Dante-enabled Product Catalog

The AQZ32 integrates AquaControl™ with Dante® Embedded technology, significantly enhancing its I/O capabilities making it a versatile addition to any audio setup. Powered by Ashly’s next-generation AquaControl™ DSP engine, the AQZ32 offers an intuitive, web-browser based interface that requires no specialized training, getting you up and running quicker and smoother. It includes 16 analog mic/line inputs each with software controlled 48V Phantom Power and mic gain settings up to 66dB. It also has 16 analog balanced line outputs. Dante inputs 1-16 can be independently selected instead of each corresponding analog input number. So, you can choose if you need processed analog or Dante sources. Dante inputs 17-32 are available on each mixer to expand the mix with these Dante sources. Dante outputs 1-32 allow you to route processed inputs and outputs to amps and other devices. A pair of AQZ32s can be setup to provide a 32x32 audio system.

The DCS-AMP12004 is a high-performance power amplifier that combines Class-D amplification technology with an SMPS switched-mode power supply and is designed for large-scale audio system applications. Its compact size, low power consumption, and high efficiency deliver reliable performance for demanding installations. Intelligent power sharing further enhances system flexibility by dynamically allocating available power across channels for maximum utilisation. One of the key strengths of the DCS-AMP12004 is its intelligent power sharing capability. A total output of 1200 W can be flexibly distributed across four channels, with up to 600 W available per channel. This allows the system to adjust channel output dynamically according to actual load requirements, avoiding the inefficiencies associated with conventional fixed-power designs. It is particularly well suited to complex applications such as public address systems, multi-zone music systems, airports, metro systems, nuclear power plants, and sports venues.    

The Flex EIGHT-N joins the miniDSP Flex family as a Dante‑capable audio processor that natively streams Dante/AES67 audio. It delivers eight Dante/AES67 AoIP channels plus four analog outputs for seamless, low‑latency multi-channel distribution within any Dante ecosystem.

Featuring multiple digital inputs, eight digital outputs and four analog outputs, the Flex EIGHT-N can drive active speakers up or be set up as a multi-zone processor via our flexible routing. Every output has miniDSP’s full processing suite: parametric EQ, IIR and FIR filters, configurable crossovers, advanced biquad programming and precise delay.

Digital inputs include USB, S/PDIF (RCA and optical) and Bluetooth (LDAC/aptX). An OLED front panel and encoder/volume knob provide local control, and firmware supports upgrades such as Dirac Live room correction.

In short, the Flex 8N is a compact, Dante‑ready powerhouse for modern active speaker and subwoofer systems, offering direct network streaming and sophisticated loudspeaker processing.

The Microflex MXN-6 Networked Pendant Loudspeaker provides high-quality speech reproduction for AV conferencing applications. The PoE-powered MXN-6 is the ideal loudspeaker solution when used as part of a Shure networked conferencing audio system. Combined with Microflex Advance Array Microphones and IntelliMix® DSP solutions, the full Shure Microflex Ecosystem delivers premium audio performance with an unmatched ease of deployment.

AX Center It's the audio interface that dreams are made of, as it unites the distinguished audio quality of pristine Digital Audio Denmark AD & DA conversion with the renowned transparency and musicality of the DAD mic preamps. It's all about capturing the source truthfully and get it into your DAW exactly as it is. But that's only the beginning of this journey through the various applications covered by AX Center. On the analog side, two XLR / 1/4" Jack combo connectors on the front for connecting microphones and instruments, you can add two 8 ch. mic/line A/D cards for 16 mic inputs and you have the perfect interface for a small studio or location recording. Or add one 8 ch. D/A card for an additional 8 analog outputs and you have the perfect interface for mixing in 7.1.4 ATMOS format. With one 8 ch. line A/D card and one 8 ch. D/A card, you have the perfect mastering converter. And all along you have two headphone connections where all sources in the system can be easily monitored. Included on the rear panel are a host of connectivity, covering Thunderbolt, Dante AoIP, MADI, ADAT, S/PDIF and Word Clock. In total - and fully expanded - you can max out AX Center with a staggering 984 input and output channels.

Nova supports three main functions. Trinnov’s Optimizer is Nova’s main feature – Trinnov’s exclusive room correction and acoustic optimization technology. Nova can process and optimize up to six channels, including bass management. More than a digital acoustic processor, Nova is also a monitoring controller. NOVA differs from the previous product generation in that it’s PC-less. Trinnov’s new Audio Core uses an ARM processor that provides the computational power necessary to complete Trinnov processing of a limited number of channels without needing additional CPU or fans.

H2-4D , the world’s first dedicated 16 channel DOLBY ATMOS ® digital audio renderer

Primary functions:

Convert up to 16 Channels of HDMI audio simultaneously to:
DANTE® (AES67 included)
BALANCED
ANALOG
HDMI® (Audio Only)
AES/EBU (AES3)

Render:
DOLBY Atmos®
DTS X ®

Decode:
DOLBY® - Dolby Digital, Dolby Digital Plus, Dolby TrueHD , Dolby Surround (from Atmos
DTS® - DTS, ES, HD Master Audio, HD High Resolution Audio, Express HD LBR, Neural:X
PCM - 2ch, PCM Multi channel, Downmix

Dolphin系列数字音箱管理器内置高性能DSP处理芯片，支持多路模拟、Dante信号输入输出，自带31段参数均衡器，增益控制，压缩器，延迟等功能。

Dolphin系列数字音箱管理器内置高性能DSP处理芯片，支持多路模拟、Dante信号输入输出，自带31段参数均衡器，增益控制，压缩器，延迟等功能。

Dolphin系列数字音箱管理器内置高性能DSP处理芯片，支持多路模拟、Dante信号输入输出，自带31段参数均衡器，增益控制，压缩器，延迟等功能。

Dolphin系列数字音箱管理器内置高性能DSP处理芯片，支持多路模拟、Dante信号输入输出，自带31段参数均衡器，增益控制，压缩器，延迟等功能。

DAP0306_D4 is a 3in6out analog professional audio processor, which can achieve a variety of functions as a DSP processor including an integrated compressor, limiter, frequency divider, decelerator, equalizer, mixing matrix, supports 7 brands input and 7 brands output PEQ. It also can be debugged and monitored quickly by exquisite intuitive Mconsole software, for the construction of professional audio amplification system and provides a wide operation space.

★96K24BIT sample rate

★4 input channels DEQ and 1024 FIR, 8 output channels 512*FIR

★Control interface: USB and RJ45 by TCPIP protocol, configuring RS232, RS485 central control interface, supporting GPIO external expansion control interface     

★Support Mconsole and app TCON network control 

★Support presets archiving and locking, hide setting parameters"

PA1203-D2 is a DSP class-D amplifier module used in active speakers. Based on DSP signal processing technology, network technology, class-D amplifier and switching power supply technology. Configured with 2-channel Dante input interface to easily integrate into Dante network audio system. 

* 2*500W4Ω or 1*1000W8Ω

*DSP process module: DA14CL/D

*support 180-265VAC input

*IPS colorful display

*Connet by 2*RJ45 of network or USB free-driver

PA1203-D2 is a DSP class-D amplifier module used in active speakers. Based on DSP signal processing technology, network technology, class-D amplifier and switching power supply technology. Configured with 2-channel Dante input interface to easily integrate into Dante network audio system. 

* 2*500W4Ω or 1*1000W8Ω

*DSP process module: DA14CL/D

*support 180-265VAC input

*IPS colorful display

*Connet by 2*RJ45 of network or USB free-driver

PA2310-D2 is a DSP class-D amplifier module used in active speakers. Based on DSP signal processing technology, network technology, class-D amplifier and switching power supply technology. Configured with 2-channel Dante input interface to easily integrate into Dante network audio system. 

* 2*500W4Ω+ 1*1000W8Ω

*DSP process module: DA24CL/D

*support 180-265VAC input

*IPS colorful display

*Connect by 2*RJ45 of network or USB free-driver

The SNA DANTE is a sound level limiter system for large music hall and nightclubs. it is the first sound level limiter on the market that integrates this technology.

The sound level pressure is measured by the acoustic sensor, then it is transmitted to the processor that integrates music peaks with a very gentle slope, and adjustable rise time and fall time, thus no limitation degrades the signal dynamics like a traditional compressor.

Selection of the A and/or C weighting for sound analysis.

An Embedded Web Server allows consulting the sound level history and equipment setting. The memory is dimensioned to store the values of acoustic pressure 24 hours a day and 7 days a week, for 6 months.

USB Host for connecting a USB stick for collecting history.

8 audio channels could be processed simultaneously

Acoustic and electric Auto-verification of the measuring sensor.

Real time display Leq1s and Leq15min.

Rack mountable 16-port 40w PoE audio quality switch designed for use in Dante Networks. In addition to the 16 Ethernet ports, the STNet-Switch II includes two SFP ports that can be adapted to fiber or high speed Ethernet connections. The two SFP ports can be turned into a high speed optical fiber ports or copper Ethernet ports with the addition of a Gigabit SFP transceiver module.

Features:

• 10M/100M/1G RJ45 PORT: QTY 16.
• 100M/1G SFP PORT: QTY 2
• PROPRIETY (PoE+ 40W)
• STORAGE TEMPERATURE: -20C to 70C
• IEEE802.af (PoE 15.4W)
• IEEE802.3at (PoE+ 30W)
• AVAILABLE PoE POWER: 640W
• OPERATING TEMPERATURE: 0C to 40C
• OPERATING HUMIDITY: 10 to 90% RH
• STORAGE HUMIDITY: 10 to 90% RH
• AC INPUT: 100V-240V

The Dlogixs DNS-7216 is a network speaker selector.

DLM-808 is an audio matrix processor, with 8 input and 8 output channels, with high-definition LCD to display current status at real time, with network port to expand network devices. It is available for large-scale place, such as theater, broadcast hall, gymnasium and conference center and so on. Its function can be summary as below:

1. All input channels are equip with GATE/EXP/CROSSOVER/PEQ/DELAY/COMPRESSOR function .
2. All input channels are equip with Gain/Crossover/PHASE/PEQ/DELAY/COMPRESSOR function
3. 8*8 audio matrix
4. With 8 channels in and 8 channels out audio interface expansion card (DANTE).
5. Match with PC and App operation software, which is convenient for user.
6. User can on-line update DSP and MCU Firmware via internet.

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.

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. 

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.

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.

The HDLM 8 Loudspeaker Management Controller is capable of managing more complex sound reinforcement systems. The HDLM 8 Loudspeaker Management Controller is capable of managing more complex sound reinforcement systems. A web browser based user interface with multi-client capability allows real time access from a PC or a Mac to user-critical parameters for multiple users simultaneously. Excellent dynamics, a low distortion factor and filters with outstanding sound properties ensure first class audio quality. This device can be ordered with an integrated Dante Brooklyn Card. 

The UNiKA DSP-428 Loudspeaker Management Processor utilizes state of the art 64-bit digital processing technologies with 24-bit delta-sigma data converters in 128x oversampling. The DSP-428 is also a versatile DSP Audio Processor with 4 analog inputs, 1 AES/EBU input (2-channel), 1 DANTE audio network input, 8 analog outputs, Its processing capability includes Gain, Parametric EQ, Shelving Filters, Time Delay, Crossover Functions, Compression, Limiting, and Matrix Routing. All analog inputs and outputs are precisely balanced with XLR connectors. With 64-bit digital processing and the high quality converters available on the market today, the DSP-428 offers sound quality which can compete with whatever your reference unit of choice is.

The NT4-DSP offers a 4in 4out topology. With 4 x Analog In, 4 x AESin and 4 x Dante in, FIR Filter and all available latest DSP technology, It provides a networkable, highly sophisticated DSP solution. This ensures worldwide optimal performance with every Loudspeaker System. With four fully balanced inputs, and four balanced outputs, the NT4-DSP offers increased flexibility. It is designed to operate at a native sampling rate of 96 kHz with 24-bit converters on the inputs and outputs, to give a bandwidth of over 30kHz and a dynamic range in excess of 115 dB. Available on each input channel is a 10 band full parametric EQ, every output has 12 fullparametric EQ´s, Peak & RMS Limiter + 1000 FIR Taps. The Inputs can be used analog, via AES/EBU or Dante Audio.

• 4in 4out DSP with matrix routing
• Analog, Digital Inputs
• Dante™ Audio
• Presets for all AD-Systems products on board
• 3rd party presets could be edited
• Peak + RMS Limiting
• FIR Filter
• Future-upgradeable

Due to the Dante™ interface, it is possible to conveniently integrate the speaker management system DSM-48DT as a 2-channel receiver into a Dante™ audio network. Both of the Dante™ signals are available for the input channels IN3 and IN4, alternatively to the analogue inputs and the digital input. Due to the Dante™ interface, it is possible to conveniently integrate the speaker management system DSM-48DT as a 2-channel receiver into a Dante™ audio network. Both of the Dante™ signals are available for the input channels IN3 and IN4, alternatively to the analogue inputs and the digital input. Thus, the DSM-48DT allows you to receive one stereo audio signal or two mono audio signals from other Dante™-enabled devices which are located in the same subnetwork and to process this/these in the same way as an analogue signal or AES/EBU signal. Speaker management systems featuring analogue inputs as well as outputs and additionally provide a Dante™ interface are called Breakout Boxes. In order to use the DSM‑48DT as a receiver unit, it has to be connected to at least one Fast Ethernet switch (100 Mbps Ethernet).