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AI Noise Reduction and Feedback Suppression Processor(Dante)ANP-401D
sketch:
1.Equipped with a 2inch display paired with a rotary knob for easy, on-device host parameter configuration.
2.Supports 4CH analog and 2CH Dante(AES67 optional) input, AI-processed and mixed to 1CH analog or Dante/AES67 output for clear audio.
3.AI-Powered Audio Clarity
(1)Deep Noise Cancellation: Up to 25dB suppression of environmental noise.
(2)Smart Feedback Control: 18dB adaptive feedback suppression gain.
(3)Intelligent Recognition: Deep learning distinguishes human voice/instruments from background noise (steady-state & transient).
(4)Customizable: Switchable AI features with 5 adjustable levels for any meeting scenario.
(5)Natural Sound Quality: Zero noticeable distortion or "pumping effect" after processing.
4.High-Resolution Audio Processing
Capable of 24-bit audio restoration at a 48kHz sampling rate, the system maintains a minimal processing low latency less than 15ms.
5.Interface Compatibility
4×XLR female/6.35 mm balanced combo connectors with per-channel switchable 48 V phantom power, plus 4 × Phoenix terminal block inputs. This design ensures compatibility with a wide range of audio devices, including microphones, mixing consoles, and sound cards.
6.Control Protocol Support
Supports UDP, RS232, and RS485 control protocols, enabling flexible system integration and remote device management.
7.AES67 Version
The AES67 version is optional, providing compatibility with modern audio-over-IP networks for flexible deployment in professional settings.
overview Function and characteristic Technical Parameter
1.Equipped with a 2inch display paired with a rotary knob for easy, on-device host parameter configuration.
2.Supports 4CH analog and 2CH Dante(AES67 optional) input, AI-processed and mixed to 1CH analog or Dante/AES67 output for clear audio.
3.AI-Powered Audio Clarity
(1)Deep Noise Cancellation: Up to 25dB suppression of environmental noise.
(2)Smart Feedback Control: 18dB adaptive feedback suppression gain.
(3)Intelligent Recognition: Deep learning distinguishes human voice/instruments from background noise (steady-state & transient).
(4)Customizable: Switchable AI features with 5 adjustable levels for any meeting scenario.
(5)Natural Sound Quality: Zero noticeable distortion or "pumping effect" after processing.
4.High-Resolution Audio Processing
Capable of 24-bit audio restoration at a 48kHz sampling rate, the system maintains a minimal processing low latency less than 15ms.
5.Interface Compatibility
4×XLR female/6.35 mm balanced combo connectors with per-channel switchable 48 V phantom power, plus 4 × Phoenix terminal block inputs. This design ensures compatibility with a wide range of audio devices, including microphones, mixing consoles, and sound cards.
6.Control Protocol Support
Supports UDP, RS232, and RS485 control protocols, enabling flexible system integration and remote device management.
7.AES67 Version
The AES67 version is optional, providing compatibility with modern audio-over-IP networks for flexible deployment in professional settings.
Function and characteristic



Technical Parameter
Bit depth
24BIT;
Noise reduction range
-10DB to -40DB adjustable continuously;
Frequency response
50HZ to 20KHZ (+0.5DB);
Total harmonic distortion + noise
≤0.01% (1KHZ, 0DB gain);
Processing delay
≤15MS (48KHZ);
Gain adjustment range
independent adjustment of -10DB to +10DB for each channel;
Phantom power
+48V / 10MA MAX;
Power supply specification
AC110V - 220V;
Operating environment
temperature 0°C to 45°C, humidity 10% to 85% (no condensation);
Form size
482MM (width) x 220MM (depth) x 44MM (height) 1U rack;
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