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  • RM-D1616-正.jpg

16×16 Digital Audio Media Processor RM-D1616


Full-featured mixing. It provides flexible and simple signal routing operations to users. A button can adjust the routing path and level.

Product number:RM-D1616

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  • Description
  • Technical Specs
  • Connection
  • Ø Analog input channel: 16

    Ø Analog Output channel: 16

    Ø Processor :ADI SHARC 21489@450 MHz SIMD;

    Ø DSP processing capacity :400 MIPS, 1.6 GFLOPS;

    Ø Sampling rate :48 kHz, ± 100 ppm;

    Ø THD+N:<-94dB @17dBu

    Ø Input dynamic range: 110dB

    Ø Output dynamic range: 112dB

    Ø The processing module is configurable and it can be replaced as required.

    Ø Provide 8 machines operation on line at the same time;

    Ø Built-in One In One Out USB sound card. It supports music play, recording and soft video conferencing (such as: ZOOM, Tencent conference, nail conference, etc.);

    Ø Bus AEC; Tail time: 512msConvergence rate: 60dB/S; Echo cancellation amplitude: 60dB;

    Ø AFC (feedback suppression) for each independent channel. It adopts Notch algorithm, and sound transmission gain increase: 10dB;

    Ø Noise suppression (ANS); Signal-to-noise ratio increase 18dB.

    Ø Ducker

    Ø Noise gain compensator (ANS)

    Ø Gain Sharing Automatic mixing (AMC) and Gate Mixer.

    Ø 16 balanced section of the British parameters. It provides 5 kinds of filter options: Parametric, Lowshelf, Highshelf, Lowpass, Highpass;

    Ø Provides end-user customized operation interface. Maximum 30 devices can be managed on the same interface;

    Ø It has central control function. And it can control the power supply, signal switching, environmental control, audio and so on in the system as a whole.                                                    

    Ø It should provide the customed user interface;

    Ø It should have built-in USB sound card, and support recording and teleconference;

    Ø The processor chip adopts ADI architecture, not less than 40bit DSP floating point arithmetic engine. And it provides free configuration software architecture;

    Ø Analog input and output channel is not less than 16x16

    Ø Input and output quantization is not less than 48KHz/24bit;

    Ø The input channel should have:

    Ø Input gain is 3dB, not less than 16 gears;

    Ø Not less than 12 PEQ segments and not less than five filter types.

    Ø Adaptive echo cancellation (AEC), noise suppression (ANS);

    Ø Gain shared Automatic mixing (AMC), Gate Mixer;

    Ø Automatic gain (AGC)

    Ø Ducker

    Ø The noise gain compensator (ANC)

    Ø Each channel should have no less than 16 points of adaptive feedback control (AFC).

    Ø The output channel should have:

    Ø Not less than 12 PEQ, 31 GEQ, Frequency divider, Delay, Limiter.

    Ø Not less than 12 x17 Matrix;

    Ø Not less than 16 sets of presets;

    Ø Support input/output channel LINK and Grouping function;

    Ø The input dynamic range must be at least 110dB.

    Ø The output dynamic range must be at least 112dB

    Ø Frequency response range is not less than 2020kHz (±0.15dB) ;

    Ø THD+N: no less than -94dB @17dBu                                                                                                                                                             Ø ProcessorADI SHARC 21489

    Ø Sampling rate/Quantization Bits48K/24bit

    Ø 40bit DSP Floating point arithmetic engine

    Ø Analog input and output channels: 16x 16

    Ø Input Gain: 0/3/6/9/12/15/18/21/24/27/30/33/36/39/42/45 dBu

    Ø Phantom power: +48V/10mA max

    Ø Frequency response(20~20kHz):±0.15dB

    Ø Maximum level: +18dBu

    Ø THD+N:<-94dB @17dBu

    Ø Input dynamic range: 110dB

    Ø Output dynamic range: 112dB

    Ø Channel isolation @1kHz:108dB

    Ø Input impedance (balanced connection):5.4KΩ

    Ø Output impedance (balanced connection):600Ω

    Ø System delay:<3ms

    Ø Working power: AC110~240V,50Hz/60Hz

    Ø Dimension (W x D x H):482 x 260 x 45mm

    Ø Weight:4KGS

    Keywords:
    • Conference Microphone System
    • Video Conference System

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