Full Function Matrix Remix 16in 16out Audio Matrix DSP Audio Processor With Echo Canceller
Main Features:
- USB Background music playback and recording function
- Support mobile phone, tablet control and distributed cloud control
- DSP audio processing, built-in automatic mixer, feedback elimination(AFC), echo elimination(AEC), noise elimination (ANC) function
- Input per channel:Front stage amplifier, signal generator, expander, compressor, 5 stage parameter equalization
- Output per channel:31 Section diagram equalizer, delay device, frequency divider, limiter
- Full function matrix remix
- Built-in automatic camera tracking function
- Support for scenario presets;
- Automatic memory protection when power off
- 1U Whole aluminum chassis
- Channel:8CH,12CH,16CH

Specification: Model No. | MX-1616 | DSP Processing | Ti 456MHz FLOPS DSP | Number of analog channels | 16 Input +16 Output | Core Algorithm | Automatic mixing, feedback elimination(AFC), echo elimination(AEC), noise elimination(ANC) | GPIO | 8 ( Including input and output) | RS232/RS485 | 1 | RJ45 control interface | 1 | USB Port | 1 | RJ11 phone interface | 0 | DANTE network interface | 0 | Simulated maximum gain | 51dB | Digitalizing bit | 24bit | Sampling Rate | 48k | Frequency Response(20~20KHz) | ±0.2dB | Analog-to-digital dynamic range (A-weighted) | 114dB | Digital-to-analog dynamic range (A-weighted) | 120dB | Input to output dynamic range | 108dB | Total harmonic distortion + noise | <0.003% @1KHz ,4dBu | Floor noise (A- weighted) | -90dBu | Delay storage | 2s | Analog input to output system delay | 3ms | Input impedance (balance -type) | 20KΩ | Output impedance (balance -type) | 100Ω | Maximum Input Level | +18dBu,balance | Maximum Output Level | +18dBu,balance | Equivalent input noise EIN(20-20kHz,A-weighted). | ≤-131dBU | Phantom power (per input) | 48V | Input common mode rejection,60Hz | 70dB | Channel Isolation,1kHz | 104dB | Dimensions (W*D*H) | 482*258*45(mm) | Weight | 3Kg | Power Consumption | <40W | Operating Temperature | -10 - 50℃ | Working Power Supply | AC110V-220V,50Hz/60Hz | |
Audio Processing Core Algorithm:
Efficient and comprehensive algorithm is the basis of perfect sound quality, but also the crystallization of engineers' experience and wisdom. The built-in core algorithm is the soul of the processor.
AUTOMIXER
- Improve the transparency and clarity of speech;
- The feedback, reverberation and comb filtering effects are significantly reduced.
- Automatic adjustment, simplified Settings, plug and play;
- It can solve common problems such as insufficient gain before feedback and unclear speech.
- Each input channel has a dual-band equalizer.
- The adaptive noise threshold allows each input channel to distinguish between continuous background noise (such as air conditioning) and changing sound (such as voice), and constantly adjusts the channel activation threshold, so that the channel can only be activated when the voice volume is higher than the background noise;
- Lock the last mic until the next mic is activated, ensuring that background ambient sounds are present (without the last mic lock, a long pause in the conversation shuts down all the microphones, as if the audio signal is missing);
- Precisely control the priority of each microphone and lock down key speakers.
Automatic Echo Cancellation (AEC)
- Using subband algorithm, it has less MIPS consumption.
- The length of echo path can be set, the maximum echo off tail can be supported up to 512ms, suitable for all kinds of large, medium and small meeting rooms;
- Using the stable Double Talk detection method, it is effective even in the environment of strong background noise and nonlinear distortion, and the residual echo will not increase during the simultaneous speech of both sides.
- Strong robustness, can work in all possible applications and environments;
- The embedded noise suppression algorithm can eliminate the additional noise in the noise environment.
- The variable step size and post-processing algorithm greatly improve the rate of convergence and the echo rejection ratio (ERLE) of the nonlinear distortion of the terminal speaker.
Automatic feedback elimination (AFC)
- Multi-point filtering and multi-subband frequency shifting keep the harmonic property of the original pitch period without causing sound distortion.
- Through acoustic modeling of room feedback path, the acoustic feedback can be eliminated adaptively.
- It can quickly track the feedback path changes and greatly enhance the ability to suppress the noise. The microphone transmission gain can be increased by 6-18db, greatly enhancing the microphone gain, suitable for various large, medium and small meeting rooms.
Automatic noise elimination (ANC)
- It is a noise suppression technique to deal with noisy speech signals.
- It decompositions the input signal into a series of frequency subbands, estimates the environmental noise and signal level in each subband, and then attenuates the subband signal according to the real-time SNR. The output signal is synthesized by smoothing and post-processing of these processed subband signals.
- Because of the unique post-processing algorithm, the noise suppression algorithm can track the environmental noise changes quickly and accurately while maintaining good output sound quality. Noise suppression reaches -30db, speech is almost completely distortion free.
FAQ:
Q1: Are you manufacturer or trade company?
A: We are manufacturer,which located in Shenzhen, Guangdong, China.
Q2: Do you offer OEM&ODM service?
A: Yes, We can provide OEM or ODM Solution depends on client requirements
Q3: What's the quality standard of your products?
A: Our products has approved CE, ROHS certification.
Q4: Which place will use FHB equipment?
A: Our products are widely used in International Conference, Governments, Banks, Churches, Airports, Parliaments, Education institutes, Hotel, Private corporate conference hall etc,.
Q5: What’s the delivery time?
A:Sample order: During 5days,Bulk order:10-15days
Q5: Why choose FHB Audio Products?
A:Pls refer to below details:
FHB Audio is driven by innovation. We are continuously developing new technologies that provide time saving methods that benefit you and our dealer network. Our innovative solutions focus on everyday use cases and how our products can cut down installation times, reduce maintenance requirements, increase productivity, and provide you long-lasting reliability.
- Wide & Deep Products Line
We provide over 200 audio & Video products, as well as a series of Dante products, such as Digital Signal processor, Digital microphone system,Dante signal processor, 2CH Dante AVUO,2CH Dante wall plate,4CH Dante box., Dante Bluetooth panel, Dante microphone, and Dante speaker etc…
Our industry-leading support team makes it easy for you to choose the right product and provide the right system solutions.
- Professional & Fast Service,
- 24 hours online services to provide suggestions and solutions for customer.
- After-Sales Technical Support /Product Maintenance
- Wiring/Installation/Debugging Instruction
Professional System Designers to assist in product selection and system design to meet your specific needs.
Check Products one by one, to make sure all products are working excellent quality..