Paulkitson DSP0408RTS Professional Stage Digital Audio Processor 96KHz 32Bit DSP 16Band EQ With AES FIR Speaker Audio Equipment

This device is a high-performance digital audio processor with a sampling rate of 96KHz. A 24-bit AD/DA analog-to-digital/digital-analog conversion chip and a high-speed processing DSP chip are used to collect and process audio. Support 2 analog/1 AES/EBU input, 4 analog/2 AES/EBU output. Each input and output channel supports 16-band parametric EQ, equipped with output FIR filter, high and low pass filter, signal generator, noise gate, gain control, input compression, output limiter, scene preset and up to 2000ms per channel It has powerful functions such as adjustable delay and is equipped with 100M Ethernet port, USB and RS232 control port, which can realize fast configuration and debugging. Carry out a series of optimization and adjustment for high-quality sound reinforcement system.

Features 1:

●●32-bit DSP chip processing, sampling rate 96KHz, 32bit ADC and 24bit DAC;
The input processing part includes: 2-way AES/EBU digital signal input, signal generator, noise gate, 16-segment equalization, high and low pass filter, all pass filter, gain control, mute, phase, 2000ms adjustable delay and Linkage adjustment and other processing functions;
The output processing part includes: 4-way AES/EBU digital signal output, 16-segment equalization, FIR filter, high and low pass filter, all-pass filter, gain control, mute, phase, 2000ms adjustable delay and linkage adjustment, etc. processing function;
●● All inputs and outputs can be freely allocated in matrix, and the name of each input and output channel can be changed;
●●All input and output channels have independent phase curve adjustment function;
●●All input and output channels have high and low pass filters, the type of frequency divider can be selected: Butterworth, Bessel, Linkwich-Rayleigh, and the slope can be selected from -6dB/Oct to -48dB/Oct;
Threshold, ratio, attack time, recovery time and compensation gain are adjustable for all input channel pressure limits;
●●All output channel limiters have threshold value and adjustable recovery time;
●●All input and output channels have adjustable delay time up to 2000ms;
Parameter settings between any channels can be freely copied, and any channel can be adjusted in linkage;
●●You can see the PEQ curve adjustment diagrams of all current input or output channels in any input or output channel;
●●Built-in test signal generator, pink noise, white noise, frequency sweep and 20Hz-20kHz sine wave can be selected as the input mode, and the signal amplitude can be adjusted;
The front panel is equipped with input and output level indicators, USB control port, dual network port design with cascading function, and the rear panel is equipped with RS232 control port. One-button connection makes the user's operation easier and faster;
●● Support 128 preset scenes, the status of the whole machine and each preset can be stored and recalled separately, and it also has a password protection function to make the device more secure;
●●Applications: professional performances, conference rooms, courtrooms, auditoriums, multi-function halls and other scenarios.

Features 2:

●Sampling rate 96KHz
●Analog input 4 XLR balanced
●Input impedance 10KΩ balanced/20KΩ unbalanced
●Analog output 8 XLR unbalanced
●Output impedance 50Ω balanced/100Ω unbalanced
●Frequency response 20Hz – 20KH±0.1dB
●The default output level is 0dBu
●AD & DA conversion 32Bit / 24Bit
●Maximum input level 22dBu
●Maximum output level 22dBu
●THD+N distortion ≤0.0025% @ 4dBu 20Hz-20KHz
●System delay ≤1.85ms
●Bottom noise ≤-91dBu
●Input to output dynamic range ≥113dBu
●S/N signal-to-noise ratio ≥111dBu ●DSP processing: TI’s OMAP-L138 high-performance DSP is used, with computing power up to 3648 MIPS and 2746 MFLOPS per second Built-in multiple audio algorithms, the highest sampling rate supports up to 96KHz
Parametric Equalizer Input and output channels support up to 16-band equalization.
6 Filter Types Peak, Notch, 2nd Order All Pass, 1st Order All Pass, High Shelf, Low Shelf
●Filter gain range -30dB ~+30dB, step accuracy: 0.1dB
●Input/output gain range -72dB ~ +12dB, step accuracy: 0.1dB
●The center frequency can be adjusted with a step accuracy of 1Hz within the frequency range of 20Hz~20kHz
●Filter Q value/broadband peak, notch, second-order all-pass, Q value: 0.01～5
●High and low pass filter Butterworth slope: 6/12/18/24/30/36/42/48dB/Oct
●Bessel slope: 6/12/18/24/30/36/42/48dB/Oct
Linkwich-Rayleigh: 12/24/36/48dB/Oct
●Input noise gate Threshold range: -100dBFS~0dBFS;
●Start control time: 1ms ~ 1000ms;
●Recovery time: 1ms ~ 1000ms
●Input compression threshold range: -84dBFS ~ 0dBFS
●Rate: 1 to 20
●Startup time: 1ms ~ 1000ms
●Recovery time: 1ms ~ 1000ms
●Compensation gain: -24dB ~ 30dB
●Output limit Threshold range: -84dBFS ~ 0dBFS
●Recovery time: 0ms ~ 100000ms
●Delay: Each input and output channel has an adjustable delay up to 2000ms
●High and low pass filter: Butterworth Slope: 6/12/18/24/30/36/42/48dB/Oct
Bessel slope Bessel slope: 6/12/18/24/30/36/42/48dB/Oct
Linkwich-Rayleigh: Linkwich-Rayleigh: 12/24/36/48dB/Oct
●Input noise gate: threshold range: -84dBFS ~ 0dBFS;
Start control time: 1ms ~ 1000ms;
Recovery time: 1ms ~ 1000ms
●Input compression: threshold range: -84dBu ~ 0dBu
Ratio: 1 ~ 20
Startup time: 1ms ~ 1000ms
Recovery time: 1ms ~ 1000ms
Compensation gain: -24 ~ 30
●Output limiter: Threshold range: -84dBu ~ 0dBu
Recovery time: 100ms ~ 1000ms;
Delay: Each input and output channel has an adjustable delay of up to 2000 ms