Xta808 is a high-performance 8-in-8-out digital media matrix, which can meet various vocal requirements, especially the application of conference system and other applications requiring multiple microphones and speakers. Xta808 includes 8 line / MIC inputs and 8 line output channels. Line / MIC input can be switched independently with phantom power supply. All inputs and outputs are managed by a powerful Marani ® exclusive DSP engine, plus a high-end 24 bit AD / DA converter. Xta808 supports full matrix mixing mode, and the input channel can be routed / mixed to any output channel in any proportion. Each mic / line input channel is equipped with a high / low-pass filter, 3-stage parametric equalizer, noise gate function and gain control. In addition, MIC input channel also has a feedback suppression function, which is based on a powerful "frequency shift" algorithm, which can be perfectly applied in the human voice environment. Each output is equipped with a parameter equalizer of up to 5 segments. The slope range of the frequency division filter is 6dB / oct-24db / Oct. in addition, RMS compressor, peak amplitude limiting and adjustable delay are included.
Xta808 can be connected to remote control PC software through Ethernet, USB or RS485 port. In addition, the device can be controlled by special software on the Apple iPad ®. For more complex control systems, 4 digital input and output ports integrate GPIO, allowing xta808 to connect to different devices. Through RS485 interface, xta808 can also be connected to CP4 wall controller to select preset and set main volume.
Specifications
● maximum input level: Line: + 18dbu; microphone: - 18dbu
● maximum output level: + 18dbu
Thd + n distortion: 0.005% @ 1kHz 0dBU
● s / N SNR: > 104dba
● frequency response: 20Hz - 20kHz + / - 1dbu
● AD & DA conversion: 24 bits
● DSP processing: Marani DSP exclusive algorithm, 24X32 bit filtering processing, 96 bit precision intermediate data operation results
● input Equalization: three equalization filters are provided for each input channel
● output Equalization: five equalization filters are provided for each output channel
● filter type: symmetrical bell or high / low pass Chevron to the second order
Filter gain: step: 0.5dBu, Baer or chevron gain range: ± 12dBu
● center frequency: select within ± 20Hz based on 1 / 24th octave resolution step
● filter Q &#118alue / Broadband: Q / BW step (q): 0.1 resolution Q / BW range: 0.4/3.59 ~ 10 / 0.0312
● high and low pass filtering: from the first order (Butterworth - 6dB / OCT) to the fourth order (Butterworth, linquet or Bessel - 24dB / OCT)
● threshold: 18dbu ~ - 12dB
● detection time range: 5ms ~ 200ms (1ms resolution ~ 20ms; 10ms resolution ~ 100ms 20ms ~ 200ms)
● release time range: 0.1s ~ 3S (0.1s resolution)
● scale: 1:1 (through) to 32:1 (hard limiter) calculated with 0.1 step accuracy log table
● inflection point: soft / hard
● delay: each output channel can increase / decrease the delay in 21 US steps up to 380.998 MS
● feedback activation: available only when microphone input is selected
● automatic mixing function: NOM attenuation, gain sharing algorithm and priority avoidance processing
● preset parameters: 6 user calls; 4 S1-S4 digital input interfaces
● size: 483x44x229mm 1RU
● net weight: 3.5kg