PRO1608 network digital audio matrix
The PRO 1608 is a powerful digital audio processor and touch screen operation is used to control interface display. ADI SHARC 4th generation of ADSP-21489 floating-point audio DSP chip provides the supreme performance 400 MHz/2200 MFLOP processor capacity. The ultra-low ground noise pre-amplification circuit, low-distortion analog circuit and 114dB audio AD and DA provide high-quality sound in field. The front panel of the device displays volume and volumes of all input 16 and output 8 channels; the LCD of front panel of the device displays the current device IP addresses and the presetting numbers and presetting names used the processor for the administrator’s quick identification and identifying the current processors in the system. The front panel is composed of network connection status indicator and error alarm indicator. Where gross error occurs, red alarm indicator twinkles.
Powerful DSP processor capacity
Based on powerful ADSP-21489 and powerful DSP processing capacity and our unique core DSP algorithm, the built-in standalone 8-channel ultra-low-distortion self-adaptable feedback inhibitor, 16-channel noise gateway, 16-channel input compression limiter, 16-channel 16-band PEQ, 16-channel input 48dB slope high-pass-low-pass filtering, 8-channel 1-second time delayer mean that you can set each input audio channel precisely and exquisitely. The 16×08 full matrix mix function mixes any input channels as you like. The 8 output channels are configured with standalone 8-band PEQ, compression limiters, 48dB slope high-pass-low-pass and 2-second time delayer.
Built-in sinewave, pink noise and white noise signal generators store 32 presetting values.
Rich audio channels and control interface
1 TCP/IP telecommunication port, 1 RS-232 telecommunication port, 1 USB telecommunication port. A third-party telecommunication protocol is opened to meet large, medium and small-sized professional audio projects. Applications of public amplification systems for theatre, music hall, remote video conference, stadium, church, conferencing center and theme park are satisfied.
LCD status display:
Front 1602LCD displays IP addresses, current presetting names and duration etc.
Easy-to-use control software
The software interface is developed through senior sound engineers and professional tuner’s in-depth communication and commissioning and the operators’ practice. The control software is popular and easily understand and can be operated without reading the operating manual. Each input numeral is directly entered by the keyboard. To get the precise value, such as -12.2dB, directly enter -12.2. Operate the volume bar and press Shift+ selected key and press the up and down keys to attain 1dB step forwards and backwards. As for PEQ, Limiter and such complicated parameter adjustment, these parameters can be quickly copied and pasted. You can finish multi-channel data copying easily and conveniently.
Open RS-232 and TCP/IP telecommunication protocols
A third-party device is used to control volume, call modes and set mute and TCP/IP protocol is used to read level meters in batch before and after mixing for being integrated with a third-party software.
High-performance floating-point DSP processing chip;
16-channel balanced input audio;
16-channel supports MIC input, each of which supports 48V phantom power supply;
8-channel balanced input audio;
8 channel standalone self-adaptable feedback inhibitor
8 channel automatic mixing
ADC CS5368 114dB dynamic, AC CS4385 114dB dynamic
Input per channel: pre-amplification, noise gateway, compressor, 16-band parameter equilibrium, delayer and automatic mixing console
Output per channel: 8-band parameter equilibrium, frequency divider, high- and low-pass filtering, compression limiter and time delayer
Built-in signal generator: sinewave signal, pink noise and white noise
Front panel 1602 display indicates IP address and current presetting values
RS-232 and TCP/IP protocols are opened for a third-party control
Video recording tracking code output is used for video linking function via a third-party centralized control
32 scenes presetting function is supported and called via TCP/IP and RS-232 protocols.
Connected to Android system and smart phone or tablet computer operation and control software package is supported.
Specifications and parameters
Audio input: 16 channels of balanced input, Phoenix plug
Nominal input level: +4 dBu line or -40 dBu microphone level
Microphone pre-amplification gain: 0-40dB analog gain, 20dB digital gain
Maximum input level: +20dBu
Input impedance: >5 kΩ balanced, >3kΩ unbalanced
Common mode rejection ratio: >70dB @ 1 kHz
Pre-amplification equivalent input noise (EI N): <-125 dBu, 2 2Hz - 22kHz
Phantom power supply: +48 V DC
Audio output: 8 channels of balanced line level, Phoenix plug
Nominal output level: 0 dBu line level
Output impedance: 600 Ω balanced
DSP frequency, processing capacity: 400Mhz, 400MIPS,2200 MFLOP
Sampling frequency/quantification: 48 kHz，24Bi t ADC，24Bi t DAC
Frequency response: 20 Hz – 20 kHz, + /- 0. 5 dB
Dynamic range: 114 dB，ADC, DAC
THD: <0. 01%；20Hz～20kHz@+4dBu
Inter-channel crosstalk: >-80 dB @ 1 kHz, typical
Equipment dimensions WxDxH:1U, 482 mm x 464mm x 44mm
Power supply: 100～240VA, 50/60 Hz, 75W