Dante-enabled Product Catalog

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

Dante数字音频传输技术，无压缩、无损耗、极低延时；以网络IP化架构设计，实现网络化、数字化，具有传输距离远、无干扰及低损耗等特点；集网络传输、纯数字功放的全频音箱。以太网供电和音频数据传输整合，实现线缆布局简单化，且符合以太网工程规范；具有高效率高声能输出、出色的音质、完善的保护功能和良好的电磁兼容特性；自带DSP网络有源音箱可在PC端远程对有源音箱的EQ、喇叭两端电压、电流进行监测控制；具有完善的保护功能（短路保护、过流保护、过欠保护、直流保护、功率压限、温度压限、长期输出功率限制、削波限幅器；免去了AP插座和适配器，使工作环境更安全整洁；设备移动方便，可移到任何有局限网线的地方可使用；以网络IP化架构设计具有高扩展性和音频组网能力。

The new ALM series transmits clear intelligible live announcements and precise messages. Thanks to the digitally controllable directional effect, the entire frequency spectrum can be heard in the desired sound coverage area. The combination of a long range and a wide horizontal opening angle has the advantage that just a few ALM loudspeakers cover a large area. Therefore, the ALM loudspeakers are particularly suitable for installations where a high level of speech intelligibility is required, for example in lecture halls, places of worship, airport terminals or shopping center. The ALM series is also equipped with a 24 volt power supply and is compatible with an EN 54 certifie emergency power supply system.

Configuring line array speakers has never been easier

Because of it´s modular structure of up to three lengths, the ALM series is easy to install. The automatic detection of your own IP address in the PAN C software accelerates the configuration process. The intuitive and uncomplicated PAN C software makes the configuration of line array speakers easier than ever. The software calculates the adjustable sound alignment as well as all internal parameters in real time.

Each ALM loudspeaker is designed as a real two-way system and has vertically arranged broadband loudspeakers for the low and mid-range as well as 4 tweeters for a silky, brilliant reproduction of the high-frequency range. Each individual loudspeaker is powered and controlled by a separate power amplifier and DSP channel. Using a complex computational algorithm in the DSP units, the vertical dispersion can be continuously controlled in the range between 5° – 40°. The desired range (focus distance) can also be adjusted continously.

Two analog channels, two DANTE® channels plus a 100 V input are available as inputs. The input DSP offers an adjustable limiter, a gate, delay function and a 10-band parametric Equalizer with IIR filters for each input. Due to the discrete visual design and the appealing housing the ALM series can be easily integrated into architecturally demanding environments. The variable sound dispersion makes difficult acoustic situations easy to handle and can be installed just with little effort.

The new ALM series transmits clear intelligible live announcements and precise messages. Thanks to the digitally controllable directional effect, the entire frequency spectrum can be heard in the desired sound coverage area. The combination of a long range and a wide horizontal opening angle has the advantage that just a few ALM loudspeakers cover a large area. Therefore, the ALM loudspeakers are particularly suitable for installations where a high level of speech intelligibility is required, for example in lecture halls, places of worship, airport terminals or shopping center. The ALM series is also equipped with a 24 volt power supply and is compatible with an EN 54 certifie emergency power supply system.

Configuring line array speakers has never been easier

Because of it´s modular structure of up to three lengths, the ALM series is easy to install. The automatic detection of your own IP address in the PAN C software accelerates the configuration process. The intuitive and uncomplicated PAN C software makes the configuration of line array speakers easier than ever. The software calculates the adjustable sound alignment as well as all internal parameters in real time.

Each ALM loudspeaker is designed as a real two-way system and has vertically arranged broadband loudspeakers for the low and mid-range as well as 4 tweeters for a silky, brilliant reproduction of the high-frequency range. Each individual loudspeaker is powered and controlled by a separate power amplifier and DSP channel. Using a complex computational algorithm in the DSP units, the vertical dispersion can be continuously controlled in the range between 5° – 40°. The desired range (focus distance) can also be adjusted continously.

Two analog channels, two DANTE® channels plus a 100 V input are available as inputs. The input DSP offers an adjustable limiter, a gate, delay function and a 10-band parametric Equalizer with IIR filters for each input. Due to the discrete visual design and the appealing housing the ALM series can be easily integrated into architecturally demanding environments. The variable sound dispersion makes difficult acoustic situations easy to handle and can be installed just with little effort.

The new ALM series transmits clear intelligible live announcements and precise messages. Thanks to the digitally controllable directional effect, the entire frequency spectrum can be heard in the desired sound coverage area. The combination of a long range and a wide horizontal opening angle has the advantage that just a few ALM loudspeakers cover a large area. Therefore, the ALM loudspeakers are particularly suitable for installations where a high level of speech intelligibility is required, for example in lecture halls, places of worship, airport terminals or shopping center. The ALM series is also equipped with a 24 volt power supply and is compatible with an EN 54 certifie emergency power supply system.

Configuring line array speakers has never been easier

Because of it´s modular structure of up to three lengths, the ALM series is easy to install. The automatic detection of your own IP address in the PAN C software accelerates the configuration process. The intuitive and uncomplicated PAN C software makes the configuration of line array speakers easier than ever. The software calculates the adjustable sound alignment as well as all internal parameters in real time.

Each ALM loudspeaker is designed as a real two-way system and has vertically arranged broadband loudspeakers for the low and mid-range as well as 4 tweeters for a silky, brilliant reproduction of the high-frequency range. Each individual loudspeaker is powered and controlled by a separate power amplifier and DSP channel. Using a complex computational algorithm in the DSP units, the vertical dispersion can be continuously controlled in the range between 5° – 40°. The desired range (focus distance) can also be adjusted continously.

Two analog channels, two DANTE® channels plus a 100 V input are available as inputs. The input DSP offers an adjustable limiter, a gate, delay function and a 10-band parametric Equalizer with IIR filters for each input. Due to the discrete visual design and the appealing housing the ALM series can be easily integrated into architecturally demanding environments. The variable sound dispersion makes difficult acoustic situations easy to handle and can be installed just with little effort.

The Aimline PoE+ series includes column loudspeaker with an integrated power amplifier that uses Power over Ethernet (PoE+). There is no need for any additional power supply. Thanks to highly efficient power electronics and a specially developed power amplifier an output of 30 W / 60 W peak (when using a PoE+ supply) can be achieved. Two channels DANTE® are available with 10-Bd IIR filters, delay and the 24 internal presets offer all necessary configurations. It is controlled and monitored using the IP-based PAN C software.

All active AIMLINE digital systems can therefore be controlled via just one software interface.

Easy connection – just one cable

This system will solve the issue of having to provide a separate amplifier or I/O interfaces for speaker installations using just a single CAT cable over long distance. This assembly can be used in wide range of projects for paging and background music applications. The AIMLINE AL 4.1 PoE+ is ideal for hotels, education, huddle rooms, hospitals, retail stores, restaurants, airports, houses of worship, and boardrooms.

The 4-channel loudspeaker controller module with digital signal processing for amplifiers of the KS AUDIO series

CA 4D and TA 4D. A unique convincing solution for perfect sound and easiest handling. The KS D MOD controller module complements the F MOD modules for applications where no quick change of the loudspeaker systems is required - i.e. especially in the installation area.

The one-time configuration of the loudspeaker system is done on the module by a USB stick transmitting the data while the system is switched on. During that process, both the signal inputs and the power outputs automatically receive the correct function:

from four individual channels

via two channels in Bi-Amp

or high-power bridge mode

to a single-channel 3-way active loudspeaker system with woofer-bridge mode output, and optionally “BI-Amp” control.

a SAT mode is also available in which the L+R signal is summed up to mono sub-low.

In addition to digital signal processing with FIR filter tech- nology, the special feature of the D MOD is its complete operability and controllability via standard networks. An integrated gigabit switch allows daisy-chaining to the next device. The KS REMOTE PC software enables easy ma- nagement and graphically, easy to understand, operation of an unlimited number of controller amplifiers arranged in groups.

All inputs can be controlled either digitally or analogously. Simultaneous digital and analog operation is also possible. A network-audio transmission using DANTETM protocol is available as an option.

Despite the many new possibilities and state-of-the-art technology, the D MOD can replace an existing analog or digital controller by simple front-panel installation. This puts your public address system back on the cutting edge of technology.

The CA 4D is a 4 channel MOSFET amplifier with controller card system configuration technology, using digital controller cards D MOD or F MOD. Both exclusively supporting the KS AUDIO FIRTEC™ finite impulse response technology. When changing controller cards or when loading a new program from the internal memory both the inputs and the outputs are appropriately configured to reflect the correct set up.

Whether or not the system is configured as 4 single channels – 2 channel / 2 way – 2 channel with mono Sub woofer – 2 channel mono bridged high power sub woofer – 1 channel 3 way - L and R SAT configuration, nearly every combination of KS AUDIO speaker systems is possible. This allows for a wide field of application, from small and medium size concert sound reinforcement to speech reproducing at the highest quality.

The CA 4D is equipped with 4 independent analog FET power amp modules. Two of them deliver a continuous sinewave power of 700 watts into 4 ohms each. While the other two modules deliver half that power. The controller card and an applicable program determines the distribution of power to always deliver the maximum available. The high performance of the CA 4D amplifier is achieved by three balanced supply voltages. The power losses are substantially less than with any other conventional analog power-amplifier designs. A total of 40 quality selected power MOSFET transistors, deliver a maximum headroom. Soft clipping circuits working in conjunction with

the operating voltage, limit the NF signals and therefore enable the power amps to always work with a high damping factor as well as remaining fully regulated.

The power supply is a state-of-the-art, heavy duty wide-range switch mode design, with Power-Factor-Correction (PFC) built in. The start-up current is limited to assure maximum protection. A second power supply is used to independently supply the controller card electronics along with the low voltage cooling fans and to operate the stand-by switch. Excess temperature is dissipated by 2 fans. The air is taken in from the front panel inlet thru an easy to clean filter. It is passed through to the back panel outlet, while passing over the heat fins and the power transformer to cool these units down to nominal operating temperatures.

On the front panel there are 4 mute switches corresponding to the 4 channels of the amplifier. Each of these switches has a red LED below it to indicate the mute status. On the rear panel there are 4 Neutrik speakON output connectors. Their status is indicated by 4 green LEDs. The mains cable is connected via a PowerCon.

The housing is constructed of 1.5mm thick steel while the 19′′ front plate is constructed of 4mm thick anodized aluminum and has 2 handles for ease of use.

Linea’s 88C Series of eight channel amplifiers are optimised for use in installations. Combining massive power and peerless audio performance with unique DSP, the 88C Series also incorporates many features that ease it’s integration on to both new and legacy systems, the 88C Series represents an unmatched advancement in amplifier technology.

Sharing the same feature set and form factor as Linea's 44C Series of four channel amplifiers the 88C Series is available in models that develop between 400Watts and 2,500 Watts per channel (up to 5,000 Watts for a bridged pair). Unusually these are not, ‘transient’, ‘burst’ or ‘peak’ power figures, they are the power levels that are delivered by all channels simultaneously even when driven with the same dense program material.

With Linea amplifiers there is no need to ration power between channels, you can use all the power you have paid for all the time. This has been independently verified.

Linea can do this because state of the art components and a finely optimised design results in an amplifier that is between 7% and 10% more efficient than any competitive product. This efficiency gain directly translates in to the generation of about half the wasted energy (heat) which is clearly significant. In addition to reducing energy costs high efficiency results in generous power reserves ensuring that pristine sound quality is maintained even under the most extreme conditions.

Ethernet is used for remote control and monitoring while the tamper proof front panel allows easy verification of the operating status and signal levels. Control via Linea’s intuitive Podware PC application provides many convenient and time saving features such as very flexible management of presets, and Overlay grouping of Mutes, Gains, Delays and EQ across an entire system. Control from 3rd party systems is also possible via TCP/IP, RS232 and RS485 while comprehensive relay and contact closure ports ensure that basic control can be achieved very easily.

The advanced 96kHz DSP is fully integrated into the product and provides an array of unique and genuinely useful features. Because all aspects of the DSP algorithms are designed and hand crafted in house by Linea, they are not available on products from any other manufacturer. For example, LIR linear phase crossover filters which are as easy to use as more conventional filter shapes, and have less latency than FIR filters. Also the powerful VX Limiter suite which sports Peak, RMS and Excursion limiting, as well as multiband limiting for passive 2-way enclosures.

MondeF 2-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 2 Dante® outputs, 8 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 2-zone network audio server (Dante®)

2 Dante® outputs

8 Dante® inputs

Completely pre-configured, including installed software and drivers

Miniature server, slimline design, flat plastic housing

Operating system: Windows 10 Professional, 64-bit

Processor: Intel Celeron 3865U, 1.8 GHz, Intel HD Graphics 610, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted either as a wall/tabletop unit or directly on the rear of the monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 32-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 32 Dante® outputs, 16 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 32-zone network audio server (Dante®)

32 Dante® outputs

16 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 4-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 4 Dante® outputs, 8 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 4-zone network audio server (Dante®)

4 Dante® outputs

8 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 64-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 64 Dante® outputs, 32 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 64-zone network audio server (Dante®)

64 Dante® outputs

32 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 8-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 8 Dante® outputs, 8 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 8-zone network audio server (Dante®)

8 Dante® outputs

8 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

MondeF 16-zone network audio server (Dante®) for professionally streaming Dante® audio data. Equipped with 16 Dante® outputs, 16 Dante® inputs. The server is completely pre-configured with MondeF software, drivers and the Audinate Dante® software for use with Dante® audio products.

The installed software includes Internet radio (120 channels max. simultaneously), announcement feature, (programmable) alarms/announcements, scheduling, playlists etc.

Furthermore, the server provides a variety of connection facilities, such as USB (4 x), RJ45 network connection, HDMI™ output and DisplayPort for connecting up to 2 optional monitors or touch screens, thus allowing for a convenient operation of all network components.

MondeF 16-zone network audio server (Dante®)

16 Dante® outputs

16 Dante® inputs

Completely pre-configured, including installed software and drivers

Mini server, slim design, slimline plastic housing

Operating system: Windows 10 Professional, 64-bit version

Intel Celeron 3865U processor, 1.8 GHz, Intel HD610 Graphics, supports 2160p/60 Hz

Wireless LAN 802.11n, internal antenna

Can be mounted as a wall device/desktop device or directly to the rear side of a monitor/touch screen (VESA mount)

Supplied with power supply, supporting feet and VESA mount (75/100 mm)

The brand Dante® as well as the Dante® software which may have been integrated into these products are licenced by Audinate Pty Ltd.

A 1U 19inch rack mounted AoIP monitoring and mixing unit.

4 models are available the RL64, RL128, RL192 and RL256 and each unit can be configured at time of purchase to be Dante or Ravenna

RL Series units allows two modes of operation a mixing mode which can mix up to 256 channels at 48kHz (128 channels at 96kHz) plus microphone input (via headphone connector) and AES input to create a 259 into 2 mixer. Alternatively, it can be set into a simple Listen mode where touching a channel instantly routes to the LR output and cancels any previous selection.

The outputs of the LR bus are presented to the user via the inbuilt 4mm x 9mm full range speakers or via the 3.5mm headphone jack. The speakers are automatically muted when a headphone jack is inserted. Simultaneous outputs are also available in the form of the stereo balanced line outputs and AES output accessed via XLR connectors on the rear of the RL Series Unit. The mix output is also routed to the AoIP output.

The RL Series units are operated and controlled via the large letterbox 6.6” diagonal (168mm) 1440 x 240 24-bit colour LCD and capacitive touch panel.

A simple and elegant menu system has been designed to be intuitive to navigate allowing the user to access all the features of the RL Series unit very quickly.

The high-resolution screen is easy to read presenting comprehensive metering, routing and status information to the user.

Firmware updates of the RL Series unit are achieved via a dedicated USB micro AB connector on the front panel. There is also a dedicated 3.5mm jack socket for the headphone output

Stacks Image 30

On the rear of the RL Series unit there are 4x Primary and Secondary ports for up to 4x AoIP inputs. 2x balanced line outputs on 3 pin male XLRs and AES output also on 3 pin male XLR. These outputs can be configured to follow the main volume settings of the speakers/headphones or they can also be set to follow the main Mute control.

Additionally, there is an AES input via 3 pin female XLR connector

The RL Series unit can be powered via either of the two 2.5mm DC input connectors requiring 24V @ 2A. These inputs also have a screw lock to prevent accidental unplugging. Using both inputs provides redundancy.

A 1U 19inch rack mounted AoIP monitoring and mixing unit.

4 models are available the RL64, RL128, RL192 and RL256 and each unit can be configured at time of purchase to be Dante or Ravenna

RL Series units allows two modes of operation a mixing mode which can mix up to 256 channels at 48kHz (128 channels at 96kHz) plus microphone input (via headphone connector) and AES input to create a 259 into 2 mixer. Alternatively, it can be set into a simple Listen mode where touching a channel instantly routes to the LR output and cancels any previous selection.

The outputs of the LR bus are presented to the user via the inbuilt 4mm x 9mm full range speakers or via the 3.5mm headphone jack. The speakers are automatically muted when a headphone jack is inserted. Simultaneous outputs are also available in the form of the stereo balanced line outputs and AES output accessed via XLR connectors on the rear of the RL Series Unit. The mix output is also routed to the AoIP output.

The RL Series units are operated and controlled via the large letterbox 6.6” diagonal (168mm) 1440 x 240 24-bit colour LCD and capacitive touch panel.

A simple and elegant menu system has been designed to be intuitive to navigate allowing the user to access all the features of the RL Series unit very quickly.

The high-resolution screen is easy to read presenting comprehensive metering, routing and status information to the user.

Firmware updates of the RL Series unit are achieved via a dedicated USB micro AB connector on the front panel. There is also a dedicated 3.5mm jack socket for the headphone output

Stacks Image 30

On the rear of the RL Series unit there are 4x Primary and Secondary ports for up to 4x AoIP inputs. 2x balanced line outputs on 3 pin male XLRs and AES output also on 3 pin male XLR. These outputs can be configured to follow the main volume settings of the speakers/headphones or they can also be set to follow the main Mute control.

Additionally, there is an AES input via 3 pin female XLR connector

The RL Series unit can be powered via either of the two 2.5mm DC input connectors requiring 24V @ 2A. These inputs also have a screw lock to prevent accidental unplugging. Using both inputs provides redundancy.

PENTA 721s-DNT is a Dante to Dante converter with built-in Samplerate Converter, MADI and AES, which provides a versatile and flexible solution for signal distribution of Dante audio between two different networks as well as routing via AES/EBU and MADI and optical fibre networks.

It is a compact 1U digital audio interface with a dual set of Dante IP Audio network ports, Pro|Mon and dual power supply, providing routing and selection of sample rate between two separated Dante networks and individual domains. One of the domains has a capacity of 128 Dante channels and 128 audio flows, and the other domain has a capacity of 64 Dante channels and 32 audio flows.

All Dante channels can be routed to and from any of the digital interfaces of the unit. It comes with eight channel AES/EBU input/output channels, MADI and two IP Audio Ethernet in/outs, as well as an optional mini-module slot with dual SFP MADI optical in/out connection.

PENTA 721s-SDI is a Dual SDI embedder/de-embedder with onboard Samplerate Converter, MADI, AES and Dante, which provides a versatile and flexible solution for signal distribution of embedded audio via AES/EBU, MADI, as well as routing via IP Gigabit Ethernet and optical fibre networks.

Penta 721s-SDI is a compact 1U digital audio interface with a dual set of 3G SDI input/output ports, Pro|Mon and dual power supply. It provides embedding and de-embedding of 16 audio channels per port where samplerate conversion can be enabled if desired.

All embedded channel can be routed to and from any of the digital interfaces on the unit. It comes with eight channel AES/EBU input/output channels, MADI and Dante IP Audio Ethernet in/outs, as well as an optional mini-module slot with dual SFP MADI optical in/out connection.

DX32R is your versatile digital audio Swiss knife - an interface and matrix designed for multiple audio applications such as multi-room recording studios, post-production facilities and audio distribution installations. It allows you to route all digital input and output and manage them on a mono channel basis within a single unit or a combination of units.

With near zero latency, the DX32R can be used to expand I/O channel count and interface digital components, providing a high quality bridge between AES/EBU, MADI and the Dante IP audio network.

DX32R include a 384x384 multiplexer, enabling all digital inputs and outputs to be routed in any combination; this enables the unit to also split signals for advanced signal distribution.

The DX32R IP Audio protocol is based on the robust tried-and-tested Dante™ digital audio network technology, and will interoperate with products of other brands that comply with Dante.

The protocol is compliant with the AES67 standard for AoIP. The Dante and NTP IP Audio formats provide fast, flexible, and economical audio routing via IP and are compatible with the AX32 AD/DA converter, and other Dante devices. The IP Audio routing provides low latency, tightly synchronized, transport of uncompressed audio over Gigabit IP Ethernet Layer 3 networks using off the shelf switches, and routers for audio routing via one or more sub-nets.

A total of 512 channels can be routed on a 1 Gigabit network, and more if the network capacity is higher.

DX32R is TCP/IP controlled via one of the two Ethernet ports vi the DADman Control software available for both Windows and OSX.

The routing of the IP audio via Dante is done by using the Audinate Dante controller software.