The functional characteristics of the SYSTEM the FEATURES 

S fully automatic detection field noise point functions, through the DSP system for filtering, low voice 

Sound compensation automatic mixing, intelligent high speed feedback processing; Back to the fully digital maximum eliminated 

To lose. Automatically adapt to the acoustic environment, without debugging, fast calibration function, ensure the quality, 

To reduce latency. Provide three file input level selection, prevent the peak input distortion. At the same time 

When using more than ten microphone, can effectively increase the microphone pickup from 30-100 cm. is 

Often use the machine without debugging, automatically ADAPTS to the acoustic environment (about utomatically adapt to the process 

Need to 10 seconds), the whole system will be an additional 6 ~ 12 db gain. If use conditions 

(the system) after repeatedly appeared in the process of debugging the noise, abnormal sound quality please send appropriate volume 

Reduce, then native fast calibration button, at this time will be randomly generated 1 seconds - 6 db 

Signal sound, sound quality back to normal. 

Technical parameters of the SPECIFICATIONS 

Rated voltage: 220 v + 10% for 50 hz 

Professional sampling rate: 32 KHZ 

Frequency response: 125 hz to 15 KHZ (sound mode) 

20 hz to 15 KHZ (music) 

Distortion: < 0.1% @ 1 KHZ 

Signal to noise ratio: > 90 db 

Signal delay: 7 m/s (music) 11 m/s (voice) 

Input impedance: Ω 20 k 

Temperature range: - 10 ~ 55 ℃ 

Weight: 2.5 Kg 

Size: 480 * 220 * 44 mm 

Don't need to debug automatically adapt to the acoustic environment 

Maximum 6 db, 12 db can be used in different environment is very good