Nuoxun – LM848RTS (96K+FIR+AES/EBU In/Output)

Product Description

PRODUCT FEATURES

● 32-bit DSP processing, 96KHz sampling rate, 32 bit AD and 24 bit DA converter;

● The input processing includes: signal generator, noise gate, 16-band EQ, HPF, LPF, all-pass filter, gain control, mute, phase, 2000ms adjustable delay, linkage adjustment and other processing functions;

● The output processing includes:16-band EQ, HPF, LPF, FIR filter, all-pass filter gain control, mute, phase, 2000ms adjustable delay, linkage adjustment and other processing functions;

● All inputs and outputs channels can be freely routing, 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 HPF and LPF, can be selected: Butterworth, Bessel, LinkwitzRiley, the slope is optional from -6dB/Oct to -48dB/Oct;

● All input compressor have thresholds, ratios, start-up time, recovery time and compensation gain adjustable;

● All output limiter have thresholds for limiting amplitude, and the recovery time is adjustable;

● The delay of all input and output channels with 2000ms delay;

● Parameter settings between any channels can be freely copied, and any channel can be linked adjustment.

● You can see the PEQ curve adjustment chart of all the current input or output channels on any input or output channel.

● Built-in signal generator, input mode can choose pink noise, white noise, sweep frequency and 20Hz-20KHz sine wave adjustable, signal amplitude adjustable.

● The front panel has input and output level indicator lights, the rear panel has 232 control ports. One-button connection makes the user’s operation easier and faster;

● 128 preset scenes, the state of the whole machine and each preset can be stored and recalled separately. This device has a password protection function to make the device more secure;

● Application: professional performances, conference rooms, courtrooms, auditoriums, multi-function halls, etc.

TECHNICAL PARAMETERS

Sample Rate：96KHz

Analog Input：4 x XLR Balanced

Input Impedance：10KΩ Balanced

Input Impedance：20KΩ Unbalanced

Analog Output：8 x XLR Balanced

Output Impedance：50Ω Balanced

Output Impedance:100Ω Unbalanced

Frequency Response：20Hz–20KHz±0.33db

AD & DA Converter：32 Bit AD and 24 Bit DA Converter

Default Output Level：0dBu

Maximum Input Level：20dBu

Maximum Input Level：20dBu

THD+N Distortion：<0.0025% @ 4dBu 20Hz-20KHz

System Delay：2.2ms

Noise Floor：-90dBu

Input To Output Dynamic Range：≥110dBu

S/N Ratio：≥110dBu

DSP PROCESSING TECHNICAL PARAMETERS

Use TI OMAP-L138 high-performance DSP, with a computing capacity of up to 3648 MIPS and 2746 MFLOPS per second, built-in multiple audio algorithms, 96KHz sampling rate.

Parametric equalizer：input and output channels with 16-band EQ adjust

6 filter types：Peaking, notch, all pass2nd, all pass 1st, high self and low self

The filter gain range：-15dB ~+15dB, step accuracy: 0.1dB

Input and output gain range：-72dB ~ +12dB, step accuracy: 0.1dB

Center frequency：adjustable within the frequency range of 20Hz~20kHz with step accuracy of 1Hz

Filter Q value Bandwidth：Peaking, Notch, All-pass 2^nd, Q value: 0.18～144.27;

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

LinkwitzRiley: 12/24/36/48dB/Oct

Noise gate：Threshold range: -100dBFS ~ 0dBFS;

Start control time: 1ms ~ 1000ms;

Recovery time: 1ms ~ 1000ms

Input compressor：Threshold range: -84dBFS ~ 0dBFS

Ratio: 1 ~ 20

Start-up time: 1ms ~ 1000ms

Recovery time: 1ms ~ 10000ms

Compensation gain: -24dB ~ 30dB

Output limiter：Threshold range: -84dBFS ~ 0dBFS

Recovery time: 100ms ~ 100000ms;

Delay：Each input and output channel has up to 2000ms delay adjustable