16x16 Digital Audio Processor

16x16 Digital Audio Processor
Details:
ARES Q1616 digital audio processors are powered by the ADSP-21569 core and offer a variety of commonly used audio processing functions such as AEC (Acoustic Echo Cancellation), AFC (Adaptive Feedback Cancellation), AM (Automatic Mixing), Ducker, EQ (Equalization), and more, fulfilling the audio processing needs of various sound systems. They are suitable for applications in conference rooms, churches, five-star hotels, auditoriums, conference centers, multifunctional halls, and digital courtrooms.
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Description
Specification

Shenzhen Tendzone Intelligent Technology Co., Ltd. is one of the most reliable manufacturers and suppliers of 16x16 digital audio processor in China. Please feel free to wholesale high quality 16x16 digital audio processor made in China here from our factory. We also accept customized orders.

 

16x16 Digital Audio Processor

16×16 Digital Audio Processor - AEC · AFC · AGC · Auto-Mixing · Web-Based Control

 

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Product Overview

 

The ARES Q1616 is a professional digital audio processor built on the powerful ADSP-21569 floating-point DSP core. It provides a complete suite of audio processing functions - AEC (acoustic echo cancellation), AFC (adaptive feedback cancellation), ANS (noise suppression), AGC (automatic gain control), auto-mixing, ducker, limiter and full parametric EQ - covering the sound-reinforcement needs of most audio systems.

 

The 16 × 16 configuration offers 16 channels of balanced microphone/line inputs with +48 V phantom power and 16 channels of balanced line outputs, at a 24-bit / 48 kHz sampling rate. A B/S-architecture software platform allows PC or mobile devices to log in remotely through a browser to adjust parameters and download manuals, while the embedded system design keeps the processor free from virus interference for outstanding operating stability.

 

The ARES Q1616 is widely used in conference rooms, churches, star-rated hotels, auditoriums, conference centers, multi-function halls, digital courtrooms and other venues that demand reliable, high-quality sound reinforcement.

 

 

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Key Features
 

 

 

Audio I/O & Sound Quality

  • 16 channels of balanced microphone/line inputs with 48 V phantom power soft switch.

    16 channels of balanced line outputs.

    24-bit / 48 kHz sampling rate for outstanding audio quality.

    Optimized preamp gain with 0 dB / 10 dB / 20 dB / 30 dB / 40 dB settings, accommodating both MIC and LINE levels.

    USB 2.0 audio interface supporting recording and playback via USB flash drive (MP3 / WAV, FAT32).

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Acoustic Processing

  • Adaptive Feedback Cancellation (AFC) - high-speed floating-point algorithms provide feedback suppression for each microphone channel, with 16 suppression points and manual / dynamic / fixed processing modes, effectively eliminating "howling".
  • Shared adaptive Acoustic Echo Cancellation (AEC) - processed independently on every input channel with a 512 ms tail, swiftly eliminating echoes in video conferences; built-in ANS removes up to 18 dB of noise.
  • Automatic Gain Control (AGC) - keeps system output volume stable regardless of how far the presenter is from the microphone.
  • Intelligent Ducker - microphone signals automatically duck background music, highlighting the vocals.
  • Instant-response limiter to perfectly protect downstream equipment.
  • Threshold-type and gain-sharing automatic mixing (AM) for easy management of multi-microphone combinations and clear sound reinforcement.
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Application Scenarios

  • Conference rooms and boardrooms - echo-free far-end audio for video conferencing.
  • Digital courtrooms - intelligible, stable speech reinforcement for proceedings.
  • Conference centers and multi-function halls - flexible zone routing and EQ tuning.
  • Auditoriums and lecture halls - automatic feedback suppression with multiple microphones.
  • Churches and houses of worship - natural vocals over background music via ducker.
  • Star-rated hotels - background music paging with priority ducking announcements.

 

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System Solution

 

Microphones and program sources feed the 16 balanced inputs, where the ARES Q1616 applies preamplification, AEC / AFC / ANS / AGC, auto-mixing and equalization. The processed signals are routed to power amplifiers and speakers through the 16 balanced outputs, recorded to or played back from a USB flash drive, and delivered to video conferencing terminals with far-end echo fully cancelled. Engineers can tune every parameter remotely from a browser, while third-party control systems integrate via TCP/IP or RS-232.

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About Tendzone

 

Established in 2010, Tendzone is a global leader in providing advanced audio-visual (AV) solutions and manufacturing high-quality AV products. We specialize in a wide range of cutting-edge technologies, including audio processors, microphones, speakers, power amplifiers, AV over IP systems, digital conference systems, and MIDIS Distributed Multimedia Transmission Control Systems. Our solutions are trusted across industries such as conference rooms, command centers, education, multi-functional halls, and stadiums.

 

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Our Service

 

Our dedicated customer support team offers reliable technical support, quick response times, and preventive maintenance service.

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Pre-Sales

We assess your unique requirements, provide tailored consultations and demonstrations, and design a customized AV system to ensure the perfect solution.

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In-Sales

We deliver transparent proposals, manage the full project implementation, and provide initial training to ensure a smooth and successful setup.

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After-Sales

We offer prompt technical support, customized maintenance plans, continuous software upgrades, and actively use your feedback to drive ongoing service improvement.

 

 

How To Cooperate With Us?

Tendzone, established in 2010, is a global leader in providing advanced audio-visual (AV) solutions and manufacturing high-quality AV products. We specialize in a wide range of cutting-edge technologies, including audio processors, microphones, speakers, power amplifiers, AV over IP systems, digital conference systems, and MIDIS Distributed Multimedia Transmission Control Systems. Our solutions are trusted across industries such as conference rooms, command centers, education, multi-functional halls, and stadiums.

Our address

Building 1A, Building 16, Shenzhen International Innovation Valley, Dashi 1st Road, Nanshan District, Shenzhen

Phone Number

Mobile/Wechat/WhatsApp : (+86) 13302998794

E-mail

travis.tu@tendzone.com

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Technical Specifications 

Sampling Rate

48 kHz, 24-bit

Frequency Response

20 Hz – 20 kHz (±0.5 dB)

Dynamic Range (A/D/A)

>106 dB, A-weighted (20 Hz–20 kHz @ 0 dB gain)

THD+N

≤0.005% @ 1 kHz, +4 dBu

Noise Floor (A-weighted)

-90 dBu

Channel Isolation

≤-105 dB @ +20 dBu, 1 kHz

Input CMRR

≥60 dBu @ +20 dBu, 60 Hz

Maximum Input Gain

40 dB

Maximum Input Level

20 dBu

Maximum Output Level

20 dBu

Input Impedance

2 kΩ balanced

Output Impedance

200 Ω balanced

Phantom Power

DC 48±6 V, 10 mA

AEC

Echo cancellation tail 512 ms; built-in ANS, noise reduction up to 18 dB

AFC

16 suppression points; manual / dynamic / fixed processing modes

Auto Mixing

Threshold type and gain-sharing type

Input Channel Processing

Preamplifier, signal generator, expander, compressor, 10-band parametric EQ (PEQ / NOTCH / High-Shelf / Low-Shelf / LP / HP)

Output Channel Processing

Selectable 10-band or 31-band parametric EQ, delay, high-pass / low-pass filters, limiter

Presets

50 groups

Connectors

3.81 mm Phoenix terminal blocks (analog I/O)

Front Panel Indicators

PWR / RUN / LINK

Control

LAN (TCP/IP, 10/100/1000M RJ45), RS-232, open third-party protocol; WEB UI in Chinese / English

USB

USB 2.0 - recording & playback via USB flash drive (MP3 / WAV, FAT32)

Power Supply

AC100-240V~, 50/60 Hz

Power Consumption

35 W

Dimensions (W × D × H)

483 × 180 × 44 mm (1U, standard 19" rack)

Net Weight

2.3 kg

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