Dante-enabled Product Catalog

The audio processing center is an audio processor with 16 inputs (8 analog and 8 digital) and 16 outputs (8 analog and 8 digital). The industry-leading ADI DSP SHARC21489 is used, and the maximum operating frequency is up to 450MHz. The input and output modules of DSP can be customized according to the actual situation. At the same time, it has independent AFC, AEC, ANS, AGC, gain sharing automatic mixing, threshold automatic mixing and other processing modules to meet the audio processor and transmission needs of various places. Such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.

The audio processing center is an audio processor with 24 inputs (16 analog and 8 digital) and 24 outputs (8 analog and 16 digital), using industry-leading ADI DSP SHARC21489, the highest operating frequency can reach 450MHz. The input and output modules of DSP can be customized according to the actual situation. At the same time, it has independent AFC, AEC, ANS, AGC, gain sharing automatic mixing, threshold automatic mixing and other processing modules to meet the audio processor and transmission needs of various places. Such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.

The network audio processor is a network audio processor with 28 inputs (12 analog and 16 digital) and 28 outputs (12 analog and 16 digital). It adopts the industry-leading ADI DSP SHARC21489, the highest The working frequency can reach 450MHz. The input and output modules of DSP can be customized according to the actual situation. At the same time, it has independent AFC, AEC, ANS, AGC, gain sharing automatic mixing, threshold automatic mixing and other processing modules to meet the audio processor and transmission needs of various places. Such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.

The audio processing center is an audio processor audio processing center with 32 inputs (16 analog and 16 digital) and 32 outputs (16 analog and 16 digital). It is an audio processing center with 16 analog inputs and 16-channel analog output audio processor, using industry-leading ADI DSP SHARC21489, the highest operating frequency can reach 450MHz. The input and output modules of DSP can be customized according to the actual situation. At the same time, it has independent AFC, AEC, ANS, AGC, gain sharing automatic mixing, threshold automatic mixing and other processing modules to meet the audio processor and transmission needs of various places. Such as meeting rooms, multi-function halls, conference center auditoriums, administrative centers, etc.

A network media matrix mainly used for local meetings, education and remote interactive education. It is equipped with excellent digital audio processing technology, eight-channel audio processing, including full-band adaptive echo cancellation, automatic noise suppression, and feedback suppression algorithms. Long-distance pickup microphone, easy to install, simple to operate, comes with 6+1 network ports, supports PoE, 100/1000M adaptive, 100 meters transmission distance, supports IEEE802.3af/at standard, and the maximum power of a single port is 60W. Easily meet the high-quality voice needs of local sound reinforcement, recording and broadcasting, and remote interactive education.

A network media matrix mainly used for local meetings, education and remote interactive education. It is equipped with excellent digital audio processing technology, eight-channel audio processing, including full-band adaptive echo cancellation, automatic noise suppression, and feedback suppression algorithms. Wireless microphone pickup, exquisite pen microphone, automatic frequency linking at boot, simple operation; comes with 6+1 network ports, supports PoE, 100/1000M adaptive, 100 meters transmission distance, supports IEEE802.3af/at standard, single port The maximum power is 60W. Easily meet the high-quality voice needs of local sound reinforcement, recording and broadcasting, and remote interactive education.

Features: The D series converter box is compatible with all Dante interfaces and has all the features of Dante audio network. Each conversion box can provide high-fidelity sound quality restoration, ultra-low latency, and accurate synchronization of sampling rates across the entire network. Compact design, easy to use, plug and play; conversion between analog audio signals and Dante digital audio signals; expand audio and power through a network cable; fully support Dante software, easy to configure the network and audio routing; support PoE power supply, Built-in lightning and surge protection.

The D series converter box is compatible with all Dante interfaces and has all the features of Dante audio network. Each conversion box can provide high-fidelity sound quality restoration, ultra-low latency, and accurate synchronization of sampling rates across the entire network. Compact design, easy to use, plug and play; conversion between analog audio signals and Dante digital audio signals; expand audio and power through a network cable; fully support Dante software, easy to configure the network and audio routing; support PoE power supply, Built-in lightning and surge protection.

The D series converter box is compatible with all Dante interfaces and has all the features of Dante audio network. Each conversion box can provide high-fidelity sound quality restoration, ultra-low latency, and accurate synchronization of sampling rates across the entire network. Compact design, easy to use, plug and play; conversion between USB audio signal and Dante digital audio signal; expand audio and power through a network cable; fully support Dante software, easy to configure network and audio routing; support USB power supply, Built-in lightning and surge protection; it can be used on any PC, Mac or Linux system without reconfiguring the Dante network. It can record and play dual-channel audio at the same time without any other software.

Features: ●The high-quality pre-op amplifier guarantees the output sound quality and controls the output volume of each channel ●Advanced digital reverberation modulation function ●Twelve microphone input sockets. Independent phantom power supply for each channel, independent input volume adjustment, independent switch indicator ●The device itself provides a mixing channel, mixing input twelve microphones (each microphone can adjust the volume of the output to the mixing channel), while controlling the final output mixing volume ●Each input channel and mixing channel have an independent output, and the input channel and mixing channel can be monitored locally ●The equipment allows nearby installation, and the network transmits the audio of independent channels and mixed channels to the remote power amplifier or mixer ●With device superimposition function (when there are more than twelve microphones, two sets of this device can be superimposed and used) ●Twelve channels of microphones can be adjusted individually, which can also use various types and indicators of microphones to ensure that the microphone volume input and output in the system are balanced ●Only one audio equipment of the power amplifier, with this equipment can drive multiple, even dozens of microphones to work ●The panel displays the network status, which is convenient for fault diagnosis ●RS232/RS485 remote control interface reserved ●Only one network port, daisy chain connection is not allowed ●The network IP of the device can be set independently ●Microphone connection matching: phantom or non-phantom powered microphone ●Microphone connection: 12 XLR sockets

MDA 4-7000DM is a high-performance and high-power touring-grade digital power amplifier. It has a total power output of 4x1600W under a 4-channel 4Ω load. Built-in powerful digital processor with 4 inputs and 4 outputs. The 4.3-inch high-definition touch screen clearly displays the system status, and the sufficient power is equipped with a powerful built-in DSP to enable this power amplifier to take into account both the fixed installation and the mobile performance market.

MDA4-7000DM adopts powerful Marani DSP, including 4 input and 4 output processing channels. Both DSP and AD/DA run at 96KHz sampling rate. The complete processing function provides a complete crossover solution for the speaker. With signal hot backup automatic switching function, 3 levels of priority can easily back up the input source.

From input gain/delay/noise gate/EQ/compression

/FIR /to output gain/delay/polarity/X-over/FIR/EQ/RMS compressor/Peak limiter, there are up to 13 PEQ types to choose, The output crossover filter includes the classic Linquez Rayleigh/Bessel/Butterworth, and the MARANI brand unique NXF（Northed X-over Filter）, built-in FIR filter, and newly added MIR linear phase filter can make the phase of the crossover point easier to join while maintaining a very low delay. All the functions we provide are designed to help you better restore the sound.

The Mir-F family of digital processors are Marani's new flagship system processors, including the MIR480I and MIR880F.The hardware adopts top AD/DA chips with excellent -96dBu background noise. It also has analog, digital AES/EBU, and Dante signal access capabilities, and has automatic switching function. Any signal can be set as the highest priority. As the core processor of the system, it has the redundancy and backup characteristics, and is suitable for government projects, high-end clubs, large mobile performances, theaters, radio and television and other places with higher requirements for signal processing.

The compact and portable DT20 converts two analogue inputs, via combo XLR-Jack, to two channels of Dante audio, enabling hassle-free connection of analogue sources to a Dante network for install and live applications.

The compact and portable DT02 converts two channels of Dante audio to a pair of XLR line-outputs, enabling simple break-out to analogue equipped hardware for install and live applications.

DT22 is a compact 48kHz/96kHz Dante interface enabling convenient placement of 2 mic/line inputs and 2 line outputs wherever you need to get analogue audio in and out of a Dante network. Equipped with standard Phoenix / euroblock connectors, DT22 is designed for fixed installation and can be easily mounted under a meeting table, for example in a breakout room or huddle space, under a podium for local mic/PC and speaker connection, or rack-mounted to interface with analogue equipment such as an AV switcher.

dLive CTi1500 takes our best-selling compact dLive C1500 control surface and applies 'superleggera' Italian performance car principles to set a new standard for lightweight control surfaces. CTi1500 has been engineered across the board to ruthlessly strip out as much weight as possible, while ensuring tour-grade structural integrity and a first-class user experience.

SQ Dante V2 provides any SQ mixer or AHM processor with a full 64x64 channel, 96kHz or 48kHz interface to a Dante network. An SQ fitted with the Dante option card can seamlessly connect to amplifiers, preamps and other devices on the Dante network.The SQ Dante V2 card can be switched between 96kHz or 48kHz to accommodate Dante networks running at either sample rate. In addition to facilitating system integration, the card also enables digital splits and affordable multitrack recording direct to a computer.

DANTE Audio DSP Software
Support:
1. All MONISMS DANTE DSP audio products
2. UDP protocol
3. EQ: 2*15BQs
4. DBE/DDE/DEQ
5. 3-Band advanced DRC + AGL (Automatic Gain Limiter)
6. Accept DSP software customization service

Sonia Amp is a PoE+ IP-addressable amplifier designed to power up to eight (8) Sonia C5 5.25" speakers from a single unit. Offering four (4) distinct channels that can supply Sonia C5 speakers with either the identical or different audio feeds, Sonia Amp allows a single amplifier to drive audio in anywhere between one and four different rooms. Sonia Amp is entirely CatX- based, requiring no proprietary audio connectors or traditional copper speaker wire. Instead, IT technicians can install Sonia systems with standard Category cable and RJ45 connectors, dramatically lowering time and cost of deployment. Unlike traditional amplifiers that require a dedicated equipment rack for mounting and ventilation, Sonia Amp mounts directly to the rear of any Sonia C5 speaker, eliminating in-room storage and permitting a full room speaker system to be hidden in a suspended ceiling. Sonia Amp is convection cooled without any fan noise and is Plenum rated to enable installation in air handling spaces. Sonia Amp is Dante enabled, AES67-compatible, and offers an integrated network switch with PoE to enable the powering of a ceiling-mounted microphone array. In this respect, Sonia Amp adds additional value to the IT technician by creating just one single Category cable drop to the ceiling for the full room audio system. When paired with a Xilica Solaro QR1, an entire audio system can be deployed in the drop-ceiling. Sonia Amp additionally offers fault sensing on each channel, providing the same monitoring capabilities as an individually addressable IP speaker without the cost.

A portable video recorder that uses CFexpress as a recording media. This unit supports the recording and playback of 4K 120p images on a stand-alone basis, and by connecting multiple units, you can record and play video with any extended resolution and frame rate.

Features:

● 4K YC_BC_R 4:2:2 120p can be recorded on a stand-­alone basis
● A CFexpress TypeB card (sold separately) is used for recording media
● Compatible with fixed 1/6 compression recording
● Recording times are 20 minutes with fixed 1/6 com­pression when recording 4K 120p to 640GB CFexpress
● Supports relay capturing with dual slots
● Equipped with a function to slowly playback recorded 4K YC_BC_R 4:2:2 120p video at 60p or less
● Multiple video recorders can be connected for arbi­trary extended resolutions in units of 4K (extended in the future)
● Multiple-unit connected operation enables video recording at an arbitrary extended frame rate with 60p/120p as one unit (future expansion)
● Supports 12G/6G/3G/HD-SDI, HDMl2.0 (OUTPUT)
● Supports 25G/100GbE (future extensions/options),25GbE/100GbE complies with SMPTE ST 2110-10/-20/-30/-40
● Equipped with TIMECODE input/output terminals(LTC only)
● Equipped with MADI input/output terminals
● Equipped with 2 channels of analog audio input and output connectors
● Supports Audinate Dante
● Genlock(BB/3 Synchronization Signal) input is equipped
● Supports up to 32-channel audio (uncompressed, 24-bit) recording
● Equipped with both DC power input and AC power input

The PDS-4K is a full-screen presentation switcher for smaller applications – from breakout rooms to mid-sized events. It is designed with the rental industry in mind to meet today's requirements for high-quality and fast seamless switching with digital inputs.

With a single unit you can drive up to two 4K main screens and include a backstage MultiViewer for operators to monitor the whole system.

Simplicity, ease-of-use and users’ convenience are key. Leveraging Barco’s expertise in the rental & staging industry in the workplace to enable high-end corporate presentations with unprecedented intuitiveness, outstanding image quality and reliability. Ensuring the viewing comfort of board members and guests, presenters only need one click to switch between content and access a wide range of layouts and a set of professional transitions.

The PDS-4K is compatible with all the familiar Barco visualization and collaboration solutions. It is a powerful tool for flawless presentations in combination with ClickShare. A long-lasting, reliable investment bringing the best visual experiences to wow and impress.

The BinloopX is a modular multi-channel Audio/Video platform designed to handle all of your high-quality synchronized playback needs.

The industry’s favorite synchronized A/V player has been completely reengineered with our X-Series architecture to be the ultimate A/V platform for themed attractions. By leveraging the power of our industry-standard show control engine and the precision of our advanced SyncCore synchronization system, the BinloopX integrates seamlessly with the Alcorn McBride ecosystem of products. This makes it the perfect solution to add high-quality A/V playback to any attraction.

The BinloopX itself is a 3U chassis complete with redundant power supply, BX-CON1 control module, TFT status displays, and 8 slots that can be used to install A/V modules as needed. The BX-CON1 control module is equipped with our X-Series show control platform and SyncCore system, which provides extremely versatile control features and precise synchronization with other attraction systems via PTP, NTP, GPS, SMPTE LTC or Genlock. This system also provides sample-accurate audio clocks and frame-accurate video clocks for all A/V modules installed within the BinloopX chassis.

With BX-16A audio modules, a single BinloopX chassis can provide up to 128 outputs of 24-bit 48kHz uncompressed audio with up to 512 tracks of audio playback that can be mixed to those outputs. Outputs are provided using Dante and AES67 network audio for easy integration. These modules can be installed as needed to scale the capabilities of the BinloopX to meet the needs of your attraction.

Experience the ultimate in audio and video playback with Alcorn McBride reliability.

The Model 209 Talent Console offers a unique combination of audio, remote control, and tally-output resources for use in podcast and other specialized audio applications. The unit is housed in a compact enclosure that's intended for mounting under a tabletop, within custom furniture, or as part of an on-air set.

The Model 209 supports Dante® audio-over-Ethernet digital media technology and AES67 for integration into contemporary applications. Dante Domain Manager™ software application is supported.

The unit is simple to deploy, is "pro" quality, and provides an intuitive user experience. The audio quality is excellent, with low distortion, low noise, and high headroom. Careful circuit design and rugged components ensure long, reliable operation.

The Model 209 integrates directly into Dante audio-over-Ethernet environments. A PoE connection, microphone, and a pair of headphones or earpiece create a complete audio "position." Remote control inputs and tally outputs allow the unit to integrate into complex applications.

Media Axis is the world’s first large-scale matrix switching and extension system supporting Ultra HD/4K60 4:4:4 via Native and IP architectures. With its dynamic, proprietary design features and technologies, Media Axis provides a truly adaptive solution that works where and how you need it.

No other matrix system provides a more comprehensive set of features to support scaling, routing, and processing of high bandwidth content. Media Axis is the revolutionary new breed of matrix switchers.

The MAX-AI-DA is a Dante™ Audio with HDMI 2.0 ports input card for Media Axis™ supporting 4K/60 4:4:4 resolutions and Dante™/AES67 IP audio. Each MAX-AI-DA has primary and secondary RJ-45 ports for IP audio, four HDMI ports, and four stereo unbalanced analog ports. Media Axis is the world's only 4K60 cross platform, multi frame size matrix system with Dante and AV over IP connectivity.

- Maximum supports 35.5Wx2@8Ω or PBTL mode for 71W@4Ω
- Built in Audio DSP for Volume, PEQ, Filter and Input Gain controls
- Dante™ Ultimo insides with hi performance Class D Audio DSP Amplifier
- DAN-AMP-2ch - Config Tool software to configure audio DSP
- Fully protection with OTP, OTP, OVP, Auto Recovery and Surge Protection
- Supports Dante™ Domain Manger and AES67 compliant

The SAS Luna provides 12 high quality microphone preamps, each with a very wide gain range and individually switchable +48 VDC phantom power.  Luna also provides 8 line level fully floating audio outputs.  An embedded web server allows configuration and monitoring of inputs and outputs.  The Luna pairs well with an SAS Nucleus DSP Engine to provide comprehensive dynamics processing, EQ and AGC.