller (OMC) automatically attenuates microphones that are currently not actively in use, providing unclipped speech with maximum resistance to feedback. The responsive touchscreen display enables easy access to standard functions such as preset recall. Mastertone can be configured, controlled and monitored via dedicated software using the built-in network interface. This interface also facilitates external control of the Mastertone by third party control systems. Superb audio quality is assured by the inclusion of premium convertors, state of the art processing and advanced algorithms throughout the signal processing chain.

The Model 371A Intercom Beltpack begins with the features offered by 2-channel broadcast-style party-line intercom user devices then adds support for traditional single-channel headsets. Applications that have traditionally deployed single-channel analog party-line intercom systems from firms such as Clear-Com® can now easily upgrade to a high-performance dual-channel system. To maintain compatibility and user familiarity, legacy headsets that use a 4-pin female XLR connector are directly compatible. Key Features Dante audio-over-Ethernet technology Two independent talk and listen channels Integrated call function Supports dynamic and electret microphones 4-pin male XLR headset connector Excellent audio quality Uses STcontroller for configuration AES67 and Dante Domain Manager™ support Power-over-Ethernet (PoE) powered

The Model 348 Intercom Station offers eight talk-and-listen audio channels that can be configured for intercom, talent cueing (IFB), and REMI/At-Home applications. Utilizing Dante® audio-over-Ethernet network technology, the desktop unit offers extensive resources in a compact table-top enclosure. It is ideal for traditional sports and news broadcasting, live event, corporate AV, theater, and eSports applications.

The Model 792 Central Controller with Model 793 Control Console is a StudioComm for Dante® system designed for supporting high-channel-count, advanced multichannel monitoring applications. The Model 792 Central Controller with Model 793 Control Console is a StudioComm for Dante® system designed for supporting high-channel-count, advanced multichannel monitoring applications. Features include one 16-channel and one stereo Dante digital audio input and one 16-channel balanced analog input. Also includes one 16-channel and one stereo Dante monitor output and one 16-channel balanced analog monitor output. Three configurable Gigabit Ethernet network interfaces; two can be used for transporting audio (Redundant Dante) and the third for accessing the management menu system.

The DIO816AES is an inexpensive way to get your AES audio into and out of the Dante or AES67 network. The unit has eight stereo AES inputs and eight stereo AES outputs with remote network level control. It can handle multiple sample rates and includes a secondary port for Dante Redundancy or a switch output. The DIO816AES is an inexpensive way to get your AES audio into and out of the Dante or AES67 network. The unit has eight stereo AES inputs and eight stereo AES outputs with remote network level control. It can handle multiple sample rates and includes a secondary port for Dante Redundancy or a switch output. Both ports include network activity LEDs. Level and operational adjustments may be made using the DIO816AES’s web page.

20 channels Digital Automatic Mixer with Dante

64-channel Digital Automatic Mixer with Dante

Microphone array system with Dante

WDC-903 digital wireless conference discussion system, with professional appearance design, perfect circuit composition and stable functional characteristics, can be cooperated with VTS-2000 automatic video tracking system and central control system to achieve functional diversification of modern high-tech conference.  WDC-903 digital wireless conference discussion system, with professional appearance design, perfect circuit composition and stable functional characteristics, can be cooperated with VTS-2000 automatic video tracking system and central control system to achieve functional diversification of modern high-tech conference. The data instruction of this wireless conference discussion system is transmitted by LoRa protocol. Audio works in UHF band. High fidelity circuit design can produce high quality sound effect. The system has anti-interference protection, which can effectively block the influence of computers, mobile phones and Bluetooth wireless devices in the environment. It is fully applicable to local government departments, business conference rooms, business venues and international conference centers. Practical function

• Adopt LoRa 433Mhz wireless control signal and UHF audio transmission technology.
• The system automatically searches interference-free channels and replaces them.
• Control up to 100 Wireless Discussion units, 2 chairman units and up to 5 wireless units to speak at the same time.
• With Dante interface, it can communicate with any device with Dante to transmit audio.
• Bright OLED operation screen, with Chinese and English language menu selection, perfect function, easy operation.
• Super anti-interference function can effectively block the impact of computer devices, mobile phones and Bluetooth devices in the environment.

  Flexible Speech Four conference modes, first in first out, limited time speech, application speech and voice-activated speech, are available to meet different conference needs. Expansibility

• With antenna interface, it can be used in conjunction with the antenna system of Relacart, and the transmission signal is very strong.
• Any number of wireless conference systems can be installed in the same building.
• With VTS-2000 automatic video tracking system, can achieve conference video tracking function.
• Equipped with RS-232 interface, it can connect various central control devices.

Easy installation The unique advantages of simple installation, easy and fast, and no impact on the installation site, especially for the installation or layout of historical museums, ancient buildings and other historical and cultural buildings, users need not worry about damage to ancient buildings, because they do not need wiring, nor drilling holes, providing a professional perfect solution for harsh and complex application occasions.

MultiDyne’s 12G BULLET solves the problem of high-bandwidth, single-link transport to and from any video source. At only three inches in length, the 12G BULLET connects to the back of any camera, monitor or other video source, with the ability to add more units without additional power cabling. A camera operator with four video outputs can interlock four units together, and support a multichannel production off of a single power supply without the need for multiple wall warts. The new BULLET for Dante and Ethernet adds another layer of efficiency to field and mobile production teams managing digital audio feeds over long-distances. The common interlocked mounting capability across both products means that users have a quickly scalable and problem-solving option for video, audio and Ethernet feeds in the field. That patented interlocked mounting innovation, along with centralized power distribution, are key differentiators from previous BULLET generations and competitive products. The 12G BULLET integrates cable equalization and re-clocking techniques to optimize signal integrity across all short- and long-distance transport needs, removing the presence of jitter from existing signals. Integrated OLED monitoring adds value through consistent readings of jitter rates, signal type, optimal power and more. BULLET products are also available with a special 1RU tray capable of housing up to 24 devices, and support 12G SDI rates down to 5Mbps. This makes the BULLET series ideal for virtually any broadcast and AV-related field acquisition need, from high-end sports production and pre-fibered venues to courtesy feeds, confidence monitoring and distance learning.

The NA2-IO-DPRO allows microphone-level signals and AES/EBU signals to be injected into or taken out of the Dante network. Besides acting as a Dante converter, NA2-IO-DPRO can also operate as an AES/EBU converter: microphone inputs can simultaneously be injected into the Dante network and sent to an OB van (for example) via an AES/EBU signal fed from the device’s XLR output. The NA2-IO-DPRO allows microphone-level signals and AES/EBU signals to be injected into or taken out of the Dante network. Besides acting as a Dante converter, NA2-IO-DPRO can also operate as an AES/EBU converter: microphone inputs can simultaneously be injected into the Dante network and sent to an OB van (for example) via an AES/EBU signal fed from the device’s XLR output. Digital audio networks simplify routing and distribution of audio signals for fixed installations, touring and event applications. Neutrik‘s Dante interface series and the numerous mounting options provide a complete solution for integrating legacy audio gear into the Dante network.

MDA4-750PS is a highly flexible, powerful and intelligent 4-Channel power amplifier delivering up to a total of 4x750W @ 4 ohms and 4x870W @ 2 ohms, or able to drive 70V/100v Constant Voltage Lines MDA4-750PS is a highly flexible, powerful and intelligent 4-Channel power amplifier delivering up to a total of 4x750W @ 4 ohms and 4x870W @ 2 ohms, or able to drive 70V/100v Constant Voltage Lines. The 4 output channels can be bridged into 2, and provide 1500W@8ohms power output. Designed to meet the most demanding portable and fixed installation sound systems, it provides a full set of value added features such as high output power, efficient cooling system, on board DSP and USB/Ethernet for monitoring and control via PC software. MDA4-750PS includes a highly efficient Switch Mode Power Supply, which provides power to the output stages. The 4 output stages use the well-proven Class D module-full bandwith PWM modulator obtaining ultra low distortion, high efficiency and also equipped with a full set of circuit protections. Furthermore the Clip/Limiter function provides output monitoring to prevent speaker damage with gentle gain reduction at clip threshold, in addition to the efficient heat dissipation system and Over-Heat protection which themselves ensure uncompromised reliability. MDA4-750PS is more than just an amplifier. It is also a capable and sophisticated loudspeaker processor, thanks to its powerful MARANI® DSP running 96 kHz / 24 bit [ 96 bits precision for the internal intermediate processes] and high performance 24bit AD/DA Converters. It offers 4 channels of slope up to 48dB/Oct IIR HP/LP crossover filters, or up to 512 taps FIR filters[FIR coefficients can be imported as .txt/.csv  file from external application] , RMS compressor, parametric Eqs , alignment delay and white/pink noise internal generator, everything needed to optimize a loudspeaker system.

MDA4-2000PS is a highly flexible, powerful and intelligent 4-Channel power amplifier delivering up to a total of 4x1700W @ 4 ohms and 4x2320W @ 2 ohms, or able to drive 70V Constant Voltage Lines. MDA4-2000PS is a highly flexible, powerful and intelligent 4-Channel power amplifier delivering up to a total of 4x1700W @ 4 ohms and 4x2320W @ 2 ohms, or able to drive 70V Constant Voltage Lines. Designed to meet the most demanding portable and fixed installation sound systems, it provides a full set of value added features such as high output power, efficient cooling system, on board DSP and USB/Ethernet for monitoring and control via PC software. MDA4-2000PS includes a highly efficient Switch Mode Power Supply, which provides power to the output stages. The 4 output stages use the well-proven Class D module-full bandwith PWM modulator obtaining ultra low distortion, high efficiency and also equipped with a full set of circuit protections. Furthermore the Clip/Limiter function provides output monitoring to prevent speaker damage with gentle gain reduction at clip threshold, in addition to the efficient heat dissipation system and Over-Heat protection which themselves ensure uncompromised reliability. MDA4-2000PS is more than just an amplifier. It is also a capable and sophisticated loudspeaker processor, thanks to its powerful MARANI® DSP running 96 kHz / 24 bit [ 96 bits precision for the internal intermediate processes] and high performance 24bit AD/DA Converters. It offers 4 channels of slope up to 48dB/Oct IIR HP/LP crossover filters, or up to 512 taps FIR filters[FIR coefficients can be imported as .txt/.csv file from external application] , RMS compressor, parametric Eqs , alignment delay and white/pink noise internal generator, everything needed to optimize a loudspeaker system.

MDA2-2500P is a highly flexible, powerful and intelligent 2-Channel power amplifier delivering up to a total of 2x2150W @ 4ohms. MDA2-2500P is a highly flexible, powerful and intelligent 2-Channel power amplifier delivering up to a total of 2x2150W @ 4ohms. Designed to meet the most demanding portable and fixed installation sound systems, it provides a full set of value added features such as high output power, efficient cooling system, on board DSP and USB/Ethernet for monitoring and control via PC software. MDA2-2500P includes a highly efficient Switch Mode Power Supply, which provides power to the output stages. The 2 output stages use Class D amplifier module-full bandwidth PWM modulator obtaining ultra low distortion, high efficiency and also equipped with a full set of circuit protections. Furthermore the Clip/Limiter function provides output monitoring to prevent speaker damage with gentle gain reduction at clip threshold, in addition to the efficient heat dissipation system and Over-Heat protection which themselves ensure uncompromised reliability. MDA2-2500P is more than just an amplifier. It is also a capable and sophisticated loudspeaker processor, thanks to its powerful MARANI® DSP running 96 kHz / 24 bit [ 96 bits precision for the internal intermediate processes] and high performance 24bit AD/DA Converters. It offers 2 channels of slope up to 48dB/Oct IIR HP/LP crossover filters, or up to 512 taps FIR filters[FIR coefficients can be imported as .txt/.csv file from external application] , RMS compressor, parametric Eqs , alignment delay and white/pink noise internal generator, everything needed to optimize a loudspeaker system.

LPP240A is a High End 2-IN/4-OUT digital speaker management system which maintains the same outstanding Sound Quality of the former DPA240A version, but highly improved in terms of DSP processes and added of an astonishing FIR filtering tool set. LPP240A is a High End 2-IN/4-OUT digital speaker management system which maintains the same outstanding Sound Quality of the former DPA240A version, but highly improved in terms of DSP processes and added of an astonishing FIR filtering tool set. Designed for maximum versatility, it provides all the processing and control necessary for both live and fixed installation use. Proving for any crossover configuration, it offers 2 analog inputs, 4 analog outputs and 1 AES/EBU input, managed by 2 powerful MARANI® DSP Engines, for a full 96kHz processing, in addition to 24 Bit AD/DA Converters. Each input channel provides a choice of a 13-band Parametric EQ, Gain control, Noise Gate, RMS Compressor, Internal White/Pink Noise Generator, and configurable Delay. Each output offers up to 7-band of parametric equalization, in addition to the IIR crossover filters which themselves provide slopes from 6dB/Octave up to 48dB/Octave. Each output path also features PEAK Limiter, RMS Compressor and configurable Delay. Particularly, the Output X-Over Filtering can be selected to be IIR style, with the complete set of Classic Hp/Lp filters (Butterworth, Linkwitz-Railey, Bessel) or FIR, being for each output channel available a 512 taps FIR filter which can be set as Hp/Lp/Bp. On each one of the 2 Input paths, one more 1024 taps Asymmetrical FIR is available for Phase Correction purposes, where the Coefficients Calculation can be supported by an embedded Wizard tool or can be Loaded from external sources as Third Parties Sw for the FIR filters coefficients generation. Being the phase correction FIR Asymmetrical, a powerful tool for adjusting/reducing the FIR latency is available, allowing therefore the LPP240A to be used without any problem also for Live performances still having the Phase correction FIR running together with the eventual FIR for the X-Over implementation in cascade. The LPP240A supports a full set of matrix mixing modes where Input sources can be routed/mixed to the Input paths, inputs paths may be routed/mixed in any ratio to any output and couple of outputs (1/2, 3/4) can be swapped at pleasure.

LPP260A is a High End 2-IN/6-OUT digital speaker management system which maintains the same outstanding Sound Quality of the former DPA260A version, but highly improved in terms of DSP processes and added of an astonishing FIR filtering tool set. LPP260A is a High End 2-IN/6-OUT digital speaker management system which maintains the same outstanding Sound Quality of the former DPA260A version, but highly improved in terms of DSP processes and added of an astonishing FIR filtering tool set. Designed for maximum versatility, it provides all the processing and control necessary for both live and fixed installation use. Proving for any crossover configuration, it offers 2 analog inputs, 6 analog outputs and 1 AES/EBU input, managed by 2 powerful MARANI® DSP Engines, for a full 96kHz processing, in addition to 24 Bit AD/DA Converters. Each input channel provides a choice of a 13-band Parametric EQ, Gain control, Noise Gate, RMS Compressor, Internal White/Pink Noise Generator, and configurable Delay. Each output offers up to 7-band of parametric equalization, in addition to the IIR crossover filters which themselves provide slopes from 6dB/Octave up to 48dB/Octave. Each output path also features PEAK Limiter, RMS Compressor and configurable Delay. Particularly, the Output X-Over Filtering can be selected to be IIR style, with the complete set of Classic Hp/Lp filters (Butterworth, Linkwitz-Railey, Bessel) or FIR, being for each output channel available a 512 taps FIR filter which can be set as Hp/Lp/Bp. On each one of the 2 Input paths, one more 1024 taps Asymmetrical FIR is available for Phase Correction purposes, where the Coefficients Calculation can be supported by an embedded Wizard tool or can be Loaded from external sources as Third Parties Sw for the FIR filters coefficients generation. Being the phase correction FIR Asymmetrical, a powerful tool for adjusting/reducing the FIR latency is available, allowing therefore the LPP260A to be used without any problem also for Live performances still having the Phase correction FIR running together with the eventual FIR for the X-Over implementation in cascade. The LPP260A supports a full set of matrix mixing modes where Input sources can be routed/mixed to the Input paths, inputs paths may be routed/mixed in any ratio to any output and couple of outputs (1/2, 3/4, 5/6) can be swapped at pleasure.

LPP260A is a High End 2-IN/6-OUT digital speaker management system which maintains the same outstanding Sound Quality of the former DPA260A version, but highly improved in terms of DSP processes and added of an astonishing FIR filtering tool set. LPP480A is a High End 4-IN/8-OUT digital speaker management system which maintains the same outstanding Sound Quality of the former DPA480 version, but highly improved in terms of DSP processes and added of an astonishing FIR filtering tool set. Designed for maximum versatility, it provides all the processing and control necessary for both live and fixed installation use. Proving for any crossover configuration, it offers 4 analog inputs, 8 analog outputs and 2 AES/EBU inputs, managed by 2 powerful MARANI® DSP Engines, for a full 96kHz processing, in addition to 24 Bit AD/DA Converters. Each input channel provides a choice of a 13-band Parametric EQ, Gain control, Noise Gate, RMS Compressor, Internal White/Pink Noise Generator, and configurable Delay. Each output offers up to 7-band of parametric equalization, in addition to the IIR crossover filters which themselves provide slopes from 6dB/Octave up to 48dB/Octave. Each output path also features PEAK Limiter, RMS Compressor and configurable Delay. Particularly, the Output X-Over Filtering can be selected to be IIR style, with the complete set of Classic Hp/Lp filters (Butterworth, Linkwitz-Railey, Bessel) or FIR, being for each output channel available a 512 taps FIR filter which can be set as Hp/Lp/Bp. On each one of the 4 Input paths, one more 1024 taps Asymmetrical FIR is available for Phase Correction purposes, where the Coefficients Calculation can be supported by an embedded Wizard tool or can be Loaded from external sources as Third Parties software for the FIR filters coefficients generation. Being the phase correction FIR Asymmetrical, a powerful tool for adjusting/reducing the FIR latency is available, allowing therefore the LPP480A to be used without any problem also for Live performances still having the Phase correction FIR running together with the eventual FIR for the X-Over implementation in cascade. The LPP480A supports a full set of matrix mixing modes where Input sources can be routed/mixed to the Input paths, inputs paths may be routed/mixed in any ratio to any output and couple of outputs (1/2, 3/4, 5/6, 7/8) can be swapped at pleasure. For remote configuration and control the LPP480A can be connected via USB/RS485/TCP-IP connections. The control remote PC software allows simultaneous control up to 32 units, setting all parameters and showing real time levels. 

LPP480F is a high end 4-IN/8-OUT digital speaker management system which is MARANI® flagship controller, adding astonishing hardware and software features on the basis of the LPP series. LPP480F is a high end 4-IN/8-OUT digital speaker management system which is MARANI® flagship controller, adding astonishing hardware and software features on the basis of the LPP series. Designed for maximum versatility, it provides all the processing and control necessary for both live and fixed installation use. Proving for any crossover configuration, it offers 4 analog inputs, 8 analog outputs, 2 Stereo AES/EBU inputs, and 4 Dante™ inputs, managed by 3 powerful MARANI® DSP Engines, for a full 96kHz processing, in addition to 24 Bit Texas Instruments AD/DA Converters. Each input channel provides a choice of a 13-band Parametric EQ, with plenty of filter types, Gain control, Noise Gate, RMS Compressor, Internal White/Pink Noise Generator, and configurable Delay. Each output offers up to 7-band of parametric equalization, in addition to the IIR crossover filters which themselves provide slopes from 6dB/Octave up to 48dB/Octave. Each output path also features PEAK Limiter, RMS Compressor and configurable Delay. Output X-Over Filtering can be selected to be IIR style, with the complete set of Classic Hp/Lp filters (Butterworth, Linkwitz-Railey, Bessel) or FIR, being for each output channel available a 512 taps FIR filter which can be set as Hp/Lp/Bp. On each one of the 4 Input paths, one more 1024 taps FIR is available for Phase Correction purposes, where the Coefficients Calculation can be supported by an embedded Wizard tool or can be loaded from external sources as Third Parties software for the FIR filters coefficients generation. Being the phase correction operated by an FIR Filter which can be Asymmetrical other than Symmetrical, a powerful tool for adjusting/reducing the FIR latency is available, allowing therefore the LPP480F to be used without any problem also for Live performances still having the Phase correction FIR running together with the eventual FIR for the X-Over implementation in cascade. The LPP480F supports a full set of matrix mixing modes where Input sources can be routed/mixed to the Input paths, inputs paths may be routed/mixed in any ratio to any output and couple of outputs (1/2, 3/4, 5/6, 7/8) can be swapped at pleasure. For remote configuration and control the LPP480F can be connected via USB /TCP-IP Interfaces. Moreover, with front panel dedicated buttons, user can go directly to the editing pages like Xover, channel editing windows, while can monitor the real-time meters and other parameters from the LCD screen.

Manage and control your broadcast events quickly, easily and efficiently from any global location with the ST150. Manage and control your broadcast events quickly, easily and efficiently from any global location with the ST150. Ideally suited for HD studio broadcast contributions where there are limited on-site technical resources available. The camera, lights and audio can be controlled from any Chrome browser using the Quicklink Manager Portal to ensure the highest quality, in frame broadcast content is received. The central operator fully manages the broadcast and can choose both file recording as HD-SDI and live output to air.

• Rack or desktop
• Use with your own monitor/teleprompt
• Use your own Panasonic PTZ camera

Manage and control your broadcast events quickly, easily and efficiently from any global location with the ST500. Ideally suited for HD studio broadcast contributions where there are limited on-site technical resources available. The camera, lights and audio can be controlled from any Chrome browser using the Quicklink Manager Portal to ensure the highest quality, in frame broadcast content is received. The central operator fully manages the broadcast and can choose both file recording as HD-SDI and live output to air.

• Compact and portable
• Inbuilt professional Panasonic PTZ Camera
• Inbuilt lighting with remote controllable brightness

Musicrown's DL Dante Connect Box series is an I/O interface. Musicrown's DL Dante Connect Box series is an I/O interface.

• High quality A/D(24-Bit), D/A(32-Bit) convert, 48KHz sampling rate.
• DL2.2 is 2 channel balanced MIC/LINE XLR input, 2 channel balance XLR output.
• DL2.0 is 2 channel balanced MIC/LINE XLR input.
• DL0.2 is 2 channel balance XLR output.
• Dante digital network port configuration is a double standard network interface, which fully supports the function of unite switching and hot backup. Digital audio channel can support 2IN/2OUT, using 24-bit precision and extremely low network latency.
• Each input channel with independent signal light and gain control, also with +48V phantom power controlled by switch. Input channel 1 has high resistance switch select, support guitar, bass, keyboard musical instruments equipment input; input channel 2 extra with two group stereo input port (pair RCA input and one 3.5 socket), supporting media player, notebook computer and other equipments signal input.
• Each output channel with independent signal light.
• Dante can be used in the standard existing network interchanger, has highly interoperability: perfect cooperation with other start using Dante device.
• Outer connect +12VDC power supply, Power switch independently controlled and power indicator light.

DL Series:

• DL2.2 is 2 channel balanced MIC/LINE XLR input, 2 channel balance XLR output
• ●  DL2.0 is 2 channel balanced MIC/LINE XLR input
• ●  DL0.2 is 2 channel balance XLR output