500556 - Processor MuxLab - Free user manual and instructions

Product Type	Dante 4-Channel Audio DSP (Digital Signal Processor)
Brand	MuxLab
Model	500556
Dimensions	142 x 157 x 46 mm (5.6 x 6.2 x 1.8 in)
Power Supply	DC 12V or PoE (IEEE 802.3af)
Dante Interface	2 Ethernet ports (Dante digital audio)
Analog Audio I/O	4 balanced inputs, 4 balanced outputs
Phantom Power	48V DC per input (switchable)
Sampling Rate	48 kHz, 24-bit
Frequency Response	20 Hz – 20 kHz (±0.25 dB)
Total Harmonic Distortion	≤0.005% @ +4 dBu
Noise Floor (A-weighted)	-88 dBu
Channel Isolation	100 dB @ 1 kHz
System Delay	1 ms
Maximum Input Level	+10 dBu (balanced)
Maximum Output Level	+14 dBu (balanced)
Input Impedance	100 Ω
Gain Range (analog)	-6 dB to +36 dB (simulated)
Control Interface	RS485, TCP/IP (API)
DSP Functions	AEC, AFC, ANC, Compressor, Limiter, Graphic EQ, Parametric EQ, Matrix Mixer
Software	Windows PC software (download via device web interface)
Operating Temperature	0°C to 50°C
Warranty	2 years
Safety & Cleaning	Unplug before cleaning; do not use liquid cleaners; keep away from moisture; refer servicing to qualified personnel.
Included Accessories	2 mounting brackets, 4×6-pin plugs, 2×3-pin plugs, 1 screwdriver, user manual (download)

Frequently Asked Questions - 500556 MuxLab

How do I connect the 500556 to a Dante network?

Connect one of the two Ethernet ports on the rear panel to a PoE-capable network switch using a standard CAT5e or higher cable. The device will obtain power via PoE and appear in the Dante Controller software.

What is the default IP address and how do I access the web interface?

The default IP address is 192.168.1.200 with subnet mask 255.255.255.0. Ensure your PC is on the same network segment, then enter the IP in a browser to download the PC software.

How do I restore the device to factory default settings?

In the installed PC software, go to Settings > Reset. This will restore all parameters to factory defaults, including IP address and password.

Can I use phantom power on all input channels?

Yes, each of the 4 balanced inputs can be individually set to provide 48V phantom power via the software's Input Settings tab. Toggle the 48V switch on/off as needed for condenser microphones.

What DSP processing options are available?

The unit includes Parametric EQ (5-band), Graphic EQ (31-band), Compressor, Limiter, Acoustic Echo Cancellation (AEC), Acoustic Feedback Cancellation (AFC), Noise Suppression (ANS), and a Matrix Mixer for flexible routing.

How do I update the firmware?

In the PC software, navigate to Settings > Upgrade. You can upload a firmware file obtained from MuxLab support. Do not power off during the update.

Is the 500556 compatible with third-party controllers?

Yes, it provides an API for RS485 and TCP/IP control. You can integrate it with external control systems like Crestron or AMX using the RS485 port or network commands.

What should I do if there is no audio output?

Check that the device is powered (LED indicators on front panel), the network cable is properly connected, and the Dante streams are subscribed in Dante Controller. Also verify that mute buttons are disabled and gain is set appropriately in the PC software.

How do I clean the device?

Unplug the power before cleaning. Use a dry, soft cloth to wipe the exterior. Do not use liquid or aerosol cleaners, as they may damage the electronics or cause electrical shock.

What is the maximum cable length for analog audio connections?

For balanced analog audio, use shielded twisted-pair cable. The maximum recommended length is 100 meters (328 feet) to avoid signal degradation, although shorter lengths are preferred for optimal performance.

USER MANUAL 500556 MuxLab

Dante 4-Channel Audio DSP

User Manual

500556

Table of Contents

1. Safety Precautions....3
2. Introduction......4
3. Features....4
4. Package Contents......4
5. Specifications....5
6. Hardware Interface....5
   6.1 Front Panel....5
   6.2 Rear Panel....6
7. Software Interface....6
   7.1 Software Download....6
   7.2 Connecting to the Unit....7
   7.3 Settings....8
   7.4 Pre-Pocessing....8
   7.4.1 Input Settings....8
   7.4.2 Parametric Equalizer....9
   7.4.3 Compressor....10
   7.4.4 AFC/AEC/ANS....11
   7.5 Matrix Mixing....13
   7.6 Post-Processing....13
   7.6.1 Graphic Equalizer....13
   7.6.2 Limiter....14
   7.7 Output Settings....14
8. Application Diagram....15

1. Safety Precautions

To ensure the best performance from the product, please read all instructions carefully before using the device. Save this manual for future reference.

• Follow basic safety precautions to reduce the risk of fire, electrical shock, and injury.
• Do not dismantle the housing or modify the module. It may result in electrical shock or burns.
• Do not open or remove the housing of the device as you may be exposed to dangerous voltage or other hazards.
• To prevent fire or shock hazard, do not expose the unit to rain, moisture and do not install this product near water. Keep the product away from liquids.
• Spillage into the housing may result in fire, electrical shock, or equipment damage. If an object or liquid falls or spills on the housing, unplug the module immediately.
• Do not use liquid or aerosol cleaners to clean this unit. Always unplug the power to the device before cleaning.
• Using supplies or parts not meeting the product specifications may cause damage, deterioration or malfunction.
  • Refer all servicing to qualified service personnel.
  • Install the device in a place with adequate ventilation to avoid damage caused by overheat.
• Unplug the power when left unused for a long period of time.
• Information on disposal of devices: do not burn or mix with general household waste, please treat them as normal electrical waste.

Copyright Notice

All contents in this manual are copyrighted, and cannot be cloned, copied, or translated without the express permission of MuxLab Inc. Product specifications and information in this document are for reference only, and the content may be updated from time to time without prior notice.

2. Introduction

The Dante 4-Channel Audio DSP (Model 500556) is a product developed for the transmission, routing, and processing of audio signals in Dante networks. It can perform high-quality, low-delay audio transmission, supports 4-way input and output balanced analog channels, and can be powered via PoE from any PoE network switch or via a 12V DC power supply. This processor supports 4-way analog audio signal balance input and output, 4-way Dante digital audio signal input of 2 Dante devices, and the 4-way Dante digital audio signal output. The device has one of the best DSP algorithms in the industry that includes options such as AEC, AFC, ANC, Compressor, Limiter, Mixer and Graphic Equalization among others; In addition, the product provides PC version control software to monitor and operate all the functions on the DSP in an easy and intuitive way.

3. Features

• Built-in Dante module
• Supports PoE power from PoE network switches or via a 12V DC power supply
- Features 4 balanced inputs and 4 balanced outputs
- The input supports 48VDC phantom power supply
- DSP processing functions include gain adjustment, stage parameter equalization, compressor, mixer, graphic equalization, limiter, setting, etc
- Provides software for the Windows platform to manage all audio operations
• API provided for RS485 and TCP/IP control

4. Package Contents

• One (1) Dante 4-Channel Audio DSP
- Two (2) Cabinet mounting bracket
• Four (4) 6 pin connection plug
- Two (2) 3 pin connection plug
• One (1) Screwdriver
• One (1) User manual (available via download)

Notes: Confirm that the product and accessories are all included. If not, please contact the supplier from which you purchased the unit.

5. Specifications

Specification
Dante Interface	4
Simulated maximum gain	-6~36dB
Digitalizing bit	24bit
Sampling Rate	48kHz 24bit
Frequency Response	20~20KHz (±0.25dB)
Phantom power (per input)	48V
Maximum level (input/output)	+10dBu /+14dBu , balance
Total harmonic distortion	≤0.005% @4dBu
Noise floor (A- weight)	-88dBu
Input impedance	100Ω
System delay	1ms
Channel isolation @1kHz	100dB
Working Power Supply	DC 12V & PoE
Working Temperature	0-50°C
Dimensions 142 X 157 X 46mm
Warranty	2 years
Order Information	500556 Dante 4-Channel Audio DSP (UPC: 627699005569)

6. Hardware Interface

6.1 Front Panel

1. LED display: Indicates the current IP address information of the device.
2. Channel indicator: Indicates In/Out when the channel has a signal flowing through.
3. Power indicator light: Indicates device is powered on.
4. System indicator light: When the device is running normally, the indicator light is flashing

6.2 Rear Panel

1. 12VDC Power interface (optional, when PoE not present)
2. Dante PoE network interface: Two network port interfaces, for Dante digital signal transmission or can connect to Dante Controller control software.
3. Input signal interface: 4 x Input Analog Audio signals
4. Output signal interface: 4 x Output Analog Audio signals
5. RS485 interface: used for control.

7 Software Interface

7.1 Software Download

The source file of the installation software is embedded in the device. To download the software, you only need to enter the factory default IP address (default IP: 192.168.1.200 Subnet Mask: 255.255.255.0) in the browser url address bar, and press Enter to navigate to the download interface. According to the content information of the web interface. Just click to download the software, and please note that before installing the PC software, please ensure that the PC client has installed Microsoft .Net Framework3.5 or above.

Note: Make sure the IP address of the client PC and the device are on the same network segment; otherwise it will not be accessible.

Install the "Dante 4-Channel DSP" software you just downloaded so you can control all the DSP options.

7.2 Connecting to the unit

Main Interface

Steps:

1. From the main interface, click search (1) to discover DSP units on the network.
2. Double-click the device name in the device list, the Sign In login window will pop up.
3. Enter the user name and password: (default user name: admin, password: 123456) click the [OK] button to enter.
4. If you need to modify the device IP address information, right-click Device name, and the Setting window will pop up, and you can change it according to the requirements of the use environment.

7.3 Settings

From the main interface, click Setting:

Upgrade: for the device firmware upgrade if necessary.

Reset: to restore the factory default parameter state.

About: to view device serial number information

7.4 Pre-Processing

7.4.1 Input Settings

1. 48V: Turn on or off the 48V phantom power supply of the analog channel,
2. Mute: mute button.
3. Channel fader: the channel gain value range (0\~-72dBFS), can be controlled by the fader.

7.4.2 Parametric Equalizer

Parametric equalizers are digital filters used in audio for adjusting the frequency content of a sound signal. Parametric equalizers provide capabilities beyond those of graphic equalizers by allowing the adjustment of gain, center frequency, and bandwidth of each filter.

Equalizer type: 5-terminal parametric equalizer

Passthrough/Enable: enable/disable the equalizer;

Band Pass/Enable: Enable/disable the equalizer for the current band

Reset: restore the current parameters to the default state

Center frequency: the center frequency that needs to be equalized

Gain: the gain/attenuation value of the frequency center point

Bandwidth: That is, the range of influence of the segment around the center frequency, the larger the value, the greater the bandwidth and the greater the range of influence.

7.4.3 Compressor

Compressors are used to reduce the dynamic range of signals above a user-determined threshold, the signal levels below the threshold remain unchanged.

Threshold: Threshold of the compressor.

Ratio: The input and output compression ratio of the compressor.

Attack Time: The start time of the compressor.

Recovery Time: The recovery time of the compressor.

Reset: reset to default parameters

Enable: Compressor enable indicator or control

Acoustic Feedback Cancellation (AFC)

- Multi-point filtering and multi sub-band 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.

Audio Echo Cancellation(AEC)

• Using sub-band 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.

Acoustic Noise Suppression (ANS)

- It is a noise suppression technique to deal with noisy speech signals.

- It decompositions the input signal into a series of frequency sub-bands, estimates the environmental noise and signal level in each sub-band, and then attenuates the sub-band signal according to the real-time SNR. The output signal is synthesized by smoothing and post-processing of these processed sub-band 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.

7.5 Matrix Mixing

Controls the mixing logic.

Columns: Input Channels

Row: Output Channels

7.6 Post Processing

7.6.1 Graphic Equalizer

The gain of 31 frequency points can be adjusted individually, to achieve the purpose of strengthening or weakening certain frequency points and achieve different effects.

Bypass/Enable: Enables and disables the equalizer.

Gain: Gain/attenuation at the frequency center point.

Flat: restore the gain of all frequency bands to 0dB.

Narrowband: A type of bandwidth that is lower than normal bandwidth.

Common: Commonly used ordinary bandwidth.

Broadband: the widest

7.6.2 Limiter

Thru/Enable: Enables or disables the limiter.

Threshold: The initial level of the limiter. When the signal is higher than this limit value, the limiter processing module is started.

Recovery time: When the input signal is lower than this setting value, the sound channel will not be turned off immediately, but the closing time will be delayed according to this setting value. During this time, as long as there is a signal above the "threshold" limit value, the sound channel can be continuously opened.

Compression: The difference between the signal processed by the limiter and the input signal.

7.7 Output Settings

You can and set the mute inversion of the output terminal and the output audio gain

1. Application Diagram
   

graph TD
    A["CD Player"] -->|500556| B["Switch"]
    C["Mic"] -->|500817| D["Powered Speakers"]
    E["Switch"] -->|4 In 4 Out Dante| F["Power Amplifier with Speakers"]
    F --> G["Mux-DSP Configurator"]
    G --> H["Laptop"]
    style A fill:#f9f,stroke:#333
    style C fill:#f9f,stroke:#333
    style E fill:#f9f,stroke:#333
    style F fill:#ccf,stroke:#333
    style G fill:#cff,stroke:#333
    style H fill:#ffc,stroke:#333