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M-7132A network type digital audio matrix

Product number:1065212205814403072


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M-7132A is a powerful digital audio processor, using ADI SHARC fourth-generation ADSP-21489 floating-point audio DSP chip, providing the highest performance 400 MHz/2200 MFLOP processor capability.




  • Product Description
  • M-7132A is a powerful digital audio processor, using ADI SHARC fourth-generation ADSP-21489 floating-point audio DSP chip, providing the highest performance 400 MHz/2200 MFLOP processor capability. Ultra-low noise floor preamplifier circuit, low distortion analog circuit, 118dB audio AD, DA, provide high-quality sound quality for the scene. The volume status display on the front panel of the device can display the volume status of all input 16 and output 16 channels; the LCD display on the front panel of the device can cyclically display the current IP address of the device, the preset number and preset name currently used by the processor, which is convenient for the system When there are multiple processors in the system, it is convenient for the administrator to quickly identify and distinguish the current processor. The front panel has a network connection status indicator and an error alarm indicator. When a serious error occurs, a red alarm can be flashed.

    Powerful DSP processor capability
    Based on the super powerful DSP processing capability of ADSP-21489, and based on our unique core DSP algorithm, a built-in independent 16-channel ultra-low distortion adaptive feedback suppressor, 16-channel noise gate, 16-channel input compressor, 16-channel 8-band PEQ, 16-channel input 48dB slope high-pass-low-pass filter, and 16-channel 1-second delay means you can make precise and meticulous settings for each input audio channel. 16x16 full matrix mixing to achieve free mixing of any input channel. The 16 output channels have independent 8-band PEQ, compressor, 48dB slope high-pass-low-pass, and 2-second delay.
    Built-in sine wave, pink noise, white noise signal generator. 32 presets can be stored.

    AEC echo cancellation algorithm (optional function)
    The algorithm-based fast adaptive echo cancellation and unique local benefit mode bring clear voice to remote video conferences. The echo cancellation function can be switched on and off with one key, and users can quickly compare the echo cancellation effect and reduce the complexity of debugging.

    Rich audio channels and control interfaces
    1 TCP/IP communication port, 1 RS-232 communication port, open third-party control protocol. It can meet the use of various large, medium and small professional audio projects. It can meet the application requirements of public sound reinforcement systems such as theaters, concert halls, remote video conferences, stadiums, churches, conference centers, theme parks, etc.

    LCD screen status display
    The 1602 LCD screen on the front panel displays important information such as IP address, current preset name, and usage time.

    Easy-to-use control software
    We visited experienced senior audio engineers and professional tuners, communicated in-depth debugging and application personnel's operating habits, and developed software interfaces that meet industry applications. The control software is easy to understand and can be operated quickly without manuals. Each input value can be directly input with the keyboard to get the exact value you want, such as - 12. 2dB directly input - 12. 2. Click the volume fader to select the channel, and press the up/right, down/left direction keys on the keyboard to achieve 0.1dB step and step reduction. For complex parameter adjustment such as PEQ and Limiter, the parameters can be quickly copied and pasted, and you can easily realize multi-channel data copying, which is convenient to operate.

    Open RS-232, TCP/IP communication protocol
    Realize third-party equipment to control volume, call mode, set mute, and batch read pre-mixing level meter and post-mixing level meter through TCP/IP protocol, which is convenient for third-party software integration.

    High resolution Android, Apple IOS app
    Support mainstream high-definition Android tablet computer, Android phone resolution to 1920x1080 high-definition resolution
    Automatically discover audio processors on the network
    Can display 16 input, 16 output channel signal level meters
    Can control 16 input, 16 output channel volume, mute
    The App can recall 32 presets, and can display the names and status of all presets on the host
    The device list page can display key information such as IP addresses, ARM-DSP firmware version numbers, and device descriptions of all processors on the network.

    ● High-performance floating-point DSP processing chip;
    ● 8~16 channels balanced input audio channels
    ● All input channels support MIC input
    ● All input channels support 48V phantom power supply
    ● 8 to 16 balanced audio output channels
    ● Independent adaptive feedback suppressor for each channel
    ● 8 to 16 channels automatic mixing
    ● ADC PCM4204 118dB dynamic
    ● DAC PCM4104 118dB dynamic
    ● Input per channel: preamp, noise gate, compressor, 8-band parametric EQ, delay, automatic mixer
    ● Output per channel: 8-segment parametric equalization, high and low pass filters, compressors, delays
    ● Built-in signal generator: sine wave signal, pink noise, white noise
    ● Front panel 1602 display shows IP address, current preset
    ● Open RS-232, TCP/IP protocol to realize third-party control
    ● With camera tracking code output, it is convenient to realize camera linkage function through third-party central control
    ● Supports 32 sets of scene preset functions, which can be called through TCP/IP and RS-232 protocols
    ● Built-in USB playback and recording functions, can identify Chinese song names
    ● 8 custom function GPIO

    Specifications and parameters
    DSP frequency, processing capacity: ADI ADSP-21489 400Mhz, 400MIPS/2200 MFLOP
    Core algorithm: matrix mixing, automatic mixing, optional: echo cancellation
    Sampling frequency/quantization: 48 kHz, 24Bi t ADC, 24Bi t DAC
    ADC, DAC dynamic range: ADC PCM4204 118dB, DAC PCM4104 118dB
    Total harmonic distortion: <0.0035% @0dBu 20Hz~20kHz
    Frequency Response: 20 Hz – 20 kHz, ±0.3dB @0dBu
    Common Mode Rejection Ratio (CMRR): >80dB @ 1 kHz MIC Gain 20dBu
    Channel Crosstalk Crosstalk: 100dB ±5dB @ 0dBu 1Khz
    Maximum input level: +18.5dBu
    Audio interface standard: 8 to 16 balanced input, phoenix plug
    Default input and output levels: +0 dBu
    Microphone preamp gain: 0-40dB analog gain, 12dB digital gain
    Input Impedance: >5kΩ balanced, >3kΩ unbalanced
    Output impedance: 600Ω
    Preamp Equivalent Input Noise (EIN): <-125 dBu, 22Hz - 22kHz
    Output noise floor: -93dBu (unweighted)
    Phantom Power: +48 VDC
    Power supply: 100~240VA, 50/60 Hz, 75W
    Working environment: maximum ambient operating temperature 40°C



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