To control the MXA910, use Shure Designer software. After completing this basic setup process, you should be able to:
Before you get started, you'll need:
Note: If Designer isn’t available, you can use a web application to control the MXA910 and Dante Controller to route audio. Download the Shure Web Device Discovery application to access your device’s web application.
The easiest way to route audio and apply DSP is with Designer's Optimize workflow. Optimize automatically routes audio signals, applies DSP settings, turns on mute synchronization, and enables LED logic control for connected devices.
The MXA910 includes IntelliMix® DSP that can be applied to the automix channel output.
For this example, we'll connect an MXA910 and an ANIUSB-MATRIX.
You can also route audio manually in Designer outside of the Optimize workflow, or use Dante Controller.
After you have coverage set up, you can send audio from the ANIUSB-MATRIX to other Dante devices or analog sources.
The Microflex®Advance™ Ceiling Array is a premium networked array microphone for AV conferencing environments, including boardrooms, huddle rooms, and multi-purpose spaces. The ceiling array uses Shure's Steerable Coverage™ with Autofocus™ technology: 8 highly directional pickup lobes capture participant audio from overhead, continually fine-tuning the position of each lobe in real time as participants lean back in their chairs or stand up .
The microphone also includes the IntelliMix DSP suite, which includes AEC, noise reduction, automatic mixing, and more. Control the microphone with Shure Designer software, or a browser-based web application. The microphone integrates seamlessly with Dante™ digital networked audio, AES67, and third-party preset controllers, including Crestron and AMX, to deliver a high-quality AV conferencing experience that appeals equally to integrators, consultants, and meeting participants.
Configurable Coverage
IntelliMix DSP
Software Control
Network Connectivity
Professional Design
① Dante audio, power, and control
Each array microphone connects to the network over a single network cable, which carries Dante audio, Power over Ethernet (PoE), and control information to adjust coverage, audio levels, and processing.
② Analog audio (microphone to network)
Analog equipment, such as a wireless microphone system or a gooseneck microphone on a podium, connects to the Dante audio network through a Shure Network Interface (model ANI4IN) for a completely networked conferencing system.
③ Far-end audio (network to loudspeakers)
Dante-enabled loudspeakers and amplifiers connect directly to a network switch. Analog loudspeakers and amplifiers connect through a Shure Network Interface (model ANI4OUT), which converts Dante audio channels into analog signals, delivered through 4 discrete XLR or block connector outputs.
④ Device control and Dante audio
Control: A computer connected to the network controls the microphone with Shure Designer software. You can remotely adjust coverage, muting, LED behavior, lobe settings, gain, and network settings.
Audio: Route audio with Dante™ Controller or Shure Designer software. Dante Virtual Soundcard enables audio monitoring and recording directly on the computer.
The network port carries all audio, power, and control data. It is located on the back panel as shown.
① Network Port
RJ-45 jack for network connection.
② Network Status LED (Green)
Off = no network link
On = network link established
Flashing = network link active
③ Network Speed LED (Amber)
Off = 10/100 Mbps
On = 1 Gbps
The LED on the microphone indicates whether the microphone is active or muted, identifies the hardware, and provides confirmation of firmware updates.
| Microphone Status | LED Behavior / Color |
|---|---|
| Active | Green (solid) |
| Mute | Red (solid) |
| Hardware identification | Green (flashing) |
| Firmware update in progress | Green (progresses along bar) |
| Reset |
Network reset: Red (progresses along bar) Factory reset: Triggers device power-up |
| Error | Red (split, alternate flashing). See the device event log for details. |
| Device power-up | Multi-color flash, Blue (moves quickly back and forth across bar) |
Note: When the LED is disabled, the LED still illuminates while the device is powering up and when an error state occurs.
Custom LED brightness, colors, and behaviors are assignable in the control software. They can also be controlled through an external control system:
The lighting for mute and active microphone states is configurable to match the behavior of other devices in conference rooms. In the LIGHT BAR PROPERTIES menu, use the drop-down menus to select LED settings.
To dim or turn off the LED, use the brightness fader.
This device requires PoE to operate. It is compatible with both Class 0 and Class 3 PoE sources.
Power over Ethernet is delivered in one of the following ways:
The hardware reset button is located inside a grille hole and can be pushed with a paperclip or other small tool. The hole is identified with a gray circle. When looking at the Shure logo, it is the second hole in the fourth row from the top.
Network reset (press button for 4-8 seconds)
Resets all Shure control and audio network IP settings to factory defaults. Red LED along bar.
Full factory reset (press button for longer than 8 seconds)
Restores all network and web application settings to the factory defaults. Multicolor flash, then blue LED along bar.
Reboot Device (): Power-cycles the device as if it were unplugged from the network. All settings are retained when the device is rebooted.
Default Settings (): Restores audio settings back to factory settings (excluding device name, IP settings, and passwords).
Designer's Optimize workflow speeds up the process of connecting systems with at least 1 microphone and 1 audio processor. Optimize also creates mute control routes in rooms with MXA network mute buttons. When you select Optimize in a room, Designer does the following:
The settings are optimized for your particular combination of devices. You can customize settings further, but the Optimize workflow gives you a good starting point.
After optimizing a room, you should check and adjust settings to fit your needs. These steps may include:
Compatible devices:
To use the Optimize workflow:
If you remove or add devices, select Optimize again.
Designer allows administrators and technicians to control:
Coverage: Adjust lobe width and location, select templates, save or load presets, customize light bar settings, and run automatic setup.
Channels: Adjust and monitor channel levels, mute channels or channel groups, configure automix settings, and adjust equalizer settings.
Settings: Control network IP settings, device name, passwords, languages, firmware identification, and device reset.
Think of each lobe as an individual microphone. If there were 8 microphones on the table, each one could be physically moved according to seating arrangements, and would be plugged into a mixer with independent gain and channel controls. With the Microflex Advance Ceiling Array Microphone, Designer delivers control over the physical coverage and audio channel settings, with user presets to quickly switch between configurations. Each lobe is moved according to seating arrangements, with three width settings to change the size of the coverage area. Independent mixer channels control the level and audio properties for each lobe.
Each lobe is represented graphically and can be dragged into place. A corresponding mixer channel provides control over audio settings for each lobe.
To configure the MXA910, follow these steps:
Select the device and set the properties:
You'll see the lobes moving in Designer's coverage map as participants shift positions, which is the Autofocus technology in action. Autofocus fine-tunes each lobe's position in real time, even if meeting participants lean back or stand up.
Independent width control makes it possible for some channels to capture individual talkers (narrow), while others cover multiple talkers (wide).
To change a channel width:
Width Settings:
Channel widths for the three settings with the microphone 6 feet above a table
You can use Auto position to correctly position the lobe for a selected channel:
This microphone uses built-in Autofocus technology to fine-tune each lobe's position in real time, even if meeting participants lean back or stand up. You'll see the lobes moving in Designer's coverage map as participants shift positions. Autofocus only responds to in-room sound sources.
For best results with Autofocus, always route a reference source to the microphone's AEC Reference In channel. Even if you're only using direct outputs from the microphone and a different DSP, route a reference signal to the microphone's AEC Reference In channel to take full advantage of Autofocus.
Gain levels on MicroflexAdvance microphones must be set for each saved coverage preset to ensure an optimized gain structure for all seating scenarios. Always adjust the levels before making any changes to automix settings to ensure the best performance.
There are 2 different gain faders that serve different purposes:
Channel Gain (Pre-Gate)
To adjust, go to Channels. These faders affect a channel's gain before it reaches the automixer and therefore affect the automixer's gating decision. Boosting the gain here will make the lobe more sensitive to sound sources and more likely to gate on. Lowering gain here makes the lobe less sensitive and less likely to gate on. If you're only using direct outputs for each channel without the automixer, you only need to use these faders.
IntelliMix Gain (Post-Gate)
To adjust, go to . These faders adjust a channel's gain after the lobe has gated on. Adjusting the gain here will not affect the automixer's gating decision. Only use these faders to adjust the gain of a talker after you are satisfied with the automixer's gating behavior.
Maximize audio quality by adjusting the frequency response with the parametric equalizer.
Common equalizer applications:
To turn off all EQ filters, select Bypass all EQ.
If you’re using Shure Designer software to configure your system, please check the Designer help section for more about this topic.
Adjust filter settings by manipulating the icons in the frequency response graph, or by entering numeric values. Disable a filter using the check-box next to the filter.
Filter Type
Only the first and last band have selectable filter types.
Parametric: Attenuates or boosts the signal within a customizable frequency range
Low Cut: Rolls off the audio signal below the selected frequency
Low Shelf: Attenuates or boosts the audio signal below the selected frequency
High Cut: Rolls off the audio signal above the selected frequency
High Shelf: Attenuates or boosts the audio signal above the selected frequency
Frequency
Select the center frequency of the filter to cut/boost
Gain
Adjusts the level for a specific filter (+/- 30 dB)
Q
Adjusts the range of frequencies affected by the filter. As this value increases, the bandwidth becomes thinner.
Width
Adjusts the range of frequencies affected by the filter. The value is represented in octaves.
Note: The Q and width parameters affect the equalization curve in the same way. The only difference is the way the values are represented.
These features make it simple to use effective equalizer settings from a previous installation, or simply accelerate configuration time.
Use to quickly apply the same PEQ setting across multiple channels.
Use to save and load PEQ settings from a file on a computer. This is useful for creating a library of reusable configuration files on computers used for system installation.
Export
Choose a channel to save the PEQ setting, and select Export to file.
Import
Choose a channel to load the PEQ setting, and select Import from file.
Apply Automix EQ to make system-wide changes, such as a treble boost to improve speech clarity. Use Channel EQ to make adjustments to a specific channel. For example, to reduce unwanted noise picked up by only one channel.
Conferencing room acoustics vary based on room size, shape, and construction materials. Use the guidelines in following table.
| EQ Application | Suggested Settings |
|---|---|
| Treble boost for improved speech intelligibility | Add a high shelf filter to boost frequencies greater than 1 kHz by 3-6 dB |
| HVAC noise reduction | Add a low cut filter to attenuate frequencies below 200 Hz |
| Reduce flutter echoes and sibilance | Identify the specific frequency range that "excites" the room:
|
| Reduce hollow, resonant room sound | Identify the specific frequency range that "excites" the room:
|
Use the built-in equalizer contours to quickly apply EQ changes to all channels. EQ contours are separate from the per-channel EQ settings. Applying both EQ contours and per-channel EQ has a cumulative effect, meaning that the EQ changes stack on top of each other.
To enable a contour, open the web application and select a contour in the device options section.
Click Bypass all EQ to quickly bypass any EQ contours or channel EQ settings.
Use presets to quickly save and recall settings. Up to 10 presets can be stored on each device to match various seating arrangements. A preset saves all device settings except for the Device Name, IP Settings, and Passwords. Importing and exporting presets into new installations saves time and improves workflow. When a preset is selected, the name displays above the preset menu. If changes are made, an asterisk appears next to the name.
Note: Use the default settings preset to revert to the factory configuration (excludes Device Name, IP Settings, and Passwords).
Open the presets menu to reveal preset options:
| save as preset: | Saves settings to the device |
| load preset: | Opens a configuration from the device |
| import from file: | Downloads a preset file from a computer onto the device. Files may be selected through the browser or dragged into the import window. |
| export to file: | Saves a preset file from the device onto a computer |
Add channels to a Mute group or Fader group to link the corresponding controls together. For example, if channels 1, 2, and 3 are added to a Mute group, muting any of those individual channels will mute all of the grouped channels.
If you’re using Shure Designer software to configure your system, please check the Designer help section for more about this topic.
Mute sync ensures that all connected devices in a conferencing system mute or unmute at the same time and at the correct point in the signal path. Mute status is synchronized in the devices using logic signals or USB connections.
To use mute sync, make sure logic is enabled on all devices.
Designer's Optimize workflow configures all necessary mute sync settings for you.
Compatible Shure logic devices:
To use mute sync, route the microphone’s signal to a processor that has logic turned on (P300, ANIUSB-MATRIX, or IntelliMix Room software). Microphones always have logic turned on.
For help with specific mute sync implementations, see our FAQs.
Optimal microphone placement is determined by the seating arrangements and infrastructure. Follow these guidelines for the best possible results:
The maximum mounting height that can be set is 30 feet (9.14 meters). In a typical acoustic environment1, the microphone maintains an "A" rating based on the STIPA2 (Speech Transmission Index for Public Address systems) international standard at distances up to 16 feet between the microphone and talker. In better acoustic environments, the STIPA "A" rating may extend beyond 16 feet.
Consider the following when determining a mounting height:
[1] Room conditions: RT60 (reverb time) = 500 ms @ 1kHz, A weighted room noise = 40dBSPL(A)
[2] IEC-602682-16 standard
The intelligibility scale objectively compares the acoustic performance of the array microphone with a cardioid gooseneck microphone at various distances. This information is useful for predicting how the array microphone will perform at a given distance and to determine an ideal mounting height. The data in the intelligibility scale table is derived from measuring the microphones to meet an equivalent value from the Speech Transmission Index IEC-602682-16 standard.
| Ceiling Array Microphone (Distance to Talker) | Cardioid Gooseneck Microphone (Distance to Talker) |
|---|---|
| 6 ft (1.83 m) | 3.75 feet (1.14 m) |
| 8 ft (2.44 m) | 5 feet (1.52 m) |
| 10 ft (3.05 m) | 6.25 feet (1.91 m) |
| 12 ft (3.66 m) | 7.5 feet (2.29 m) |
Data was collected in a typical huddle room with the following measurements:
Note: These values are specific to the described room. In a well-controlled acoustic environment, the array microphone may perform with equivalent Speech Transmission Index values at even greater distances. In highly reverberant rooms, the performance is less predictable.
A = Distance between array microphone and talker
B = Distance between cardioid microphone and talker
In this example, the acoustic performance of the array microphone mounted (A) feet from the talker matches the cardioid gooseneck microphone placed at a distance of (B) feet from the talker.
The array microphone mounts directly in a ceiling grid, or can be attached using other methods.
Before you begin:
IMPORTANT: Do not install the 60 cm model in a 2 ft (609.6 mm) ceiling grid.
| Model | Ceiling Grid Size | Color |
|---|---|---|
| MXA910B | 2 x 2 ft (60.9 x 60.9 cm) | Black |
| MXA910W | 2 x 2 ft (60.9 x 60.9 cm) | White |
| MXA910AL | 2 x 2 ft (60.9 x 60.9 cm) | Aluminum |
| MXA910B-60CM | 60 x 60 cm (23.6 x 23.6 in) | Black |
| MXA910W-60CM | 60 x 60 cm (23.6 x 23.6 in) | White |
| MXA910AL-60CM | 60 x 60 cm (23.6 x 23.6 in) | Aluminum |
| MXA910W-A | 2 x 2 ft (60.9 x 60.9 cm) | White |
| MXA910W-US | 2 x 2 ft (60.9 x 60.9 cm) | White |
Optional: Before installing the microphone in the ceiling, attach the included rubber pads on the corners of the microphone to prevent scratching.
Note: An optional junction box accessory (model A910-JB) mounts on the microphone to directly connect conduit.
Optional: Before installing the microphone in the ceiling, attach the included rubber pads to the corners of the microphone flange to prevent scratching.
Note: An optional junction box accessory (model A910-JB) mounts on the microphone to directly connect conduit.
Note: Depending on the width of the ceiling grid T-bars, you may need to remove or adjust one side's T-bar to install the MXA910W-A.
Optional: Before installing the microphone in the ceiling, attach the included rubber pads to the corners of the microphone flange to prevent scratching.
Note: An optional junction box accessory (model A910-JB) mounts on the microphone to directly connect conduit.
① Wire Suspension Hanging Points (4 mm hole size)
② VESA Mounting Holes
Secure the microphone to the ceiling using braided metal cable or other high-strength wire. Use the 4 integrated hanging points on the back of the microphone to securely attach the cable. The hole size in the hanging points is 4 mm (0.15 in).
The rear plate on the microphone has 4 threaded holes for attaching the microphone to a VESA mounting device. The mounting holes follow the VESA MIS-D standard:
You can mount the microphone in hard ceilings without a tile grid using the A910-HCM accessory.
Learn more at www.shure.com.
The grille and frame of the array microphone can be painted to blend in with room design. Some basic disassembly is required for painting.
Important: Do not remove the screws that are farthest in the corner and recessed into the panel (see graphic).
Important: Do not paint the foam.
(1.1) Removing the screws to detach the main assembly
(1.4 - 1.5) Removing screws and detaching one side of the frame
(2.1) Masking off the extrusion (highlighted in black)
Note: The label on the assembly is placed on the corner that corresponds to the LED. Use it for reference to ensure proper orientation during reassembly.
To keep the Ethernet cable out of view, use the appropriate method based on the installation type.
| Installation | Cable Solution |
|---|---|
| Ceiling grid | Run the cable above the ceiling grid |
| VESA (pole mounting) | Guide the cable through the pole to run it above the ceiling grid |
| 4-point wire suspension | Use cable ties to attach the CAT5 cable along one of the hanging wires |
| Hard ceiling | Run the cable above the ceiling |
Note: If using conduit to contain the cable, the optional junction box accessory (model A910-JB) mounts directly to the rear panel of the microphone.
The A910-JB junction box mounts on the microphone, enabling conduit connections for cable runs. Refer to local building codes and regulations to determine if the junction box is necessary. There are three punch-out sections on the junction box for attaching conduit.
Important: Punch out the necessary holes in the junction box prior to installing it onto the microphone.
To install:
You can control this device using Shure Designer software. Designer enables integrators and system planners to design audio coverage for installations using MXA microphones and other Shure networked components.
With Designer, you can:
To access your device in Designer:
Learn more and download at www.shure.com/designer.
The Shure Web Server Discovery application finds all Shure devices on the network that feature a web-based GUI. Follow these steps to install the software and access the web application:
① Install the Shure Discovery application
Download and install the Shure Discovery application from www.shure.com. This automatically installs the required Bonjour device discovery tool on the computer.
② Connect the network
Ensure the computer and the hardware are on the same network.
③ Launch the Discovery application
The app displays all Shure devices that feature a GUI.
④ Identify the hardware
Double-click on a device to open its GUI in a web browser.
⑤ Bookmark the device's web application (recommended)
Bookmark the device's DNS name to access the GUI without the Shure Discovery app.
The web application is compatible with all HTML5-supported browsers. To ensure the best performance, disabling hardware acceleration and unused plug-ins is recommended.
If the Discovery application is not installed, the web application can be accessed by typing the DNS name into an internet browser. The DNS name is derived from model of the unit, in combination with the last three bytes (six digits) of the MAC address, and ending in .local.
Format Example: If the MAC address of a unit is 00:0E:DD:AA:BB:CC, then the link is written as follows:
MXA910: http://MXA910-aabbcc.local
Applies to Designer 4.2 and newer.
Before setting up devices, check for firmware updates using Designer to take advantage of new features and improvements. You can also install firmware using Shure Update Utility for most products.
To update:
When updating firmware, update all hardware to the same firmware version to ensure consistent operation.
The firmware of all devices has the form of MAJOR.MINOR.PATCH (e.g., 1.2.14). At a minimum, all devices on the network, must have the same MAJOR and MINOR firmware version numbers (e.g., 1.2.x).
When you update an MXA910 from 3.x to 4.x firmware, you'll be able to use IntelliMix DSP features optimized for MXA.
Here's what changes with 4.x firmware:
IntelliMix DSP Added
New Automixer Added
| 3.x Automixer Setting | New 4.x Automixer Setting |
|---|---|
| Classic | Gating |
| Smooth | Gain Sharing |
| Manual | Manual |
| Custom | Gating |
New Autofocus Feature Added for All Lobes
Echo Reduction Removed
Other Changes
Shure offers a range of connectivity options for conferencing. MXA microphones, audio processors, and network interfaces all use Dante to send audio over standard IT networks. You can use Shure's free Designer software to control most Shure devices and route audio between them.
| Device | Purpose | Physical Connections | Dante I/Os |
|---|---|---|---|
 |
Ceiling array microphone with IntelliMix DSP |
|
|
  |
Linear array microphone with IntelliMix DSP |
|
2 Foot:
|
 |
Table array microphone |
|
|
 |
Audio processor with IntelliMix DSP and matrix mixer |
|
|
 |
Audio processing software with IntelliMix DSP and matrix mixer | Varies depending on device |
|
 |
Matrix mixer with USB and analog input/output |
|
|
 |
Converts analog signals to Dante signals |
|
|
|
|
Converts Dante signals to analog signals |
|
|
 |
|
|
|
 |
Networked ceiling loudspeaker powered by PoE |
|
|
 |
PoE-powered network mute button for Shure devices |
|
n/a |
| MXA910 and ANI22 | MXA910 and ANIUSB | MXA910 and P300 | |
|---|---|---|---|
| Room size | Small or medium | Small or medium | Medium or large |
| Mobile I/O | No | No | Yes |
| Analog I/O | 2x2 | 1x1 | 2x2 |
| USB I/O | No | Yes | Yes |
| Power | PoE | PoE | PoE+ |
| Logic control | No | No | Yes |
| Soft codec mute sync | No | Yes | Yes with P300 firmware 3.1.5 or later |
| Accommodates multiple MXA910s | No | No | Yes |
| Accommodates additional analog mics | No | No | Yes |
For more analog inputs, use ANI4INs to convert analog signals to Dante signals. For more analog outputs, use ANI4OUTs to convert Dante signals to analog signals.
For an easy soft codec solution, use the ceiling array with an ANIUSB-MATRIX or a P300.
Equipment List:
Equipment List:
In larger installations, you can use multiple MXA910s and a P300 for a distributed DSP approach that makes installation simpler. For best results, use a maximum of 3 MXA910s.
Equipment List:
This channel automatically mixes the audio from all selected channels to deliver a convenient, single output. To adjust the automix channel settings, select the IntelliMix tab. All IntelliMix DSP blocks can be applied to the automix channel.
To use the automix channel, do the following:
Gating mode delivers fast-acting, seamless channel gating and consistent perceived ambient sound levels. The off attenuation setting is applied to all inactive channels, regardless of the number of active channels.
Gain sharing mode dynamically balances system gain between open and closed channels. The system gain remains consistent by distributing gain across channels to equal one open channel. The scaled gain structure helps to reduce noise when there is a high channel count. When fewer channels are used, the off attenuation setting is lower and provides transparent gating.
Manual mode sums all active tracks and sends the summed signal over a single Dante output. This provides the option to route an individual signal for reinforcement or recording, without enabling automixing. The settings from the faders in the standard monitoring view apply to the summed output.
Leave Last Mic On
Keeps the most recently used microphone channel active. The purpose of this feature is to keep natural room sound in the signal so that meeting participants on the far end know the audio signal has not been interrupted.
Gating Sensitivity
Changes the threshold of the level at which the gate is opened
Off Attenuation
Sets the level of signal reduction when a channel is not active
Hold Time
Sets the duration for which the channel remains open after the level drops below the gate threshold
Maximum Open Channels
Sets the maximum number of simultaneously active channels
Priority
When selected, this channel gate activates regardless of the number of maximum open channels.
Always On
When selected, this channel will always be active.
Send to Mix
When selected, sends the channel to the automix channel.
Solo
Mutes all of the other channels
Automix Gain Meter
When enabled, changes gain meters to display automix gating in real time. Channels that gate open will display more gain than channels that are closed (attenuated) in the mix.
Classic mode emulates the Shure SCM820 automixer (in its default settings). It is renowned for fast-acting, seamless channel gating and consistent perceived ambient sound levels. Off-attenuation in this mode is fixed at -20 dB per channel, regardless of the number of open channels.
In Smooth mode, Off-attenuation settings for each channel are scaled, depending on the number of open channels. The scaled gain structure helps to reduce noise when there is a high channel count. When fewer channels are used, the lower off-attenuation provides transparent gating.
| Number of channels enabled | Off-attenuation (dB) |
| 2 | -5 |
| 3 | -8 |
| 4 | -10 |
| 5 | -12 |
| 6 | -13 |
| 7 | -14 |
| 8 | -15 |
Custom mode provides control over all automixing parameters. This mode is useful when adjustments must be made to one of the preset modes to fit a particular application. If parameters are changed in smooth or classic mode, custom mode automatically activates.
Manual mode sums all active tracks and sends the summed signal over a single Dante output. This provides the option to route the signal for reinforcement or recording, without enabling automixing. The settings from the faders in the standard monitoring view apply to the summed output.
This device contains IntelliMix digital signal processing blocks that can be applied to the automix channel output. The DSP blocks include:
To access the DSP blocks, select the IntelliMix tab. When enabled, each DSP block will be colored.
Selecting Bypass IntelliMix will bypass the following DSP blocks: AEC, AGC, noise reduction, compressor, and delay.
In audio conferencing, a far-end talker may hear their voice echo as a result of a near-end microphone capturing audio from loudspeakers. Acoustic echo cancellation (AEC) is a DSP algorithm which identifies the far-end signal and stops it from being captured by the microphone to deliver clear, uninterrupted speech. During a conference call, the AEC works constantly to optimize processing as long as far-end audio is present.
When possible, optimize the acoustic environment using the following tips:
To apply AEC, provide a far end reference signal. For best results, use the signal that also feeds your local reinforcement system.
P300: Go to Schematic and click any AEC block. Choose the reference source, and the reference source changes for all AEC blocks.
MXA910, MXA920, MXA710: Route a far-end signal to the AEC Reference In channel.
IntelliMix Room: Go to Schematic and click an AEC block. Choose the reference source. Each block can use a different reference source, so set the reference for each AEC block.
Designer's Optimize workflow automatically routes an AEC reference source, but it's a good idea to check that Designer chooses the reference source you want to use.
Reference Meter
Use the reference meter to visually verify the reference signal is present. The reference signal should not be clipping.
ERLE
Echo return loss enhancement (ERLE) displays the dB level of signal reduction (the amount of echo being removed). If the reference source is connected properly, the ERLE meter activity generally corresponds to the reference meter.
Reference
Indicates which channel is serving as the far end reference signal.
Non-Linear Processing
The primary component of the acoustic echo canceller is an adaptive filter. Non-linear processing supplements the adaptive filter to remove any residual echo caused by acoustic irregularities or changes in the environment. Use the lowest possible setting that is effective in your room.
Low: Use in rooms with controlled acoustics and minimal echoes. This setting provides the most natural sound for full duplex.
Medium: Use in typical rooms as a starting point. If you hear echo artifacts, try using the high setting.
High: Use to provide the strongest echo reduction in rooms with bad acoustics, or in situations where the echo path frequently changes.
Noise reduction significantly reduces the amount of background noise in your signal caused by projectors, HVAC systems, or other environmental sources. It is a dynamic processor, which calculates the noise floor in the room and removes noise throughout the entire spectrum with maximum transparency.
The noise reduction setting (low, medium, or high) represents the amount of reduction in dB. Use the lowest possible setting that effectively lowers noise in the room.
Automatic gain control automatically adjusts channel levels to ensure consistent volume for all talkers, in all scenarios. For quieter voices, it increases gain; for louder voices, it attenuates the signal.
Enable AGC on channels where the distance between the talker and the microphone may vary, or in rooms where many different people will use the conferencing system.
Automatic gain control happens post-gate (after the automixer), and will not affect when the automixer gates on or off.
Target Level (dBFS)
Use -37 dBFS as a starting point to ensure adequate headroom, and adjust if necessary. This represents the RMS (average) level, which is different from setting the input fader according to peak levels to avoid clipping.
Maximum Boost ( dB)
Sets the maximum amount of gain that can be applied
Maximum Cut ( dB)
Sets the maximum attenuation that can be applied
Tip: Use the boost/cut meter to monitor the amount of gain added or subtracted from the signal. If this meter is always reaching the maximum boost or cut level, consider adjusting the input fader so the signal is closer to the target level.
Use delay to synchronize audio and video. When a video system introduces latency (where you hear someone speak, and their mouth moves later), add delay to align audio and video.
Delay is measured in milliseconds. If there is a significant difference between audio and video, start by using larger intervals of delay time (500-1000 ms). When the audio and video are slightly out of sync, use smaller intervals to fine-tune.
Use the compressor to control the dynamic range of the selected signal.
Threshold
When the audio signal exceeds the threshold value, the level is attenuated to prevent unwanted spikes in the output signal. The amount of attenuation is determined by the ratio value. Perform a soundcheck and set the threshold 3-6 dB above average talker levels, so the compressor only attenuates unexpected loud sounds.
Ratio
The ratio controls how much the signal is attenuated when it exceeds the threshold value. Higher ratios provide stronger attenuation. A lower ratio of 2:1 means that for every 2 dB the signal exceeds the threshold, the output signal will only exceed the threshold by 1 dB. A higher ratio of 10:1 means a loud sound that exceeds the threshold by 10 dB will only exceed the threshold by 1 dB, effectively reducing the signal by 9 dB.
Audio is encrypted with the Advanced Encryption Standard (AES-256), as specified by the US Government National Institute of Standards and Technology (NIST) publication FIPS-197. Shure devices that support encryption require a passphrase to make a connection. Encryption is not supported with third-party devices.
In Designer, you can only enable encryption for all devices in a room in live mode: .
To activate encryption in the web application:
Important: For encryption to work:
AES67 is a networked audio standard that enables communication between hardware components which use different IP audio technologies. This Shure device supports AES67 for increased compatibility within networked systems for live sound, integrated installations, and broadcast applications.
The following information is critical when transmitting or receiving AES67 signals:
| Shure Device Supports: | Device 2 Supports: | AES67 Compatibility |
|---|---|---|
| Dante and AES67 | Dante and AES67 | No. Must use Dante. |
| Dante and AES67 | AES67 without Dante. Any other audio networking protocol is acceptable. | Yes |
Separate Dante and AES67 flows can operate simultaneously. The total number of flows is determined by the maximum flow limit of the device.
All AES67 configuration is managed in Dante Controller software. For more information, refer to the Dante Controller user guide.
Third-party devices: When the hardware supports SAP, flows are identified in the routing software that the device uses. Otherwise, to receive an AES67 flow, the AES67 session ID and IP address are required.
Shure devices: The transmitting device must support SAP. In Dante Controller, a transmit device (appears as an IP address) can be routed like any other Dante device.
Dante digital audio is carried over standard Ethernet and operates using standard internet protocols. Dante provides low latency, tight clock synchronization, and high Quality-of-Service (QoS) to provide reliable audio transport to a variety of Dante devices. Dante audio can coexist safely on the same network as IT and control data, or can be configured to use a dedicated network.
This device is compatible with Dante Domain Manager software (DDM). DDM is network management software with user authentication, role-based security, and auditing features for Dante networks and Dante-enabled products.
Considerations for Shure devices controlled by DDM:
See Dante Domain Manager's documentation for more information.
Dante flows get created any time you route audio from one Dante device to another. One Dante flow can contain up to 4 audio channels. For example: sending all 5 available channels from an MXA310 to another device uses 2 Dante flows, because 1 flow can contain up to 4 channels.
Every Dante device has a specific number of transmit flows and receive flows. The number of flows is determined by Dante platform capabilities.
Unicast and multicast transmission settings also affect the number of Dante flows a device can send or receive. Using multicast transmission can help overcome unicast flow limitations.
Shure devices use different Dante platforms:
| Dante Platform | Shure Devices Using Platform | Unicast Transmit Flow Limit | Unicast Receive Flow Limit |
|---|---|---|---|
| Brooklyn II | ULX-D, SCM820, MXWAPT, MXWANI, P300, MXCWAPT | 32 | 32 |
| Brooklyn II (without SRAM) | MXA920, MXA910, MXA710, AD4 | 16 | 16 |
| Ultimo/UltimoX | MXA310, ANI4IN, ANI4OUT, ANIUSB-MATRIX, ANI22, MXN5-C | 2 | 2 |
| DAL | IntelliMix Room | 16 | 16 |
QoS settings assign priorities to specific data packets on the network, ensuring reliable audio delivery on larger networks with heavy traffic. This feature is available on most managed network switches. Although not required, assigning QoS settings is recommended.
Note: Coordinate changes with the network administrator to avoid disrupting service.
To assign QoS values, open the switch interface and use the following table to assign Dante®-associated queue values.
| Priority | Usage | DSCP Label | Hex | Decimal | Binary |
|---|---|---|---|---|---|
| High (4) | Time-critical PTP events | CS7 | 0x38 | 56 | 111000 |
| Medium (3) | Audio, PTP | EF | 0x2E | 46 | 101110 |
| Low (2) | (reserved) | CS1 | 0x08 | 8 | 001000 |
| None (1) | Other traffic | BestEffort | 0x00 | 0 | 000000 |
Note: Switch management may vary by manufacturer and switch type. Consult the manufacturer's product guide for specific configuration details.
For more information on Dante requirements and networking, visit www.audinate.com.
PTP (Precision Time Protocol): Used to synchronize clocks on the network
DSCP (Differentiated Services Code Point): Standardized identification method for data used in layer 3 QoS prioritization
Switches and cables determine how well your audio network performs. Use high-quality switches and cables to make your audio network more reliable.
Network switches should have:
Ethernet cables should be:
For more information, see our FAQ about switches to avoid.
Use the following best practices when setting up a network to ensure reliable communication:
This Shure device uses 2 IP addresses: one for Shure control, and one for Dante audio and control.
To access these settings in Designer, go to .
Configure IP
Sets IP mode of the selected network interface:
IP Settings
View and edit the IP Address, Subnet Mask, and Gateway for each network interface.
MAC Address
The network interface's unique identification.
IP configurations are managed in Shure Designer software. By default, they are set to Automatic (DHCP) mode. DHCP mode enables the devices to accept IP settings from a DHCP server, or automatically fall back to Link-Local settings when no DHCP is available. IP addresses may also be manually set.
To configure the IP properties, follow these steps:
To manually assign IP addresses, follow these steps:
Latency is the amount of time for a signal to travel across the system to the outputs of a device. To account for variances in latency time between devices and channels, Dante has a predetermined selection of latency settings. When the same setting is selected, it ensures that all Dante devices on the network are in sync.
These latency values should be used as a starting point. To determine the exact latency to use for your setup, deploy the setup, send Dante audio between your devices, and measure the actual latency in your system using Audinate's Dante Controller software. Then round up to the nearest latency setting available, and use that one.
Use Audinate's Dante Controller software to change latency settings.
| Latency Setting | Maximum Number of Switches |
|---|---|
| 0.25 ms | 3 |
| 0.5 ms (default) | 5 |
| 1 ms | 10 |
| 2 ms | 10+ |
Packet bridge enables an external controller to obtain IP information from the control interface of a Shure device. To access the packet bridge, an external controller must send a query packet over unicast UDP* to port 2203 on the Dante interface of the Shure device.
Note: The maximum accepted payload 140 bytes. Any content is allowed.
| Bytes | Content |
|---|---|
| 0-3 | IP address, as 32-bit unsigned integer in network order |
| 4-7 | Subnet mask, as 32-bit unsigned integer in network order |
| 8-13 | MAC address, as array of 6 bytes |
Note: The Shure device should respond in less than one second on a typical network. If there is no response, try sending the query again after verifying the destination IP address and port number.
*UDP: User Datagram Protocol
The packet bridge does not allow cross-subnet command strings.
| Port | TCP/UDP | Protocol | Description | Factory Default |
|---|---|---|---|---|
| 21 | TCP | FTP | Required for firmware updates (otherwise closed) | Closed |
| 22 | TCP | SSH | Secure Shell Interface | Closed |
| 23 | TCP | Telnet | Not supported | Closed |
| 53 | UDP | DNS | Domain Name System | Closed |
| 67 | UDP | DHCP | Dynamic Host Configuration Protocol | Open |
| 68 | UDP | DHCP | Dynamic Host Configuration Protocol | Open |
| 80* | TCP | HTTP | Required to launch embedded web server | Open |
| 443 | TCP | HTTPS | Not supported | Closed |
| 2202 | TCP | ASCII | Required for 3rd party control strings | Open |
| 5353 | UDP | mDNS† | Required for device discovery | Open |
| 5568 | UDP | SDT (multicast)† | Required for inter-device communication | Open |
| 57383 | UDP | SDT (unicast) | Required for inter-device communication | Open |
| 8023 | TCP | Telnet | Debug console interface | Closed |
| 8180 | TCP | HTML | Required for web application (legacy firmware only) | Open |
| 8427 | UDP | SLP (multicast)† | Required for inter-device communication | Open |
| 64000 | TCP | Telnet | Required for Shure firmware update | Open |
*These ports must be open on the PC or control system to access the device through a firewall.
†These protocols require multicast. Ensure multicast has been correctly configured for your network.
See Audinate's website for information about ports and protocols used by Dante audio.
In audio conferencing, a talker may hear their voice echo as a result of the microphone capturing far-end audio from loudspeakers.
The echo reduction feature prevents the far-end signal from activating the microphone. Ideal for installations in which per-channel DSP echo cancellation is not within a project budget, echo reduction is highly effective for connecting directly to a computer or video codec which hosts a single-channel echo canceller.
An echo reference signal from the far end is routed through Dante Controller software to the microphone's processing algorithm. The processor uses this signal to prevent the microphone from gating on and capturing audio from the loudspeakers.
This device receives logic commands over the network. Many parameters controlled through Designer can be controlled using a third-party control system, using the appropriate command string.
Common applications:
A complete list of command strings is available at:
| Problem | Solution |
|---|---|
| Software lags in Google Chrome browser | Problem is browser-related. Turn off hardware acceleration option in Chrome. |
| Sound quality is muffled or hollow |
|
| Microphone does not show up in device discovery |
|
| Audio is not present or is quiet/distorted |
|
| No lights | Check if brightness is disabled or if any Light Bar settings are turned off. |
| Auto-positioning identifies incorrect location | If talker is in a corner or very close to a wall, acoustic reflections may interfere with localization accuracy. Try automatic positioning again, and if the issue persists, manual positioning may be necessary. |
| Microphone does not power on |
|
The equipment is intended to be used in professional audio applications.
Note: This device is not intended to be connected directly to a public internet network.
EMC conformance to Environment E2: Commercial and Light Industrial. Testing is based on the use of supplied and recommended cable types. The use of other than shielded (screened) cable types may degrade EMC performance.
Changes or modifications not expressly approved by Shure Incorporated could void your authority to operate this equipment.
Industry Canada ICES-003 Compliance Label: CAN ICES-3 (B)/NMB-3(B)
Authorized under the verification provision of FCC Part 15B.
Please follow your regional recycling scheme for batteries, packaging, and electronic waste.
This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the manufacturer's instruction manual, may cause interference with radio and television reception.
Notice: The FCC regulations provide that changes or modifications not expressly approved by Shure Incorporated could void your authority to operate this equipment.
These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:
This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:
The CE Declaration of Conformity can be obtained from: www.shure.com/europe/compliance
Authorized European representative:
Shure Europe GmbH
Global Compliance
Jakob-Dieffenbacher-Str. 12
75031 Eppingen, Germany
Phone: +49-7262-92 49 0
Email: info@shure.de
www.shure.com
This product meets the Essential Requirements of all relevant European directives and is eligible for CE marking.
The CE Declaration of Conformity can be obtained from Shure Incorporated or any of its European representatives. For contact information please visit www.shure.com
All specifications measured from narrow lobe width. Values for all widths are within ± 3 dB of these specifications unless otherwise noted.
| Adjustable | Narrow | 35 degrees |
| Medium | 45 degrees | |
| Wide | 55 degrees |
RJ45
Power over Ethernet (PoE), Class 0
9W, maximum
5.3 kg (11.75 lbs)
| MXA910xx | 603.8 x 603.8 x 54.69 mm (23.77 x 23.77 x 2.15 in.) H x W x D |
| MXA910xx-60CM | 593.8 x 593.8 x 54.69 mm (23.38 x 23.38 x 2.15 in.) H x W x D |
| MXA910W-A | 603.8 x 603.8 x 54.69 mm (23.77 x 23.77 x 2.15 in.) H x W x D |
| MXA910W-US | 603.8 x 603.8 x 54.69 mm (23.77 x 23.77 x 2.15 in.) H x W x D |
HTML5 Browser-based or Shure Designer
UL 2043 (Suitable for Air Handling Spaces)
IEC 60529 IP5X Dust Protected
−6.7°C (20°F) to 40°C (104°F)
−29°C (-20°F) to 74°C (165°F)
180 to 17,000 Hz
| Channel Count | 10 total channels ( 8 independent transmit channels, 1 Automatic mixing transmit channel, 1 AEC reference in channel) |
| Sampling Rate | 48 kHz |
| Bit Depth | 24 |
at 1 kHz
0.75 dBFS/Pa
Relative to 0 dBFS overload
93.25 dB SPL
Ref. 94 dB SPL at 1 kHz
83 dB A-weighted
| Direct outputs | 7 ms |
| Automix output (Includes IntelliMix processing) | 18 ms |
11 dB SPL-A
82.25 dB
| MXA910 firmware 4.x or newer | Automatic mixing, Acoustic Echo Cancellation (AEC), Noise Reduction, Automatic Gain Control, Compressor, Delay, Equalizer (4-band Parametric) , Mute, Gain ( 140 dB range) |
| MXA910 firmware 3.x or older | Automatic mixing, Echo Reduction, Equalizer (4-band Parametric) , Mute, Gain ( 140 dB range) |
Equivalent acoustic performance, compared to a cardioid gooseneck microphone (environment dependent)
Cardioid distance multiplied by 1.6
Up to 250 ms
Cat 5e or higher (shielded cable recommended)
Polar response measured directly on-axis from a distance of 6 feet (1.83 m).
Frequency response measured directly on-axis from a distance of 6 feet (1.83 m).
The edge of the blue coverage area for each channel in the web application represents where the sensitivity reaches -6 dB. Understanding how lobe sensitivity is displayed helps to:
Measured at 1 kHz, on-axis
| Junction box accessory | A910-JB |
| Hard ceiling mount | A910-HCM |
| MXA910W-60CM frame and grille assembly | RPM904 |
| MXA910AL-60CM frame and grille assembly | RPM905 |
| MXA910B-60CM frame and grille assembly | RPM906 |
| MXA910W-US frame and grille assembly | RPM901W-US |
| Rubber pad set | 95A28365 |
| Cable management clip | 95A29877 |
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