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  • Basic digital audio processor
Basic digital audio processor

The host adopts the world-famous media matrix audio processing algorithm. The host set has anti-interference, no howling, and other core algorithm technologies to support 24 hours of uninterrupted work.

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  • Features
  • Specifications
  • Factory&Case
1, ★ host uses the world-famous media matrix audio processing algorithm, the host set has anti-interference, no howling, and other core algorithm technology, support 24 hours of uninterrupted work
2, 8/12/16 balanced microphone / line input, using bare wire interface terminals, balanced connection
3, 8/12/16 balanced output, using bare wire interface terminals, balanced connection
4, 48kHz A / D, D / A conversion, up to 96kHz sampling rate
5, two high-speed DSP processing chip TI L138 3648MIPS/2746MFLOPS 64bit processing core
6, group control function
7, channel copy, paste, joint control function
8, adaptive POE external panel control interface
9. Input per channel: preamplifier, signal generator, expander, compressor, 5-band parametric equalization, automatic mixing console
10, output per channel: speaker manager (8-band parametric equalization, delay, frequency divider, high and low pass filter), limiter
11, ★ built-in dodge function (DUCK) (AFC) adaptive feedback cancellation function
12, built-in sound pressure level control function (SPL)
13, full-function matrix mixing function
14, M-LAN* multi-purpose data transmission and control port, support SNMP / SMTP remote management
15, GPIO programmable control interface
16, built-in automatic camera tracking function
17, RS-232 bidirectional serial control interface is used to control other external devices such as: video matrix and other RS-232 devices, or to receive third-party RS-232 control, such as AMX, Meloarte central control: RS-485 camera control port;
18, the use of the environment: for small and medium-sized professional communication conferences, training rooms, academic lecture halls, multi-purpose halls, banquet halls, star hotels, courts and other aspects of sound reinforcement needs 
Phantom power (per input)


+48 VDC, 10 mA

Input source input mode can switch balanced microphone or line, using Phoenix plug interface
Input quantization 48KHz/24bit
A/D dynamic range ≧110dB
Phantom Power DC 48V
Input common mode rejection @ 60Hz 105 dBu @ +20dBu , 60Hz
Input impedance 20k Ω balance, 10k Ω unbalance
Maximum input gain 54dB
Output quantization 48KHz/24bit
D/A dynamic range 112dB
Input to output dynamic range


Channel isolation 112dB
Frequency response 20 to 20 kHz (±0.5 dB)


Total harmonic distortion (THD+N)


≤0.005% @1kHz, +4dBu

Output impedance

 100 Ω

Maximum output 20dBu
Working power supply AC 110V~230V AC, 50Hz / 60Hz
Operating temperature 0 - 40 °C


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