IntelliMix Room is digital signal processing (DSP) software designed to optimize the performance of Shure networked microphones with videoconferencing software, resulting in better conference room audio all around. It's designed to run on the same computer as any videoconferencing software, which reduces the amount of equipment in the room.
IntelliMix Room requires other software and hardware to work in your room. You'll need the following:
IntelliMix Room software is sold based on the number of IntelliMix DSP channels you need. For example, an 8-channel license has 8 channels with all of the IntelliMix DSP blocks (AEC, NR, and AGC).
The software always includes 8 auxiliary Dante input channels without IntelliMix processing. You shouldn't count these when deciding on your channel needs.
To choose a channel count:
|Possible equipment combinations||
|Channel count||8 channels||16 channels|
To purchase, contact your local Shure sales representative (find yours at shure.com). For each installation, you can choose from 8 or 16 channels of IntelliMix DSP. Licenses are available for 3- and 5-year durations.
After purchasing, you'll receive an email with instructions for creating a software.shure.com account, where you can find your license ID. The license ID activates all purchased installations.
Before purchasing, you can try out a 16-channel version of IntelliMix Room. The software runs exactly like a purchased version during the trial.
To download a trial version, contact your local Shure sales representative (find yours at shure.com).
Once you set up a trial license, you'll get a license ID for the software. Enter that ID in Designer to activate the trial.
For best results, do not run IntelliMix Room and Designer on the same PC. Devices that don't meet the system requirements are not supported. Virtual machines are not supported.
|Windows 11 22H2||12/31/2025|
|Windows 10 22H2||12/31/2025|
|Windows 11 21H2||10/08/2024|
|Windows 10 21H2||06/11/2024|
|Windows 10 21H1||12/13/2022|
|Windows 10 20H2||05/09/2023|
These are the recommended Windows settings for conference room audio processing:
Additionally, follow these system best practices:
Before installing, make sure you have admin rights for all devices.
You can deploy IntelliMix Room using standard software deployment tools. See below for available command line and silent install arguments.
|NIC index||0||The 0th found NIC using the lookup GetEnabledNetworkAdaptersIds|
|Analytics opt out||False||Users have data collection enabled by default.|
|Disable push notifications||True|
|Optimize power plan||True|
During installation, the software modifies your firewall to allow access for all Shure .exes. These changes are required to run the software.
To use IntelliMix Room, you need Shure Designer software installed on a computer with a network connection to all IntelliMix Room installations.
Designer has some basic concepts that are important to understand as you start using IntelliMix Room:
Each installation of IntelliMix Room appears as a separate device in Designer. Each installation's name matches the computer's network name.
To find any online installations:
If you can't find some installations:
Note: IntelliMix Room doesn't show up in Shure Update Utility or Shure Web Device Discovery.
Before uninstalling, make sure you have admin rights and an internet connection for all devices.
Devices must have an internet connection to release their licenses.
As of version 3.3, there are 2 ways to update IntelliMix Room:
To update using Designer:
Important: Don't close Designer while updating firmware.
On the PC running IntelliMix Room, open IntelliMix Room settings from the Windows system tray and go to .
Download the 2.x version from shure.com and run the installer over the 1.x version. The 2.x version replaces the 1.x version.
Note: Updating from version 1.x to version 3.x and above does not work. Update sequentially from 1.x to 2.x, and then use the 2.x and newer update process.
To activate the software, you need:
IntelliMix Room must be installed on a device before you can activate that license.
After installing IntelliMix Room on all devices, use Shure Designer software to activate your licenses. Designer is usually installed on a separate computer since it manages all installations of IntelliMix Room.
There are a couple terms to know as you manage licenses for IntelliMix Room:
Here's an example workflow for the whole process:
To renew your IntelliMix Room licenses, contact your Shure sales representative.
When you renew, your license ID stays the same. You won't need to make any changes to any existing room setups. All installations will continue running normally.
After an initial grace period, you will hear periodic audio interruptions that remind you to renew your license.
You will receive email reminders to renew your license near the expiration date.
After purchasing, you might need to install IntelliMix Room on a different device than it was originally installed on. Make sure all devices have an internet connection before attempting to reassign a license.
To reassign a license to a new device:
Deactivating the license for an IntelliMix Room installation causes that installation to stop passing audio. Make sure the device has an internet connection before deactivating.
After deactivation, the license is available to be used again on another installation of IntelliMix Room.
To deactivate licenses:
In some situations, you will need to contact Shure support to deactivate a license. These include:
To see information about available licenses and your account, sign in at software.shure.com. Use the username and password you set up during purchase.
IntelliMix Room initially requires an internet connection to activate your license, but it doesn't require one after activation.
When you receive an IntelliMix Room license from Shure, some of your information is collected and stored.
The information stored includes the following:
This information is stored in data centers that are in Santa Clara, CA and Elk Grove Village, IL.
To connect IntelliMix Room to videoconferencing software, select IntelliMix Room Echo Cancelling Speakerphone as the speaker and the microphone in your videoconferencing software. Do the same thing in the computer's sound settings.
The microphone setting sends signals to the videoconferencing software from any microphone connected to IntelliMix Room.
The speaker setting sends a far-end signal from the videoconferencing software to IntelliMix Room. This is how IntelliMix Room gets an AEC reference and a signal for local sound reinforcement.
If you choose a different source as the speaker, you won't be able to get far-end audio from the videoconferencing software into IntelliMix Room to use as an AEC reference.
To route your microphone's signal to IntelliMix Room for processing, use Designer.
This example reflects a small conference room with:
To route signals to the DSP:
You can also manually route audio and apply DSP settings outside of the Optimize workflow if you prefer.
Note: If you're using a non-Shure Dante microphone, use Dante Controller to route the near-end signal to IntelliMix Room.
To use acoustic echo cancellation (AEC), you need to route a far-end signal to the software. The AEC uses that far-end signal as a reference and blocks it from being sent back to the far end as echo.
Each input channel can use a different AEC reference source. If all channels use the same source, select the AEC reference source on each input channel.
If far-end participants hear echo artifacts, the display may be introducing latency. See Troubleshooting for help.
Dante Loudspeakers: Route the signal to the loudspeaker(s) in Dante Controller.
Analog Loudspeakers: Connect the loudspeakers to an ANI22 or ANI4OUT, and route the signal to the ANI in Dante Controller.
To apply DSP blocks:
DSP blocks also get applied during Designer's Optimize workflow.
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:
To use the Optimize workflow:
If you remove or add devices, select Optimize again.
The schematic view in Designer provides an overview of the entire audio signal chain, with the ability to adjust settings and monitor signals.
Right-click an input, output, or processing block to access the following options:
Copy / paste
Copy and paste settings between items. For example, set the equalizer curve on the USB output, and then use the same setting for the analog output. Or, copy the gain and mute status from one input channel to several others.
Mute / unmute
Mutes or activates the channel
Enable / disable
Turns processing on or off (does not apply to matrix mixer or automixer)
Opens the dialog to adjust parameters
Global (right-click in blank area)
Mute all inputs
Mutes all input channels
Mute all outputs
Mutes all output channels
Unmute all inputs
Unmutes all input channels
Unmute all outputs
Unmutes all output channels
Close all dialogs
Clears all open dialogs from the workspace
Create a custom environment to monitor and control a set of inputs, outputs, and processing blocks from a single screen. There are two ways to break out dialogs:
Open as many dialogs as you need to keep important controls available.
A meter appears underneath each input and output to indicate signal levels (dBFS).
The lines connecting inputs and outputs to the matrix mixer appear colored when connections are established. When a signal is not routed, the line appears gray. Use these tools to troubleshoot audio signals and verify connections and levels.
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 contours to quickly apply EQ changes to any of the Dante input channels. Applying both EQ contours and other channel EQ filters has a cumulative effect, meaning that the EQ changes stack on top of each other.
Listen to and test your system as you make EQ changes.
Off: Turns off any active EQ contours
MXA910 High Pass: 300 Hz low-cut filter
MXA910 Low Shelf: 960 Hz, -10 dB low-shelf filter
MXA910 Multi-Band: 200 Hz low-cut filter, parametric 450 Hz, -10 dB, 2.87 Q, ½ octave parametric, 900 Hz, -10 dB, 2.87 Q, ½ octave parametric
MXA310 Low Cut: 180 Hz low-cut filter
MXA710 Low Shelf: 300 Hz, -6 dB low-shelf filter
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.
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 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.
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.
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.
Automatic gain control 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.
Automatic gain control is post-fader, and adjusts the channel level after the input level has been adjusted. Enable it 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.
Target Level (dBFS)
Represents the level that you want the gain to reach. This level is different from adjusting the input fader according to peak levels to avoid clipping. Suggested starting points:
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, adjust the input fader so the signal is closer to the target level.
The AI denoiser identifies and reduces the level of noises such as typing, shuffling papers, or slamming doors. When the denoiser detects noise, it reduces the noise so your voice comes through clearly.
The denoiser adjusts for random noises that aren’t always present in your audio signal, whereas noise reduction helps control constant background noise. For best results, use both the denoiser and noise reduction.
Use the meter to check that the denoiser is actively suppressing noise. The denoiser is applied to the automix output signal.
Shure trains the AI denoiser with thousands of audio files. These include speech samples, noise samples, and samples that have mixtures of speech and noise. During this training, the denoiser learns how to identify patterns of speech and non-speech content in the frequency spectrum. It can then identify and preserve the speech content and reduce the non-speech content.
AI training or listening only happens at Shure’s labs. You will receive any improvements to the AI denoiser algorithm when you update to the latest software version.
Settings indicate how much the denoiser lowers the level of noises. High significantly reduces noises, while low minimally reduces noises. Using the denoiser can affect speech levels if noise and speech happen at the same time. The effect varies with the type of noise and how loud it is.
Listen to and test the different settings to find one that works best in your space.
Designer's Call status feature uses microphone LEDs to show if you're in a videoconferencing call or not. This is a location-level feature, so it applies to all microphones in a Designer location.
When Call status is enabled:
Call status is compatible with the following codecs:
Note: If your codec is running on a computer with a Chrome operating system, call status will not work.
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 turn on mute sync:
For help with specific mute sync implementations, see our FAQs.
You can link channels to each other to create groups for muting and fader controls. You link channels by clicking for the channels and the controls that you want linked. For example, if channels 1, 2, and 3 are linked for Mute, muting any of those individual channels mutes all of the linked channels.
The Inputs tab controls a channel's gain before it reaches the matrix mixer. However, you should also adjust the source's gain before it reaches IntelliMix Room.
To monitor a source's input level before IntelliMix Room processing: Set metering to Pre-gain in the Settings menu.
The 2 metering modes allow you to monitor signal levels before and after the gain stages.
There are 2 different gain faders that serve different purposes:
Input Gain (Pre-Gate)
To adjust, go to Inputs. 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 channel more sensitive to sound sources and more likely to gate on. Lowering gain here makes the channel less sensitive and less likely to gate on.
Automixer Gain (Post-Gate)
To adjust, go to Automixer. These faders adjust a channel's gain after automixer's gating decision. Adjusting the gain here will not affect the automixer's gating decision. Only use these faders to adjust the gain of a channel after you are satisfied with the automixer's gating behavior.
The matrix mixer routes audio signals between inputs and outputs for simple and flexible routing:
Crosspoint gain adjusts the gain between a specific input and output, to create separate submixes without changing input or output fader settings. Select the dB value at any crosspoint to open the gain adjustment panel.
Gain staging: Input fader > crosspoint gain > output fader
Connect inputs and outputs by selecting the box where they intersect.
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.
Note: Not all settings are available on all automixers.
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.
Mic Optimization Mode (P300 only)
Select the microphone that is used with the automixer for best performance. For best results, use Designer's Optimize workflow (this automatically selects the correct mic optimization mode).
Use the Off setting when using a Shure Microflex® Wireless system, or traditional wired microphones.
In the Automixer tab, use the menus below each channel to choose where the signal to the matrix mixer should come from.
All options include input channel gain, mute, solo, and PEQ.
Sends a signal without AEC, noise reduction, or AGC to the matrix mixer.
Sends a signal with AEC and noise reduction but without automixer gating or AGC to the matrix mixer.
Sends a signal with automixer gating, AEC, and noise reduction but without AGC to the matrix mixer.
Sends a signal with automixer gating but without AEC, noise reduction, or AGC to the matrix mixer.
Note: Direct out tap points are not available on all Shure automixers.
IntelliMix Room works with many different types of networks. Here are some of the most common network setups:
This software sends 2 types of data over the network: Shure control data and Dante audio data. You can use the same NIC (network interface card) for both, or use 2 different NICs to separate the traffic.
During installation, you will be asked to choose a network for each one.
Shure control NIC:
Dante audio NIC:
To change the NICs after installation, click the IntelliMix Room icon in the system tray of the computer running IntelliMix Room.
To change Designer's NIC, go to Settings.
During installation, the software prompts you to choose a NIC (network interface card) to use for licensing identification. The NIC must be permanent and not removable. Do not use NICs that could be removed, such as a USB-to-Ethernet adapter or a docking station.
To see the current licensing NIC, click the IntelliMix Room icon in the system tray of the PC running IntelliMix Room. Go to .
This setting cannot be changed after installation.
Shure IntelliMix Room underwent penetration testing by a reputable third-party security assessment company. The test results placed IntelliMix Room above the average of all applications tested. Going forward, Shure will continue to internally and externally test the security of IntelliMix Room. For more information on the test results, please contact email@example.com.
If there's a conflict with the default port, IntelliMix Room automatically selects a port from the specified range.
|Protocol||Port||Range||Inbound or outbound?||Description||Applicable .exe|
|UDP||n/a||319-320||Both||PTP clocking||C:\Program Files\Shure\IntelliMixRoom\dal\exe\ptp.exe|
|UDP||n/a||34440 - 38000||Both||Audio routing||C:\Program Files\Shure\IntelliMixRoom\dal\exe\apec3.exe|
|UDP||5353||n/a||Both||mDNS||C:\Program Files (x86)\Audinate\Shared Files\mDNSResponder.exe|
|UDP||n/a||38801 - 45000||Both||ConMon channels (unicast)||C:\Program Files\Shure\IntelliMixRoom\dal\exe\conmon_server.exe|
|UDP||n/a||8700 - 8708||Both||ConMon channels (multicast)||C:\Program Files\Shure\IntelliMixRoom\dal\exe\conmon_server.exe|
|UDP||n/a||34336 - 34439, 38001 - 38800||Both||Audio channels||C:\Program Files\Shure\IntelliMixRoom\Shure_IntelliMix_Room.exe|
|UDP||4321||n/a||Both||Audio (multicast)||C:\Program Files\Shure\IntelliMixRoom\Shure_IntelliMix_Room.exe|
|UDP||5568||n/a||Both||Session Data Transport (SDT), part of ACN||C:\Program Files\Shure\IntelliMixRoom\Shure_IntelliMix_Room.exe|
|UDP||n/a||49152-65535||Both||ACN dynamic ports||C:\Program Files\Shure\IntelliMixRoom\Shure_IntelliMix_Room.exe|
|UDP||8427||n/a||Both||Shure SLP (multicast)||C:\Program Files\Shure\IntelliMixRoom\Shure_IntelliMix_Room.exe|
|TCP||n/a||1025-65535||Inbound||Shure Command Strings||C:\Program Files\Shure\IntelliMixRoom\Shure_IntelliMix_Room.exe|
If a device can't connect with the Flexera Cloud License Server after allowing the process through the firewall, allow this domain on your network: flexnetoperations.com.
Alternatively, you can manually allow blocks of IP addresses.
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.
Note: IntelliMix Room is not compatible with Dante's device lock feature.
If Dante Domain Manager isn't accessible, you can manually remove the IntelliMix Room installation from the domain. DDM can become inaccessible if an installation of IntelliMix Room gets moved from one network to another and it doesn't get removed from the Dante domain first.
To remove the installation from the domain:
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|
To control IntelliMix Room with third-party control systems, turn on command strings in Designer. You must choose an open port on the computer running IntelliMix Room to send and receive command strings.
Default port: 2202
Port range: 1025-65534
To test if the port is available:
A complete list of command strings is available at pubs.shure.com/command-strings/IntelliMixRoom.
|IntelliMix Room installations won't show up in Designer||
|Licenses won't activate in Designer||
|Clicked IntelliMix Room icon in system tray, and the screen loads continuously||This means IntelliMix Room isn't running properly. Try the following solutions:
|IntelliMix Room won't pass audio||
|Far end hears whispy echo sounds coming from room using IntelliMix Room||
This is likely caused by latency introduced to the signal chain after the AEC reference signal. We have observed devices like soundbars or displays adding extra latency to the signal after it leaves IntelliMix Room. This difference between the AEC reference signal and what comes through the speakers causes problems for the echo canceler's training, and results in a whispy sound.
To fix this problem, delay your AEC reference signal so that it's closer to the signal coming out of the speakers.
There are 2 different event logs to help with troubleshooting: the IntelliMix Room-level event log, and the Designer-level event log. Each one logs different types of events.
IntelliMix Room Event Log
To access, open IntelliMix Room from the system tray and select Event log.
IntelliMix Room's event log collects detailed information about the license status, CPU performance, audio performance, and other possible issues. If you need to contact Shure's support team, this event log provides the most detail about each installation.
Designer Event Log
To access, open Designer and select Event log.
Designer's event log collects high-level information about all devices and software controlled by Designer. This event log isn't as detailed as the device-specific event logs.
Event logs collect up to 1,000 entries. Select Export log to create a CSV (comma separated values) document to save and sort the log data.
An action or event has been successfully completed.
An action cannot be completed, but overall functionality is stable.
A problem has occurred that could inhibit functionality.
Provides details on events and errors, including IP address and subnet mask
Time since most recent boot-up
Indicates event type for internal reference
Restore factory defaults: Resets all application and license settings to the factory defaults. Network settings remain the same.
Automatic mixing, matrix mixer, acoustic echo cancellation (AEC), noise reduction, automatic gain control, compressor, delay, AI denoiser, equalizer (4-band parametric) , mute, gain (140 dB range)
|Dante Digital Audio or Virtual Audio Device||20 to 20,000 Hz|
|Dante Digital Audio or Virtual Audio Device||48 kHz|
|Dante Digital Audio||24 bit|
|Virtual Audio Device||24 bit|
|Virtual Audio Input||Sums to single mono channel|
|Virtual Audio Output||Dual mono channels|
|PC Input||Sums to single mono channel|
|PC Output||Dual mono channels|
Not including Dante latency
|Auxiliary Dante Inputs to Dante Out||17.9 ms|
|Dante Mic Inputs to Dante Out (AEC enabled)||24.6 ms|
|Dante Mic Inputs to Dante Out (AEC and NR disabled)||17.9 ms|
|AI Denoiser Enabled||45.3 ms|
Up to 300 ms
Didn't find what you need? Contact our customer support to get help.