DF16.16MNE Series Network Digital Matrix Process

● 32 bit SHARC DSP, 96kHz sampling rate, 24 bit AD/DA convert.
● 48x48 channels core matrix design, including 16 local analog input channels, 16 local analog output channels, 32 Dante network input channels and 32 Dante network output channels.
● Fully supports scalable DANTE digital network audio card and AEC automatic echo noise cancellation card, the users can freely configure and choose according to their requirements.
● Dante digital network audio card configuration is a double standard network interface, which fully supports the function of unite switching and hot backup, hardware system can distinguish automatically, can use immediately when it inserts. Software internal and local analog signal can realize fully sound mixing matrix processing. Each channel has independent DSP processing function.
● Optional AEC automatic echo cancellation card has powerful automatic echo cancellation function, automatic feedback cancellation and system environmental noise filtering. Effectively reducing the pickup of the speakers and the terrible echo which caused by the reflection of wall. It can also eliminate environmental noise that generated by the feedback of pickup, moreover the noise of air conditioner and fan. Making sure any environment have a noise-free, high definition audio and video system for many conference applications......

Model : DF16.16MNE Series
Name : Network Digital Matrix Process
Price : Welcome to negotiate
Size : L*W*H 482*254*89MM








 




 

DF16.16MNE    Network Digital Matrix Process

● 32 bit SHARC DSP, 96kHz sampling rate, 24 bit AD/DA convert.
● 48x48 channels core matrix design, including 16 local analog input channels, 16 local analog output channels, 32 Dante network input channels and 32 Dante network output channels.
● Fully supports scalable DANTE digital network audio card and AEC automatic echo noise cancellation card, the users can freely configure and choose according to their requirements.
● Dante digital network audio card configuration is a double standard network interface, which fully supports the function of unite switching and hot backup, hardware system can distinguish automatically, can use immediately when it inserts. Software internal and local analog signal can realize fully sound mixing matrix processing. Each channel has independent DSP processing function.
● Optional AEC automatic echo cancellation card has powerful automatic echo cancellation function, automatic feedback cancellation and system environmental noise filtering. Effectively reducing the pickup of the speakers and the terrible echo which caused by the reflection of wall. It can also eliminate environmental noise that generated by the feedback of pickup, moreover the noise of air conditioner and fan. Making sure any environment have a noise-free, high definition audio and video system for many conference applications.
● Each analog input channels with +48V phantom power, MIC/LINK input gain switchable, MIC input sensitivity adjustable.
● Input including low cut, independent feedback inhibition, PEQ, noise gate, gain, mute, phase, linkage adjustment and volume adjustment.
● Output including X-over, PEQ, Gain, mute, compressor/limiter, phase, delay, linkage adjustment and volume adjustment.
● All PEQ Gain, bandwidth, frequency continues adjustment. Type can be select by PEAK, L-SHELF, H-SHELF, LOW CUT,
HIGH CUT, ALLPASS1, ALLPASS2.
● All high cut, low cut type can be select by Butterworth, Link witz-Riley, Bessel, slope can be chosen.
● Noise gate’s threshold, time, ratio can be adjustment for inputs, compressor, limiter, ratio, time can be adjustment for outputs.
● Maximal delay time 680mS for all output channels.
● Pre settings can copy for every single channel, every channel can do link adjust.
● Inside single generator(pink, white noise and 20-20K sine wave, amplitude adjustable).
● Front panel has level indicator for input/output, USB port, standard Ethernet remote control RJ45 port and RS232&485 at rear panel. IP and ID address can set, network and unite management can realize maximal 255 units link control, but also has remote visit password protected function, make the system more stable and safe.
● Design has thorough control code, fully support the third-party centre control and management, including all volume control, preset scene recall, parameter inquire and return code, current level return display, every input and output channels control. Etc.
● 12 user presets, device and each preset can be save and recall alone. System can chose work at immediate storage status, adjusted parameter can store at preset immediately.
 
System specification Frequency Response 20Hz-20kHz,-0.3dB
Dynamic range 110dBu
T.H.D <0.006% at 1kHz(0dBu)
Crosstalk >70dBu,20Hz-20kHz
C.M.R.R >75dBu 1KHz
MIC input section Type Balance
Phantom power +48V DC
Gain 50dBu
Impedance 2k ohm
Music input section Type Balance
Gain 35dBu
Max input level +18dBu
Impedance >10k ohm
Output section Type Balance
Max output level +18dBu
Impedance <500Ω
Digital Processing 24 bit sigma-delta A/D、D/A convert
32 bit DSP,96kHz sampling rate
Power Supply AC 95V-250V  50/60Hz
Dimension(L*W*H) 482*254*89MM