To control the P300, use Shure Designer software to adjust settings and route audio between Shure devices. After completing this basic setup process, you should be able to:
Before you get started, you'll 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.
For this example, we’ll connect an MXA910 ceiling array microphone. The process is the same for any networked Shure microphone in Designer.
You can also route audio manually in Designer outside of the Optimize workflow, or use Dante Controller.
The final steps vary depending on what other devices you connect to the P300. Regardless of what devices you connect, the final steps are to check DSP settings and route signals in the matrix mixer.
Use the channel that carries audio to loudspeakers as the AEC reference. If your room has an analog loudspeaker system or built-in display speakers, Analog -- To Speaker is the most common source. If you're using Dante loudspeakers, one of Dante outputs is the reference.
Learn more about the P300 in the Designer Help section or in the complete user guide at pubs.shure.com/guide/P300.
The P300 IntelliMix Audio Conferencing Processor offers IntelliMix DSP algorithms optimized for audio/video conferencing applications, featuring 8 channels of acoustic echo cancellation, noise reduction and automatic gain control to ensure a high-quality audio experience.
The P300 provides Dante (10 in/8 out), analog (2 block in/2 block out), USB (1 in/out) and mobile (3.5 mm) connectivity options that makes connecting to room systems and collaborating with laptops and mobile devices easier than ever.
Rear panel
Front panel
① Mobile Input
TRRS mobile input connects to a mobile device. Supports bidirectional audio with a TRRS cable, or sends audio into the P300 with a TRS cable.
Note: See the cable requirements for additional information.
Pin Assignments:
Tip | Audio Input (Left) |
Ring 1 | Audio Input (Right ) |
Ring 2 | Ground |
Sleeve | Audio Output (To Phone) |
Note: Left and Right audio signals are summed to a mono signal.
② Audio Inputs (Block Connector)
Balanced audio input connects to an analog audio device. Set the analog input level to match the output level of the analog device.
Input sensitivity:
Line (+4 dBu)
Aux (-10 dBV)
Block Pin Assignments:
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Audio + |
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Audio - |
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Audio ground |
③ Chassis Ground Screw
Provides an optional connection for microphone shield wire to chassis ground.
④ Audio Outputs (Block Connector)
Balanced audio output connects to an analog device. Set the output level to match the input sensitivity of the analog device (Line, Aux, or Mic level).
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Audio + |
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Audio - |
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Audio ground |
⑤ USB Port
Connects to a computer to send and receive audio signals. Use the matrix mixer to sum any combination of signals from the P300 into a single mono channel and send through the USB output.
⑥ Dante Network Port
Connects to a network switch to connect Dante audio, Power over Ethernet (PoE), and data from the control software.
⑦ Reset Button
Resets the device settings back to the factory default.
⑧ LED Indicators
Power: Power over Ethernet Plus (PoE+) present
Note: Use a PoE+ injector if your network switch does not supply PoE+.
Network: Ethernet connection active
Network Audio: Dante audio present on the network
LED Status | Activity |
---|---|
Off | No active signal |
Green | Device is operating successfully |
Red | Error has occurred. See event log for details. |
Encryption:
LED Status | Activity |
---|---|
Off | Audio not encrypted |
Green | Encryption enabled |
Red | Encryption error. Possible causes:
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USB Audio
LED State | Status |
---|---|
Off | No USB device connected |
Green | USB device operating successfully |
Red (flashing) | Problem detected with connected USB audio device |
Note: Error details are available in the Event Log.
⑨ Level Indicators (Signal/Clip)
Tri-color LEDs indicate the audio signal level for the analog channels. Adjust output levels to avoid clipping.
LED State | Audio Signal Level |
---|---|
Off | less than -60 dBFS |
Green | -59 dBFS to -24 dBFS |
Yellow | -23 dBFS to -1 dBFS |
Red | 0 dBFS or more |
Note: The input and output LEDs stay off when metering is set to Post-gain and the channel is muted.
This device requires PoE Plus to operate. It is compatible with both Class 4 PoE+ sources.
Power over Ethernet is delivered in one of the following ways:
Always use Cat5E cable or higher.
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.
KIT, HARDWARE, P300-IMX | 90D33522 |
BRACKET, HALF RACK UNIT | 53A27741 |
USB cable | 95A39698 |
The reset button is located inside a small hole in the rear panel. Use a paperclip or other small tool to press the button.
There are 2 hardware reset functions:
Network reset (press button for 4-8 seconds)
Resets all Shure control and audio network IP settings to factory defaults.
Full factory reset (press button for longer than 8 seconds)
Resets all network and Designer settings to the factory defaults.
Reboot Device ( ): Power-cycles the device as if it were unplugged from the network. All settings are retained when the device is rebooted.
Restore Factory Defaults (
): Restores all network and Designer settings to the factory defaults. This is the same as performing a full factory reset using the reset button on the device.Default Settings (
): Resets audio settings back to the factory configuration (excluding device name, IP settings, and passwords).Designer's Optimize workflow speeds up the process of connecting systems with at least 1 microphone and 1 audio processor. Optimize also creates mute control routes in rooms with MXA network mute buttons. When you select Optimize in a room, Designer does the following:
The settings are optimized for your particular combination of devices. You can customize settings further, but the Optimize workflow gives you a good starting point.
After optimizing a room, you should check and adjust settings to fit your needs. These steps may include:
Compatible devices:
To use the Optimize workflow:
If you remove or add devices, select Optimize again.
Two mounting solutions are available for installing the P300:
CRT1 19" Rack Tray (optional accessory): Supports up to 2 devices (two P300s or one P300 and one ANI4IN, ANI4OUT, ANI22, or ANIUSB); mountable in a rack or under a table
Single-unit Mounting Tray (included accessory): Supports a single device for mounting under a table
Use the included screws from the mounting hardware kit to secure each P300 or Audio Network Interface (ANI). Devices can be mounted to face either direction. Insert the screws from the bottom in the appropriate holes, according to the following diagrams:
Align the holes as shown for securing a single device in the single-unit mounting tray
Align the holes as shown for securing up to two devices in the 19" rack tray.
The adjustable rack ears support mounting in a standard equipment rack or underneath a table.
Applies to Designer 4.2 and newer.
Before setting up devices, check for firmware updates using Designer to take advantage of new features and improvements. You can also install firmware using Shure Update Utility for most products.
To update:
When updating firmware, update all hardware to the same firmware version to ensure consistent operation.
The firmware of all devices has the form of MAJOR.MINOR.PATCH (e.g., 1.2.14). At a minimum, all devices on the network, must have the same MAJOR and MINOR firmware version numbers (e.g., 1.2.x).
6 Dante Outputs Added
Automixer Direct Out Tap Points Added
Support for Dante Domain Manager
Audio Encryption Improvements
Web Applications Removed
Shure offers a range of connectivity options for conferencing. MXA microphones, audio processors, and network interfaces all use Dante to send audio over standard IT networks. You can use Shure's free Designer software to control most Shure devices and route audio between them.
Device | Purpose | Physical Connections | Dante I/Os |
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Ceiling array microphone with IntelliMix DSP |
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Linear array microphone with IntelliMix DSP |
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2 Foot:
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Table array microphone |
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Audio processor with IntelliMix DSP and matrix mixer |
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Audio processing software with IntelliMix DSP and matrix mixer | Varies depending on device |
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Matrix mixer with USB and analog input/output |
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Converts analog signals to Dante signals |
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Converts Dante signals to analog signals |
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Networked ceiling loudspeaker powered by PoE |
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PoE-powered network mute button for Shure devices |
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n/a |
Input | Output |
---|---|
Automix | USB output |
USB input | Analog - To Speaker or Dante output |
When you use a USB-connected codec with a P300 or ANIUSB-MATRIX, you may need to set the USB device type on the Shure device.
The USB device type setting tells the codec if acoustic echo cancellation (AEC) is needed or not. The codec can then turn on or turn off its own AEC (if supported).
For example: You have an MXA710 routed to an ANIUSB-MATRIX and want to use the MXA710's AEC. Set the USB device type to Echo-canceling speakerphone to tell the codec to turn off its AEC.
To change USB device type:
ANIUSB-MATRIX only: Designer's Optimize workflow automatically sets the USB device type. You can also change it manually.
The USB port connects the host computer to the entire room audio system, including microphones and loudspeakers.
When the P300 is connected for the first time, the computer recognizes it as a USB audio device. You may need to select it as the input/output (recording/playback) device to pass audio. Assign the P300 as the default device to ensure it passes audio every time it is plugged in. Refer to the manual for your computer to configure the audio settings.
This device is compatible with USB-B to USB-C adapters. Using an adapter is only recommended for desktop and laptop computers, as many mobile devices do not support bi-directional audio through USB or lightning ports.
Input | Output |
---|---|
Automix | Analog - To codec |
Analog - From Codec | Analog - To Speaker or Dante output |
In this example, when the phone is plugged in, the built-in microphone and speaker are disabled -- the phone simply carries the call. The MXA310 microphone captures near-end audio, and the loudspeaker delivers audio from the far end of the call.
Input | Output |
---|---|
Automix | Mobile output |
Mobile input | Analog - To Speaker or Dante output |
A 1/8-inch TRRS cable is required to connect a phone to the P300. Avoid using cables with a metal flange, as it may create an electrical connection to the exterior of the phone and interrupt the signal.
To ensure proper operation, only use:
Note: If necessary, a TRS (tip/ring/sleeve) cable may be used to plug a stereo device into the P300, but the device will only be able to send audio to the P300. The Enable auto-mute feature on the mobile input channel must be turned off in this case.
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:
Per Channel
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)
Edit
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.
Use presets to quickly save and recall settings. Up to 10 presets can be stored on each device to match various signal processing requirements, room types, and microphones used. A preset saves all device settings except for the Device Name, IP Settings, and Passwords. Importing and exporting presets into new installations saves time and improves workflow. When a preset is selected, the name displays above the preset menu. If changes are made, an asterisk appears next to the name.
Note: Use the default settings preset to revert to the factory configuration (excludes Device Name, IP Settings, and Passwords).
Open the presets menu to reveal preset options:
save as preset: | Saves settings to the device |
load preset: | Opens a configuration from the device |
import from file: | Downloads a preset file from a computer onto the device. Files may be selected through the browser or dragged into the import window. |
export to file: | Saves a preset file from the device onto a computer |
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.
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 and eliminates echoes to deliver clear, uninterrupted speech. The P300 features 8 channels of AEC, with independent processing on each channel for maximum effectiveness.
For best results, improve the acoustic environment when possible:
Training is the process where the AEC optimizes processing based on the acoustic environment. It only trains when far-end audio is present and near-end talkers are quiet. The AEC is constantly adapting, so if the acoustic environment changes, the AEC automatically adjusts.
To adjust acoustic echo cancellation settings, open the AEC menu in the schematic view or inputs tab.
Reference Meter
Use the reference meter to visually verify the reference signal is present.
ERLE
Echo reduction loss enhancement displays the dB level of signal reduction (the amount of echo being removed). If connected properly, the ERLE meter activity generally corresponds to the reference meter.
Reference
Select the channel that carries audio to the loudspeakers as the reference. Analog - To Speaker is the most commonly used channel, for configurations with an analog loudspeaker system or a display with a built-in speaker.
Note: Selecting a reference on any channel applies that same reference to all channels with AEC.
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.
Medium: Use in typical rooms as a starting point. If echo artifacts appear, 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.
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.
Use the delay feature on the analog and USB outputs to synchronize audio and video. When a video system introduces latency (where you hear someone speak, and their mouth moves later), simply add delay to the analog outputs to align with the video. Delay can also be used in larger rooms to align the arrival time or phase between multiple speakers.
The 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 it is closer to full synchronization, use smaller intervals to fine-tune.
The USB output channel features delay to ensure the near-end camera and near-end audio are synchronized.
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.
The default configuration enables calling to multiple far ends with near-end Shure microphones. Connections are established for operating hardware codecs, software codecs, and mobile phones simultaneously.
Input/Source Channel | Output/ Destination Channels |
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Automix (summed Dante input channels) |
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Analog - From Codec (Analog input 1) |
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USB input |
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Mobile input |
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Near-end audio from Dante microphones (Shure MXA 310) and the mobile phone are both routed to the video codec and sent to the far end. The mobile phone is simply carrying the audio from the remote caller -- its built-in microphone and speaker are disabled.
Far-end audio from the video codec is routed to a powered loudspeaker or amplifier (analog or Dante-enabled). It is also routed to the mobile phone (connected to the P300) to relay the signal to the remote caller.
The remote caller (far end) receives audio from both the near-end and far-end locations. The P300 connects all locations by routing both near and far-end audio sources through the mobile output. The audio from the remote caller is routed to the mobile input, and then sent to the loudspeakers in the near-end room and through the video codec to the far-end room.
Input / Source Channel | Output / Destination Channel |
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Automix (four summed Dante input channels from MXA310) |
Analog - To Codec (analog output 1) Mobile output |
Analog - From Codec (analog input 1) |
Analog - To Speaker (analog output 2) Mobile output |
Mobile input |
Analog - To Codec (analog output 1) Analog - To Speaker (analog output 2) |
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:
Use this setup to mute the P300 by pressing the mute button on the MXA310. In the P300’s signal chain, muting happens after the DSP so that the AEC stays converged.
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.
To use:
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.
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 the P300.
To monitor a source's input level before it reaches the P300: Set metering to Pre-gain in the Settings menu.
To adjust channel gain in the P300's Inputs tab: Set metering to Post-gain in the Settings menu.
Note: The Matrix mixer tab lets you adjust crosspoint gain, which controls the levels for the separate submixes being sent to the different outputs.
The mobile device input gain is optimized for most devices when the fader is set at 0 dB. It provides adequate volume with sufficient headroom. As a general target, the audio signal received by the P300 from the phone should reach an average level of approximately -24 dBFS.
If the signal being sent to the far end is too quiet, check and adjust the gain levels for the near-end microphones and the automixer.
Mute Groups | Check the Mute group box to add the channel to a group. Muting any channel within the Mute group mutes all channels in the group. |
Fader Groups | Check the Fader group box to add the channel to a group. All faders within the group are linked, and move together when a single fader is adjusted. |
Maximize audio quality by adjusting the frequency response with the parametric equalizer (PEQ). Use the input equalizers to make adjustments to specific channels, while using the output equalizers to adjust frequency response of all signals that are summed through a given output.
Common equalizer applications:
Note: If you're connecting a microphone that has a built-in equalizer (such as an MXA310), disable any EQ on the mic and use the P300's EQ instead.
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
Each band has a selectable filter:
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)
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
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.
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:
|
If you’re using Shure Designer software to configure your system, please check the Designer help section for more about this topic.
Tip: Set the output metering in the settings menu to ensure accurate metering.
Adjust faders in the Outputs section as high as necessary, but make sure to avoid clipping (when the signal reaches 0 dBFS). Always adjust the input gain and crosspoint gain in the matrix mixer before the output gain.
Analog output level:Select Line, Aux, or Mic level output signal to match the sensitivity of the receiving device.
The 2 metering modes allow you to monitor signal levels before and after the gain stages.
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.
Gating Sensitivity
Changes the threshold of the level at which the gate is opened
Off Attenuation
Sets the level of signal reduction when a channel is not active
Hold Time
Sets the duration for which the channel remains open after the level drops below the gate threshold
Maximum Open Channels
Sets the maximum number of simultaneously active channels
Priority
When selected, this channel gate activates regardless of the number of maximum open channels.
Always On
When selected, this channel will always be active.
Send to Mix
When selected, sends the channel to the automix channel.
Solo
Mutes all of the other channels
Automix Gain Meter
When enabled, changes gain meters to display automix gating in real time. Channels that gate open will display more gain than channels that are closed (attenuated) in the mix.
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.
Pre-Processing/Pre-Gate
Sends a signal without AEC, noise reduction, or AGC to the matrix mixer.
Post-Processing/Pre-Gate
Sends a signal with AEC and noise reduction but without automixer gating or AGC to the matrix mixer.
Post-Processing/Post-Gate
Sends a signal with automixer gating, AEC, and noise reduction but without AGC to the matrix mixer.
Pre-Processing/Post-Gate
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.
Encryption operates at the room level, meaning that all devices included in the room must have these settings. 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. Encryption is not supported with third-party devices.
To activate encryption:
The other options allow you to re-key the encryption or disable it if encryption had previously been enabled and you no longer want it.
Important: For encryption to work:
Note: Encryption will not work between devices on 3.x and 4.x firmware. Update all devices to same the major firmware version to use encrpytion.
When connecting Shure devices to a network, use the following best practices:
This Shure device uses 2 IP addresses: one for Shure control, and one for Dante audio and control.
To access these settings in Designer, go to
.Configure IP
Sets IP mode of the selected network interface:
IP Settings
View and edit the IP Address, Subnet Mask, and Gateway for each network interface.
MAC Address
The network interface's unique identification.
IP configurations are managed in Shure Designer software. By default, they are set to Automatic (DHCP) mode. DHCP mode enables the devices to accept IP settings from a DHCP server, or automatically fall back to Link-Local settings when no DHCP is available. IP addresses may also be manually set.
To configure the IP properties, follow these steps:
To manually assign IP addresses, follow these steps:
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.
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.
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+ |
To send a device name to appear in Dante Controller, go to Settings>General and enter a Device Name. Select Push to Dante to send the name to appear on the network.
Note: names appear in Dante Controller with "-d" attached.
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.
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: Applies to firmware 4.1.x and newer.
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 |
Packet bridge enables an external controller to obtain IP information from the control interface of a Shure device. To access the packet bridge, an external controller must send a query packet over unicast UDP* to port 2203 on the Dante interface of the Shure device.
Note: The maximum accepted payload 140 bytes. Any content is allowed.
Bytes | Content |
---|---|
0-3 | IP address, as 32-bit unsigned integer in network order |
4-7 | Subnet mask, as 32-bit unsigned integer in network order |
8-13 | MAC address, as array of 6 bytes |
Note: The Shure device should respond in less than one second on a typical network. If there is no response, try sending the query again after verifying the destination IP address and port number.
*UDP: User Datagram Protocol
The packet bridge does not allow cross-subnet command strings.
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.
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:
The event log provides a detailed account of activity from the moment the device is powered on. The log collects up to 1,000 activity entries and time-stamps them relative to the last power cycle. The entries are stored in the internal memory, and are not cleared when the device is power-cycled. The Export feature creates a CSV (comma separated values) document to save and sort the log data.
Refer to the log file for details when troubleshooting or consulting with Shure Systems Support.
To view the event log:
Severity Level
Information
An action or event has been successfully completed
Warning
An action cannot be complete, but overall functionality is stable
Error
A problem has occurred that could inhibit functionality.
Log Details
Description
Provides details on events and errors, including IP address and subnet mask.
Time Stamp
Power cycles:days:hours:minutes:seconds since most recent boot-up.
Event ID
Indicates event type for internal reference.
Tip: Use the filter to narrow down results. Select a category heading to sort the log.
Problem | Solution |
---|---|
Software lags in Google Chrome browser | Problem is browser-related. Turn off hardware acceleration option in Chrome. |
Sound quality is muffled | Use equalizer to adjust frequency response. See the equalizer applications for the appropriate use. |
Audio sounds too high or too low in pitch | Make sure that the sample rate settings for Playback and Recording are the same in your computer's sound settings. If these sample rates do not match, the audio may sound too high or too low in pitch. |
Hardware does not show up in device discovery |
|
No audio |
|
Cannot route Dante audio channels | Install latest version of Dante Controller from Audinate, available at www.audinate.com. |
Hardware does not power on |
|
Didn't find what you need? Contact our customer support to get help.
Input | (2) 3-pin block connector (Active Balanced) |
Output | (2) 3-pin block connector (Impedance Balanced) |
Mobile | (1) TRRS 3.5 mm (1/8″) |
(1) USB 2.0, Type B
Single port carries 2 input and 2 output channels (Summed mono)
(1) RJ45
10 Dante Inputs, 8 Dante Outputs
Non-inverting, any input to any output
802.3 at Type 2 (PoE Plus), Class 4
17.5 W, maximum
1710 g (3.8 lbs)
H x W x D
4 x 21 x 22.6 cm ( 1.6 x 8.3 x 8.9 in.)
Shure Designer
−6.7°C (20°F) to 50°C (122°F)
−29°C (-20°F) to 74°C (165°F)
Maximum | 17.5 W ( 60 BTU/hr) |
typical | 14.6 W ( 50 BTU/hr) |
+1, -1.5 dB
20 to 20,000 Hz
Sampling Rate | 48 kHz |
Bit Depth | 24 |
Sampling Rate | 48 kHz |
Bit Depth | 16, 24 |
Does not include Dante latency
Firmware 4.1 and newer | Dante 1-8 in to Dante out (AEC enabled) | 15.4 ms |
Dante 1-8 in to Dante out (AEC disabled) | 8.7 ms | |
Dante 9-10 in to Dante out | 3.4 ms | |
Analog in to Analog out | 3.8 ms | |
Firmware 3.1 and older | Dante 1-8 in to Dante out (AEC enabled) | 12.5 ms |
Dante 1-8 in to Dante out (AEC disabled) | 5.8 ms | |
Dante 9-10 in to Dante out | 1.8 ms | |
Analog in to Analog out | 2.2 ms |
Up to 300 ms
20 Hz to 20 kHz, A-weighted, typical
Analog-to-Dante | 113 dB |
Dante-to-Analog | 117 dB |
20 Hz to 20 kHz, A-weighted, input terminated with 150Ω
Line | -86 dBV |
Aux | -98 dBV |
@ 1 kHz, 0 dBV Input, 0 dB analog gain
<0.05%
150Ω balanced source @ 1 kHz
>50 dB
9.6 kΩ
Line | +27 dBV |
Aux | +15 dBV |
80 Ω
Line | +20 dBV |
Aux | +0 dBV |
Mic | -26 dBV |
Tip | Audio Input (Left) |
Ring 1 | Audio Input (Right ) |
Ring 2 | Ground |
Sleeve | Audio Output (To Phone) |
20 Hz to 20 kHz, A-weighted, typical
Analog-to-Dante | 99 dB |
Dante-to-Analog | 90 dB |
20 Hz to 20 kHz, A-weighted, input terminated with 20Ω
-95 dBV
@ 1 kHz, 0 dBV Input, 0 dB analog gain
<0.05%
3.7 kΩ
+4 dBV
1.4 kΩ
Output terminated with 2.2 kΩ
-20 dBV
Cat 5e or higher (shielded cable recommended)
Tip | Audio Input (Left) |
Ring 1 | Audio Input (Right ) |
Ring 2 | Ground |
Sleeve | Audio Output (To Phone) |
Note: The audio input (tip and ring 1) are summed to a mono signal in the P300, to send the signal to any destination on a single channel.
19" rack tray | CRT1 |
Note: Model information and power ratings are labeled on the bottom of the unit.
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This symbol indicates that dangerous voltage constituting a risk of electric shock is present within this unit. |
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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.
Dante is a registered trademark of Audinate Pty Ltd.
This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the manufacturer's instruction manual, may cause interference with radio and television reception.
Notice: The FCC regulations provide that changes or modifications not expressly approved by Shure Incorporated could void your authority to operate this equipment.
These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:
This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:
The CE Declaration of Conformity can be obtained from: www.shure.com/europe/compliance
Authorized European representative:
Shure Europe GmbH
Global Compliance
Jakob-Dieffenbacher-Str. 12
75031 Eppingen, Germany
Phone: +49-7262-92 49 0
Email: info@shure.de
www.shure.com
This product meets the Essential Requirements of all relevant European directives and is eligible for CE marking.
The CE Declaration of Conformity can be obtained from Shure Incorporated or any of its European representatives. For contact information please visit www.shure.com