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DSP Audio Matrix Processor

DSP

SEADA DSP audio matrix processor is an advanced, professional audio matrix switcher that includes multi−channel DSP, AEC (Acoustic Echo Cancellation), HDMI de−embedding, and class compliant USB audio interface. The comprehensive and user−friendly graphic interface makes configuring every detail of your audio system intuitive and easy.

 

Key Features

  • ADI SHARC platform with semi-open architecture and ETHERNET support.
  • Integrated management of multiple devices offers open user interface.
  • Supports good sound quality (at 24bit/48kHz.)
  • Each channel has its own independent adaptive feedback suppression which can find and suppress audio feedback automatically.
  • Supports Acoustic Echo Cancellation (AEC) and Adaptive Feedback Cancellation (AFC)
  • Support Active Noise Suppression (ANS).
  • Gain sharing Auto Mixer Control (AMC).
  • Automatic Gain Control (AGC).
  • Audio Ducking (Ducker) *.
  • Ambient Noise Compensation (ANC) *.
  • Full-featured audio mixing matrix supports adjustable input level.
  • Provides up to 16 presets, each can work independently.
  • Provides 8 GPIOs, each can be independently configured as input or output, and can be used as independent ADC when configured as input *.
  • Supports audio channel duplicating, linking, and grouping.
  • Supports RS-232 and UDP central control. Provides the flexibility to assign UDP port number, with the control commands available through the control software.
  • Built-in dual channel USB sound card can be used for audio recording, audio broadcasting, and video conference.

 

*These features are not supported on SD-0404(D)

SEADA’s DSP audio matrix processor is an advanced, professional audio matrix switcher that includes multi-channel DS, AEC (Acoustic Echo Cancellation), HDMI de-embedding, and class compliant USB audio interface. The comprehensive and user-friendly graphic interface makes configuring every detail of your audio system intuitive and easy.

SEADA’s DSP audio matrix processor has great advantages on its core Algorithms which include AEC (Echo Cancellation), AFC (Feedback Cancellation), ANS (Noise Suppression), and AGC (Automatic Gain Control). The Echo Cancellation algorithm can be applied to signals from local analog inputs or remote digital channel, while Automatic Gain Control ensures that microphone levels remain at an optimum level, Noise Suppression removes steady state noise, and Feedback Suppression enables more flexible microphone, loudspeaker placements and speech reinforcement, delivering audio where you want. The signal processing is fully customizable via the DSP control software.

MODELS SD-0404(D) SD-0808(D) SD-1212(D) SD-1616(D)
Sampling Rate 48K/24BIT
Input Gain 0/6/12/18/24/30/36/42/48 DBU 0/3/6/9/12/15/18/21/24/27/30/33/26/39/42/45/48 DBU
Phantom Power 48V/10mA max
Frequency Response 20Hz~20kHz (±0.3dB)
Maximum Level ±18dBu
HD + Noise 0.0003%@4DBU 0.0003%@ 1kHz, 4DBU
Input Dynamic Range 110dB 113dB
Output Dynamic Range 112dB 115dB
Channel Isolation 108DB @ 1kHz
Input Impedance (Balanced connection) 5.4 KΩ
Output Impedance (Balanced connection) 600Ω
System latency <3MS
Number of Analog In/Out 4 x 4 8 x 8 12 x 12 16 x 16
Audio over IP In/Out (D only) 4 x 4 8 x 8 12 x 12 16 x 16
Dimensions (W X D X H) 215 X 162 X 45MM 482 X 260 X 45MM
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