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 download the software and 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.
You can install IntelliMix Room on any physical device that meets these requirements:
If you install IntelliMix Room on a hub-type device running Windows, consult the manufacturer's documentation for how to access the operating system and install software.
Devices that don't meet these requirements are not supported. Virtual machines are not supported.
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.
Optional CLI Arguments
Installation and IntelliMix Room Settings
Installer Default Settings
|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|
|Disable auto update||True|
|Optimize power plan||True|
|Disable network throttling||True|
When a new version of IntelliMix Room is available, you'll receive an email about the release. To install, download the .exe and follow the instructions.
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.
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 activate the software, you need:
IntelliMix Room must be installed on a device before you can activate that license.
To renew your IntelliMix Room licenses, contact your Shure sales representative.
After you renew your licenses, 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.
IntelliMix Room stops passing audio when your license expires.
You will receive email reminders to renew your license 90 days before it expires.
Deactivating the license for an IntelliMix Room installation causes that installation to stop passing audio.
After deactivation, the license is available to be used again on another installation of IntelliMix Room.
To deactivate licenses:
After purchasing, you might need to install IntelliMix Room on a different device than it was originally installed on.
To reassign a license to a new device:
IntelliMix Room uses a cloud license server managed by Flexera. 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 requires a continuous internet connection to verify each installation's license status. Every 12 hours, the installation checks in with the license server to validate its license. If a device running IntelliMix Room can't contact the cloud license server for 7 days, the installation becomes unlicensed and audio stops passing.
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:
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.
Shure offers a range of connectivity options for conferencing. MXA microphones, audio processors, and network interfaces all use Dante to send audio over standard IT networks. You can use Shure's free Designer software to control most Shure devices and route audio between them.
As you plan out a system, think about what other devices you need to connect to and whether you'll need extra inputs/outputs in the future.
|Device||Purpose||Physical Connections||Dante I/Os|
|Ceiling array microphone with IntelliMix DSP||
|Table array microphone||
|Audio processor with IntelliMix DSP and matrix mixer||
|Audio processing software with IntelliMix DSP and matrix mixer||Varies depending on device||
|Matrix mixer with USB and analog input/output||
ANI4IN (block or XLR connectors)
|Converts analog signals to Dante signals||
ANI4OUT (block or XLR connectors)
|Converts Dante signals to analog signals||
ANI22 (block or XLR connectors)
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:
Optimize audio speeds up microphone and DSP configuration. After routing audio from a compatible microphone to a compatible DSP, select Optimize audio. Designer then optimizes microphone and DSP settings for all Shure devices.
You can customize settings further, but Optimize audio gives you a good starting point for further customization.
To use Optimize audio:
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
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.
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.
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.
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.
|Virtual Audio Input||Dante Output 1|
|Automix Out||Virtual Audio Output|
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.
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, route microphone channels to the processing device and click Auto configure (this automatically selects the correct mic optimization mode).
Use the Off setting when using a Shure Microflex® Wireless system, or traditional wired microphones.
Gate Inhibit (P300 only)
Enable gate inhibit to prevent far-end audio from gating on near-end microphone channels.
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.
|Protocol||Port||Inbound or outbound?||Description||Applicable .exe|
|UDP||319, 320||Both||PTP clocking||Shure PTP.exe|
|UDP||34441||Both||ARCP (audio routing)||apec3.exe|
|UDP||34455||Both||DBCP (audio routing)||apec3.exe|
|UDP||5353||Both||mDNS (used by mDNSResponder.exe)||Shure mDNSResponder.exe|
|UDP||38801-38810||Both||ConMon channels (unicast)||Shure Common_Server.exe|
|UDP||8702, 8703, 8708||Both||ConMon channels (multicast)||Shure Common_Server.exe|
|UDP||5568||Both||Session Data Transport (SDT), part of ACN||Shure_IntelliMix_Room.exe|
|UDP||8427||Both||Shure SLP (discovery) (multicast)||Shure_IntelliMix_Room.exe|
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.
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.
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
|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.
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, 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 or 16 bit|
Not including Dante latency
|Auxiliary Dante Inputs to Dante out||28 ms|
|Dante Mic Inputs to Dante out (AEC enabled)||34.7 ms|
|Dante Mic Inputs to Dante out (AEC and NR disabled)||28 ms|