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Multifunction with automatic mixer


Our digital audio matrices with DSP (Digital Signal Processing) processors reach up to 12 independent inputs and 8 independent outputs.

Professional digital processors have multiple processing sections from automatic mixers, matrix, equalizers, filters, auto gain, antilarsen, compressors, delay lines, RS 485 presets, etc. etc. Among all these sections, the flagship of our audio matrices is certainly the automixer mixer segment, it has one of the most advanced algorithms, capable of operating the section of automatic mixer or auto mixer, in short, automatic mixing without operator perfectly, even in highly reverberant environments, with panoramic microphones.

Automixer section and its algorithm

The automatic mixer is equipped with a powerful algorithm, which reasons based on multiple parameters, based on multiple active thresholds and is able to precisely manage the opening and closing of microphones or sources based on the parameters that are set during the setting. The more precise the setting parameters are, the more punctual the automatic mixer will be to intervene automatically. The sound will also be totally transparent to the hearing in reverberating rooms even in the presence of high reverberation or in cases where we have background noise. The intelligent management of the automatic closing or opening of the microphones allows to obtain an output gain that in some installations can reach up to 8dB compared to a traditional mixer. This highly professional technology is now indispensable in all structures where multiple microphones must coexist in a single environment, both reverberant and with background noise.


Matrix and Mixer Section

What with traditional audio systems seemed impossible with our professional audio matrices is a breeze. It allows to address each single input on each of the 8 outputs any combination independently. Group management is possible, with the possibility of adjusting the volume directly on the cross of the matrix in order to obtain homogeneous group of inputs with processing modules of groups before distributing them to the matrix outputs.


Parametric equalizers and filters section

Each micro line input is equipped with a three-band parametric equalizer with two additional high / low pass filters if necessary. at each line output there is a six-band parametric equalization module with two additional high / low pass filters if necessary. The management of the equalizers is easy and intuitive thanks also to the graphic mode of the control software.

Antilarsen section or feedback suppressor

Within this section we find two possible antilarsen systems, in reality they are two algorithms for controlling feedback. Both sophisticated, expressly designed to reduce the acoustic feedback between microphones and speakers to avoid the so-called Larsen effect. Unlike other feedbacks, these algorithms are specific for speech. The work process as the effectiveness changes considerably compared to those who work with notch filters, we explain why: notch filter systems work when a certain frequency reaches a predetermined level which is a larsen and at that moment intervenes (at that point the system has already whistled and often in reverberant environments another frequency starts). Our antilarsen systems are phase shift or frequency shift, unlike the former they are always active, in operation we have a constant gain that can be obtained thanks to these feedbacks which will depend on the environment and the system in which it is used , which will vary from 3 to 6dB. The algorithm leaves the voice signal unchanged without introducing any distortion or limitation of the frequency response. These particular types of antilarsen are the most effective in churches, conference rooms or councils, they are the most suitable when most of the use of the professional audio system is speech.


Delay Lines Section

On each output there is a sound delay delay which can be activated with settings expressed in meters. This feature allows you to temporally "align" the speakers or speaker lines placed with respect to the main or reference front line. Thanks to this technology, in most part of professional audio systems we improve speech intelligibility and cancel the unwanted comb effect.




Signal Compressor and Noise Gate Section

In the division of the input signals where you can decide practically everything for input through software: micro or line, mute, phase, phantom, automixer, gain, control, etc. We also find audio signal compressors available, which can be activated with independent configurable parameters. It is possible to monitor the compression level in real time in order to better adjust the parameters that determine its operation; while only in the outputs we find the noise gate section, useful for automatically closing the desired output, when we decide, through the control parameters expressed in dB, that no signal must pass through that output if it does not reach at least a set parameter.

Connectivity and User Interface Section

All professional audio matrices must have remote management and control of all system parameters. The supplied easy-to-use control software is available for PC-Windows. Connected to the audio matrix, it is possible to manage all the parameters during the calibration phase. Through other ports, presets (scenarios) configurable per user can be recalled. Provided we have a convenient preset rack recall for a maximum of 6 configurations, but if we enable the RS485 port we can interface with the professional world through the appropriate configuration strings, then we can define almost every parameter. As an option we offer an interface panel Color LCD touch-screen display, MCONTROL configurable with all customer requests.

These professional audio processors are suitable for various installations, thanks to their flexibility, the designers of professional audio systems insert them in: churches, conference rooms, multimedia rooms, multifunctional buildings, places of worship and theaters, bingo, multi-zone commercial installations, hotels. They remain particularly suitable for very reverberant environments and in the presence of high ambient noise.