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 control for connected devices.
For this example, we'll connect an MXA920 and a P300.
The default setting is a 30 by 30 foot (9 by 9 meter) dynamic coverage area. Any talker inside has coverage, and anything outside that area won't be picked up.
To add more coverage areas:
You can also turn off automatic coverage in Settings to manually position up to 8 lobes.
After completing this basic setup process, you should be able to:
You will need:
The default setting is a 30 by 30 foot (9 by 9 meter) dynamic coverage area. Any talker inside has coverage, and anything outside that area won't be picked up.
To add more coverage areas:
You can also turn off automatic coverage in Settings to manually position up to 8 lobes.
To route audio to other Dante devices, use Dante Controller.
Customize LED color and behavior in Designer: .
| Microphone Status | LED Color/Behavior |
|---|---|
| Active | Green (solid) |
| Muted | Red (solid) |
| Hardware identification | Green (flashing) |
| Firmware update in progress | Green (progresses along bar) |
| Reset |
|
| 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.
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:
| SKU | Description |
|---|---|
| MXA920W-S | White square microphone |
| MXA920W-S-60CM | White square microphone (60 cm) |
| MXA920AL-R | Aluminum round microphone |
| MXA920B-R | Black round microphone |
| MXA920W-R | White round microphone |
Find MXA920 audio codec certifications at shure.com/mxa920.
| Square or round array microphone | MXA920-S or MXA920-R |
| Square or round hardware kit Square:
Round:
|
Square: 90A49117 Round: 90A49116 |
The reset button is behind the grille. To push it, use a paper clip or other tool.
Button locations:
There are 2 ways to control the MXA920:
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 device settings using Shure Web Device Discovery.
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:
To access coverage map settings:
To control automatic coverage, go to .
For most rooms, Shure recommends:
These numbers also depend on your room's acoustics, construction, and materials. With automatic coverage on, the default coverage area is a 30 by 30 foot (9 by 9 meter) dynamic coverage area.
When you use automatic coverage, the microphone captures talkers you want to hear and avoids areas you tell it to avoid. You can add a mix of up to 8 dynamic and dedicated coverage areas per microphone.
If you turn off automatic coverage, you can manually steer up to 8 lobes.
With automatic coverage on or off, the MXA920 uses Shure's Autofocus™ technology to fine-tune coverage in real time as talkers shift positions or stand. Autofocus is always active, and you don't need to adjust anything for it to work.Automatic coverage = On
When you open Coverage, there's a 30 by 30 foot (9 by 9 meter) dynamic coverage area ready to use. Any talker inside has coverage, even if they stand up or walk around.
Select Add coverage to add more coverage areas. You can use up to 8 coverage areas per microphone, and you can mix both types as needed. Drag and drop to move coverage areas.
Dynamic coverage areas have flexible coverage, which means that the microphone intelligently adapts to cover all talkers in the coverage area. Change the size to fit your space, and any talker within the boundaries of the coverage area will have microphone coverage (even as they move).
Dedicated coverage areas have microphone coverage at all times. They have a set size of 6 by 6 feet (1.8 by 1.8 meters) and work best for talkers that are in one position most of the time, like at a podium or a whiteboard.
As you set up coverage, you may want to block unwanted sounds from your microphone signal (such as doorways or HVAC equipment). There are 2 ways to block unwanted sounds in part of a room:
| Muted Coverage | No Coverage | |
|---|---|---|
| How does it sound? | Great rejection for unwanted sounds | Good rejection for unwanted sounds |
| Can unwanted sound get picked up? | No. Sounds inside muted coverage areas won’t be picked up by active coverage areas. | Possibly. Sounds outside coverage areas can be picked up at low levels by active coverage areas. |
| Does it use coverage areas? | Yes | No |
To use the muted coverage method:
To use the no coverage method:
Automatic coverage = Off
To use steerable lobes, turn off automatic coverage in . You can manually position up to 8 microphone lobes. This mode is best for when you need direct outputs, like for a multi-zone voice lift system.
The microphone doesn't use coverage areas when automatic coverage is off.
Before adjusting levels:
In this mode, there are 2 sets of gain faders:
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.
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.
There are many ways to install MXA920 microphones. See below for details about the mounting and accessory options for square and round array microphones.
Square mounting options:
Round mounting options:
Before you begin:
IMPORTANT: Do not install the 60 cm model in a 2 foot (609.6 mm) ceiling grid.
The A910-JB junction box mounts on square ceiling array microphones to connect conduit. There are 3 knockouts on the junction box for attaching conduit. See local regulations to determine if the junction box is necessary.
Note: Install the junction box on the microphone before installing the microphone in the ceiling.
To install:
The rear plate has 4 threaded holes for attaching the microphone to a VESA mounting device. The mounting holes follow the VESA MIS-D standard:
The VESA mounting holes work with Shure's A900-PM and A900-PM-3/8IN accessories to mount the microphone on a pole.
Suspend the microphone using your own equipment, or with Shure's A900-GM kit (includes mounting cables and hooks).
To mount using your own equipment, you will need:
Attach array microphones to the ceiling with the A900-CM mounting kit.
Refer to the A900-CM user guide to learn how to install on other ceiling materials.
You can mount square ceiling array microphones in hard ceilings without a tile grid using the A910-HCM accessory.
Learn more at www.shure.com.
The automatic coverage setting changes the number of Dante outputs on the MXA920.
Note: When automatic coverage is on, Dante Controller shows 8 transmit channels and the automix output. The automix output is the only channel that sends audio with automatic coverage on.
This device contains IntelliMix digital signal processing blocks that can be applied to the microphone's output. The DSP blocks include:
To access, go to the IntelliMix tab.
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 does 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 (not available on all microphones) to monitor the amount of gain added or subtracted from the signal. If the meter is always reaching the maximum boost or cut level, adjust 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.
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 checkbox 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 (+/− 18 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 EQ contour to quickly apply a high-pass filter at 150 Hz to the microphone's signal.
Select EQ contour to turn it on or off.
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:
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:
For more information, see our FAQ about switches to avoid.
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 .
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+ |
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
| 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.
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:
Refer to 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 |
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.
You can paint the grille and frame of square ceiling array microphones to blend in with a room's design.
Important: Do not remove the 4 recessed screws in each corner.
Important: Do not paint the foam.
Note: The label on the assembly is in the corner that corresponds to the LED.
The grille and back cover of round array microphones can be painted to blend in with a room's design.
Let the paint dry before reassembling.
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:
MXA920 microphones provide information about talker position, lobe position, and other settings through command strings. You can use this information to integrate the microphone with camera control systems.
See the list of commands for camera systems to learn more.
| Problem | Solution |
|---|---|
| Audio is not present or is quiet/distorted |
|
| Sound quality is muffled or hollow |
|
| Microphone does not power on |
|
| Microphone doesn't show up in Designer or Shure Web Device Discovery |
|
| Flashing red error LED | Go to to export the device event log. Designer also has an event log in the main menu that collects information for all Designer devices. Use the event logs to get more information, and contact Shure if necessary. |
| No lights | Go to . Check if brightness is disabled or if any other settings are turned off. |
| Web application lags in Google Chrome browser | Turn off hardware acceleration option in Chrome. |
For more help:
Automatic or steerable
Power over Ethernet (PoE), Class 0
10.1 W maximum
Designer or web application
| MXA920-S | UL2043 (Suitable for Air Handling Spaces) |
| MXA920-R | Not rated |
IEC 60529 IP5X Dust Protected
−6.7°C (20°F) to 40°C (104°F)
−29°C (−20°F) to 74°C (165°F)
Cat5e or higher (shielded cable recommended)
RJ45
| Channel Count | Automatic coverage on | 2 total channels (1 output, 1 AEC reference in channel) |
| Automatic coverage off | 10 total channels (8 independent transmit channels, 1 automix output, 1 AEC reference in channel) | |
| Sampling Rate | 48 kHz | |
| Bit Depth | 24 | |
at 1 kHz
−1.74 dBFS/Pa
Relative to 0 dBFS overload
95.74 dBSPL
Ref. 94 dBSPL at 1 kHz
75.76 dB A-weighted
Does not include Dante latency
| Direct outputs (Automatic coverage off) | 15.9 ms |
| Automix output (Includes IntelliMix processing) | 26.6 ms |
18.24 dB SPL-A
77.5 dB
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
125 Hz to 20,000 Hz
Frequency response measured directly on-axis from a distance of 6 feet (1.83 m).
 |
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.
This device is to be connected only to PoE networks without routing to the outside plant.
Note: This device is not intended to be connected directly to a public internet network.
Changes or modifications not expressly approved by Shure Incorporated could void your authority to operate this equipment.
Note: 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.
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:
This Class B digital apparatus complies with Canadian ICES-003. Cet appareil numérique de la classe B est conforme à la norme NMB-003 du Canada.
CAN ICES-003 (B)/NMB-003(B)
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
| 部件名称 | 有害物质 | |||||
| 铅 | 汞 | 镉 | 六价铬 | 多溴联苯 | 多溴二苯醚 | |
| 电路模块 | X | ○ | ○ | ○ | ○ | ○ |
| 金属模块 | X | ○ | ○ | ○ | ○ | ○ |
| 线缆及其组件 | X | ○ | ○ | ○ | ○ | ○ |
| 外壳 | ○ | ○ | ○ | ○ | ○ | ○ |
| 电源适配器* | X | ○ | ○ | ○ | ○ | ○ |
| 电池组* | X | ○ | ○ | ○ | ○ | ○ |
|
本表格依据SJ/T11364的规定编制。 O: 表示该有害物质在该部件所有均质材料中的含量均在GB/T26572规定的限量要求以下。 X: 表示该有害物质至少在该部件某一均质材料中的含量超出GB/T26572规定的限量要求。 注：本产品大部分的部件采用无害的环保材料制造，含有有害物质的部件皆因全球技术发展水平 的限制而无法实现有害物质的替代。 *:表示如果包含部分 | ||||||
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