To control MXA710 microphones, use Shure Designer software. After completing this basic setup process, you should be able to:
You will need:
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 MXA710 includes IntelliMix® DSP that can be applied to the automix channel output.
For this example, we'll connect an MXA710 and an ANIUSB-MATRIX.
Designer prompts you to choose an installation method for the MXA710. You can change this setting later in Coverage map.
You can also route audio manually in Designer outside of the Optimize workflow, or use Dante Controller.
These templates are designed and tested to fit most common installations, but you can adjust lobe position and width as needed.
After you have coverage set up, you can send audio from the ANIUSB-MATRIX to other Dante devices or analog sources.
The Shure Microflex®Advance™ MXA710 Linear Array Microphone represents the next evolution in Shure array microphone technology, designed for high-quality audio capture in premium AV conferencing environments. The linear form factor of the MXA710 allows for placement virtually anywhere in a meeting space, including on a wall, around a display, on a ceiling, or in a conference room table. Available in 2- and 4-foot lengths in 3 colors, the MXA710 includes proprietary IntelliMix DSP and Autofocus™ technology that provides all the processing needed for echo- and noise-free audio.
Customize LED color and behavior in Designer by going to:.
|Microphone Status||LED Color/Behavior|
|Hardware identification||Green (flashing)|
|Firmware update in progress||Green (progresses along bar)|
|Error||Red (split, alternate flashing)|
|Device power-up||Multi-color flash, then blue (moves quickly back and forth across bar)|
Note: If LEDs are disabled, they will still turn on when the device powers up or when an error state occurs.
Sits behind the microphone grille. To access, find a grille hole that aligns with the left edge of the mute status LED and the "S" of the Shure logo. Use a small paperclip or other tool to press and hold the button. You may need to try a few different holes to press the reset button.
RJ-45 jack for network connection. Power over Ethernet is required to power the microphone.
Use to attach the microphone to the wall-mounting bracket.
Use to attach eyelet screws to hold braided metal cable or other high-strength wire for suspension mounting.
Route the Ethernet cable here to keep it flush with the microphone.
Use to attach the desk stand, the microphone stand mount, or other VESA MIS-B-compatible adapters.
|MXA710B-2FT||Black 2-foot microphone (60 cm)|
|MXA710W-2FT||White 2-foot microphone (60 cm)|
|MXA710AL-2FT||Aluminum 2-foot microphone (60 cm)|
|MXA710B-4FT||Black 4-foot microphone (120 cm)|
|MXA710W-4FT||White 4-foot microphone (120 cm)|
|MXA710AL-4FT||Aluminum 4-foot microphone (120 cm)|
This device requires PoE to operate. It is compatible with Class 0 PoE sources.
Power over Ethernet is delivered in one of the following ways:
Always use Cat5E cable or higher.
|2-foot or 4-foot linear array microphone||MXA710-2FT or MXA710-4FT|
|Wall-mounting bracket (2 or 4-foot)||RPM710-2M or RPM710-4M|
|Hardware kit with:
The reset button is behind the grille and can be pushed with a small paperclip or other tool. To access the button:
The MXA710 is an extremely versatile microphone. You can install it in many places in a conference room and easily get good coverage for all talkers.
|Room size||Small to medium||Medium to large|
|Maximum number of lobes||4||8|
|Recommended distance from talkers||2 to 16 feet||4 to 20 feet|
Best Practices for Installation
|Accessory||Install location||Other hardware required?|
|Wall-mounting bracket||Wall||Drywall mounting screws and anchors|
|Display mount kit||Attach to display mount||Peerless Universal Sound Bar Kit, Chief Thinstall Center Channel Speaker Adapter, or other similar adapter with VESA MIS-B compatibility|
|A710-TB Tile Bridge||Drop ceiling tile||A710-TB Tile Bridge|
|A710-FM Flush Mount||Table, wall, or hard ceiling||No|
|A710-MSA Mic Stand Accessory||Mic stand||Mic stand|
|A710-DS Desk Stand||Credenza or other flat surface||No|
To get started, you will need:
* Not included
Shure also sells the A710-TB tile bridge, which attaches to the microphone's screw holes like the suspension cable in step 1 above. Use the hardware included with the tile bridge to attach to the microphone.
You can mount the bracket directly over a junction box, or at any other cable exit on the wall. The bracket works positioned vertically or horizontally.
To get started, you will need:
Other mounting options:
Shure also sells the A710-FM flush mount kit, which attaches to the microphone's mounting keyholes like the wall bracket in step 7 above.
The 4 screw holes (for M4 x 10 mm screws) on the bottom of the microphone are compatible with VESA MIS-B mounting products, such as the Peerless Universal Sound Bar Kit or the Chief Thinstall Center Channel Speaker Adapter.
In certain installations, covering the microphone or the mounting hardware with fabric may be desirable. Shure has tested the acoustic performance of this microphone with some fabrics from Guilford of Maine and Kvadrat.
In our tests, there was little effect on the microphone's acoustic performance if the fabric met one of the following specifications. To cover the microphone, the fabric should meet at least one of these specifications:
These are examples of fabrics that met our specifications at the time when Shure evaluated fabrics: Guilford of Maine's BeeHave, and Kvadrat's Ginger, Mi Casa, Casita, and Time.
For best results:
To control this device's settings, use Shure Designer software. Designer enables integrators and system planners to design audio coverage for installations using MXA microphones and other Shure networked devices.
To access your device in Designer:
Learn more at shure.com/designer.
You can also access basic device settings using Shure Web Device Discovery. Full control is available in Designer.
Applies to version 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.
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).
To control microphone coverage, use Designer. Microphone coverage is at the location level, meaning that there is one coverage map for all microphones in a location.
These coverage templates are designed and tested to fit most common installations.
The solid blue line in each lobe represents where the coverage is the strongest. The edge of the blue coverage area for each lobe represents where the lobe's sensitivity reaches -6 dB.
Autofocus technology fine-tunes each lobe position in real time, even if meeting participants lean back or stand up.
Use these images to understand how the coverage patterns work in different installations. Always listen to lobes as you move them into position. Have someone talk from each lobe position to make sure that you have good coverage.
1 lobe. Go to Open side view to adjust the vertical angle.
3 lobes. Some are bidirectional in certain positions.
3 lobes. Some are bidirectional in certain positions.
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.
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 locations with MXA network mute buttons. When you select Optimize in a location, 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 location, you should check and adjust settings to fit your needs. These steps may include:
To use the Optimize workflow:
If you remove or add devices, select Optimize again.
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: 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.
Use the reference meter to visually verify the reference signal is present. The reference signal should not be clipping.
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.
Indicates which channel is serving as the far end reference signal.
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 noise in the signal caused by projectors, HVAC systems, or other environmental noise. 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.
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.
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.
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.
Changes the threshold of the level at which the gate is opened
Sets the level of signal reduction when a channel is not active
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
When selected, this channel gate activates regardless of the number of maximum open channels.
When selected, this channel will always be active.
Send to Mix
When selected, sends the channel to the automix channel.
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.
Gating mode delivers 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.
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 lower off-attenuation 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.
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:
Maximize audio quality by adjusting the frequency response with the parametric equalizer.
Common equalizer applications:
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.|
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 contour to quickly apply EQ changes to all channels. The EQ contour is separate from the per-channel EQ settings. Applying both the EQ contour and per-channel EQ has a cumulative effect, meaning that the EQ changes stack on top of each other.
To use, open the microphone in Designer and click EQ contour to turn it on or off.
Click Bypass all EQ to quickly bypass any EQ contours or channel EQ settings.
Mute synchronization 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 synchronization, enable logic on connected devices using the web application or Shure Designer software. Many Shure devices have logic enabled automatically.
If you use Designer's Optimize workflow, Designer configures all of the necessary mute synchronization settings for you.
Shure logic devices:
When connecting Shure devices to a network, use the following best practices:
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:
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|
|0.5 ms (default)||5|
MicroflexAdvance devices transport two types of data over the network: Shure Control and Network Audio.
The Shure Control carries data for the control software operation, firmware updates and 3rd party control systems (AMX, Crestron).
This network carries both the Dante digital audio and the control data for Dante Controller. The network audio requires a wired, gigabit Ethernet connection to operate.
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.
|High (4)||Time-critical PTP events||CS7||0x38||56||111000|
|Medium (3)||Audio, PTP||EF||0x2E||46||101110|
|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
Dantetm 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)||MXA910, MXA710, AD4||16||16|
|Ultimo/UltimoX||MXA310, ANI4IN, ANI4OUT, ANIUSB-MATRIX, ANI22, MXN5-C||2||2|
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.
|21||tcp||FTP||Required for firmware updates (otherwise closed)||Closed|
|22||tcp||SSH||Secure Shell Interface||Closed|
|68||udp||DHCP||Dynamic Host Configuration Protocol||Open|
|80*||tcp||HTTP||Required to launch embedded web server||Open|
|2202||tcp||ASCII||Required for 3rd party control strings||Open|
|5353||udp||mDNS†||Required for device discovery||Open|
|5568||udp||SDT†||Required for inter-device communication||Open|
|8023||tcp||Telnet||Debug console interface||Closed|
|8180||tcp||HTML||Required for web application||Open|
|8427||udp||Multcast SLP†||Required for inter-device communication||Open|
|64000||tcp||Telnet||Required for Shure firmware update||Open|
|162||udp||SNMP||Used by Dante|
|2203||udp||Custom||Required for packet bridge|
|4321, 14336-14600||udp||Dante||Dante audio|
|[4440, 4444, 4455]*||udp||Dante||Dante audio routing|
|5353||udp||mDNS†||Used by Dante|
|[8700-8706, 8800]*||udp||Dante||Dante Control and Monitoring|
|16000-65536||udp||Dante||Used by Dante|
*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.
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.
A complete list of command strings is available at:
All specifications measured from narrow lobe width. Values for all widths are within ± 3 dB of these specifications unless otherwise noted.
Power over Ethernet (PoE), Class 0
10 W maximum
|MXA710-2FT||2 lbs (0.91 kg)|
|MXA710-4FT||3.7 lbs (1.67 kg)|
|MXA710-2FT||0.87 x 2.36 x 25.04 in. (22.09 x 60 x 636 mm) H x W x L|
|MXA710-4FT||0.87 x 2.36 x 49.12 in. (22.09 x 60 x 1247.76 mm) H x W x L|
UL2043 (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)
100 Hz to 20 kHz
|Channel Count||MXA710-2FT||6 total channels (4 independent transmit channels, 1 Automix output, 1 AEC reference in channel)|
|MXA710-4FT||10 total channels (8 independent transmit channels, 1 Automix output, 1 AEC reference in channel)|
|Sampling Rate||48 kHz|
at 1 kHz
Relative to 0 dBFS overload
|MXA710-2FT||101.4 dB SPL|
|MXA710-4FT||101.9 dB SPL|
Ref. 94 dB SPL at 1 kHz
71.2 dB A-weighted
Does not include Dante latency
|Direct Outputs||8.7 ms|
|Automix output (Includes IntelliMix processing)||19.3 ms|
|MXA710-2FT||22.9 dB SPL-A|
|MXA710-4FT||22.8 dB SPL-A|
Automatic mixing, Acoustic Echo Cancellation (AEC), Noise Reduction, Automatic Gain Control, Compressor, Delay, Equalizer (4-band Parametric), Mute, Gain (140 dB range)
Up to 250 ms
Cat 5e or higher (shielded cable recommended)
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 represents where the sensitivity reaches -6 dB. Understanding how lobe sensitivity is displayed helps to:
Measured at 1 kHz, on-axis
Didn't find what you need? Contact our customer support to get help.
|This symbol indicates that dangerous voltage constituting a risk of electric shock is present within this unit.|
|This symbol indicates that there are important operating and maintenance instructions in the literature accompanying this unit.|
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 device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:
Note: 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. 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:
The CE Declaration of Conformity can be obtained from: www.shure.com/europe/compliance
Authorized European representative:
Shure Europe GmbH
Headquarters Europe, Middle East & Africa
Department: EMEA Approval
75031 Eppingen, Germany
Phone: +49-7262-92 49 0
Fax: +49-7262-92 49 11 4
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