Composer Management Software Tech Tips

Composer’s docking and panel resizing features can save time and let you focus on maximizing the design and performance of your Site File. With any site file opened (Composer 3.0 or later), simply double click any panel to undock it.

Double click the rim of the panel or slide it to any one of the four sides of the software to dock it. The Search Bar is shown docked in red below.

If you are having trouble seeing all of the items in the Toolkit, such as a full list of control modules or Dante flow names, simply grab the Toolkit panel edge and stretch it to the desired width. Notice the toolkit is stretched in Diagram B

Interface A and B

One other useful feature is the ability to dock the browser and toolkit in the same location, in which case the two panels are tabbed, and switching between them is as easy as clicking the respective tab.

Interface C and D

Click the Tab between Toolkit and Browser to swap between the two panels.

Once Composer has been closed, the software will remember the panel size and docking preference the next time it is opened.

Certain environments have a requirement for network-based IR control. This tech tip provides step-by-step instructions for connecting Global Cache IP2IR to Symetrix Composer.

Tools and Resources:

• Symetrix Composer Software: Download and install the latest version of Composer from the Symetrix website: https://www.symetrixinc.com/download-composer/
• Global Cache IP2IR Intelligent Module (IM): Download and extract the IP2IR Intelligent Module from the Symetrix website: https://www.symetrix.co/wp-content/uploads/sym-software/intelligent_modules/Symetrix_MODULE_GlobalCache-IP2IR_v1-0.zip
• Global Cache iHelp: This tool helps you discover and configure your Global Cache IP2IR device on the network. Download it from the Global Cache website: https://www.globalcache.com/downloads.html#collapse1_17
• Global Cache iTest: Use iTest to send IR commands and verify that your IP2IR device is working correctly. Download it from the Global Cache website: https://www.globalcache.com/downloads.html#collapse4_17
• Control Tower IR Database: This online database provides a vast collection of IR codes for various devices. Access it at: https://irdb.globalcache.com/Home/Database
• Network Connection: Ensure your computer is running Composer and the Global Cache IP2IR device are on the same network.
• Target Device: Have the device you want to control (e.g., TV, projector, receiver) powered on and ready.

Connecting the Global Cache IP2IR to Symetrix Composer:

Obtain IR Codes for Your Device:

1. Visit the Global Cache Control Tower IR Database: https://irdb.globalcache.com/Home/Database
   • Search for your device by brand and model number.
   • Download the IR specific function code, or the full code set for your device. (Complete code sets are available to Global Cache verified users only).

• Codes and code sets will be email from Global Cache
  • Power Toggle Function Code example:

function: POWER TOGGLE

code1: sendir,1:1,1,38000,1,1,170,170,20,63,20,63,20,63,20,20,20,20,20,20,20,20,20,20,20,63,20,63,20,63,20,20,20,20,20,20,20,20,20,20,20,20,20,63,20,20,20,20,20,20,20,20,20,20,20,20,20,63,20,20,20,63,20,63,20,63,20,63,20,63,20,63,20,1798

hex code1: 0000 006D 0000 0022 00AA 00AA 0014 003F 0014 003F 0014 003F 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 003F 0014 003F 0014 003F 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 003F 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 0014 003F 0014 0014 0014 003F 0014 003F 0014 003F 0014 003F 0014 003F 0014 003F 0014 0706

• Discover Your IP2IR Device:
  • Use Global Cache iHelp to locate your IP2IR device on the network and note its IP address.

• Add the IP2IR Module to Composer:
  • Open your Symetrix Composer site file.
  • Select the DSP the IP2IR module will be added to, and open Design view.
  • In the Toolkit tab:
    • Intelligent Modules > Create or Open Existing > browser to the location the IP2IR Intelligent Module was extracted.
    • Open Symetrix_MODULE_GlobalCache-IP2IR_v1-0.mod.
    • The module will automatically be added to the Design view

• Configure the IP2IR Module Connection:
  • Enter the IP Address found when the IP2IR was discovered in the iHelp app.
  • Go online with the site to verify the IP2IR is connected.

• Add the IR function codes:
  • In the command text boxes enter a specific function code, one function per text box.
    • Ex. sendir,1:1,1,38000,1,1,170,170,20,63,20,63,20,63,20,20,20,20,20,20,20,20,20,20,20,63,20,63,20,63,20,20,20,20,20,20,20,20,20,20,20,20,20,63,20,20,20,20,20,20,20,20,20,20,20,20,20,63,20,20,20,63,20,63,20,63,20,63,20,63,20,63,20,1798

• Test the Connection:
  • Click the send button next to the Function command to send that command to the IP2IR device which will then send the command to the TV.
• Digit A is the port on the back of the IP2IR device which the IR transmitter is plugged into. There are 3 options.
  • Digit B is the number of times the command will be triggered.

A SPL (Sound Pressure Level) Computer adjusts the level of program material based upon a measurement of the ambient noise. It is typically applied where background, or foreground music must be slightly louder than a variable level crowd noise. Composer has (2) different types of SPL computer
modules available; Gap Sensing and Continuous. The term ‘program material’ will be used to refer to whatever audio signal is sent through the audio path of these modules. Depending on the application, this could be background/foreground music, paging signals, or a mixture of page and music, possibly with ducking. The ambient noise measurement is taken by the SPL Computer through its “Sense In” audio input. In most applications Symetrix recommends using any omni-directional microphone for the sense input.

Gap-Sensing SPL Computer

The Gap-sensing SPL Computer module only listens to the ambient sense microphone during program material gaps, i.e. when the program level is below a certain threshold. This prevents the module from hearing its own output which avoids run-away feedback problems.

Mono and stereo versions have the same controls. The only difference is that with stereo versions, stereo program material is adjusted. The same gain is applied equally to both channels.

Inputs: In or In-L and In-R. Program material inputs. Connect the mono or stereo program material to these inputs.

Sense In. Sense microphone input. Connect a signal from your sensing microphone to this input.

Outputs: Out, or Out-L and Out-R. Program material outputs with SPL volume applied. Connect these to your main outputs.

Note: Any equalization, dynamics processing, or level changes to the program material must be done before it is sent to the SPL computer. The SPL module should be the last thing in the signal processing chain. The gain stage from the SPL computer to the speakers must remain constant after calibration. This includes analog output block gain settings, power amplifier levels, speaker attenuators, etc… User-controlled loudspeaker attenuators should not be used when a continuous SPL computer is present.

Control: This module has a control signal output. The control signal reflects the dynamic gain change being applied by the module and changes in real time. This control signal can be used to easily create linked multi-channel dynamics modules.

Note: The control signal is scaled so that 1.0 represents the maximum gain that can be applied by the SPL computer

Controls & Indicators:

Gain: The current gain being applied by the module (assuming it is “Active”) is displayed at the top of the window.

Maximum Gain: dB that the SPL computer is allowed to apply to the signal. Use this control to put a limit on how loud the output can get with very loud ambient levels. Adjust using the slider or click in the text entry box to specify a numerical value.

Note: The Maximum Gain cannot be set to a value less than the Minimum Gain.

Minimum Gain: dB that the SPL computer is allowed to apply to the signal. Use this to put a limit on how soft the input can get with very quiet ambient levels. Adjust using the slider or click in the text entry box to specify a numerical value.

Note: the Minimum Gain cannot be set to a greater than the Maximum Gain.

Gain-Sense Ratio: Controls the change in gain versus the change in ambient level. Setting this to 1.0:1 means that for every 1dB increase/decrease in the ambient level, the SPLs gain is increased/decreased by 1dB. Higher values such as 2.0:1 mean that the gain is increased by more than the ambient increase, which allows out-shouting the crowd. A setting of 0.5:1 means that the gain is only changed by 0.5dB for every 1dB ambient change giving a more subtle effect.

Speed: Controls the rate in which the module changes the gain, specified in seconds. Longer times can cause a very gradual fade up or down in response to changing ambient levels.

Gap Threshold: Controls the level under which the program material must be in order for it to be considered a gap. When the program material is less than this level for at least the Gap Time, the Gap Detect LED will light and the SPL will respond to the ambient level. When the program material is above this level, the SPL will not make adjustments based on the ambient level. This level should be set so that the Gap Detect LED lights whenever the program material is soft enough that the ambient sense microphone does not pick up a significant amount of program material, i.e. the ambient noise dominates the program pick-up. By positioning the sense microphone in order to minimize pick-up of the program material, a higher gap threshold can be set and then take sense measurements more often, e.g. in softer musical passages. If the program material has regular gaps (e.g. a paging signal or background music with clear breaks between songs) set this threshold quite low to just above the noise floor of the program material. In doing so, it will only be sensing during actual gaps in the music/paging signal.

Gap Time: Controls the length of time in milliseconds that the program material must be silent (below the gap threshold) to be considered a gap. This setting can be used to compensate for the reverberation time of a live room. It can prevent the algorithm from responding to the reverberation tail of a page or other program material. Settings in the 100-200ms range are good for most environments, though very live rooms may require settings of 1 second or more.

Max Gap Interval: If no gaps occur in the program material in this amount of time, the module forces a gap by momentarily muting the audio. Averaging Count: Averages the number of designated sense readings before calculating a gain change. The Max Gap Interval is used as the sampling rate for the moving average.

Gap Detect LED: This LED lights when a gap is detected and the module is responding to the sense input.

Force Gap Now button: Pushing this button forces a gap a sense immediately by momentarily muting the audio.

Active button: When engaged, this button activates the module so that the gain adjustments are applied to the audio in the signal path. When inactive, SPL calculations are inhibited.

Sense controls:

Threshold: The threshold sets the sense input level at which the module applies unity gain. This should be set to the average or typical ambient level of the room during calibration. The meter displays the average level of the sense input as a reference. Use the fader, or numerical entry to adjust the threshold.

Calibrate button: Forces the threshold to be set to the current meter reading. Ideally, calibration should be done with an average ambient level and no program material playing.

Sense Statistics: Shows the highest and lowest sense values logged since the last reset. Sense values are only logged during gaps.

Reset button: Clears statistic values. Use this feature to monitor the ambient noise levels in a room over time.

Gap-Sensing SPL Composer Calibration Procedure:

When calibrating the Gap-Sensing SPL make sure no program material is playing. Use the Sense Level meter and Threshold fader to select and apply unity gain. Press the Calibration button to force the threshold to be set to the current meter reading. Once unity gain is set, adjust the faders for Maximum Gain (how much louder from unity gain should the SPL Computer be allowed to raise the program material gain), Minimum Gain (how much quieter should the SPL Computer be allowed to lower the program material gain), Gain-Sense Ratio, Speed, Gap Time, Max Gap Interval, and Averaging Count to achieve the desired performance of the SPL computer.
 

Note: Maximum Gain will put a limit on how loud the output can get with very loud ambient levels, and Minimum Gain will put a limit on how soft the input can get with very quiet ambient levels. Maximum Gain cannot be set to a value less than the Minimum Gain, and Minimum Gain cannot be set to a greater than the Maximum Gain.

Continuous SPL Computer

Continuous SPL Computer listens to the ambient noise level and makes adjustments continuously. The Continuous SPL Computer can therefore be used in environments where there are no gaps in the program material. The module can also be used with program material that contains gaps, but it is recommended that the gap-sensing module be used instead. The gap sensing module is easier to calibrate, can make more appropriate adjustments according to crowd noise, and uses fewer DSP resources. Mono and stereo versions have the same controls. The only difference is that with stereo versions, stereo program material is adjusted. The gain is applied equally to both channels.

Inputs: In or In-L and In-R. Program material inputs. Connect your mono or stereo program material (foreground/background music or page) to these inputs.

Sense In: Sense microphone input. Connect a signal from your sensing microphone to this input.

Freeze control signal input: This control signal input that can be used to inhibit SPL calculations and hence freeze the SPL gain at the current level. When the Freeze input is at or above 50%, the SPL gain will be frozen (though changes to the Output Trim will still take effect). This feature may be useful for
example in some paging applications to prevent gain changes during a page. If this feature is not required, this input may be left open.

Outputs: Out or Out-L and Out-R. Program material outputs with SPL volume applied. Connect these to your main outputs.

Note: Any equalization, dynamics processing, or level changes to the program material must be done before it is sent to the SPL computer. The SPL module should be the last thing in the signal processing chain. The gain stage from the SPL computer to the speakers must remain constant after calibration. This includes analog output block gain settings, power amplifier levels, speaker attenuators, etc… User-controlled loudspeaker attenuators should not be used when a continuous SPL computer is present.

Sense Out: This is the signal that the module is using to sense ambient level changes. It is the same signal as Sense In except filtered by a voice-band (300Hz – 4kHz band-pass) filter. This signal may be used as a diagnostic to hear what the module is hearing.
 

Control: The control signal reflects the dynamic gain change being applied by the module and changes in real time. This control signal can be used to easily create linked multi-channel dynamics modules.
 

Note: The control signal is scaled so that 100% represents the maximum gain that can be applied by the SPL Computer.

Controls & Indicators:

Gain: The current gain being applied by the module (assuming it is Active) including the output trim is displayed at the top of the window.
 

Note: Before calibration, the display shows Gain: Not calibrated.
 

Maximum Gain: dB that the SPL computer is allowed to apply to the signal. Use this to put a limit on how loud the output can get with very loud ambient levels. Adjust using the slider or click in the text entry box to specify a numerical value.
 

Note: the Maximum Gain cannot be set to a value less than the Minimum Gain.
 

Minimum Gain: dB that the SPL computer is allowed to apply to the signal. Use this to put a limit on how soft the input can get with very quiet ambient levels. Adjust using the slider or click in the text entry box to specify a numerical value.
 

Note: the Minimum Gain cannot be set to a value greater than the Maximum Gain.
 

Gain-Sense Ratio: Controls the change in gain versus the change in the ambient level. Setting this to 1.0:1 means that for every 1dB increase/decrease in the ambient noise level, the SPL’s gain is increased/decreased by 1dB. Higher values such as 1.5:1 mean that the gain is increased by more than the ambient increase, which allows “out-shouting the crowd.” A setting of 0.5:1 means that the gain is only changed by 0.5dB for every 1dB ambient change giving a more subtle effect. Use the lowest setting that works for your application, and use caution with gain: sense ratios above 1:1, since these are more likely to be unstable.
 

Up Speed: Controls the rate at which the module increases the gain, specified in seconds. Longer times can cause a very gradual fade up in response to increasing ambient levels. Technically, the time indicates how long it takes for the gain to change by 10dB, e.g. a setting of one second means a 10dB/ second gain change.
 

Down Speed: Controls the rate at which the module decreases the gain, specified in seconds. Longer times can cause a very gradual fade down in response to decreasing ambient levels. Technically, the time indicates how long it takes for the gain to change by 10dB, e.g. setting of one second means a 10dB/second gain change.

Note: The settings of Up and Down Speed also have an effect the averaging time of the module. The lesser of the two speed settings is used to control the averaging time.

Link button: When depressed, this button links the up and down speeds so that they can be moved together. This button only applies to changes made from Composer, not from external control. In many installations the up and down speeds will be the same. Separate controls are provided for situations where the noise level changes asymmetrically, e.g. a train terminal that fills slowly, but empties quickly.

Trim: This parameter allows for manual gain adjustments. This output trim is applied on top of the gain that the SPL algorithm dictates, i.e. the actual gain applied is the SPL calculated gain plus the output trim, limited by the maximum and minimum gain applied is the SPL calculated gain, plus the output trim,
limited by the maximum and minimum gain settings. The effect of the output trim is indicated in the Gain display. Before and during calibration, this setting is ignored and treated as if it were set to zero. When the SPL calculations are frozen, changes to output trim will still take effect. If you need to give an end
user some control over volume with an SPL computer module in use, this is the place to do it.

Active button: When engaged, this button activates the module so that the gain adjustments are applied to the audio in the signal path. When inactive, SPL calculations are inhibited, just as if the Freeze control input was on.

Sense controls:

Sense Level Meter: This meter displays the current RMS level of the sense input as a reference.

Calibrate button: Press this button to initiate the calibration process. The SPL Calibration Wizard opens and steps you through calibration.
 

Sense Statistics: shows the highest and lowest sense values logged since the last reset. Sense values are logged continuously, and this value represents the raw sense input consisting of both ambient noise and program material pickup.
 

Reset button: Clears these values. Use this feature to monitor the ambient noise levels in a room over time.

How the Continuous SPL Computer Works:

Since a continuous SPL computer makes adjustments while the program material is playing, it must be able to distinguish what is program material and what is ambient noise. In an ideal world, the SPL sense input would be a signal that contains nothing but ambient noise.. However, in the real world, physical
constraints make this impossible. To determine how much of the signal at the sense inputs ambient noise vs. program material a calibration procedure is used. The calibration procedure makes a known change in the program level and measures the corresponding change in the sense input level. By comparing these values it determines the feedback gain, that is, the gain between the SPL outputs and the sense input. (For this reason, the gain between the SPL module outputs and the sense input should remain constant during and after calibration. Remember, a trim control on the Continuous SPL Computer allows for end
user/integrator program material gain changes that will not negatively affect the calibrated operation) During normal operation, the SPL computer algorithm monitors its own output signal, and knowing the feedback gain, it knows how much of its own signal it is receiving back at the sense input. After accounting for that, the remaining signal at the sense input is the ambient noise component.

Sense Input Considerations:

The sense input is typically generated by one or more microphones. If multiple microphones are used, mix them together before sending to the sense input. The microphones and speakers should be configured so that the microphones hear a maximum of ambient noise and a minimum of program material. Directional microphones with the speakers in their rejection axis may be useful. While the
algorithm can discriminate between noise and program material within reason, the less program material pickup there is, the more accurate and stable operation will be. In the perfect situation with 100% ambient noise and 0% program material, gain adjustments will be perfectly accurate and stable. If the program material becomes louder than the ambient noise, the module’s operation is degraded. It is
recommended that the program material level be equal to the ambient noise level, or at worst be no more than 6dB louder than the ambient noise level for proper operation. In an extreme case where the sense microphone picks up nearly 100% music/page (0% ambient noise), the module will not be able to extract enough information to use for gain control and calibration will fail.

Continuous SPL Composer Calibration Procedure:

Composer steps you through calibrating the SPL Computer module. Calibration needs to be performed before the module will function. Before calibration, the module applies unity gain and the gain display shows “Gain: Not calibrated”. (Gain: Not calibrated displayed on the Continuous version only.)

Calibration should be performed when the ambient noise level is at a typical volume. This noise level will become the unity gain point, i.e. the ambient noise level at calibration corresponds to unity gain by the module. Also, must have valid signals at both the “In” and “Sense In” inputs to calibrate. Make sure the
microphone is working and generating a reasonable level, which can be verified using the Sense Level meter. Also make sure that there is typical program material preset as well. If program material is not available, use a pink noise generator as the program material. (In fact pink noise is a very good signal to
calibrate with since it contains a broad range of frequencies and is at a constant level.) The program material should remain at a relatively constant level during the calibration procedure. To initiate calibration, press the Calibrate button. The SPL Calibration Wizard opens, then walks through the calibration process. The first screen allows canceling out of the process if calibration was started accidentally. Hitting the Start button begins the calibration in earnest. The first thing that happens is
that the gain is set to unity and measurements of the program and ambient noise levels are made. After this, adjust the Maximum Gain slider to the desired setting. You will hear the program audio increasing as you increase this setting. You should set the value to at least +6dB for the calibration procedure to work.
(This maximum gain can be adjusted later if necessary.) After the maximum gain setting, hit the Next button. If calibration succeeds, a success message will be presented. If it fails, the reason for the failure will be listed. Some common examples are insufficient level at either the program or ambient inputs.

Calibration Step 1

Calibration Step 2

Calibration Step 3

The ambient noise level should remain constant during calibration so that the change due to the increased program material can be clearly heard. If during calibration, the crowd starts yelling or someone drops a platter of dishes, recalibrate! Once calibration is performed, the calibration data will be retained even through re-downloading to the hardware. If the unit is due to be power cycled, be sure to check the “Last Known Operating State” bubble under Power On Control States in the Tools/Site Preferences dialogue of Designer, in order to retain the calibration settings through the power cycle. In Composer go to Tools->Site Preferences and check the “Last Known Operating State” option for the Power On Control States section.

Tips for Difficult Situations:

Great efforts have been made to ensure that the Continuous SPL Computer module does not run-away, that is hear its own signal, increase the gain, leading to hearing more of its own signal, etc. until the gain becomes stuck at maximum. However, in difficult situations where the sense input contains a very large portion of program material, some instability may result. If the gain changes unpredictably or tends to get dramatically too loud or soft, try these tips.

• Re-calibrate. If anything in the sound system has changed since calibration, this may be throwing off the algorithm. Re-calibrating with normal ambient and program material levels will often fix the problem.
• Adjust the microphone/loudspeaker placement to minimize program material pickup. Fixing things acoustically is usually the best remedy. Switching to a directional microphone or adding sound-absorbing material may help. Get a rough idea of what the sense input is hearing just by listening to the Sense Out of the module. If there is a lot of program material and very little else, that is a sign of trouble. Also, at the end of calibration, some statistics are presented that give a more exact indication of noise vs. program material pickup.
• Use a lower Gain-Sense ratio. Lower ratios mean less potential for wild gain changes. Ratios above 1:1 should especially be used with caution. The default of 0.75:1 is good for many applications. Use the lowest ratio you can get away with.
• Use slower speeds. Using larger times for the up and down speeds causes the module to average out level changes over a longer period of time. The slower the speed, the more stable the module will be. Use the slowest speeds that can get away with. Also, if the module is tending to increase gain too much, try making the Down Time shorter than the Up time.

Continuous SPL Technical Notes:

If the calibration procedure succeeds, some statistics are presented showing parameters measured during calibration. These are intended to be diagnostics to help tech support solve problems, but a brief description is given here for the curious. The Music/page level and Noise level values tell how much of the sense signal was calculated to be program material (music/page) vs. ambient noise during the unity gain portion of calibration. Ideally, the noise should be louder than the music page, and the greater the difference the better the module will work. This number can give an indication of how well the microphones are placed to hear the ambient noise and reject the program material. A feedback gain parameter in dB is also presented. This shows the overall “loop gain” from the SPL module output, through the loudspeakers, through the sense microphones, and back to the SPL module sense input.

If the calibration procedure succeeds, some statistics are presented showing parameters measured during calibration. These are intended to be diagnostics to help tech support solve problems, but a brief description is given here for the curious. The Music/page level and Noise level values tell how much of the sense signal was calculated to be program material (music/page) vs. ambient noise during the unity gain portion of calibration. Ideally, the noise should be louder than the music page, and the greater the difference the better the module will work. This number can give an indication of how well the microphones are placed to hear the ambient noise and reject the program material. A feedback gain parameter in dB is also presented. This shows the overall “loop gain” from the SPL module output, through the loudspeakers, through the sense microphones, and back to the SPL module sense input.

What You Should Not Do After Continuous SPL Calibration

Do not adjust the level of any volume controls in the sound system signal path following (downstream from) the Continuous SPL Computer module. Doing so will cause the module to yield erroneous results. This includes controls both inside the unit and those outside of it such as equalizer settings, amplifier input level controls, wall-mounted L-Pad style speaker attenuators, etc. Also, do not insert compressors into the signal path after the Continuous SPL Computer module. If adjustment is needed for amplifier input levels or equalizer settings, re-calibrate the module afterward so the module can adjust to the new levels. If trim is needed for the overall system level up or down after the calibration, the preferred method is to adjust the module’s Output Trim control, through adjusting the signal level upstream from preferred method is to adjust the module’s Output Trim control, though adjusting the signal level upstream from the Continuous SPL Computer module works as well. Do not adjust the gain downstream from the SPL module! Obviously, user-controlled speaker attenuators should not be used with a Continuous SPL computer. Similarly, equalizers after the SPL module should remain fixed after calibration. If adjustable equalization is needed, e.g. treble/bass controls, place the filter modules prior to (upstream from) the SPL computer so it knows about them.

Things to Consider for Both Gap-sensing and Continuous SPL Computer Operation:

Location of Microphone:
Much more important than the type of microphone is its location. The sensing microphone needs to “hear’ the ambient sound within the controlled space. It is vital that the microphone is placed where it primarily picks up a majority of noise rather than the paging or music that is going through the system. Do not locate the sensing microphone near a localized noise source that is not typical of the ambient noise level of the controlled zone. For example, the noise from a large machine of some sort, a kids play area, or a video game, may cause the module to think that the zone is noisier than it really is. The best sense mic placement ensures that the majority of the signal picked up by the sensing microphone is ambient noise. As mentioned above, some programs material pickup is acceptable, but the less there is, the better the module will work.

Signal Path Considerations:
The best place to put the SPL Computer module in the module signal chain is that the very end, right before the output. If this is not possible for some reason, make sure the signal path after the SPL computer is fixed after calibration. For example, if a graphic EQ is used to flatten room response,
complete the EQ tuning procedure before calibrating the SPL computer module and then leave the EQ controls fixed afterward. Speaker protection limiters are OK as well, as long as they are set up so they are rarely reducing the gain (i.e. they are not consistently being driven to the point of limiting).

When to choose Gap-sensing vs Continuous SPL Computers:

Gap-sensing SPL Computer modules only listens to the ambient sense microphone during program material gaps and is best suited for a restaurant or bar type environments. The SPL Computer module will use the ambient noise level of the room between program material and then increase or decrease the
overall program level. Since an averaging count can be set to determine the gain change applied by the SPL Computer, the program material gain changes can be set to change with the trending ambient noise SPL levels rather than adjusting to a specific ambient noise measurement. For instance, the averaging
count could be set to 5 to insure a honking horn next to a restaurant in NYC does not cause the SPL computer to raise the program material gain to it upper limit just because the Gap-sensing SPL Computer took a sample of ambient noise at the time of the honking car.

Continuous SPL Computer modules make adjustments while the program material is playing and is best suited for a stadium type environment where ambient noise level is constantly changing while the program material is playing.

This document describes an easy method for controlling select Juice Goose iP-series power management devices from any Symetrix Composer-based DSP. Using the Super-modules included in Composer 5.6 and later, it is possible to control each outlet individually, all at once, and sequence each up or down. In addition, any standard Composer control can be used to trigger the power conditioner, such as:

• ARC remote wall panels
• ARC-WEB
• SymVue
• Control Server
• Preset recall
• Event Scheduler
• External Control Inputs (GPIO)

This Tech Tip is relevant to the following Juice Goose models:

• IP 1500 Series
• IP 1
• IP PD1-4

The Super-modules can be found in the \\Documents\Composer x.x\Super-modules\examples\3rd Party Control:
Juice Goose iP 1.smfx
Juice Goose iP 1500 Series.smfx
Juice Goose iP PD1-4.smfx

This example uses the IP 1520 model.

1. First, enable TCP control on the Juice Goose Management interface.

1. Make note of the Juice Goose IP address, as this will be needed in a later step. You may wish to configure a static IP for longevity in permanent installations.
2. Import the appropriate Super-module into a Composer Site File.

4. Add the appropriate Super-module to a Symetrix DSP’s Design:

1. Double-click on the Super-module to view its user interface, and copy/paste the Juice Goose IP address into all fields (up to 24x):

6. Push your Site File to the system.

The purpose of this Tech Tip is to provide information on creating SymVue Dialer Control Screens for both the 2 Line Analog Telephone Interface Card and 2 Line VoIP Interface Card. Step by step instructions will be given on how to create the Control Screens and export them to SymVue.

SymVue is a real-time user control panel application that displays Control Screens exported from Composer functioning as a multiuser, multi-point control environment for Symetrix systems.

SymVue runs on any Windows XP or newer compatible device, including touch screen enabled PCs and tablets. The computer communicates directly with Symetrix hardware over a network connection. The desired user control interface is created in Composer as a Control Screen then exported to one or many Windows devices for tailored operation of the Symetrix system.

The Input Modules for both the 2 Line Analog Telephone Interface Card (ATI) and 2 Line VoIP Interface Cards can be exported to Control Screens. These Control Screens can be used to provide remote control interfaces (Dialers) for the ATI and/ or VoIP cards without the need or use of complicated 3rd party control systems. SymVue Dialers can be custom tailored to perform any or all of the functionality of the ATI and VoIP modules. These functions can include, but are limited to:

• Detect and answer incoming calls
• DTMF tone dialing
• Speed-dialing (edit and recall)
• Redial
• Do not disturb
• Caller ID
• Call transfer
• Call hold
• Call reject
• Local three-way audio conferencing
• Conferencing and splitting of call appearances

Here are some examples of the different styles of Dialers that can be created:

Instructions

1 Make sure the ATI or VoIP Interface Card has been properly installed into the Radius AEC or Edge Hardware. 

install

Once the card has been properly installed, the Input Modules will appear on the Design View screen of the site file.

Note: The Input Module will reflect the card slot location (A, B, C, or D). The SymVue Dialer being created will be linked to that specific card slot.
Note: SymVue Dialers can be created without having the ATI or VoIP card installed. Simply right-click the Radius or Edge in the Site View screen of the site file and select “Configure I/O Cards”. Then select the correct card for the specific card slot.

2, Double click and open the Input Module for the ATI or VoIP Interface.

3. Right-click on an open section of the module and select “Copy Entire Layout to Control Screen”.

4. Select “New Control Screen”, unless a Control Screen has already been created and it is being added onto.

Note: individual pieces can be selected by right-clicking on the desired piece (i.e. button or fader)
The pre-built example SymVue Dialer has been tailored to use buttons instead of faders for volume control. A “2 Button Momentary” module is used connected to a “Button Ramp” Super Module (available in Super Module Tools folder). The Super Module is then connected to “Output Control Number” modules. The control numbers used by the “Output Control Number” modules are assigned to the volume fader. The “On” buttons for the “2 Button Momentary” module are copied to the control screen.

5. The functions of the Input Module have now been copied to the Control Screen and can now be tailored for specific look and operation.

export

6. Once the Control Screens have been created go to Tools>Control Screen
Manager and export the Control Screens to SymVue.

For additional information on creating SymVue Control Screens click here.

Due to the inherent nature of touchscreens, the use of momentary buttons on control screens in SymVue may result in some unexpected behavior – when the user touches a button on the screen, the “on” action isn’t sent to the DSP until the user actually releases their finger from the button. This can make the use of momentary buttons somewhat confusing for the end-user, in that a swiping action is required to trigger them on touchscreens. For some users, simply letting them know that a “swiping action” is required is good enough. For others, a workaround may be needed to give them expected touch functionality. Fortunately, this default touchscreen behavior does play nicely with latching buttons.

We’ve outlined a procedure that uses latching buttons in the place of momentary buttons as triggers – these latching buttons ultimately will act as if they are momentary buttons. This is accomplished by the use of a single Preset Trigger module which is triggered every time a latched button is pressed. It fires a “button off” preset to reset the state of the latching buttons to their off state. The preset is fired so quickly after the touch that the latched button appears to act as a momentary button.

For most applications, this workaround will do the trick nicely. The only drawback to creating a momentary button in this fashion is that you cannot hold it down. Therefore, this process is best used for controlling modules that are triggered via an impulse, such as preset triggers.

1. Drag the following modules into the design from the Toolkit (all are found
under the Control Modules heading):
a. 8 button Latched
b. 8 Input Logic
c. Delay Logic
d. Preset Trigger

2. Wire them up as below:

3. Next, take a snapshot of the latched buttons in their off states. Double-click the 8 button Latched module to bring up its GUI. Making sure the button is in its off state, right-click directly on the first button and store it to an un-used preset of your choice. Repeat for the rest of the buttons, making sure to use the same preset number for each.

4. All buttons should appear as below (with whichever preset number you chose). If the green indicators are not appearing over the buttons, go to the Tools menu in Composer and be sure “Super-impose Assigned Controller Numbers” is checked.

5. Open the 8 Input Logic module and set the logic operation to OR.

6. Next, double-click the Delay Logic Module. Set its delay time to .08 seconds and its hold time to .01 seconds.

7. In the Preset Trigger module, enter the preset number from step 3.

8. Wire in some modules to be controlled. In this example, Preset Triggers are used.

9. The 8 Button Latched module contains the buttons to be controlled from SymVue. Re-open this module in Composer and copy the “On” buttons over to a new or existing control screen. These buttons will now function without the need for a “swipe” motion to engage them.

The title of the tech tip says it all. Simply put, having all the DSP in the world is no substitute for proper gain structuring in an audio installation. This is because the gain structure is single handedly responsible for maximizing the dynamic range between the program audio and the noise floor. When the gain structure is set incorrectly, even the best audio equipment with unlimited DSP resources will have audible noise ranging from annoying to unacceptable by the end user. If the gain structure is set correctly, the noise floor should be completely inaudible to the human ear.

The gain structure could be defined as the relationship between various gain stages in the audio system. In a Symetrix DSP system the gain structure is composed of various gain stages within the DSP, the output level of the sources feeding the DSP inputs, as well as the analog input trim on the amplifier. As such, it is important to have a clear understanding of how to correctly adjust each gain stage in the DSP as well as the input trim of the amplifier in order to maximize the dynamic range between the program audio and the noise floor.

When properly adjusting the gain structure, it is important to step through each gain stage, starting at the beginning of the signal path and working to the end.
 

This is important to note because an older line of thinking, which is responsible for noise in many audio systems, was to start by turning the input trim of the amp to 100% and working backwards through the various gain stages turning them down to compensate for the amp. This method kept unwanted hands from adjusting the amp after the system was tuned, but it also maximized the noise in the system by increasing the noise floor at the amplifier.

Typically there are 3 digital gain stages in the DSP software: input gain, end user gain control, and output gain. Additionally, there may be 2 analog stages outside of the DSP; source gain and amp input trim (depends of the brand of amp), which may need adjusting. Prior to following this step by step tutorial, all gain stages should be left at 0db, which also known as “unity” as this setting does not attenuate the audio up or down.

This tech tip will step you through 9 simple steps to proper set the gain structure in your next audio installation.

Step 1:
Turn off the amp, as it is not necessary to hear the audio when adjusting the first few gain stages.

Step 2:
If needed, set the analog gain of the device feeding the Symetrix DSP input. In most cases the level may not be adjustable; it may be statically set as a mic or line level output. (See the unit’s documentation) For instance, a CD player is a line level output that often has an unbalanced (RCA) connection to the DSP. In the case of a device such as an external mic pre, there may be some gain adjustments that need to be made. In such a case turn the gain up as high as possible without the audio clipping during use. (more on clipping in step 3).

Step 3:
Determine whether the source feeding the DSP input has a line level or mic level signal and make the corresponding selection at the input section of the appropriate software. With the Zone Mix 761, Jupiter, and Solus hardware only mic or line is selected at the input stage. In most SymNet Designer and all Symetrix Composer hardware, 5 selections are available at each input. Line level signals can be balanced (+4dBu) or unbalanced (-10dBV) and have respective settings. Mic level can be -20dBu, 40dBu, and -50dBu. Phantom power should also be turned on when a condenser microphone is used. Radius
units are set to “Switch Mode” and the Dante ports are daisy chained between devices.

gain 1

Zone Mix 761

gain 2

Composer

Step 4:
Adjust the input trim in order to maximize gain at the input. Rule #1 is that the input should never clip, which is indicated by the red in the meter. Some audio meters, such as those in the Jupiter software have a range of -48dBu to +24dBu. In SymNet Designer and Symetrix Composer the audio meters are in DBFS and range from -72dbfs to 0dbfs. Clipping is at +24dBu and 0dbfs respectively. When the audio is clipping the signal will be distorted and will almost certainly sound bad. Even worse, clipping audio has the potential to damage hardware including the amps and speakers.
A general rule is that the program audio RMS level, or average level, should reside in the amber portion of the meter while at the same time not clipping. Depending on the meter style, this is somewhere around unity, which is +4dBu or -20dbfs. This setting usually leaves sufficient headroom between the
program audio and the point of clipping so that louder portions of the program audio do not clip, even when the user has the system turned up loud. If clipping does occur, then the input trim should be turned down until the audio stops clipping.

gain 3

Zone Mix 761 Input meter

gain 4

Composer Input meter

Step 5:
Determine which gain control will be given to the end user. Most often this gain control will be in the “middle processing”. Gain adjustments before the input processing can negate or skew processing such as, but not limited to; Compression, AGC, Limiting, and Feedback Elimination. Gain adjustments
after the output processing are not protected by processing such as the Hard Limiter. As such, end user gain control should typically be located between the input and output processing in modules such as a Mixer, Automixer, Room Combiner, or a Matrix.

Once this gain stage has been located, turn it up to +12dB. This will be the “loudest” setting possible that the end user will be able to set the system to. By tuning the system to this “loudest” gain setting, the end user will never be able to turn up the system to the point it causing clipping or damages the amp or speakers.

gain 5

Zone Mix 761 zone volume

gain 6

Composer gain module

Step 6:
Go to the analog output section and set the “output level” to +4dBu for balanced connections or
-10dBV for unbalanced connections based upon the input type of the downstream device. These downstream devices vary depending on application but can included hardware such as an amplifier, assisted listening system, or media recorder.
 

Step 7:
It is almost time to turn on the amp, however, before doing this we want to do one of two things;
1) turn the amp’s input trim to the lowest setting or off
2) if there is no input trim on the amp, use the output gain in the DSP to turn the audio extremely low or completely off.
 

This will prevent the speakers from being damaged with audio, which is at its loudest setting due to the end user gain control being set to +12dB, from suddenly playing when the amp is turn on.

Step 8: (amp does not have an input trim)
Turn on the amp and using the output gain fader, turn up the system until the audio is audibly the loudest it should ever be.

Step 8: (amp has an input trim)
First, optimize the DSP output by adjusting the output gain fader if needed, similar to how the input was tuned in Step 4. In other words, turn the output gain up so that the audio is as loud as possible on the meters without clipping, such that the RMS level of the audio resides in the amber. If the output meter indicates clipping, use the output gain to attenuate the level down until the audio stops clipping. Finally, turn on the amp and using the amp’s input trim, turn up the input trim until the audio is audibly the loudest it should ever be.
 

Step 9:
Now return to the end user gain control in the middle processing and adjust it to the appropriate level for the current conditions. If the above 9 steps were followed, the system should have the maximum dynamic range between the program audio and the noise floor, not to mention that even without audio playing the noise floor should be inaudible. Additionally, the customer can be given access to the end user gain control without the possibility that the gain can be turned up any louder than the loudest setting that was determined in Step 8. This means the customer cannot accidentally damage the system by turning it up too loud.

The purpose of this document is to provide updated information on the Push, Pull, and Dante Device Sync features for Composer 6.0 and later.

When pushing a Site File to a system with versions of Composer prior to 6.0, Composer would configure the Dante routing and naming of all the devices in the Site File. When pulling a Site File, there were two options; Synchronize to Changes or Abandon Changes. These were all the run-time changes made since the last time that the Site File was archived. Choosing “Yes” would bring those changes into Composer. Choosing “No” would bring the archived Site File into Composer as it was last archived.

Each section below explains how the features are currently handled.

Push

When pushing, Composer can either configure the network audio (Dante) or let Dante Controller manage it.

Composer’s Site Preferences window contains a check box to Configure Network Audio. When the box is checked, Composer will configure the network audio routing and naming. If the box is unchecked, all network audio routing and naming is managed with Dante Controller.

Push

By default, the Site Preferences window will automatically open for push confirmation. Site Preferences are also located in the Tools menu.

Pull

When pulling a Site File from an existing system into Composer, there are two options; Synchronize to Changes or Abandon Changes.

It is common to modify events and/or control positions while the system is online and since the Site File was last archived (pushed). If the file is open without choosing to synchronize to these changes, the file pulled will not reflect the exact state of the online system. The pulled file will be a copy of what was last archived.

Here is a list of all the options that may be selected when synchronizing changes. All options are selected by default.

Pull

• Presets
• Events
• Control Values
• Dante routing and naming

Dante Device Sync

When locating a Dante device in Composer, there is an option to synchronize the design to the device’s Dante configuration. This sync is performed separately and is not affected by the Configure Network Audio setting in the Site Preferences.

Here is a list of all the options that may be selected when synchronizing changes. Device name and channel labels are selected by default.

sync

• Device Name
• Channel Labels
• Routing/Subscriptions

confirm

Once the Dante hardware has been selected, the Sync Confirm window will open. From this window you may choose which specific features of the located hardware are synced with Composer. All features are selected by default. Remove checks from any unwanted sync features.

Overview

This Tech Tip provides information and instructions pertaining to the combined Analog and Remote Mic Switch Super-Module that can be located in the Tools folder of the Super-Module Library within Symetrix Composer software. Composer is an award winning CAD program used to create site file designs perfectly suited to each and every application.

This Super-Module can be used to combine the mic switch operation from an analog input and remote control device (ie, ARC remote, ARC-WEB, or third-Party control). Normally when using the external analog control inputs, the analog control values will supersede and override software control. 

com 1

This Super-Module will allow a single mic switch to be controlled from the analog input and software. The Super-Module also will allow for the switch or mute state to be viewed by the remote control device and drive an external control output (ie, LED).

Implementing this Super-Module

1. Wire a 1 Button Momentary module to the Mic input of the Super-module.

2. Assign the “On” button of the momentary button module to the analog mic switch.

3. Wire a 1 Button Latched module to the Remote input of the Super-module.

4. Assign control number to the “On” button of the 1 Button Latched module. This control assignment will be used for the remote control assignment of the remote controller (ie, ARC remote, ARC-WEB, or third-party control).

5. Click the “On” button of the 1 Button Latched module so the button is on and save the on state to a preset. Label this preset as Mute. This example used preset #1.

6. Click the “On” button of the 1 Button Latched module again so the button is off and save the off state to a preset. Label this preset as Unmute. This example used preset #2.

7. Open the Super-module design and make sure the Mute preset trigger matches the preset that was assigned to the 1 Button Latched module. This example uses Preset #1 for Mute.

8. Make sure the Unmute preset trigger matches the preset that was assigned to the 1 Button Latched module. This example uses Preset #2 for Unmute.

9. Wire a Logic Output module to the Out/R of Super-module. The Logic Output could be a Local Logic Output or Remote Logic Output. This output will be used to drive the red LED of the analog mic switch. This example uses Local Logic Output #1 for the red LED.

10. Wire a Logic Output module to the Out/G of Super-module. The Logic Output could be a Local Logic Output or Remote Logic Output. This output will be used to drive the green LED of the analog mic switch. This example uses Local Logic Output #2 for the green LED.

11. Open either a Gain Module or Automixer used in the signal processing and routing of the microphones.

12. Right-click on the channel Mute button.

13. Select “Set Up Remote Control.”

14. Select “Control Signal Assignment” for the Remote Control Device. Then click “Select.”

15. Expand Combined Analog & Remote Mic Switch.

16. Select Out/R and click OK.

17. The mute button of the Gain or Automixer will now be controlled by the state of the red LED of the Super-Module. When the red LED is active, that channel will be muted. When the red LED is inactive, the channel will be unmuted.

Mute

Indication of a microphone’s status can be accomplished in just a few quick steps by employing simple logic and control signal assignments. This programming can be used in courtrooms or public speaking venues to indicate, on a control screen for example, when certain microphones are active or un-muted.

In this example, all we’ll need is the microphone channel, represented by a gain module, a one button momentary module, and a two input logic module.

First, open up the gain and momentary button modules and place them to clearly view both.

Next, right click on the mute button in the gain module and select set up to remote control.

Select control signal assignment. The intention is for the gain module’s mute button to take its queue from the momentary button’s “ON” button.

This could also be accomplished by assigning both buttons the same control number. However, then either button could affect each other. Doing it this way only allows the mute button to follow the “ON” button.

From control signal assignment, click the select button. Now expand the one button momentary folder and select Button 1. Click ok.

Now whenever the “ON” button is active, the gain module’s mute button will be active. Click ok in the remote control wizard.

Open up the 2 input logic module and move into a place to be seen. Change the logic mode to “OR”. This will allow either input to make the module True.

Right click on the False LED and choose copy false LED to control screen, either one already built or a new one.

With the LED selected, change the display type in the properties panel to symbol and then resize the LED boundary box to fit just the light.

Because the mute button takes its queue from the “ON” button, when the “ON” button is active it makes the logic module true. If we wanted to use the True state LED for the indicator, we would need to change the logic inputs to “INVERT”.

We could be finished here if we put this LED next to a fader control or only needed a light indication. We can, however, go a step further and offer some label indication as well.

Right click on the momentary “ON” button and copy it to the control screen. Resize it to fit the words “mic active” and “mic inactive”, or whatever verbiage makes the most sense for your installation.

Now, in the properties panel, change the transparent parameter to True in the background color area. Then in the Text area, change the “ON” text to “Mic Inactive” and the “OFF” text to “Mic Active”. Finally, change the text color to black.

Now, when the “ON” button is in the disengaged state it will display “Mic Active” because the channel will be un-muted. When the button is in the engaged state it will display “Mic Inactive” because the channel will be muted.

Logic programming doesn’t work when not online. Push to go online and test this programming.

This programming would work the same way when incorporating any of the button processor super modules.

Be careful to assign the control signal from the correct module using the enumerator label.

Note the super module also has an input logic module with a false LED.

Many systems include an emergency mute function so external emergency sirens are easier to hear. These control systems tend to be hard-wired with a physical switch. Some systems may also include a paging system that can require hardware “and” software control.

In Composer-based systems, analog, or hardware control will always take priority over software control which can make things difficult when trying to combine an analog microphone switch along with a button on a control screen. However, using some simple logic we can combine both hardware and software control.

The purpose is to have the master mute always active unless the microphone analog switch or a control screen button are engaged, essentially creating a push to talk control. Open up your site file and drag in a one button momentary and a one button latched from the toolkit under control modules, control inputs.

Then drag in a two input logic module from the control logics folder. Finally, drag in a one output remote control number module from the control outputs folder, and wire all modules as shown.

Open up the two input logic module and change the logic mode to OR. Now, open up the remote control number output module and the master gain-sharing auto-mixer modules.

Right click on the master channel mute button and select set up to remote control. Assign the mute button an unused control number and click ok. Then enter that same control number into the remote control number output module. This examples simply uses control number one, but yours should reflect a currently unused number.

Next, open up the two button modules. Right click on the on button in the one button momentary and select set up to remote control. Choose local analog input, then choose an available analog switch input. This example uses switch two-a because one-a is already used by the emergency system mute. Click ok to save the assignment.

Right-click the on button in the one button latched and choose copy channel one latched button to control screen. This example will place this button on the control screen for zone one. Re-size and label the button as necessary, and that’s it.

To summarize what is happening. The microphone analog switch is connected to the momentary button so while it’s active, the momentary button in Composer is active. The latched button is copied to the control screen and is a direct copy of the latched button in the programming, they are active at the same time. The two input logic being in the “or” state means that if “either” the latched “or” momentary buttons are active, the logic is true.

Currently the logic button is sending 100% control signal out of the False node, which activates the master mute button in the auto-mixer. So, activating either the microphone or control screen button will change the logic to true, which then changes the false output to 0%, de-activating the master mute button.

Logic function does not work in Composer while not online. Push to go online and test your programming.

This Tech Tip features the Ducker Module in Composer. It also covers the differences between how it was used in SymNet Designer versus Composer.

A ducker is used in a scenario where one source (the side-chain) needs to “duck,” or lower, the volume of second source (program audio). Most often a ducker is used in paging applications to lower background music when a paging mic is used. However, a ducker can be used anytime sources need to be prioritized.

When the “side-chain” input senses audio, the Ducker will lower the program audio by a user-defined amount, known as the “depth” (20 dB in this example). After the side-chain input no longer senses audio above the threshold, the program audio will come back up to its previous level after a specified amount of time (Hold and Release settings). The ducker depth can be set to lower the program audio partially or completely depending on the application.

Occasionally, in SymNet Designer, there was confusion due to the fact that the side-chain input only triggered ducking of the program audio, but did not mix the side-chain back into the program audio. In other words, a page would cause the BGM to lower, but the page was not heard over the top of the BGM. In order to hear the page over the top of the BGM, the page/side-chain input needed to be mixed or summed to the output of the Ducker. Additionally, to control the level of the page/side-chain relative to the BGM/program audio, a gain stage needed to be added to the side-chain signal path.

Here is a programming example of a Ducker being used in SymNet Designer.

In Composer, using a ducker has been greatly simplified. The Ducker module has built in side-chain mix controls. By default, the sidechain input is muted. To mix the side-chain signal back into the program audio, simply turn off the mute and adjust the level for which the side-chain should be mixed into the program audio.

Here is a programming example of a Ducker being used in SymNet Composer. Notice the “Side Chain Mix” control which includes a volume fader and mute button. Additionally, the Ducker GUI now has a graphical representation of the parameters, useful for setup.

From the two examples, you can see how simple and intuitive it is to utilize the Ducker Module in Composer.

Applies to Radius NX, Edge, Prism xControl, Jupiter, and Zone Mix 761

This tech tip will explain how to properly integrate the Logic Outputs of the above DSP units into your installation. Typically these outputs would be utilized in a couple of ways – driving LEDs in order to give visual feedback to an end user, or controlling an external relay for switching other equipment, such as a projector screen or rack of other equipment. In order to do this is as seamlessly as possible, it is first necessary to know some basic facts.

First, each of these logic outputs is the open collector of a switching transistor that has its emitter tied to ground. What does this mean to you? These are not dry contacts that are simply open or closed. When the transistor is inactive, 5V is present at the logic output. When the transistor is activated, the 5V is shunted to ground through the transistor’s emitter, which results in 0V at the logic output.

Here are the specs for the logic outputs that we’ll be referring to in this tech tip:

• The logic output is pulled high (5V) when inactive.
• The logic output goes low (0V) when active.
• The maximum logic output source current is 10mA.
• The maximum external power supply voltage is 24 VDC.
• The maximum external power supply current sinking is 50mA.

How to Drive an LED

With a max output current of 10mA, it is possible to drive an LED directly from the logic output without needing a current-limiting resistor (there is an internal 500 ohm resistor). This of course depends on the forward voltage and forward current of the LED you choose (check the datasheet for your LED). In this case, simply connect as below:

If you have an LED that requires a higher voltage/current demand, an external power supply will be needed. As stated above, the max external power supply voltage is 24 VDC with 50 mA sinking current. Hook it up as below:

You can calculate the resistor’s value by using Ohm’s law:

Vs = Supply Voltage
Vf = LED forward voltage drop
I = LED forward current (in Amps)

Round up your value to the nearest standard resistor value.

Note: Various styles of LEDs (from standard through-hole to panel-mounted) in a seemingly endless variety of values are readily available. The best approach would be to identify your needs in terms of LED type, then use the extensive search functions of sites like Digikey.com or Mouser.com to see what is available.

Driving Relays

There are two types of relays we’ll work with to control external devices, the most common being a non latching mechanical relay. Taking into consideration the 10 mA output current of the logic outputs, this type of relay will typically need to have its coil driven by an external power supply. As noted earlier, the external supply should not exceed 24 VDC, while the relay coil current should not exceed 50 mA. A relay such as the Omron G5LE-1A4 DC12 should do nicely.

Take note of the flyback diode placed in parallel across the relay coil. This provides a path for discharge current to flow when the coil is switched off. Without this diode, there is the risk of damaging or destroying the internal transistor of the Symetrix device. Think of a flyback diode as the cheapest equipment insurance policy you’ll find anywhere. Use a 1N4004 or equivalent.

Another relay option would be to use a Solid State Relay (SSR), which typically has a lower current requirement for activation. Most installers use mechanical relays, but some of the advantages of SSRs are worth noting:

• Low turn-on requirements. There is no inductive coil to drive in an SSR. Instead there is an internal LED that toggles the relay, which typically requires very little current to turn on. If you choose one that requires less than 10 mA to activate, there is no need for the external power supply that you might need to power a mechanical relay coil.
• No mechanical wear-and-tear, arcing, or contact bouncing.

For a general use SSR, try a Panasonic AQV252G (max load voltage 60 VDC/VAC, max current of 2.5 A).

Triggering the Logic Outputs in SymNet Composer (Radius, Edge and xControl)

As a basic example, we’ll set up a logic output to be toggled on and off by an external device such as a Crestron or AMX controller.

1 In Composer’s Design View, drag in a single Latched Button from the Toolkit.

2. Drag in a “Local Logic Output #1” Module from the Toolkit. To use an xControl’s logic outputs, select the “Remote Logic Output” module instead.

3. Wire the output of the latched button module to the input of the logic output module.

4. Right-click the “On” Button in the latched button module and click “Set Up to Remote Control.”

5. Select “Generic Controller Number Assignment” from the drop-down menu. Either keep the “Auto-assign controller number” checkbox selected, or un-check to type in your own controller number. Click OK, then push the site file to hardware.

6. You will now be able to control the button with your external controller.

• To enable the button, send this command to the DSP: CS <CONTROLLER NUMBER> 65535 <CR>
• To disable the button: CS <CONTROLLER NUMBER> 0 <CR>

Be sure to download the Composer Control Protocol from our website for full command details.

Triggering Logic Outputs for Jupiter and Zone Mix 761

Use the “External Controller Wizard” in the software to walk through programming your logic outputs.

Limiting a fader to a specific dB range is easy within the Composer environment, whether using a Symetrix T-Series touchscreen, ARC-Series remote, or W-series remote. Third-party control, however, can be a bit more difficult if the third party doesn’t have their own inherent way to accomplish this task. Thankfully, with a small bit of logic circuitry we can emulate this range control. In this example, we are trying to limit the control of this gain module’s fader.

1. First, drag in a control fader from the Control Inputs folder in the toolkit.
   
    
2. Then, drag in a scaler from the control processes folder. Finally, drag in a one output remote control number module from the control outputs folder.
   
    
3. Open each module, including the gain, and set the windows to be able to view all.
   
    
4. In the one output remote control module, choose and enter an available remote control number. Then assign that same control number to the actual fader needing to be controlled.
   
    
5. Now right-click on the logic fader and set it up to remote control, choosing another available remote control number. Note the chosen control number cannot be the same as the gain fader.
   
    
6. We can’t see logic activity live while offline. If we push online we will see as we move the control fader from zero to 100% the gain fader moves as well. The point is to limit the effective movement of the gain fader in relation to the full scale movement of the logic fader.
   
    
7. We can allow the control fader to fully scale on the scaler’s input. But if we adjust the output to a desired degree, we can limit the movement of the gain fader. In this case we’ll set the low output to 50% and the high output to 85.7%. This will give us a range of 0 to -30 dB. The scaler out values can be adjusted to match your system requirements
   
    

The control fader is what would be set up for the end user to control in the programming. It in turn controls the gain fader where the actual audio is passing through. Be aware that the representation will be in percentage, not dB.

While the dB level box from the gain module can be set up to remote control, the third-party controller should be consulted about how it will represent this value.

Limiting a fader to a specific dB range is easy within the Composer environment, whether using a Symetrix T-Series touchscreen, ARC series remote, or W-series remote. Third-party control, however, can be a bit more difficult-if the third party doesn’t have their own inherent way to accomplish this task.

Thankfully, with a small bit of logic circuitry we can emulate this range control. In this example, we are trying to limit the control of a gain module’s fader.

First, drag in a control fader from the Control Inputs folder in the toolkit. Then, drag in a scaler from the control processes folder. Finally, drag in a one output remote control number module from the control outputs folder.

Open each module, including the gain, and set the windows to be able to view all.

In the one output remote control module, choose and enter an available remote control number. Then assign that same control number to the actual fader needing to be controlled.

Note the control number output and the gain module fader are the same control number.

Now right-click on the logic fader and set it up to remote control, choosing another available remote control number. Note the chosen control number cannot be the same as the gain fader.

Note the control fader has a different remote control number assignment than the gain module fader.

We can’t see logic activity live while offline. If we push online we will see as we move the control fader from zero to 100% the gain fader moves as well. The point is to limit the effective movement of the gain fader in relation to the full scale movement of the logic fader.

Note when the control fader is at 0% the gain fader is also at the lowest position, “off”.

We can allow the control fader to fully scale on the scaler’s input. But if we adjust the output to a desired degree, we can limit the movement of the gain fader.

limit 1

In this case we’ll set the low output to 50% and the high output to 85.7%. This will give us a range of 0 to negative 30 dB.

The scaler output values can be adjusted to match requirements of dB range(s).

The control fader is what would be set up for the end user to control in the programming. It in turn controls the gain fader where the actual audio is passing through. Be aware that the representation will be in percentage, not dB.

While the dB level box from the gain module can be set up to remote control for push value, the third-party controller should be consulted about how it will represent this value.

In Symetrix DSPs, the word “active” isn’t quite accurate to describe how presets work. Presets are simply snapshots of a given set of parameters in a given state and more recalled than they are active. This means that if a preset is recalled and one of those parameters is adjusted independently, the dropdown text box will still show the last preset recalled.

This Tech Tip will assume you already have some presets set up. In this example, there is a 6 channel gain module controlling zone volumes and a Matrix Mixer controlling source routing.

Then, drag in a Radio button from the toolkit in Control Modules > Control Inputs > # Button Radio. Label the selections accordingly. This example will not offer Preset 4 to the end-user as it would be intended for system admin use only.

Add the selections on the Radio Button to the appropriate presets.

Now that these radio button selections are included in the presets, copy the drop down box to the control screen.

Open the control screen and move preset trigger buttons onto the screen, placing them appropriately.

Now, copy the dropdown box to the control screen and resize it appropriately.

In the dropdown properties panel, set the Font, Horizontal Alignment, User Adjustable, Show Button, and Show Item List settings to match these.

Now, take the control screen out of edit mode and test the preset trigger buttons. Not only should parameters change according to the preset, but the drop down display should also change accordingly.

Remember, presets are simply snapshots of a given set of parameters and not “active”. If a preset is recalled and one of those parameters is adjusted independently, the dropdown text box will still show the last preset recalled.

Mute buttons in Composer are essentially latched buttons, toggling on and off active and inactive states. However, a client or end-user may request the use of a momentary style, physical button to toggle a mute. This can be programmed in Composer using some simple logic modules.

Drag in a 1-Button Momentary from the Control Inputs folder, a Flip-flop module from the Control Logics folder, and a 2-Output Remote Control Number module from the Control Outputs folder and wire them as shown.

This Tech Tip will exhibit how to set up this control for the unit analog output mute, but this can be used for any latched style button.

Set up the mute button to remote control – in this case remote control number 1.

Next, edit the remote control number outputs to both have the same number that was assigned to the mute button.

Now, assign the 1-Button Momentary ON button to analog remote control.

Logic programming is only functionally viewable when Composer is online. Push and go online and test the logic programming.

Every time the momentary button activates, it triggers the flip-flop module to toggle back and forth between Q and NOT Q, which in turn triggers the remote control numbers. Because the same remote control number is in both outputs, it effectively toggles the mute button off and on.

This article will demonstrate Composer control logic for automatically hanging up a call if no DTMF signal is received within a period of time. The logic is designed to function with both VoIP and ATI option cards for Radius NX and Edge.

Logic Demonstration

How It Works

There are five key modules used in this design. This section will go through them one by one:

• The Flip-Flop  module keeps track of whether or not a DTMF signal has been received from the far end. Normally, the “Set” input would be wired to the “DTMF#1” output of the 2 Line VoIP Interface module, but here it is simulated by a 1 Button Momentary module. The “Reset” input is wired to the “Hook Status#1A” output of the 2 Line VoIP Interface module, with an inverter in between. This will reset the Flip-Flop after the call ends.
• The 2 Input Logic module outputs “True” when the call is active and a DTMF signal has not been received from the caller. Otherwise, the module outputs “False”. The “In#1” input is wired to the “NOT Q” output of the Flip-Flop. The “In#2” input is wired to the “Hook Status#1A” output of the 2 Line VoIP Interface module. The logic type of this module should be set to “AND”.
• The Ramp Processor module takes in the control signal from the “True” output of the 2 Input Logic module and outputs a control signal that ramps up over a specified period of time. Here, it is set to 10 seconds, but this can be set to any desired value. This represents the amount of time the caller will have to enter a DTMF signal before the call automatically hangs up.
• The Threshold Detector module takes in the ramping control signal from the Ramp Processor module, but only outputs a control signal once the ramping control signal reaches 100%. In order to do this, the “Threshold A” value must be set to “100%”.
• The 1 Output Remote Control Number module takes in the control signal from the “True” output of the Threshold Detector module and outputs a high (100%) control signal to Remote Control Number 1. Note that the “Call/End” button in the 2 Line VoIP Interface module has been set up to Remote Control Number 1. This button will be activated when the 1 Output Remote Control Number sends its control signal, ending the call.

In some situations, you may need to avoid unintentional button presses on a control screen or external controller. This could be to trigger a preset, power down a system, or a myriad of other cases.

This could translate into the user needing to hold a button for a given amount of time before a control will activate. We can achieve this with a ramp processor and threshold detector.

Drag in these modules and wire them as shown below.

This functionality can be used with either a momentary button or latched button. The control meter at the end of this example simply shows the 100% signal flowing through the TRUE output of the threshold detector. This could be a remote control number output, preset trigger, network string output, etc.

When the ON button is pressed and held, we can see 100% control signal inputting into the ramp processor which slowly raises its output level from 0% to 100% over the span of time set by the UP rate. We see the ramp processor output then inputting into the threshold detector’s input.

Once the ramp processors output has reached 98% or higher (as set by the Threshold A slider), the threshold detector allows the signal to pass through its TRUE pin, causing the control meter to reflect 100% input.

The threshold detector could also have its FALSE pin connected to something that would trigger something else when the button is in the off state.

Additional note: logic functionality doesn’t work live in Composer unless it is online with the system.

This article explains the differences between the selectable modes for the encoder, encoder button, and individual buttons for W-Series wall remotes.

Where do I find the different encoder button modes for my W-Series remote?

The available encoder button modes for each W-Series remote are found by right clicking the device in Site View, selecting “Unit Properties…”, and selecting “Edit Remote Settings…”.

Important: Not all encoder button modes are available on all W-Series remote models. This will be further explained below.

Breakdown of Encoder Button Modes

Encoder Button Menu: Single Encoder Menu

The “Single Encoder Menu” mode is the simplest of all the modes. It allows for control of a single menu with a turn of the encoder dial and a single button with a press of the encoder button.

Compatible models: W1, W3, W4

Encoder Button Menu: ‘Select and Set’ Encoder Menu Using Individual Buttons

The “‘Select and Set’ Encoder Menu Using Individual Buttons” mode allows for control of up to 4 menus with W3, and up to 8 menus with W4, by pressing a push-button switch to select a menu, then turning the encoder dial to adjust the value of the selected menu. The encoder button can also be used to control a single button in the design.

Compatible models: W3, W4

Encoder Menu Select

The “Encoder Menu Select” mode allows the user to cycle through up to 8 menus by pressing the encoder button. The value of the selected menu is then adjusted by rotating the encoder dial. This mode allows for control of multiple menus with W1. It also frees up the push-button switches on W3 and W4 to be used for other functions, as demonstrated in the above example.

Compatible models: W1, W3, W4

Note: None of the Encoder Button modes apply to W2 since it does not have an encoder.

The xIO XLR series has a button on the front panel that makes this device versatile and useful for many different cases. In this Tech Tip, we’ll cover how to set the button up to be used as a paging toggle to mute/unmute a microphone.

In this site file, we can see there are five zones (Lounge, Patio, VIP, Restroom, and Kitchen). This site will be incorporating an xIO XLR 1×1 and it will be used to page to all zones in case of emergency or other site wide announcements are required.

You can see above that the red highlighted wire is where the xIO XLR signal will be going. Note that the kitchen does not have any BGM or microphone audio from the rest of the system. Also note this programming has a single gain module handling the audio level for the xIO XLR, this is where the MUTE will come in to play.

First, set the xIO XLR Button Press Function to Control Pins.

Now, enter into the Design View of the site and drag in the Intelligent Module and the shown logic modules; 8-button latched, Flip-flop, and Dual Preset Trigger, and wire them accordingly. If you like, rename the 8-button latched to reflect the LED controls. This can help stay organized when programming.

In this case, we’ll only really need the RED, GREEN, and ON controls for the IN side of the xIO XLR, however all 8 wires are connected.

Open up the 8-button latched as well as the Dual Preset Triggers. Engage the ON and RED buttons, and ensure their ON LEVEL is set to 100%. Then give the Dual Preset Triggers presets 1 and 2.

Decide which preset you would like the MUTED state to be. This programming uses Preset 2 for this. Right click in the outer area of the 8-button latched and Store Module Settings in Preset 2.

Now, open up the Gain module for the xIO XLR and right click on the mute to set it to Preset 2 as well.

Next, reset the 8-button latched so that the RED button is off and the GREEN button is on, leaving the ON button engaged. Right click and Store Modules Settings in Preset 1.

Just the same, add the xIO XLR Gain module’s MUTE button to Preset 1.

At this point, test your presets from Preset Manager (Tools > Preset Manager or CTRL + G) to ensure they are working as they should.
– PRESET 1 should recall the UN-MUTED state with the GREEN LED on.
– PRESET 2 should recall the MUTED state with the RED LED on.

Composer must be online to view logic programming working, so push the site design and go online. The xIO XLR LED should update to the current state of the logic programming (likely the muted state as the flip-flop default state is Not Q). Test the programming with the xIO XLR panel button to ensure the preset states are acting as they should.

The ARC-K1e is a simple, intuitive ARC remote that can be used for many parameters. However, the large majority of use cases are likely volume control. In this example we will couple an ARC-K1e with an ARC-EX4e that will act together as volume control and input selection.

This site file has four speaker zones; Bar, Patio, Restrooms, and VIP. Open up the first Speaker Manager that flows to the Bar zone. Right-click on the module gain fader and select Set Up Remote Control.

Scroll to the right in the Remote Control Devices area and select Add New ARC. If you already have an ARC-K1e or ARC-K1e + EX4e added, it should be displayed in this list.

Select the ARC-K1e + one ARC-EX4e from the drop-down list.

Now select Encoder #1 A. This will assign the first Speaker Manager gain fader to the A side of the K1e knob control. Do the same for the second Speaker Manager gain fader, but select Encoder #1 B to set it to the B side of the K1e knob control.

Note: the other two zones would require a second ARC-K1e. If two or more zones are to be controlled together, set the gain faders to the same control number. These faders will control in unison.

Now, open up the Mono Input Selector module, right-click on the slider at the bottom, and then choose Set Up to Remote Control.

Choose the Modular ARC from the list and assign the Input Selector slider to the Radio Buttons option. This will allow the four input selections to relate to the four buttons on the EX4e, in the same order that they appear in the module.

Push the design and program the ARC remote, and go online to test the control.

ADDITIONAL NOTE:
There are may more ways to take advantage of the ARC-K1e in combination with EX4e or SW4e. This Tech Tip is a basic example of setting up these remotes. Refer to the Help File > Module ARC Programming for more information.

Many installations include power sequencers that power down most of the greater system while a Symetrix DSP remains powered for system control. In this example, there is a SymVue control screen with system on/off momentary buttons that need to indicate the current state of the system.

In Composer, drag in a 2-Button Momentary module, two Local Logic Output modules, and a Flip-flop module, and wire them as shown below.

Local Logic Output #1 is intended to trigger the system ON and Local Logic Output #2 is intended to trigger the system OFF. Be sure not to connect either wire to the TRIGGER input of the Flip-flop as this will force the Q state to flip back and forth between Q and Not Q with every button press. While this doesn’t affect the power sequencer, it does affect how the LEDs display on the control screen. The SET input will set the module state to Q and the RESET input will set the state to Not Q, and either state will remain active until the other has been triggered.

Copy the two ON buttons from the 2-Button Momentary as well as the Q and Not Q LEDs from the Flip-flop module to the control screen. Resize the ON and OFF buttons and rename them appropriately. If you wish to edit the text of the LED label or remove the text entirely, the properties window allows you to change the display type to Symbol or simply delete the Text.

Push and go online to test this functionality – logic doesn’t work in Composer when offline.

ADDITIONAL NOTE:
Similar to using LEDs to indicate presets, momentary buttons don’t stay in a constant state unless held there and any change downstream from this logic will not reflect that state. In essence; the LED state is taking its cue directly from the inputs, coming from the momentary button pushes and this method does not take feedback from the power sequencer directly.

Occasionally, third party devices may need a control signal to last longer than a momentary button’s “press” to activate, but not as long as a press and hold situation. This Tech Tip explains how to get a latched button to act like a momentary button.

First, drag in a 1 Button Latched (can be multi-button), Delay Logic, and a Preset Trigger, and wire them as shown.

Set the latched button, in its OFF state, to a preset. In this example, we’ll use Preset #1. 

Next, set the Delay Logic’s delay time to 0.01 sec and the hold time to an appropriate time for your system. This example leaves the hold time at the default 1 sec. 

In the Preset Trigger module, the default number will be 1, but change this to the preset used for the latched button in its OFF state. 

Push and go online, and test the latched button by pressing it. It should return to the OFF state after the Delay Logic hold time has elapsed, firing through the Done stage output.

SymVue Control Screens are a powerful and flexible option for system control. This Tech Tip explains how to export a control screen to a T-5 touchscreen, Control Server, and Windows PC.

This example is for a three-zone venue with six inputs. User controls should include microphone level control, background music selection and level control, zone volume control, and an indicator if the emergency system mute is engaged.

In this example, we will only need to create two control screens; one for the T-5 touchscreen and one that can be used for both Control Server and Windows PC. Once both control screens have been populated with controls and designed according to the available space, open the touchscreen control screen.

T-Series Export:

Right click in the background area and select Export to SymVue. Then select Touchscreens and click Next.

If there are more than one control screen intended for this configuration, this Panel Selection dialogue will allow you to select which screens should be included and referenced by this screen. When preferred selections are made (or if no other control screens are included), click Next.

Note: the “Home” control screen of a configuration is the one that is first exported through right-click > Export to SymVue.

Panel Security allows you to set a PIN code for the desired control screens. This PIN will be used for all selected control screens. Control screens in a given configuration may not have varying PINs.

Hardware Connection allows for two options of locating the T-5 touchscreen. Select which is most appropriate for the installation – in the vast majority of cases, Typical is the appropriate selection.

Export Options is the last step. Export to Touchscreens should automatically be checked. The drop down provides options to export this configuration to a specific T-5 or a group of touchscreens.

Note: We strongly suggest selecting “Go Online with Composer site file” to avoid control subscription errors which can be caused by the touchscreen and Composer/DSP not sync’ing parameters.

When ready, click Finish. This will export the configuration to the T-5 and push/update the archive on the DSP and go online.

Control Server / Windows PC Export:

Open the control screen for Control Server and Windows PC. Repeat the steps for T-5 touchscreen, making the appropriate selections for either. 

Once the configuration has been exported to Control Server, open up the Control Server’s WEB GUI and log into the Admin account. 

Double-click on the Control Server or right-click > open in Site View.

Once viewing the Status page, in the top right corner, go to Menu > Management > Applications and select SymVue from the App list. This will expand the available configurations list.

Click the down Arrow (or click and drag) on your control screen – this example “ControlScreen” – and click Save.

Then go to Menu > Management > Users. If you haven’t created a user account for the Control Server, you can do so here. Click on the user that will be accessing the exported configuration.

Click on the down arrow for the configuration that this user will access to place it into the User Allows Configurations and click Save.

This user now has access to this control screen.

Additional note; Configurations, Licenses, and Applications:
A configuration is any number of single control screens grouped/exported together that act as one “set” of pages or screens. You can have any number of available configurations (sets of control screens) loaded into the Control Server. The number of licensesavailable determines how many of those configurations are allowed to be available to users. You can have any number of users with varying access to any number of the available licensed configurations. Every Control Server comes with 5 licenses, but more can be purchased if necessary.

These control screens are used through the SymVue Application, which is the same medium we use for our export to touchscreen or Windows PC. The Event Scheduler and Mixer apps you may see in Control Server or on export are not related to these control screen configurations.

In Symetrix DSPs, the word “active” isn’t quite the right term to describe presets. Presets are simply a snapshot of one or more parameters in a particular state, and are more so recalled than they are “active”.

However, when using a SymVue Control Screen, there is a way to offer what looks like an “active preset” indicator using the LEDs from an Input Logic module.  

Let’s say, for example, that we want three volume presets that can be recalled depending on the time of day for an entertainment venue – low, medium, high – which could correspond to before, during, and after a performance.  

In this example, we’ll use a 4 Channel Gain module that would control 4 zone volumes within the venue. Set the gain faders (and any other desired parameters) to the desired position for “Before” the performance.

Next, set the gain faders to the other two desired levels for “After” and “During” performance.

We can now see, with Super-impose Assigned Control Numbers engaged, that all faders have been assigned to Presets 1, 2, and 3.

Now, we will rename these presets for easier identification later.
Tools > Presets Manager or Ctrl+G. Highlight the preset and click “Rename…” near the bottom.

Next, drag in a 4-Button Radio and a 4 Input Logic module, and wire them together. We only need the first three outputs to inputs as we only have three presets. We won’t be using the fourth output on the 4-Button Radio or the fourth input on the 4 Input Logic.

Set the 4 Input Logic module to the “OR” mode and then 4-Button Radio module slider to settings 1, 2, and 3, and add it to Presets 1-3 accordingly. The image below shows the module in the Preset 3 position.

Now, when a given preset has been triggered there will be a 100% control signal sent from the 4-Button Radio to the 4 Input Logic which will light up the LED for that given input.

In your Control Screen, place Preset Recall buttons for all three of the created presets.

Next, copy each LED from Inputs 1, 2, and 3 in the 4 Input Logic module. 

The “Input” text is still showing here. We can remove that in the parameter properties window and then resize the right margin to fit to the LED size. We can also resize the LED to the desired size.
Display type can be changed to “Symbol” or we can remove the text “Input”.

Now that we have the basic functional elements in place, we can Push and Go Online to test – Logic functions (these LEDs) don’t actively show while Offline in Composer.

Additional note: While this method offers a way to indicate what looks like an “active” preset, it only shows the last recalled preset of these three that we have LEDs for. If a preset is recalled and any adjustments are made to any parameters included in these presets, the LEDs will not indicate this and will remain in the current state until the same or another Preset Recall button is pushed.

Unlike ARC Series remotes, W Series remotes do not have a built-in method for setting upper and lower boundaries for fader control. Fortunately, there is a way to do this in Composer using Control Screens.

1. Right click the desired fader and select “Copy to control screen”. Create a new control screen if necessary:

2. Go to Tools > Control Screen Manager:

3. Highlight the control screen created in the first step and select “View & Close”:

4. The fader in question will appear. Left click it to select it:

5. On the right side of the screen under “Properties”, locate “Control Range & Taper”:

6. Under “Minimum” and “Maximum”, set the minimum and maximum values for the fader:

7. From here, right click the fader, select “Set Up to Remote Control”, and set it up to the W remote as normal:

The purpose of this document is to provide an understanding of operation and configuration of the two different String Output modules available within Composer. The two different types or modules are the String Output and Network String Output.

 

These control modules send out an ASCII (text) or hexadecimal (binary) string every time its control input changes from low (less than 49%) to high (greater than 51%). These modules can be used to send commands to control a variety of third party devices, e.g. turn on a projector, change projector source, change a camera position, change channel of a media device, etc.

 

Note: The string is only sent from the communication port of the device where the DSP module resides. Strings may be up to 63 characters or bytes long.
Enter the string exactly as it should be sent out. To obtain an exact list of sting commands or control protocol for the third party device, refer to the device user’s guide or contact the manufacturer.
In ASCII mode, in addition to standard text characters, the following special characters are supported:

Where nn is the ASCII hex character code, e.g. \x0D for carriage return
Note: In binary mode, data is entered as sequences of bytes in hexadecimal separated by commas. For example, to send out an incrementing sequence of 12 values starting at 7, enter; 7,8,9,A,B,C,D,E,F,10,11,12.

String Output Module:

The String Output Module can be used to send control commands over the following ports:

RS-232

Supports baud rates between 1200 – 230400
On SymNet devices, the RS-232 baud rate is stored in non-volatile memory in each
device. It can be set differently for different devices.
To edit the baud rate from SymNet Composer use the following steps:

1. Right-click on the unit in Design View
2. Select Unit Properties

3. Select Configure Remote Control Ports…

4. Select the RS-232 Port tab

1. Then select the radio button for the desired baud rate (1200-230400)
   
   Note: The default baud rate is 57600. The baud rate should be set to match what is expected by the connected device.

UDP:
Uses UDP port 48631; only 1 controller can communicate with this port at a time

 

TCP:
Uses TCP port 48631; up to 4 controllers can communicate with this port simultaneously
These are the steps to add a String Output module to your design:

1. From the Toolkit drag in a String Output module (Control Modules>Control Outputs)

1. The Sting Output Properties windows will open automatically
2. First select the unit that will be transmitting the sting command

1. Next select the remote control port the string will be sent out (RS-232, UDP, or TCP)
2. Click OK
3. Double click the String Output module
4. Select the string to output mode (ASCII or Binary)
5. Then add the string command to the module

Here is an example using multiple String Output modules to change camera positions. This example uses 4 PPT (Push to Talk) microphone and 4 cameras. When a particular microphone is being used, the camera assigned to that microphone needs to be active. A 4 button processor Super-Module is used to set the function of the microphone to PTT.

 

This is an example using an ASCII command:

When the microphone button is pressed it will trigger the green or On LED. The String Output module is wired to the green LED.

Whenever the greed LED is lit for a particular microphone it will tell the camera assigned to that microphone to activate.

This is an example using a Binary command:

Network String Output Module:

The Network String Output module can be used to send control command to any device connected to the same network. This includes but is not limited to other DSPs. Commands can be sent over UDP or TCP ports.

 

UDP:
Uses UDP port 48631; only 1 controller can communicate with this port at a time.

 

TCP:
Uses TCP port 48631; up to 4 controllers can communicate with this port simultaneously
These are the steps to add a Network String Output module to your design:

1. From the Toolkit drag in a Network String Output module (Control Modules>Control Outputs)

1. Double click the Network String Output module
2. Select the string to output mode (ASCII or Binary)
3. Select the communication port (UDP or TCP)
4. Next enter in the IP address and network port of the device receiving the sting command from the DSP
5. Then add the string command to the module

Here is an example that uses multiple Network String Output modules to send a load configuration command to multiple DSPs in the system.
For this example a control screen was created to give the end user easy operation of this procedure

When the “ON” button is pressed the control signal goes high and sends out the string command.

A delay module is also wired to the latched button. The delay logic module sends a control signal to the preset trigger to reset the “ON” button

This is an example using an ASCII command

This is an example using a Binary command:

In a large convention center or conference room, microphones may be routed to different speaker zones in a mix-minus configuration in order to reinforce microphones from one zone to another zone while minimizing the potential of acoustic feedback. When in a “mix-minus” configuration, microphones are routed to all zones except the zone in which the microphone resides, so the microphone level can be quite loud without creating acoustic feedback with the speakers directly overhead.

In most applications the mix-minus setup is straight forward and easily accomplished using a Matrix Mixer module. However, when the convention center or conference room using mix-minus routing is part of a larger divisible venue, where two or more rooms can be combined and uncombined, then the logic for combining/uncombining the audio, the automixers, and control parameters (such as mute and volume) must be taken into consideration.

When no mix-minus routing is necessary, room combining is simple using Room Combiner modules. These modules will combine and uncombine the audio, automixers, and control parameters (gain, mute, sources selection) of 2 to 16 rooms with the push of a combine button.

Above: 2 Room BGM Automix Combiner

In the example above, when #1 Combine Button on the BGM Automix Combiner module is turned “on”, the audio, the automix, and the control parameters are shared between room 1 and 2. Any change to the room controls, such as BGM selection, volume, or mute, will affect both room’s controls. When the #1 Combine button is turned off, both rooms operate in a standalone fashion. This functionality is especially helpful when using a 3rd party control system, as no combine or uncombined logic needs to be added to the control system programming since Symetrix control will do all combine and uncombined logic automatically.

The limitation with the Combiner module in a mix-minus application is that each room has only a single room input and output for the local sources, whereas in a mix-minus configuration each room would have multiple speaker zones, each with their own unique mix of the microphones.

The solution is to use a Matrix Mixer and BGM Automix Combiner module in tandem, using linked controller assignments, to create a mix-minus, room combine system where audio, automixers, and control parameters combine and uncombine.

The following example will create a two room system with combining/ uncombining capabilities and mix-minus matrixing of the mics. There are 12 microphones in room 1 and 8 microphones in room 2. Each room has 4 amp channels. For simplicity sake, this example includes only the processing associated with automixing, combining/uncombining capabilities, and mix-minus matrixing of the microphones. A real world design would also include dynamics processing and filtering/equalization at the input and output stages. Follow these simple steps to program a mix-minus, room combine, site file:

Step 1:
Build the site file such that Slave Gain-Sharing Automixers are used for all mics in the system. Separate Slave Gain-Sharing Automixers should be used for the mics located in each room.

Step 2:
The Automixer discrete outputs should feed the inputs of a matrix mixer module instead of using the Mix output.

Step 3:
Add a BGM Automix Combiner to the site file that can accommodate the number of combinable rooms in the venue – two rooms in this example. Wire the Chain output of the Automixers to their respective Chain input on the Combiner (blue wire). Wire up the Master out of the Combiner to the respective Master input of the Automixer.

Step 4:
Create the mix-minus configuration of the microphones using the Matrix Mixer user interface for the rooms when in the uncombined state Once the routing and crosspoint gains are configured for the individual uncombined rooms, right click the Connect Matrix and select Store “Connect Matrix” in Preset to store the matrix settings to a preset. This example uses preset 1 for the “uncombined/standalone” preset.

Step 5:
Create the mix-minus configuration of the microphones using the Matrix Mixer user interface for the rooms when in the combined state. Right click the Connect Matrix and store to a preset. This example uses preset 2 for the “combined” preset.

Step 6:
To stay organized, open the Preset Manager and rename the combine and uncombined presets accordingly. Then recall both presets, checking the matrix each time, to ensure they are correctly changing the matrix.

Step 7:
Add the following logic circuit using control modules: 1 button latched, 1 inverter, 1 dual preset trigger. The top Preset Trigger-1 should use the “combine” preset, the bottom Preset Trigger-2 should us the “uncombine” preset.

Step 8:
Open the 1 Button Latch and the BGM Automix Combiner and assign the same controller number to Button 1 and #1 Combine. This example uses controller #1.

Step 9:
On the BGM Automix Combiner assign the Volume fader for Room 1 and Room 2 each a unique controller number. This example uses 10 and 20.

Command Buttons in Composer can be used to launch 3rd party applications. The purpose of this Tech Tip is to illustrate how to set up the Control Screen Command Buttons to launch and control parameters in these applications. By making use of a script language named AutoIt and an associated script editor
named SciTE, functions such as mouse clicks, cursor movement and key strokes in 3rd party applications can be executed by Command Buttons. The 3rd party application could be virtually any Windows program. Windows Sound Recorder is being controlled by Composer in the following example:

• First, set up a Control Screen in Composer.

• Place a Command Button on the Control Screen. The “Command Button Properties” automatically opens after placement, in order to label the button. Choose “Browse” in order to direct the command button to the location where the desired program’s executable lives.
  C:\Windows\System32\SoundRecorder.exe

By using the recommended script editor, SciTE, AutoIt is able to produce an .exe file which can execute a mouse click among many other functions. If one associates a Command Button with the executable file created in SciTE, the user can control the 3rd party application with the Command Buttons

• AutoIt and the recommended script Editor SciTE can be downloaded
  from the following links:
  http://www.autoitscript.com/site/autoit/downloads/
  http://www.autoitscript.com/site/autoit-script-editor/downloads/
• Start SciTE from your Start Menu/All Programs/AutoItv3/SciTE Script Editor. Check the AutoIt Help File for the language’s syntax rules. Check the SciTE Help File for tips regarding the editor’s capabilities.

• Open the program named AutoIt Window Info. Click and drag the order, or any button parameter in your chosen application. Make a note of the information on the line labeled “Advanced Mode,” which can be found on the Control tab.

  
• The first line of code entered into the SciTE editor in this example is an optional title line: Opt(“WinTitleMatchMode”, 2)
• The information required in order to activate the “Start Recording” Button can be gathered from the “Advanced Mode” line of the Control tab in the program AutoIt Windows Info, after the Finder Tool’s target has been dropped on the parameter to be controlled.

Below is the line of code to be entered into the script editor to activate the record button with a function:
ControlClick (“Sound Recorder”, “”, “[CLASS:ToolbarWindow32; INSTANCE:2]”)

ControlClick is an AutoIt script command to execute a mouse click.
http://www.autoitscript.com/autoit3/docs/functions/ControlClick.htm

• Convert the .au3 script document which you created in SciTE into an executable in the same directory by hitting F7 while the SciTE file is open. An .exe with an identical file name will be created in the existing directory, next to the .au3 file.

• Create another Command Button on your control screen in Composer which will be the “Start Recording” button execution.
• Right click on this new Command Button and choose “Command Button Properties”. Direct the browser for this button to the AutoIt executable file which you just created in SciTE.
• Once this is complete, save the Composer file. You may also export the control screen to SymVue at this time.

• When the “Launch Windows Sound Recorder” button is clicked, it will initiate the Sound Recorder. When the “Record” button is activated, Windows Sound Recorder will begin recording.

  

This Tech Tip describes the steps necessary to locate and use the Dante audio, program the touch-sensitive mute buttons, and control the LED’s on the Shure MXA310 from Composer.

The Microflex Advance Table Array is a networked array microphone ideal for AV Conferencing applications where premium audio and a low profile appearance are paramount. Shure IntelliMix® DSP Suite Steerable Coverage™ technology deploys four discrete zones of table coverage for best-in-class audio capture, configuring all parameters seamlessly through a browser-based graphical user interface. Here are the steps to create the Dante audio flow for the MXA310 within Composer.

1 Make sure the Symetrix DSP being used to locate the MXA310 is running
matching Composer firmware (must be Composer version 5.3 or newer).

2. In the Toolkit, open up Third-party Dante Devices, Shure. Drag an MXA310 into the Site files Design view.

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2. In the Toolkit, open up Third-party Dante Devices, Shure. Drag an MXA310 into the Site files Design view.

3. Locate the DSP, and then the MXA310.

4. Right click on the MXA310 and select MXA310 Unit Properties.

5. Enter the Control Interface IP for the MXA310, and then click Verify Control IP. Click OK.

Note: The Control Interface IP for the MXA310 can be obtained from the Shure Web Device Discovery software available from the Shure website: http://www.shure.com/americas/products/software/utilities/shure-web-devicediscovery-application

6. Double-click on the DSP to open the site files Design view.

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7. In the Toolkit, open Dante Transmit and Receive flows, Receive Flow Modules for Existing Flows. Drag in the MXA310 Mic Flow.

8. Wire the MXA310 Mic Flow into the design.

Program the touch-sensitive mute button for toggle, push-to-mute, push-to-talk or disable settings or to send controls to external devices. Here are the steps to utilize the button on the MXA310 from Composer.

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8. In the Toolkit, open Control Modules, Control Inputs. Drag in a 1 Button Momentary module.

9. Double-click and open the 1-Button Momentary module to open it.

10. Right-click the “On” button.

11. Select Set Up to Remote Control.

12. In the Set Up Remote Control window, select 3rd Party Remote Analog Input – ‘MXA310’. Click OK.

13. In the Toolkit, open the Super-module Library, Import super-module. In the Super-modules folder, Examples, Tools, select the 1-Button Processor.

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14. Wire the 1 Button Momentary module into the 1-Button Processor Super module.

15. Double click the 1-Button Processor to open the Super-module control screen.

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16. Select the desired microphone switch operation (Push to Talk, Push to Mute, Toggle, or Disabled).

17. Double click and open either a Gain Module or Automixer used in the signal processing and routing of the MXA310. In this example, a Gain-sharing Automixer is used.

18 Right-click on the Master Mute button, select Set Up to Remote Control.

19. Select “Control Signal Assignment” for the Remote Control Device.

20. Then Click “Select”.

21. Expand 1-Button Processor, select 1 Off/R. Click OK.

Note: The Master Mute button of the Gain-sharing Automixer will now be controlled by the state of the red LED of the super-module. When the red LED is active, that mic will be muted. When the red LED is inactive, the mic will be unmuted.

22. In the Toolkit, open up Control Modules, Control Outputs. Drag in a Remote Logic Output module.

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22. In the Toolkit, open up Control Modules, Control Outputs. Drag in a Remote Logic Output module.

23. The Remote Logic Output Properties window will open. For Remote Unit, select MXA310. For Logic Output, select Light Ring – Ring. Click OK.

24. Wire the On/G output from the 1-Button Processor Super-module into the Remote Logic Output.

25. Return to the Site files main Site View and push the Site file to Go-Online.

Composer can control the MXA310 LEDs separately or as 4 individual sections. However, the setup in the Shure web GUI has to be setup correctly first, to allow this. The default MXA310 setup won’t allow LED control from Composer. Here are the steps to control the MXA310’s LED’s from Composer.

26. Right click on the MXA310, and select Unit Properties

27. Click the “Launch Web Configuration Interface” button.

28. To control the LEDs from Composer, turn off “Display Automix Gating” and select “Ring” for Lighting Style, on the Light Ring tab.

Note: If the option for “Display Automix Gating” is not visible, select the Button Control tab and temporarily select “Local” for the Mute Control Function. Then select the Light Ring tab to uncheck the box for “Display Automix Gating”.

29. On the Button Control tab, set the Mute Control Function to “Logic Out”.

30. Set the Mute Control Mode to “Push to talk”. Push to talk should always be the selection regardless of which microphone switch operation is selected in the 1-Button Processor Super-module in the Composer site file.

Being able to troubleshoot a Symetrix system is a paramount skill in any technician’s skillset. The common understanding of signal flow is a basic concept, but there are some issues that can come up that signal flow doesn’t address. Here are some common issues and techniques our Support team uses to isolate the problem:

CANNOT LOCATE SYMETRIX DEVICE ON CONTROL (ethernet) NETWORK

Includes all DSPs, W Series remotes, T Series touchscreens, xControl, and Control Server. 

Behavior: Unit is not locating in Composer at all or is showing intermittent location status (green check is coming and going). 

Subnet/Network Mismatch

Barring advanced network setups that communicate across subnets (not recommended for Symetrix equipment), an extremely simple, but possible answer is simply that your PC is looking at the incorrect network or is not physically connected to the same network as the DSP. First, ensure that your PC is wired either directly to the DSP’s Control (ethernet) port OR into a port in the same VLAN/switch as the DSP. Note: Symetrix DSPs are programmed from the factory to boot in DHCP mode (no static IP is set in the factory) which will then resolve to a 169.254.x.y (Class B) APIPA IP address if no DHCP server is found. If a static IP was previously set on the DSP, it will hold that same IP address on reboot. 

Ensure that you have selected the correct network in Composer and also that your PC is either set into the correct network/subnet (if needing static) or set to DHCP (on the same network as the DSP) and is receiving the same address range as the DSP. The IP Address and Subnet Mask columns in the ”Select Search Network” dialogue show your PC’s current IP address and subnet. A command prompt “ipconfig” will also show your PC’s current network configuration.

There are different ways to edit your PC’s IP configuration, but one quick way is to click Start and search for ‘ncpa.cpl’. This is a shortcut that will bring you to the Network Connections panel. From here, right click on the wired connection and select Properties. Then open the TCP/IPv4 properties – where your PC’s IP settings can be edited.

An additional step would be to confirm the IP address of the DSP itself. The front of the DSP has a main menu that shows the Dante, Ethernet, and ARC status, along with its Control (ethernet) IP address.

If this menu isn’t currently displaying, push the button to the right of the display to cycle through to the correct menu – you may need to push and hold the button to return to the Dashboard menus.

The command prompt “ping” is another simple procedure and can provide important information in a few ways, specifically when “-t” is added to the end of the string; for example, “ping 192.168.150.196 -t”. Note: space between ping and the IP address, and between the IP address and -t.

The “-t” will allow the ping to continue running indefinitely, instead of the default four pings. Press CTRL+C to stop the ping. A constant ping like this can help identify network issues by showing if the return times are longer than this.

In a healthy network, without too many switch hops, we would expect ping times to be majorly equal to or sub 1 ms (=1ms or <1ms), with only very occasional small spikes if any at all, depending on overall network traffic.

common 1

If you are able to see the DSP through a ping, but still not through Composer, please reach out to our Integrator Support team for further assistance.

If a ping is responding with “Destination Host Unreachable” then a device with this IP address is either not on this network or the DSP or your PC could be experiencing a NIC issue.

If a ping is responding intermittently, response times are inconsistent, or overall, not consistently ~1 ms, here are a few scenarios that could be occurring:

• IP conflict: if the unit you’re pinging and some other device have the same IP address and are now fighting for prominence on the network. Unplug the device you intend to ping from the network and send the ping again. If you get a response, there is some other device on the network with that IP. If you don’t get a response there is likely no IP conflict and some other issue is occurring.
• Physical layer issues: if the switch/router, cabling, or ports are faulty they can interrupt network traffic. Try swapping out any of these components (where possible).
• A quick way to test for this would be connecting the PC directly to the DSP/unit in question, if not already – eliminating any greater network as a variable. If pings clear up with a direct connection, there’s more to investigate with the other components.
• Network Configuration; the configuration of a switch or greater network can unintentionally interrupt the traffic flow. Check for incorrect or unnecessary IGMP snooping configuration, port blocking, security/firewall/anti-virus, Green Ethernet is disengaged, use of STP instead of mSTP in VLAN configs, and QoS.
• A quick way to test for this would be connecting the PC directly to the DSP/unit in question, if not already – eliminating any greater network as a variable. If pings clear up with a direct connection, there’s more to investigate with the network configuration. Note: Integrator Support does not preside over on-site network(s) and cannot assist with configuration, which includes advanced networking such as crossing subnets and LANs (which requires Dante Domain Manager).
• NIC in a Bad State; the NIC on either the PC or the DSP can become impaired over time via constant plugging and unplugging to different networks or devices. Rebooting the DSP (waiting about 10 seconds before re-applying power) should clear its NIC while disabling and re-enabling the NIC on the PC should clear it (aside from rebooting the PC altogether).

CANNOT LOCATE DEVICE ON DANTE NETWORK

Includes all Symetrix DSPs, xIO Dante expanders, and third-party Dante devices.

Behavior: Unit is not locating in Composer through DSP or xIO Updater/Configurator or is showing intermittent location status (green check is coming and going).

Subnet/Network Mismatch

Barring advanced network setups that communicate across subnets, the Dante network is fundamentally the same as the Control network with regards to basic communication. All Dante devices must be in the same subnet to communicate with each other. 

Composer is intended to configure the Control network and thus unable to look at the Dante network directly in the same way that Dante Controller can. It is normally recommended to allow the Dante network to remain in DHCP mode for the simplest set up and maintenance. 

To ensure that devices are on the same network/subnet, check the Device View in Dante Controller for the given unit under the Status tab (your PC must be looking at the Dante network).

If the Dante network must be set to static IP addresses, do this within the Network Config tab of Device

If network configuration seems correct, the same ping techniques from the Control network can still apply. Much of the case will relate to something not allowing multicast traffic to flow. IGMP and low-quality switches are the largest culprits. Audinate has articles regarding IGMP Snooping that can be of benefit. 

https://www.getdante.com/blog/well-intentioned-mishaps-with-igmp-snooping

https://www.getdante.com/support/faq/multiple-leader-clocks

REMOTE TERMINAL COMMANDS

From a Composer perspective, there are some tools available that can help illuminate issues. All of these commands should be done with the intended DSP located and then going to Tools > Launch Remote Terminal, then Options > Debug Mode, and ensure the IP address of said DSP is in the upper left IP address field.

• “INFO…” Remote Terminal Command; this command coupled with a target will return different diagnostics about the DSP:
• INFO CARDS; returns a list of the installed I/O cards in the DSP. This can be used to confirm that the DSP has and is reporting having a Dante card. It will return “Brooklyn…”, “non-Dante Clock Card”, or “none”.
• Brooklyn means the DSP is accurately reporting its Dante card.
• Non-Dante Clock Card means the DSP does not have a Dante card, but has a clock card instead. This could be due to a purchase error and the Symetrix Sales or Integrator Support departments should be contacted.
• None means the DSP is expecting something in that particular card slot but is unable to recognize it. Please contact our Integrator Support team if this is what the DSP reports.
• INFO DANTE
• This will request a report of all Dante information from the DSP including card type, Primary IP address (and secondary if in redundant mode), Dante channel usage, and other diagnostic information.
• “GDBCV” Remote Terminal Command; this command requests the Dante browse information from a located DSP – what it can see on the Dante network.
• Send the command “GDBCV” (no quotes) to the DSP. This will return a report of all Dante devices the unit’s Dante card can see, with a bit of extra information describing the communication quality.
• “ACTIVE” or “ACTIVE K” is healthy network communication, and the unit should be locating in Composer. If this is reported and the unit is still not locating in Composer, please contact our Integrator Support team for further assistance.
• “QUERIED”, “UNQUERIABLE”, and “UNPINGABLE” are potential all signs of network miscommunication – the DSP can see the devices but is unable to gather all required information to make a complete handshake. If devices remain in these states for extended periods of time, double-check network configuration, if using a switch, or consider a more direct connection between the Dante device and the DSP’s Dante port (similar to PC and DSP on the Control network from earlier in his document).
• Restart & Reboot Remote Terminal Commands; these commands offer various ways of restarting either firmware alone or power cycling the unit as a whole along with restarting firmware. Note: these commands should only be used when the conditions are safe and the system is not in a critically active state. It is also highly recommended that amps be turned off before sending these commands, as a pop may occur which some devices may be sensitive to.
• “R!”; this command reboots the main processor, and power cycles the unit, but doesn’t restart the Dante card.
• “R!!”; this command reboots and reinitializes both the main processor and the Dante card, as well as power cycles the unit – this is the same as a manual power cycle by pulling the cable, just can be done from Remote Terminal instead of physically on the unit.
• “BR”; this command restarts the firmware on the Dante card but doesn’t restart the main processor firmware and doesn’t power cycle the unit.
• “R?”; this command restarts the main processor firmware but doesn’t restart the Dante card firmware and doesn’t power cycle the unit.

Restarting the Dante Discovery Service

If Dante Controller isn’t discovering any devices the Dante Discovery Service could be in a bad state.

• Close Dante Controller.
• Open Task Manager and navigate to the Services tab.
• Scroll in the window to find the “Dante Discovery” service.
• Right click on this service and select Restart.
• Give the PC some time to restart the service and re-attempt to locate devices in Controller.

DHCP RESET OF DSP NIC

Most of the time it will be easy to find the IP address of a DSP. As was covered earlier in this document, the front display of Symetrix DSPs can be cycled to show both the Control and Dante IP addresses (Dante IP is in the System Pages). For Symetrix xIO devices that don’t have a screen to display their IP it can be difficult to find this, especially if you’re unable to locate the unit on the network.

Every Symetrix device (except for the ARC series) has a factory reset button. Reference to location of these buttons can be found in another Symetrix Tech Tip document Factory and Network Interface Card Resets 

If a Symetrix device has previously been set to a static IP address, you can single short press the factory reset button to reset the NIC to DHCP. Once reset and then manually power cycled, the unit should receive a DHCP address if a server is available or resolve back to its 169.254.x.y link local address. Note: BE AWARE, a long press of the reset button will factory reset the unit. This short press should be a “good solid click”, similar to pushing a mechanical elevator button

In many A/V applications, it may be specified or simply practical to have the DSP recall a particular configuration of saved parameters, such as sources, gains, mutes, and matrix routing at a scheduled time of day or week. These stored settings are known as “presets”. Presets are a digital snapshot of a single parameter or a collection of parameters that can be triggered with one command or button press. Storing and recalling presets in a DSP is analogous to taking a snapshot of a set of parameters in the DSP, and at a later time during operation, showing the DSP the snapshot and requesting that it set the parameters back to the previous configuration exactly as they appear in the snapshot.

All Symetrix DSP hardware has the ability to trigger presets at a particular time and day when the presets are scheduled using the Event Scheduler in the DSP setup and configuration software. Once a preset has been stored, it can be scheduled to trigger on a single date or as a reoccurring event. Exclusions of dates can be made to accommodate a changing schedule.

For example, in a high school at 8am Monday through Friday a bell may be scheduled to sound; however, during spring and summer break this bell would not need to ring while students are not attending school, so these spring and summer dates can be excluded from the schedule. In this example, most of June, and all of July and August (80 dates) have been excluded from triggering the Morning Bell preset.

There is a problem that can arise when presets are scheduled to be triggered at a particular time and day, and this problem is called “clock drift”. In order for the DSP to trigger a scheduled event, the DSP must keep a real-time, internal running clock, so that it knows the current time and day. This clock is generated
by an internal oscillating crystal, which over time “drifts” ever so slightly away from the actual time of day. This drift is usually quite small, on the order of 10 ppm (20 ppm worst case) or 6 seconds/week. This means however, that after one year of operation the internal clock could drift by 314 seconds, and as such the Morning Bell preset in the previous example would be triggered 5 minutes early after one year. After 5 years the preset would sound approximately 26 minutes early, which in most cases would be unacceptable.

What can be done to fix or stop clock drift?
The best approach is to synchronize the DSP to an NTP Server.

Synch the DSP to an NTP Server:
Network Time Protocol (NTP) is a networking protocol for clock synchronization between computer systems over packet-switched, variable-latency data networks. In operation since before 1985, NTP is one of the oldest Internet protocols in use.

If the DSP resides on a network that contains a server providing NTP services, the DSPs clock can sync with that server by Enabling NTP Synchronization and entering the NTP server’s IP address. If the DSP has a valid network route to the internet, any publicly available NTP server may be used.

 

Click this link for a list of public NTP server IP addresses hosted on the Internet: http://tf.nist.gov/tf-cgi/servers.cgi

In the Symetrix Jupiter and Zone Mix 761 software, the NTP server IP field is accessed in the Event Scheduler by clicking the ‘Set Device Clock’ button and then the ‘Advanced’ button.

In Symetrix Solus, the NTP server IP must be entered using Remote Terminal and the “Write NTP” command. Locate Remote Terminal (c:>Program Files>Symetrix>SymNet
Designer 10.0) and then type “WN (Example: WN 192.168.100.23)

In Composer each DSP can be set to a NTP server by accessing the unit properties.

Note: Symetrix Legacy and Express hardware does not support NTP clock sync

Reset Clock using Set Clock:
The DSP clock can be set or reset without downloading or pushing a file to the DSP using either Designer or Composer. Make sure the DSP has been located then select “Set Clock” from the “Hardware” menu.

 

Designer:
Time, Date and Daylight Saving Time can all be set using the “Set Clock” window. Once the desired setting has been entered click the “Set Clock” button.

Composer:
Sync to PC Clock or a specific date and time can be set using the “Set Clock” window. Daylight Savings Time can also be enabled. Once the desired setting has been entered click the “Set Clock” button.

No audio on an output. Echo in a conference room. A page message playing in the wrong zone.

 

These three issues are but a handful of situations that can occur during the commissioning phase of a Composer audio system, and of which some troubleshooting steps will need to be taken in order to resolve the problem.

 

Troubleshooting a complex signal path can be time consuming; however, with the right tool troubleshooting a DSP signal path can be done easily, intuitively, and within a very short amount of time. Composer offers the “Trace Signal Path Forward” tool, which will show the complete path of any input signal with Composer.

 

To use this feature, simply right click any wire in the Composer signal path and choose “Trace Signal Path Forward”. This will cause the entire signal path for that source or mix to highlight red.

 

Here are three examples of using Trace Signal Path Forward to troubleshoot the examples problems mentioned in the first paragraph.

1) No audio on a particular output:
In this example, there are 12 mics used for an automix system. Mic input #5 can be heard in a single zone, output #4, when it should be routed to all 8 outputs of a Radius. When Trace Signal Path Forward is used, it becomes obvious that mic #5 is routed to only 1 zone via the Submix Matrix output.
Hint: follow the red wire

Since mic #5 is only present on the Submix Matrix output #4, open the user interface of the Submix Matrix. Notice that mic #5 is only routed to output 4. Click the connect button for mic#5 for all other outputs within the matrix to solve the problem of missing mic#5 audio on the other 7 Radius outputs.

2) Echo in a conference room:
In this conference room example, the far end caller is complaining of hearing echo. Typically echo in a conference room is caused by having an incorrect mix being feed to the REF input for a mic or all mics. The REF input should only ever consist of the far end caller and any local media sources. Echo is when the far end caller hears themselves talking, when their audio plays in the conference and then enters the mic and is sent back to the fall end caller. So, if they are hearing themselves echo back, first check to make sure the far end caller audio is routed to all mic REF inputs.

2) When Trace Signal Path Forward is it is easy to see that the far end caller is not routed to REF 8, so the problem in routed is quickly located. Using this feature is100 times faster than muting all mics except one, then checking for echo, and repeating this procedure until the mic that causes echo is located.

Here is a close up of the problem.

Notice how REF #8 is not receiving the far end audio. When the wires are followed back upstream it is easy to see a wire is missing on the 8th channel, between the compressor and summer, where the far end audio stops passing to the REF #8. Once this wire is placed, then REF #8 will get the far end audio and the echo problem should be solved.

3) A page message playing in the wrong zone:
In this example, the customer has complained that when a certain preset is triggered and a page is made, the page is being routed to a wrong zone. The page is only supposed to go to zones 1,2,4,5,8, however it is also sounding in zone 7. When onsite, first trigger the preset that is causing the issue for the customer, then use Trace Signal Path Forward and follow the page input. Notice the highlighted red line indicates that the page does indeed get routed to output zone 7, as well as the correct zones.

Following the page signal path from the input to the output it becomes clear that the mono distributor is incorrectly routing the page to zone 7 when this preset is triggered.

To fix this routing problem, uncheck button 7 in the mono distributor, then right click the module or button 7 and save it in the off state to the respective preset. Use the preset manager to recall the presets to insure the page will only be routed to the correct zones when this preset is triggered.

Conclusion:
There are a variety of ways in which an incorrect signal path can create a lengthy and difficult troubleshooting session during the commissioning stage; however, with the Trace Signal Path Forward tool that Composer provides, finding and fixing signal path routing errors is easier than ever.

The purpose of this Tech Tip is to provide information and instruction on using AES67 with Symetrix Dante-enabled DSPs. The AES67 standard provides interoperability between different forms of AoIP (Audio over IP). AES67 is not a networking solution in and of itself, but rather a group of interoperability specifications for connecting media streams. AES67 is supported by various IP-based audio networking systems such as Dante, Ravenna, Livewire and Q-LAN.

Because Dante supports AES67, this allows Symetrix Dante-enabled DSPs to receive and transmit audio with other IP-based audio networking systems, Q-LAN as an example. When using Symetrix Dante enabled DSPs with AES67, there a few key points to keep in mind:

• Symetrix Dante-enabled DSPs are compatible with AES67, but are not AES67 specific hardware.
• AES67 stream assignments are handled by the receiving device
  • AES67 streams will only appear as a transmitter in Dante Controller.
  • AES67 transmit streams from a Symetrix Dante-enabled DSP will NOT be assignable in Dante controller.
• Here is a link to set up AES67 receive flows with Q-SYS
  • AES67 is capable of unicast and multicast communication, however Dante’s implementation of AES67 currently only supports multicast.
  • When two Dante-enabled devices are passing audio between each other they will always use Dante for the communication, regardless of AES67 streams.
• Audinate’s Ultimo chipset does not currently support AES67
• Here is a link to the AES67 standard

AES67 Receive Stream

Here are the instructions for creating AES67 receive buses, using the generic Network Receive Modules (This example uses a Radius AEC and QSC Q-SYS Core 250i)

aes 1

1. From the Toolkit, add a Radius AEC to the Site View page.

aes 2

2. Open the Design View page by double-clicking the Radius AEC.

aes 3

3. From the Toolkit, expand Network I/O Modules, then expand Receive Modules.

aes 4

4. Double-click or drag in a New Network Receive Module.

aes 5

5. The Network Receive Module Properties window will open automatically. Click the button to “Add New Bus.”

aes 6

6. Change the type to AES67.

aes 7

7. Click the “Browse AES67” button.

aes 8

8. Select the desired AES67 multicast stream from the list.

aes 9

9. Click the “Select AES67 Stream” button.

aes 10

10. The New Bus window is now updated with the AES67 stream information (device network name and channel names).

11. The new AES67 receive bus is available in the Network Receive module Properties window.

12. Click Ok. The new receive bus has now been created.13. Push the site file and Composer will make the AES67 to Dante subscriptions.

13. Push the site file and Composer will make the AES67 to Dante subscriptions.

AES67 Transmit Stream

Here are the steps to create AES67 transmit streams:

aes 11

1. Open the site file to the Design View page.

aes 12

2. From the Toolkit, expand Network I/O Modules, then expand Transmit Modules.

aes 13

3. Add a New Network Transmit Module. The Network Transmit Module Properties window will open.

aes 14

4. Edit the name of the transmit bus. Note: Naming of transmit buses is very important for organization.

aes 15

5. Select the number of channels in the transmit stream.

aes 16

6. Select the transmit bus type.

7. Name the individual transmit channels.

8. Click OK and the transmit bus will be added to the site file.

6. Select the transmit bus type.

7. Name the individual transmit channels.

8. Click OK and the transmit bus will be added to the site file.

This is a general-purpose step-by-step guide for connecting to Symetrix digital signal processors and related hardware with a PC. Please note that Symetrix only recommends using Windows 10 and above. Other operating systems are not officially supported at this time.

Step 1 – Install the right software for the device

Symetrix site design software is used to connect to Symetrix devices and is available to download, install, and run for free. The required software will depend on the devices that needs to be accessed:

Composer:

Current Symetrix open-architecture DSPs all use Composer, which can be downloaded here. These include:

• D100
• Radius
• Prism
• Edge
• Solus NX

Other Symetrix hardware that can be accessed through Composer will include:

• Endpoints and expanders (xIn, xOut, and xIO devices)
• T Series touch panels
• W Series wall remotes
• Control expanders (xControl, Control Server)

Important: To avoid errors when going online with the hardware, please download the version of Composer that matches the DSP’s firmware revision number as closely as possible. This number can be found by cycling through the system pages on the front LCD panel of the DSP.

Integrator Series:

Software for Symetrix’s current Integrator Series (closed-architecture) DSPs can be downloaded here. These include:

• Jupiter
• Zone Mix 761

Legacy Hardware:

Legacy open-architecture DSPs such as 8×8 DSP, Express CobraLink, and original Solus (non-NX) require SymNet Designer. This software has been discontinued and is no longer supported by Symetrix, but the final version (10.7) can be downloaded here. Software for all other legacy products, such as Zone Mix 760, AirTools-series, and Lucid-series, is no longer available for download.

Step 2 – Make sure the PC is on the right network

Once the correct software has been downloaded, the next step is to connect the PC to the device’s control network. If a DSP is Dante-enabled, make sure not to confuse the Dante ethernet port for the control ethernet port. Configuration of these devices through the Symetrix software is always done through the control port.

By default, Symetrix devices will obtain an IP address automatically, either from a DHCP server or, if a DHCP server is not available, by obtaining a link-local (169.254.x.x) IP address. Most Composer-enabled devices will display their IP address on the front LCD panel. Cycling through the system pages on the front LCD will additionally display the subnet mask. If a device has previously been configured with a static IP address, it can be reset to DHCP by briefly pressing the device’s reset button, which is usually recessed in the housing on the back of the device.

ncpa

It is important that the PC’s network settings match those of the devices being used in the system. To check this, enter ‘ncpa.cpl’ in the Windows search bar to open the list of network adapters on the PC:

Right click the network adapter that will be used to connect to the device, select ‘Properties.

version

Then double click ‘Internet Protocol Version 4’:

address

The network settings of the PC’s network adapter will display. If the Symetrix device is set to DHCP, select ‘Obtain an IP address automatically.’ Alternatively, a static IP address and custom subnet mask can be set here:

Important: Ensure that both the IP subnet and subnet mask of the network adapter match that of the device. If setting the PC to a static IP address, it must be a different/unused IP address on the network. If connected directly to the DSP with a static IP address, setting the PC to an address “right next to” the DSP usually safe. Example; if the DSP IP address is 192.168.100.50, set the PC to 192.168.100.51.

Step 3 – Locate the Symetrix hardware on the network

Once the PC is on the correct network, open the appropriate Symetrix software. The next steps will depend on the software being used.

Composer:

site

If a copy of the site file is available on the PC: Select the ‘File’ menu > Open and select it from File Explorer. In Site View, all located devices will have a checkmark in the lower left corner. If there is no checkmark present, click the empty box in the lower left corner of the device to open the Locate Hardware menu:

In the Locate Hardware menu, a list of available devices will appear. If necessary, click ‘Select Network to Search…’ to ensure that the correct network adapter is being used to scan for devices. Either double click the device in the list or highlight it and select ‘Locate to Selected Hardware’ to finish locating the device:

Repeat the above process for all devices in the Site View.

If the site file needs to be pulled from the unit:Go to the ‘Hardware’ menu > ‘System Manager’ > ‘Hardware’ tab. A list of all available units on the network will display. If needed, click “Select Network to Search…” to change the network being scanned for devices. Highlight the desired unit, then select ‘Go Online (Pull from Unit…)’:

The Pull Site File From Hardware Wizard will appear. Select a location on the PC where the site file will be saved, then click ‘Next’:

Next, select either ‘Yes – Synchronize to All Changes’ to keep any changes made to the configuration while last online with this site file, or ‘No – Abandon Changes’ to revert to the archived version of the site file. ‘Show Advanced Options’ allows for more granular control over which changes are kept when synchronizing:

Select ‘Next’, then either select ‘Finish’ to go online with the site file as-is or select ‘Cancel’ to make changes to the site file before going online:

A note about Dante devices– Any Dante devices in the design must be located through a Symetrix DSP that has already been located:

As of Composer 8.5, an xIO Updater/Configurator module may be added to the site view to configure Symetrix xIO Dante devices if a Symetrix DSP is not available. Symetrix recommends using separate networks for Dante and control.

Integrator Series:

Locating an Integrator Series DSP is done in the Connection Wizard of the Jupiter or Zone Mix 761 software. This can be done either by selecting ‘Existing File on Device’ > ‘Open Connection Wizard’ from the startup menu, or by selecting the Connection Wizard icon in the top ribbon:

Once the Connection Wizard opens, select the option that best fits the connection type, then select ‘Next’. A list of the PC’s network adapters will appear. Select the one that is connected to the ethernet port of the device, then select ‘Next’. Select ‘Open Network Connections’ to show these network adapters in Windows Control Panel if any settings need to be changed:

A list of devices will appear. Any devices not compatible with the current site file will be grayed out. Select the device, then select ‘Next’. Selecting the ‘Properties…’ button will allow a static IP address to be set for the device if desired:

On the final screen, select ‘Finish’ to close the Connection wizard. To go online immediately, ensure the ‘Go online now’ box is checked:

Step 4 – Go online with the system

Composer:

online

Once all devices in the site file have been located, select ‘Go online (push site file to hardware)’:

Note: The icon with the yellow arrow is for pulling the site file from the located hardware. Please see the passage entitled “If the site file needs to be pulled from the unit” in the previous section for more information on pulling the site file from the hardware.

Next, the Site Preferences window will appear. These are generally advanced options that can be left alone, however if Dante routing is being managed in Dante Controller rather than in Composer, uncheck the box next to ‘Configure Network Audio.’ Click ‘OK’ to proceed:

dialogue

At this point, if the site file has not yet been saved to the PC, the File Explorer will appear and prompt for a filename and location to save the file to. If any ARC remotes are present in the design, a dialogue will appear and ask if all remotes should be programmed now. Regardless of whether ‘Yes’ or ‘No’ is selected here, the system will continue to push and go online:

success

Once the site file has been successfully pushed, a success dialogue will appear. After clicking ‘OK’, the system volume will gradually ramp up unless the system mute is engaged:

Now that the system is online, parameters can be changed in real time, and signal meters will display their data. However, if any modules are moved, added, or deleted, or if any wires are changed, the system will automatically go offline. The site file must be re-pushed in order to go back online.

Important: The firmware versions of all devices in a Composer site file must match the version of Composer being used before going online with the system. If this is not the case, a message will appear prompting a firmware upgrade before the system can go online. Please refer to the Updating Firmware with Composer Tech Tip for further assistance.

Integrator Series:

After finishing the Connection Wizard, select the orange ‘Off-line’ button in the top ribbon. The drop-down arrow can be selected to choose which previously located device to go online with:

A prompt will then appear allowing the user to select whether to push the currently open configuration file to the device, or to pull the configuration file off of the device and save it to the PC.

Once the system is online, parameters can be changed in real time, and signal meters will display their data.

Integrator Series devices will operate normally with the factory firmware and should not require firmware updates to go online.

FAQs and Troubleshooting

“My device does not appear in the Locate Hardware menu.”

• Double check that the PC’s NIC and the Symetrix device are on the same network.
• Double check that the selected network in the Locate Hardware menu corresponds to the intended NIC.
• Change all octets of the IP address and subnet mask being searched for to ‘255’, uncheck the box next to ‘Don’t show located and enabled units’, and check the box next to ‘Show incompatible hardware’ in order to broaden the search as widely as possible.
• If a USB to ethernet adapter is being used with the PC, connect using a standard ethernet port instead if possible.
• Power cycle both the PC and the device.
• Re-seat the ethernet cable in both the PC and the device.
• Try a different ethernet cable.
• If the device is connected to the PC through a network switch, try a different switch port, or connect directly to the PC instead.
• If all else fails, disconnect the device from the network, reset its network settings by tapping the reset button once, then directly connect it to the PC (ensuring the PC is set to automatically obtain an IP address).

“I’m getting a ‘Failed to go online’ error message.”

• Disable Windows Defender Firewall and any third-party antivirus/firewall programs that may be blocking network traffic.
• Double check that the device firmware versions for all devices in the site file match the version of Composer being used (the first two numbers are most important).
• Power cycle both the PC and the device.
• If the device is connected to the PC through a network switch, try connecting directly instead.
• If a device cannot be located and is not needed in the site file, right click it and select ‘Disable Unit’.

“I can’t locate my Dante device.”

• Double check that the DSP is Dante-enabled by going to the ‘Tools’ menu > ‘Launch Remote Terminal’ > ‘Options’ menu > enable ‘Debug Mode’, then send the command info cards to the IP address of the DSP. If ‘Non-Dante Clock Card’ is displayed in the output under ‘Audio Network Card’, then the device does not have a Dante card installed. Please contact sales@symetrix.co to purchase one. If ‘No Card Present’ is displayed instead, there may be a problem with the Dante card.
• Double check that the Dante device is connected to the Dante port of the DSP.
• Connect the Device directly to the DSP’s Dante port, bypassing any network switches. If it can be located using this method, there may be a problem with the network.
• If all else fails, connect the PC to the Dante network, or directly to the Dante device, and verify that it appears in Dante Controller. If not, then there may be a problem with the Dante device, or it may be set to a static IP address outside of the Dante network.

“What does the yellow checkmark next to a device in Composer mean?”

A yellow checkmark means that the device is muted, while a green checkmark means that the device is unmuted.

Introduction

The Symetrix xControl serves a similar purpose for Edge, Radius NX, and Prism as the Control I/O did for legacy SymNet SymLink and Express Cobra hardware. Its primary purpose is to bring the overall cost of logic I/O heavy systems down.

xControl Rear Panel

1. Ethernet: 10/100 Base-T Ethernet port for network connection to the system over IP. Features auto-crossover sensing for direct device-to-device connections. Accepts PoE IEEE 802.3af Class 1.
    
2. RS-232: Two serial communications interface for sending strings to 3rd party devices or accepting 3rd party control commands. Port Settings: 57.6 kbaud (default), 8 data bits, 1 stop bit, no parity, no flow control.
    
3. External Control Inputs: Eight (8) analog control inputs. Each analog control input can be configured to support 1 potentiometer or 2 closures (+3.3 VDC reference voltage supplied).
    
4. Logic Outputs: Sixteen (16) logic outputs with eight (8) paired common ground pins. Logic Outputs go low (0V) when active, and are internally pulled high (5V) when inactive and can drive external LED indicators directly.
    

Examples of Common Use Cases

Conferencing Push To Talk and LED Muted/Active Indications

In conferencing applications the logic outputs are typically used to either light LEDs directly or interface with something expecting a control voltage that controls the LEDs itself. Typically, they are following mutes somewhere in the Symetrix design which are linked to push-to-talk (push-to-unmute) logic.

External Relay Trigger

External relays are often driven by logic outputs for the purpose of controlling a power sequencer or controlling a “conference in session” lamp/sign.

Camera Control

Logic outputs are sometimes use to interface with the GPIO inputs of a camera PTZ control unit which essentially expects contact closures to trigger it to preset camera positions. These may be driven in our system by presets, the Gating Automixer channel “ON” LEDs, or the PTT logic detailed above. Most often this type of setup is used during video conferencing or in court room applications.

Projector Control

The dual RS-232 ports on the xControl can be configured to send any custom RS-232 string in ASCII or Binary allowing Symetrix to control 3rd party hardware. Often times a projector is used in a conference room or class room application, and must work in tandem with the audio system. Using an ARC remote or SymVue control screen as the user interface, when prompted by the host DSP the xControl can send 3rd party protocol commands to a projector, controlling common parameters such as On/Off and the selected input source.

Powering and Hookup to the Network
 

Warning: The xControl is a true PoE (power over Ethernet) device and must be connected to the host DSP through the data network. It is not an ARC network device. Do not under any circumstance plug the xControl Ethernet port into the ARC port on a Edge, Prism, Radius NX, or ARC-PSe. The ARC DC voltage may damage the xControl, which may cause a failure not covered under the manufacturer’s warranty.

Configuring IP Parameters

x 1

Locating Hardware

x 2

OR

x 3

OR

Discovery of, and connection to, xControl hardware is done with the Locate Hardware dialog found under the Hardware menu or by clicking the Connection Status box in the bottom left corner of the xControl icon.

1. IP Configuration with Composer
2. The Locate Hardware dialog will scan the network and list available units with DHCP IP addresses.
3. Select the xControl unit to assign a static IP address and click the Properties button.
4. To assign the xControl a static IP address, select “Use the following IP address” and enter the appropriate IP Address, Subnet mask and Gateway.
5. Click OK when finished.
6. Next, back in the locate hardware dialog, ensure the xControl device is highlighted and click “Select Hardware Unit” to connect the selected xControl on the network to the xControl in the Composer Site File.
7. Close the Locate Hardware dialog.

The purpose of this Tech Tip is to show you how to probe, listen to, and meter any signal by using the Selected Wire Audio Module. This is useful for commissioning a Symetrix system or troubleshooting a noise issue while onsite. The second half of this Tech Tip provides detailed instructions to accomplish this

Selected Wire Audio Module

The Meter Bar displays the physical (unit) input and output meters as well as the Selected Wire Meter. The panel may be resized and docked. Its orientation may be switched by clicking on the double-ended arrow button in the upper right corner. Each group of input or output meters may be collapsed or expanded by clicking on  the +/- buttons above their labels.

One of the great things about being able to click on any wire in the design is that  you can quickly monitor any point in the signal path. When a wire in the design is selected, it will meter audio passing as long as the unit is online. The Selected Wire Audio module can be wired directly to one of the unit outputs to be connected to headphones, near-field monitor speakers, or even a listen cue wedge for live sound applications.

wire 1

The Selected Wire Audio module can be wired to a Matrix Mixer module, so that it can be routed anywhere in the software or to any of the outputs. This function is available to anyone logged in as a user for a live monitor feed. The end user could matrix it to a monitor feed during a live broadcast.

wire 2

In this example, the Selected Wire Audio module is wired to a Dante flow so that it can be routed to outputs of another Dante enabled device such as the Symetrix xOut 12.

In this example, you can easily compare the sound pre or post Compressor, as well as pre or post EQ.

The Selected Wire Audio module can be wired into an Oscilloscope module, for testing of the signal path.

wire 3

You can wire into a Dante channel that can be routed directly to the laptop speakers using Dante Virtual Soundcard (DVS). A full set of instruction can be found in the Tech Tip: Sending Audio with Dante Virtual Soundcard to Composer hardware.

Follow these simple steps to monitor any point in the DSP signal path from the speakers in the host computer running Composer:

Necessary items:

• Composer software installed on the host computer
• Symetrix Radius NX, Prism, or Edge DSP • Dante Virtual Soundcard (DVS) installed on the host computer (www.audinate.com) • Dante Controller installed on the host computer (www.audinate.com)

Step 1:

In order to have the ability to probe the DSP signal path in Composer and have  the audio play out on the host computer speakers via Dante, it is necessary to merge the Ethernet control network with the Dante network. Simply use a short  CAT5 patch cable to connect one unused Ethernet port on the Symetrix device to  a Dante port. Plug the host PC into the other unused Ethernet port. When using Prism, an external switch is required.

WIRING KEY: Dante = GREEN, Ethernet/Control = RED, Dante/Ethernet Control  Merger = BLUE

Step 2:

In Composer create a one channel, Dante Transmit Flow and wire its input to the output of the “Selected Wire Audio” module. Also name the Flow and the channel. In the example, the Dante Flow is entitled “Monitor Send” and the channel name is “Dante Laptop Monitor.

wire 4

Note: Both the Selected Wire Audio and Diagnostic modules are located under the ‘Ins’ modules of the DSP.

Step 3:

Open the Dante Virtual Soundcard with Audio Interface set to “WDM” and audio format at 48 KHz.

Step 4:

Configure the host PC/laptop to use the DVS:

• Go to Control Panel->Sound
• On the “Playback” tab make sure the laptop speakers are the default device

• On the “Recording” tab click on DVS Receive 1-2 and click the Properties button.

• Click “Listen to this Device” and then click OK

Step 5: 

Open Dante Controller

• Expand the “laptop network name” under Dante Receivers.

Example: rcurtright-lt2

• Expand the Radius,Prism, or Edge unit under Dante Transmitters.
• Click the cross points for the laptop DVS channel 01 and 02 so they receive audio  from the DSP transmitter’s channel “Dante Laptop Monitor

• When the cross points get green checks, the Dante audio should now be received and played from the laptop.

Step 6:

Return to Composer and click on any wire to “select it” so that the wire turns red. Now the output of the “Selected Wire Audio” module will be the audio on the red selected wire, which will enter the Dante Transmit Flow “Monitor Send”. The audio then travels across the Dante network to be received by the DVS where the selected wire audio will play out of the host computer’s laptop speakers. Start testing, click, probe, monitor!

Hint: Place an Oscilloscope, found under “Meters and Analyzers” in Composer, in line with the Dante Transmit Flow. This creates the added benefit of viewing the selected wire audio on a scope while at the same time monitoring with the host computer speakers.

SNMP (Simple Network Management Protocol) is an application-layer protocol that facilitates the exchange of information between network devices. Third party SNMP software includes programs such as Aprisma Spectrum, CA UniCenter, HP OpenView, and IBM Tivoli.

Using SNMP an IT manager can monitor various pieces of hardware, all from different manufactures, for things such as IP info, network bandwidth and resources used, and even the current health of hardware and its components. Often times, SNMP allows an IT manager to address a problem before it
actually happens, or when hardware fails to be able to pinpoint the exact unit and where it is located within a venue or network.

 

Currently, Composer hardware does not support SNMP, but there are options that are worthwhile to consider.

 

First, Composer adds a ‘Diagnostic’ module within the Design View of every DSP in the system. The various features the diagnostic module displays can help with determining the health of a unit before or after a failure, the IP address and DHCP lease info, processor temp, and control indication.

With the release of Composer 1.2 the diagnostic module can be assigned controller numbers to most of the fields related to a units health and operation. Controller numbers can be assigned to individual fields one by one, or all at once by right clicking the diagnostic module and choosing “Assign All Unassigned Controls to Controllers (#-#)”.

Once controller numbers have been assigned to the diagnostic fields, the module will appear like this if ‘Tools->Super-impose Assigned Controller Numbers’ is checked.

These numbered fields can now be monitored for their state with remote controls such as an ARC-2e, SymVue, or with a smart device using ARC-WEB.

Additionally, if SNMP is required, many third party control systems such as Crestron or AMX offer full SNMP monitoring functionality. This means that the control system can monitor the diagnostic fields for all Symetrix DSP hardware across the network, acting as the SNMP intermediary between the SNMP monitoring software and system.

 

For more info on SNMP monitoring using Crestron, click to following link:
http://www.crestron.com/downloads/pdf/featured_articles/169/Bridging_AV_And_IT.pdf

 

For more info on SNMP monitoring using AMX, click to following link:
http://www.prnewswire.com/news-releases/amx-extends-netlinxtm-withjavatm-for-industrys-first-dual-language-control-system-72165242.html

Managing digital audio on a network may seem like a daunting task, particularly when the network is not a dedicated network switch, or series of switches used only for audio, but instead is shared amongst a variety of network devices, such as printers, PCs, control equipment, and servers, all sharing network bandwidth with the networked digital audio.
 

Understanding how to manage networked audio becomes extremely important when commissioning a system in which an existing corporate network is intended to provide the network infrastructure between two or more Dante capable; devices, third party consoles, or I/O end points. In such a scenario, managing the Dante audio becomes a necessary consideration; however, Dante has made the job of managing the unicast and multicast audio simple and straight forward.
 

First, it is best to understand the basics of unicast and multicast network traffic and how it relates to Dante networked audio, then cover the method for managing these two protocols.
 

Within Composer there are two types of Dante flows that can be defined; unicast and multicast.

Unicast: A unicast flow is transmitted from one Dante device and is routed to exactly one other receiving Dante device. A unicast flow is routed across the network via a destination IP address embedded within the header of the Dante network packet.

Multicast: A multicast flow is transmitted from one Dante device and is typically routed to multiple receiving Dante devices. A multicast flow, when not managed, will be transmitted to every device connected to the network.

It is recommended to use unicast Dante flows whenever possible, especially when Dante is on a shared data network. This eliminates unnecessary network traffic by ensuring the Dante audio travels directly from one IP address to another, rather than proliferating across the entire network. That being said, when a Dante channel is routed to three or more devices, it is a more efficient use of the network bandwidth to use multicast flows.
 

The fact that multicast Dante, when not managed, will be transmitted to every device connected to the network means that each network device must analyze the multicast packet of data, determine if the packet must be received, and then either receive the packet or disregard it and continue operation. For devices without Dante capabilities, receiving large amounts of multicast data can lead to slower processing speeds, sluggish network response, and other performance related issues. In fact, a type of denial-of-service-attack utilizes this exact method for sabotaging network service.

The question then becomes, is it possible to manage multicast Dante flows on the network such that these multicast flows are only routed to the LAN ports of the Dante receiving devices? The answer is “yes”, and in fact, this management is very easy to implement. This management process is called “IGMP snooping”.

IGMP snooping is a feature of a managed network switch that allows it to listen in on conversations between the multicast source, receivers, hosts, and routers. By listening in to these conversations, the switch builds and maintains a map of which links need which IP multicast streams, such that multicast streams may be filtered from the links which do not need them, and thus ports receive only specific multicast traffic they have subscribed to.

From the diagram above, it should be clear that as a standard practice IGMP Snooping should be enabled on all shared networks with multicast Dante flows.

Simply enabling the IGMP Snooping feature is all that is required, and from that point on, Dante multicast traffic will be filtered, kept from broadcasting to all devices on the network, and routed only to links containing Dante devices subscribed to the multicast flow.

Composer hardware utilizes Dante as the digital audio bus for routing audio between hardware and 3rd party Dante enabled hardware. Setting up a Dante network is made simple, quick, and easy with Dante’s ability to receive an IP address using DHCP and then auto-resolve connections between units. That being said, there are times when it will be specified or ideal for the Dante ports to have specific and unique, static IP addresses. Composer hardware allows for assigning static IP addresses to the Dante ports using Dante Controller.

Follow these easy steps to assign a static IP address to a Dante port on a device:

Step 1: Download Dante Controller from the Audinate website:
http://www.audinate.com/index.php?option=com_content&view=article&id=305

Step 2: Plug the PC’s LAN port into the Dante network.

Note: If it is desired to stay online with the system using Composer while simultaneously monitoring or assigning static IP addresses to the Dante network, use a CAT5 or CAT6 jumper cable to merge the control port with the Dante network.
See the blue wire below:

Step 3: Launch Dante Controller and verify that it locates the Dante devices on the Device Status Tab.

Step 4: Double Click on the “Device Name” of the Composer unit to open the “Device View”.

Step 5: Go to the Network Config tab and select “Manually configure an IP Address”.

Step 6: Enter the static IP Address, Netmask, and Gateway.

Note: Be sure to use unique IP settings.

Step 7: Hit “Apply” to write the new static IP information to the Dante port of the Composer hardware.

Step 8: Use the drop down at the top of the Device View page to select the next Composer unit and repeat Steps 5, 6 and 7.

Composer is intuitive, open architecture, drag-n-drop DSP software for any and all commercial or live sound applications using Symetrix Dante enabled DSP hardware. Additionally, Audinate’s Dante has arrived in full force and is quickly becoming the industry standard, cross platform, network audio buss of choice in the commercial A/V and live sound market.

This tech tip provides a clear and concise set of instructions for both initial setup of Composer to Dante enabled hardware and/or troubleshooting a Dante network in roughly 5 minutes. Simply put, when setting up or troubleshooting any complex system “Less is more”. It is easier to get to the root of any problem when all moving parts have been minimized. Start small and build up to something bigger, that way any issue becomes apparent immediately rather than being buried in a slew of variables, making it much harder to identify.

When setting up a Dante network with Composer hardware the different units may be specified to be installed in different locations across a 3rd party network. Furthermore, Dante may be specified to be in “redundant” mode. However, for initial setup and troubleshooting of a Composer / Dante network of 10 Dante enabled devices or less, it is advised to first daisy chain the units together with Dante in “switched” mode, which is the factory default.

Once audio is successfully passing between the daisy chained units, proving that all Dante devices are communicating normally and that there are no hardware failures, then and only then (if specified) configure the Dante ports to redundant mode and move the hardware to their respective locations across the 3rd party network.

Steps for initial setup of Dante enabled devices (10 or fewer Composer devices):

1) Direct connect all Composer / Dante enabled units together by daisy chaining them:
• Start by connecting the primary Dante port of the first unit to the secondary Dante port of the second unit and continue until all units are daisy chained together.

 

2) Connect the PC to the DSP’s Ethernet port:
• The PC which has Composer and Dante Controller installed should connect to the right Ethernet port on the first DSP in the daisy chain, such as an Edge or Radius, which has dual Ethernet ports.

 

3) Link the Ethernet control network and the Dante network together temporarily for setup:
• On the top unit in Step 2, connect the left Ethernet port to the secondary Dante port.

 

4) Connect any 3rd party Dante device to the primary Dante port on the bottom unit in the daisy chain:
• If there is more than one 3rd party Dante enabled device, connect them to a common switch, and plug the primary Dante port of the bottom unit in the daisy chain into the same switch.

 

5) Open up Composer and connect to all units:
• The steps for connecting DSPs in Composer software to their respective hardware units is outlined in depth in the Composer certification online training.
• It is advised to leave all units and the PC in DHCP for initial setup

 

6) Push a site file into the system and check to confirm Dante passes between all units:
• If Dante passes between all units as it should, you can now:
o Switch to Redundant mode if necessary.
o Move the units to their respective distributed locations.
o Disconnect the Ethernet and Dante networks.
o Assign static IP addresses to hardware if needed.
• If Dante does not pass between all units, proceed to Step 7:

7) If Dante is not passing audio or a unit is not seen by Composer, open Dante Controller.
• Since the Ethernet port and Dante secondary port are connected together in Step 3, the PC can run Composer and Dante Controller simultaneously.
• Check Dante Controller to see if all of the Dante enabled units are recognized.

 

8) If Dante Controller cannot recognize a particular unit:
• Check CAT-5 cable connections
• Hard reset units with the hard reset button next to the Ethernet port, and then upgrade firmware being careful not to interrupt the upgrade process.
• Reconnect the units in Composer software and check Dante Controller to see if all units are present and accounted for.

 

9) If you’re still having issues, contact Symetrix support at support@symetrix.co
• Also check these additional Symetrix Dante resources:
o Know-it-Use-It-Troubleshoot-it-Dante.pdf
o Record-Audio-Dante.pdf
o DVS.pdf

Among the hundreds of manufacturers that have adopted Dante as their networked audio bus of choice, Yamaha and Symetrix stand out as early adopters and trendsetters for developing and standardizing products based around the Dante protocol.

 

As such, both Symetrix and Yamaha have come up with some recommendations for integrating Dante when commissioning or setting up a Dante network. This tech tip will cover recommendations from both manufacturers, their differences when applicable, and a brief troubleshooting guide should problems arise.

Yamaha Recommendations:

Setup:
The Yamaha and SymNet factory default Dante mode is “daisy chain” or Switched mode as it is called in SymNet. This means all Dante ports, Yamaha and SymNet, can be daisy chained together for the initial setup and no 3rd party network switch need be used.

 

If a 3rd party network switch will be used for Dante, connect the primary port of each Yamaha and SymNet unit into the 3rd party network switch. Once the DSP and Yamaha have been programmed correctly, all Dante subscriptions should connect automatically after a site file push or power cycle, which can be verified with Dante Controller. If for some reason the subscriptions do not reconnect, then 1) the subscriptions may not have been created correctly, or 2) the 3rd party network switch may be at fault and its settings should be confirmed as optimized for Dante. If problems persist, troubleshooting steps should be taken.

 

Troubleshooting subscriptions:
https://www.symetrix.co/wp-content/uploads/2013/08/SymNet-Specifics-for-Dante-Subscriptions-3rd-Party-Dante-Sources-and-Real-time-Dante-Matrixing.pdf
Troubleshooting Network Switch Settings:
https://www.symetrix.co/wp-content/uploads/2013/01/2012-11-Know-it-Use-It-Troubleshoot-it-Dante.pdf

Also like Symetrix, Yamaha recommends switching to Redundant mode only after verifying all units in the system pass Dante via the Primary port and that all units are reporting their current Dante mode as “Redundant”. Symetrix makes the same Dante setup recommendations in the following tech tip:
https://www.symetrix.co/wp-content/uploads/2013/04/2013-2-02-Setup-Dante-in-5-Minutes-Time-or-Less.pdf

Yamaha CL and QL Series:

The Yamaha CL Series consoles (e.g., CL3) can be setup to two available options for how Dante patching/routing is controlled. There is a “Dante patch by console” and a “Dante patch by Dante controller”.

 

If Dante patch by console is selected, then the Yamaha runs many processes that normally are handled by Dante Controller to allow the Yamaha console to control the Dante routing for devices that are mounted in the console’s I/O Rack. Do not mount Symetrix devices into the console’s I/O Rack because it
will result in unwanted Dante Patch changes to Symetrix devices when “Dante patch by console” is selected.

 

To be safe, Symetrix recommends the Yamaha Dante setup should be set to “Dante patch by controller”.
To do this, on the Yamaha CL Series console go to:
Setup / Dante setup /
And select: Dante patch by controller.
Then use Composer and, when applicable, Dante Controller for all routing changes in the Dante network.

 

Be aware that Dante Patching and Dante Patch Recall through the CL console will not be available.
Note 1: This White Paper on Dante subscriptions should be consulted before using Dante Controller to change Symetrix Dante routing:
https://www.symetrix.co/wp-content/uploads/2013/08/SymNet-Specifics-for-Dante-Subscriptions-3rd-Party-Dante-Sources-and-Real-time-Dante-Matrixing.pdf

Network Switch:

As of the writing of this tech tip, Yamaha does not officially recommend any particular brand or model of network switch for Dante. That being said, Yamaha has successfully used the Cisco SG300 is a variety of Dante applications and provides detailed instructions for setting up the SG300 to use with Dante here
on their website:
http://www.yamahaproaudio.com/global/en/training_support/selftraining/dante_guide/index.jsp

 

The Yamaha SG300 setup guide covers the following topics:

• Preparing to Configure a Network Switch
• Disabling Energy Efficient Ethernet (EEE)
• Constructing a Virtual Local Area Network (VLAN)
• QoS Settings (Prioritizing the clock synchronization)
• Multicast Settings
• Setting Multiple Switches (Copying settings)
  
  Note 2: Symetrix agrees with Yamaha that setting up a network switch correctly for Dante is necessary for reliable Dante operation and also does not recommend a particular brand or model of switch, nor does Symetrix provide setup instructions for a particular model. Symetrix follows Audinate’s lead by stating that any network switch can work with Dante, but some features on some switches will allow for larger and more reliable Dante operation.
  
  Dante makes use of standard Voice over IP (VoIP) Quality of Service (QoS) switch features, to prioritize clock sync and audio traffic over other network traffic. VoIP QoS features are available in a variety of inexpensive and enterprise Ethernet switches. Any switches with the following features should be appropriate for use with Dante:
  • Gigabit ports for inter-switch connections
  • Quality of Service (QoS) with 4 queues
  • Diffserv (DSCP) QoS, with strict priority
  • A managed switch is also recommended, to provide detailed information about the operation of each network link: port speed, error counters, bandwidth used, etc.

Additionally, both Yamaha and Symetrix recommend turning off all EEE features of the network switch to prevent low power operation from impacting audio performance.

Troubleshooting a Yamaha / Symetrix Dante connection:

If experiencing Dante failures between a Yamaha console and a Symetrix DSP, check the following:

1) Ensure that Symetrix Dante subscriptions (those channels Symetrix is to receive from the Yamaha) are setup using Composer. See Note 1 for clarification. Composer has  a “Dante
Browse” feature to make creating the Dante receive flows easily from 3rd party hardware simple, quickly, and intuitive. If Dante Controller is used to patch Dante audio into a Symetrix DSP, these subscriptions
will be temporary and will be lost after a site file is pushed or the Symetrix DSP is power cycled.

2) If the Yamaha is a CL Series console, ensure that the Yamaha Dante Setup is set to “Dante patch by controller”. Be aware that Dante Patching and Dante Patch Recall through the CL console will not be
available in this mode.

3) Yamaha SG300 Setup Guide recommends turning on IGMP Snooping when multicast Dante is being used. Typically Dante will not be affected negatively by this switch feature. However, Symetrix has seen a case or two in which the IGMP Snooping caused instability. So, if multiple Symetrix units are showing as “clock master” and IGMP Snooping is enabled in the 3rd party network switch, turn off IGMP Snooping on the network switch and power cycle all Symetrix units and the network switch. If turning off this feature solves the problem, then leave it turned off, otherwise IGMP Snooping can be left enabled as per the Yamaha recommendation.

This Tech Tip explains the purpose for Dante redundancy. This Tech Tip also provides step-by-step instructions to properly change a Composer-based system from “Switched mode” to “Redundant mode,” or, from “Redundant mode” to “Switched mode.”

Dante offers a full-time redundancy option with permanent primary and secondary audio transmission. Redundancy requires a second network, either using a second physical switch (recommended) or via a VLAN isolating the network traffic.

Audio data is transmitted on both the primary and secondary networks simultaneously. In the event of a failure on one network, audio will still continue to be transmitted via the other network.

All Symetrix Dante devices ship with Dante in the default “Switched mode.” This allows units to be daisy chained, eliminating the need for third-party networking hardware. When in “Switched mode” if the Dante hardware is setup physically in a redundant network configuration, traffic from the Primary Dante port will flow out the Secondary Dante port, and the Dante traffic from the Secondary Dante port will flow out the Primary Dante port. This creates a data feedback loop on the Dante network that will crash the Dante cards in the Symetrix units. Symetrix units with only Dante connections, such as the xIn 12 and the xOut 12, may become unresponsive until they are power cycled and the redundant Dante network switch is turned off or the redundant ports are unplugged from the secondary Dante network.

The diagram below illustrates Dante setup physically in a redundant network configuration.

The red wires are for Ethernet/Control. The blue wires are for Primary Dante. The green wires are for Secondary Dante. Notice that each set of wires has its own network switch.

Changing from Switched Mode to Redundant Mode

To switch the Symetrix system to run Dante in “Redundant mode” when already in the default “Switched mode,” use the procedure below.

1 Cable the Dante network as if it were in “Switched mode,” not “Redundant mode.” In other words, if using an external switch or a direct connection between two units, make connections only to the primary jack. If more than two devices are used without an external switch, daisy chain from one unit’s primary to the next unit’s secondary. Do not complete the loop from the last unit back to the first unit.

2. In Composer, go to Tools > Dante Flow Manager > Configure Dante. Select “Redundant Network.”

1. Push the file and go on-line with the units. This will take slightly longer than usual as the Dante units change their mode.
2. Power down the units.
3. Cable the Dante network as appropriate for “Redundant mode.” Connect the primary and secondary the separate switches like the above diagram shows.
4. Power on the units.
5. Push the file and go on-line.

Note: Dante devices that do not support redundancy must be connected to the primary network only.

Changing Redundant Mode to Switched Mode

To switch a Composer-based system to run Dante in “Switched mode” when already in “Redundant mode,” use the procedure below.

1. Power down the secondary Dante network.
2. In Composer, go to Tools > Dante Flow Manager (or Tools > Network I/O Manager > Configure Dante. Select “Switched Port”.
3. Push the file and go on-line with the units. This will take slightly longer than usual as the Dante units change their mode.
4. Power down the units.
5. Cable the Dante network as appropriate for “Switched mode.”
6. Power on the units.
7. Push the file and go on-line.

This Tech Tip is to provide instruction on how to add third-party Dante devices to the User Library of Composer. Symetrix Composer contains a list of supported third-party Dante devices. When these supported devices are added to a site file, Composer can create transmit/receive subscriptions as well as provide control for some devices.

The User Library is located in the Third-party Dante Devices section of the Toolkit in Device View. Any third-party Dante device can be added to the User Library. Once a device is added to the User Library, Composer will treat it as a supported third party Dante device. There are two methods to add a third-party Dante device to the User Library:

• Browse Dante Network
• Import XML File

Browse Dante Network

Add 1

Here are the instructions to add a third-party Dante device to the User Library by browsing the Dante network: (This example uses a Radius AEC and a Windows PC running Dante Via)

1. From the Toolkit, add a Radius AEC to the Site View page.
2. Next from the Toolkit, expand Third-party Dante Devices.
3. Expand the User Library and add a New Dante Device to the Site View page.

Add 2

The Dante Device User Library Manager window will open.

4. Click the “Browse Network” button.

Add 3

5. The Locate Hardware window will open and display all available Dante devices on that network.

Add 4

6. Select the desired Dante device, then click the “Select Hardware Unit” button.

Add 5

7. The Dante Device User Library Manager will now list that Dante device. Select the desired Dante device and click the “Select Device Type” button. This will add the device to the site view page.

Add 6

8. Once added to the library, these devices are available to add to any site file. Open the Design View page by double-clicking the Radius AEC.

Add 7

9. From the Toolkit, expand Network I/O Modules, then expand Receive Modules.

Add 8

10. Bus#1 is automatically created and available. Add Bus#1 to the Design View page.*
 

11. Push the site file and Composer will make the Dante subscriptions for these channels.

Import XML File

Here are the instructions to add a third-party Dante device to the User Library by importing an XML file: (This example uses a Radius AEC and a RDL DD-BN31 wall-mounted Dante interface)

1. From the Toolkit, expand Third-party Dante Devices.
2. Expand the User Library and add a New Dante Device to the Site View page.

The Dante Device User Library Manager window will open.

Add 9

Here are the instructions to add a third-party Dante device to the User Library by importing an XML file: (This example uses a Radius AEC and a RDL DD-BN31 wall-mounted Dante interface)

1. From the Toolkit, expand Third-party Dante Devices.
2. Expand the User Library and add a New Dante Device to the Site View page.

The Dante Device User Library Manager window will open.

Add 10

3. Click the “Import” button.

Add 11

4. Navigate to the XML file to be imported.

5. Select the XML file, click “Open.”

Add 12

6. Click OK when the Select Dante Device Specification File to Import window opens, confirming the file has been imported.

The RDL DD-BN31 device is now available from the User Library.

Add 13

7. Select the device from the list of Known Types; click the “Select Device Type” button.

Ad 14

The RDL DD-BN31 will be added to the site file and is now available from the Toolkit.

Add 15

8. From the Toolkit, add a Radius AEC to the Site View page.

9. Open the Design View page by double-clicking the Radius AEC.

10. From the Toolkit, expand Network I/O Modules, then expand Receive Modules.

Add 15

11. Bus#1 is automatically created and available. Add Bus#1 to the Design View page.*

12. Push the site file and Composer will make the Dante subscriptions for these channels.

*Whenever a Symetrix Dante-enabled Analog I/O Expander, third-party Dante device, or transmit bus are added to a site file, a receive bus is automatically created, and available from the Toolkit.

Introduction

This guide covers usage of Composer and Dante Domain Manager (DDM), exploring crucial do’s and don’ts. This resource aims to optimize your experience with these powerful tools for managing your audio network. While DDM is supported in past Composer releases, it is highly suggested that when using DDM with Composer, to always use the most recent version of Composer. At the time of this writing the most recent DDM compatible version is Composer 8.5.1.

Dante Domain Manager Operation

1. Pushing Site Files: When planning to utilize DDM, it should be ensured that the system is working properly before enrolling the devices in DDM. This includes ensuring device firmware is up to date. All Dante routing and utilization of Intelligent Module controls on SymVue screens be completed and pushed before enrollment. After enrolling devices into DDM, changes to Intelligent Modules and Dante routing will not be accessible without removing devices from the Domain and rebooting the DSP. This is due to some of the limitations on Dante Device locating under DDM.
    
2. Device Enrollment: After ensuring the system is in working order and Dante routes are as planned, enroll the devices into DDM using the same process as any other system. All devices in a Site File should be in the same domain. See Known Issues in 8.5.1 (8).
    
3. Enrollment Sequence: If devices are enrolled prior to initiating the ‘Push Site File’ operation, the ‘push’ functionality will fail as the Dante devices are no longer located in Composer. To correct this, un-enroll the devices from the domain, reboot the DSPs, push the completed Site File, and re-enroll the devices.
    
4. Adjustments to Site Files: If modifications to the Site File or DSP firmware are needed, it is essential to un-enroll devices before proceeding and if needed, reboot the DSP to locate non-DSP Dante devices. This step ensures smooth incorporation of changes without disruptions to Composer-Dante communications. Then, make the changes, push to ensure correct functionality, and re-enroll the devices into the Domain.

Dante Director

Similar operation has been observed while using Composer with Dante Director. Any steps taken to use Composer with DDM can and should be applied when enrolling devices into Dante Director. 

Operational Notes Using Composer 8.5.1 or Later

1. Clock Leader in Dante Controller: When combining Brooklyn 2 and Brooklyn 3 cards in the same domain, a Brooklyn 2 card will show as Primary v1 Multicast Leader and a Brooklyn 3 card will show as Primary v2 Multicast Leader in Dante Controller. This issue can be corrected by selecting a Brooklyn 2 card device as the preferred leader.
    
2. Clock Leader in DDM: When combining Brooklyn 2 and Brooklyn 3 cards in the same domain, Dante Domain Manager will report “There are multiple (2) grandmaster devices.”

This issue can be corrected by enabling unicast clocking in the domain showing the error, once enabled, you can disable unicast clocking and the error should go away.

Both solutions (1) and (2) must be performed after the system is rebooted and each subsequent time the system is rebooted. It should be noted that although these errors are reported, there has been no reported loss of audio or control because of these errors.

1. Firmware: It is not recommended to upgrade or downgrade firmware while devices are in a domain. If a firmware change happens while devices are enrolled into a domain and there are communication errors between Composer and devices, removing the device from the domain and rebooting the device should fix the communication issues. If there are still issues, attempt the firmware upgrade again.
    
2. Stopping DDM Service: It is not recommended to stop the DDM service while devices are enrolled in a domain, stopping the DDM service can result in communication issues between Composer and the devices in the domain.
3. AES-67: AES-67 is not supported when in a domain. It creates issues when a device is enrolled/unenrolled.
    
4. Locking: Monitoring Dante device lock state from Composer is not supported for device enrolled in a domain.
    

5. Pushing: Pushing a Composer Site File is not supported while devices are enrolled into a domain.
   If attempting to push you will be presented with this error. Unenroll devices prior to pushing and re-enroll the devices after pushing is completed.

   
6. Locating in Composer: Enrolling Devices into a domain will cause non-DSP Dante devices to fail to locate. Note, after enrolling devices into a domain, non-DSP Dante devices will not locate in the Composer site view. Audio will still pass through the network but if needed, rebooting the locating DSP will show that the devices are in fact located.

Operational Notes Using Versions of Composer Prior to 8.5.1:

1. Dante Device Location: Instances have been observed where Dante devices, initially located within Composer, fail to maintain their location after enrollment into a designated domain.
    
2. Upgrading Challenges: Migrating from Composer 8.3 to either Composer 8.4 or Composer 8.5 could potentially lead to a situation where the DSP fails to acknowledge the presence of its Brooklyn card.
    
3. DDM Service Impact: Stopping the Dante Domain Manager service can trigger a scenario wherein the DSP ceases to identify its associated Brooklyn card, a pivotal component of the system’s functionality.
    

All three issues can result in the DSP not recognizing its Brooklyn card. If this is the case for you, isolating the device on the Dante network and clearing the Dante Domain Credentials can help re-establish communication with the Brooklyn card..

Composer Configuration for Sharing Audio Between Domains

It is possible, should the situation arise, to route audio from one domain to another. Follow Audinate’s suggested steps to configure shared audio groups.

In Composer, route audio and push the Site File before adding the devices to a domain, as discussed earlier in this document. Take note of the TX and RX channels that are being used prior to push.

Note that the Dante “From xIO Bluetooth RCA” is using the DSP’s RX Dante channels 6-9 (Named BT L, BT R, Analog 3, and Analog 4) and that the “To Genelec” Dante bus is using the DSP’s TX Dante channels 1-2 (Named To Genelec-C and To Genelec-C).

After configuring shared audio groups in DDM, the shared Dante channels will appear in the configured Dante domains, highlighted in green in Dante Controller. You may need to reroute the audio in Dante Controller.

The purpose of this Tech Tip is to provide instructions on using Dante’s Device Lock feature with Symetrix Dante-enabled hardware in Composer 6.0 or later. Device Lock allows you to lock and unlock supported Dante devices using a 4-digit PIN (Personal Identification Number) in Dante Controller. Audinate’s Dante Controller software must be used to lock and unlock Symetrix Dante-enabled hardware. They CANNOT be locked or unlocked using Composer. When a device is locked, audio will continue to flow according to its existing subscriptions, and it may be monitored, but it cannot be controlled or configured. Its subscriptions and configuration settings become read-only.

Dante Controller: https://audinate.com/products/software/dante-controller

Lock Status Indication

Symetrix Composer

In Composer, Site View will show the lock state of all Dante equipped units via the Dante Logo at the bottom center of the unit’s icon. That icon will be one of three
colors indicating ‘Dante Locked’, ‘Dante Unlocked’, and ‘Dante Lock Feature Unavailable’.

lock 1

Locked: Yellow

lock 2

Unlocked: Green

lock 3

Feature Unavailable: Red

Dante Controller

There are multiple locations within Dante Controller where the lock status of a device may be found.

• A small gray lock icon against the device name in the Network View > Routing tab.

• Red background when the device is hovered over in the Network View > Routing tab.

• A check in the Device Lock column in the Network View > Device Info tab.

• A red lock icon in the Device View toolbar.

Lock Symetrix Dante-enabled Hardware

There are two different locations (Network View and Device View) to lock a Dante device in Dante Controller.

Network View
1. Open Dante Controller.

Note: Dante Controller opens to the Routing tab of the Network View page.

2. Click on the Device Info tab.

3. Click the box in the Device Lock column.

4. Enter and confirm a 4-digit PIN.

5. Click the “Lock” button. The check appears in the box to confirm the device in now locked.

Device View

1. Open Dante Controller.
Note: Dante Controller opens to the Routing tab of the Network View page.

2. Double-click the device name of the device to be locked

3. Click the lock icon.

4. Enter and confirm a 4-digit PIN, then click the “Lock” button.

5. The lock icon will turn red to indicate the device is locked.

Unlock Symetrix Dante-enabled Hardware

There are two different locations (Network View and Device View) to unlock a device in Dante Controller.

Network View

1. Open Dante Controller.
Note: Dante Controller opens to the Routing tab of the Network View page.

2. Click on the Device Info tab.

3. Click the check box in the Device Lock column.

4. Enter the 4-digit PIN, then click the “Unlock” button.

5. The check has been removed from the box to confirm the device in now unlocked.

Device View

1. Open Dante Controller.
Note: Dante Controller opens to the Routing tab of the Network View page.

2. Double-click the device name of the device to be unlocked.

3. Click the lock icon.

4. Enter the 4-digit PIN, then click the “Unlock” button.

5. The lock icon is no longer red, indicating that the device is unlocked.

Forgot PIN

A forgotten PIN may be reset in order to access a locked device. The instructions must be followed very carefully, or the process will fail.

1. Isolate the device from the rest of the Dante network.

2. Disconnect and reconnect the device.

3. Wait for at least 2 minutes, then open the Unlock Device window.

4. Use the ‘Forgot PIN’ option in the Unlock Device window.

Overview
 

This tech tip will explain how to integrate the Visionary Solutions Duet Encoders/ Decoders into your Symetrix installation. The Visionary Solutions devices allow for moving 4K video over IP, bypassing the need for more traditional video matrix switching or video wall creation. The encoders and decoders come in two flavors: the Dante-enabled Duet devices (DuetE/D), and the non-Dante devices (E4100/D4100)
 

Using Symetrix DSPs along with Visionary Solutions’ Dante-enabled devices allows for total control of both the Dante audio and the video routing from one central device. For the non-Dante-enabled devices, Symetrix DSPs are able to control video source selection at the decoder, along with whatever audio is riding along with the AV Stream.
 

Before we go into working with controlling these devices from Composer, it is of paramount importance to look into the networking requirements and connections. In fact, it is highly recommended that you not connect any encoders or decoders to a switch until the below switch settings have been enabled.

Networking Requirements Switches Capabilities:

• Managed, with PoE (Visionary Solutions devices require full 15.4W PoE per port).
• Non-blocking.
• Minimum 1GbE bandwidth.
• Capable of IGMP (with IMGP Snooping).
• 8K or better Jumbo Packet capability.

Switch Settings:

• 2 VLANs – One for Video and Control traffic, the other for Dante traffic.
• Multicast must be allowed on all network ports through which video passes. DSP Ethernet ports will also need to be on this VLAN – multicast is not necessary on these ports.
• Flow Control must be removed on any network ports used for video streams.
• IGMP (Internet Group Membership Protocol): Video traffic from these devices is multicast, meaning it is broadcast across the network from a single device to all devices on the network – whether those devices want it or not. This can lead to wasted network bandwidth, as well as the potential for certain devices to be flooded. Enabling IGMP ensures that the multicast packets will only be received by those devices that are intentionally a part of that Group Membership.
• IGMP Snooping and Querier must be enabled (set Querier Version to V2 if possible).
• Enable IGMP Snooping Fast Leave: If your switch supports IGMP Snooping Fast Leave, turn it on. This lessens the amount of time it takes for a device to leave a multicast group and join another – thus speeding up the video switching time.
• Enable Jumbo Frames.
• Disable Energy Efficient Ethernet (Green Ethernet).

Cisco SG300 Example:
Two VLANs will need to be created – one for Video and Control traffic, and another for Dante traffic:

Tie the appropriate physical ports to each VLAN. In this case the first 5 physical ports will be assigned VLAN 2 (Video and Control), and the following 4 will be assigned to
VLAN 3 (Dante traffic).

For IGMP Snooping to function on the SG300, Bridge Multicast Filtering must be enabled:

Edit the Video+Control VLAN and enable IGMP Snooping Status, Immediate Leave, and IGMP Querier Status. Set Querier Version to V2.

Enable IGMP Snooping and Querier:

Enable Jumbo Frames:

Finally, disable Energy Efficient Ethernet (Green Ethernet):

VLAN 2: (Dante traffic):

• Multicast should be allowed to pass on all Dante network ports in order to allow
  multicast clock packets to pass unimpeded.
• IGMP is only needed if there is multicast Dante audio.
• Note that QoS is not needed on a Dante-only network
• Energy Efficient Ethernet (Green Ethernet) should be disabled.
  
  Now that the switch has been configured properly, here is a basic connection
  diagram, showing 2 encoders and 1 decoder, along with a Radius NX 12×8 DSP. The
  Visionary Solutions devices’ PoE LAN ports connect to VLAN 1, and their Dante ports
  to VLAN 2. The Radius NX 12×8’s Ethernet port is connected to VLAN 1, and one of
  its Dante ports connects to VLAN 2.

A video source is connected to a Visionary Solutions Encoder with HDMI. The encoder converts this into an IP stream that is transmitted across the Video/Control VLAN to one or more Decoders. This stream is then converted back to HDMI at the decoder, and sent out to the connected display.

A note on bandwidth:
If you take a look at this table provided by Visionary Solutions, note that a resolution of 1080p60 can take up 200 Mbps of bandwidth.

So if for some reason you have a Gigabit switch that can’t do IGMP properly (or otherwise know there may be an issue with multicast bandwidth management on the network), there could be an issue with having the Ethernet control port of a Symetrix DSP on the same VLAN as the video traffic. Why? On all Symetrix DSPs (aside from Radius NX), the control port is a 10/100 port. Without adequate multicast bandwidth management in the above scenario, the control port of the DSP would be flooded by multicast data, which will cause communication issues with the DSPs. It is therefore recommended that in a situation where there is questionable bandwidth management capabilities, the Radius NX DSP should be used as the preferred solution. This is due to its built-in Gigabit control ports, which will handle much more traffic.

Visionary Solutions Web Admin:
Configuring Encoder/Decoder IP Addresses:

1. Access the web interface for the encoder and decoder units. (log in with
   admin/admin)
2. Select the Network tab.
3. Set the IP.MODE to Static
4. Set the IP.ADDRESS. (e.g. 192.168.1.45)
5. Set the IP.NETMASK. (e.g. 255.255.255.0)
6. Set the IP.GATEWAY. (e.g. 192.168.1.1)
7. Click save.

Configuring the Encoder/Decoder Stream Addresses:
Visionary Solutions recommends setting the first octet to 225. Although not required, it’s helpful to set the last 3 octets to match the IP address as set above in the Network tab (e.g. 225.168.1.45).

1. Access the web interface as above.
2. Select the Configuration tab.
3. Set STREAM.MODE to Multicast.
4. For the encoder:
   a. Set STREAM.ADDRESS to a multicast IP address, such as 225.168.1.45 (to match the control IP in the above example).
   b. Click STREAM.ENABLE = True
   c. Save
5. For the decoder:
   a. The STREAM.HOST IP should be set to the IP of the encoder that the decoder should be receiving from.
   b. The STREAM.ADDRESS should also be set to the STREAM.ADDRESS of that same encoder (as set in step 4).
   Note these fields in the decoder will update while the decoder is being controlled by Symetrix Composer software. If a different encoder is selected from Composer, the Configuration tab will be updated to reflect the different encoder’s IP info.

Working with the DuetE Encoder and DuetD Decoder (Dante-enabled)

Composer Set Up:
A basic classroom design with audio being received into a Radius NX from two decoders, as well as two channels of audio being transmitted to the single decoder:

1. Locate DSP: In Composer, first drag in a Dante-based DSP (e.g. Radius NX 12×8). With your PC on the same subnet as the DSP, locate the hardware by clicking the lower-left corner of the block. Select the DSP from Available Units on Network list, and click “Select Hardware Unit”. The lower-left corner will show a green checkmark when the unit is properly located.

1. Drag in Encoder and Decoder modules: Now that communication has been established with a DSP, it’s possible to locate Dante devices through it. From the Third-party Dante Devices section in the Toolkit, drag in the Visionary Solutions encoders and decoders as needed. Do note that the maximum number of third-party Dante devices locatable by a single DSP is 24.

1. Locate Encoders and Decoders: As in step 1, locate each encoder and decoder by clicking the square in the lower left corner of each. This will open the Locate Hardware window, which shows the available units on the network. Highlight the relevant device, and click “Select Hardware Unit”:

Click “OK” on the Sync Confirm screen:

A green check mark will appear as each unit is successfully located:

1. Right-click either encoder to open the Encoder Unit Properties Window:
   a. Now that the decoder is located, the Host Control Interface IP should be auto-populated. This can be verified by clicking “Verify Host IP”.

b. The Dante Audio Reception section allows the encoder to receive up to four channels of Dante audio from any source on the Dante network. These received Dante channels can be selected to transmit over the A/V stream (see step 5c below). But first, click “Edit Source” to choose the Dante source.

c. The Dante Audio Transmission section shows the four channels of Dante audio the encoder transmits onto the Dante network. These channels contain the audio from the video source that is plugged into the encoder

1. Double-click either encoder to access the Encoder Settings window. This view provides:
   a. Various diagnostic and networking information.
   b. A video stream preview that updates approximately every second (which can be copied to a SymVue control screen for end user previewing).
   c. The “Audio” selector, which determines which audio source the encoder packages up and sends over the AV Stream to the decoder. This selector can be right-clicked and set up to be remotely controlled by any control system.
   d. Note that the Video Wall Wizard can be accessed here as well (this function is covered later

1. To receive and process the Dante audio directly from the Encoders, double-click the DSP to enter the Design View of the DSP. Expand Network I/O Modules à Receive Modules in the Toolkit, and find the Dante Receive Buses that are tied to the Encoders. Drag those in, and place them in the site file.

1. Back on Site View, right-click the decoder to open the Decoder Unit Properties window:
   a. Again, now that the decoder is located, the Host Control Interface IP should be auto-populated:

b. Use the “Dante Audio Reception” section to program the decoder to receive up to four Dante audio channels, such as the two Dante transmit channels shown below . Once this is set, it is then possible to select between these Dante channels, or the audio stream coming from the encoder (see step 8a below).

c. “Dante Audio Transmission” shows the four Dante audio channels the decoder is transmitting onto the Dante network. These names can be edited.

d. The ”Video Selector” area allows for up to 64 different encoders to be set up as video sources. Highlight a channel in the Video Selector table, and click “Edit Source”. You can then either manually enter the Host IP and Stream IP of an encoder, or click “Browse Dante Network” to pick an encoder. The IP info will then auto-populate. (Note this info can also be manually entered in the Video Source Selector area in Step 8 below).

1. Double-click the decoder to open the Decoder Settings window. This view also provides various networking info and diagnostic information for the decoder, as well as:
   a. A/V Settings:
   i. The Host IP and Stream IP fields show the encoder from which the decoder is currently receiving video. This info can be manually filled in, but it is recommended to instead enter the info into the Video Selector area as mentioned in Step 7. The Host IP and Stream IP fields under A/V Settings will then update automatically based on the video source selected in the Video Source Selector.
   ii. The “Audio” selector is used to select which audio source is played out of the decoder’s HDMI output. Choose “stream” to select the audio coming across from the selected encoder. Choose “Dante” to select the Dante audio the decoder has been programmed to receive in the decoder’s Unit Properties (Step 7b).
   b. Video Source Selector:
   i. The Video Source Selector allows the user to choose which of the available encoder streams gets picked up by the decoder. It is also possible to copy the Video Source Selector buttons to a SymVue control screen, for end-user control of video source selection. A single control number may also be assigned to the horizontal source selector fader for ARC or third-party control.
   ii. The video stream preview window updates approximately every second, and can be copied to a SymVue control screen for end-user viewing.

Creating a Video Wall:
The Video Wall Wizard can be used to lay out multiple decoders into an array of up to 4×4 decoders. There is a max of 64 possible video wall configurations, with up to 64 presets created for each.

1. Access the Video Wall Wizard from either a located encoder or decoder’s Settings window, or by going to the Tools menu in Composer and clicking “Wall Wizard (VSI Video)”.
2. Create a new video wall by clicking the Add button. Specify the name of the configuration, as well as the number of rows/columns according to the number of decoders you’d like to have as part of the video wall. Click OK.

1. The array of decoders will now appear in the center of the screen. Click on an
   Unassigned Decoder, and select the “Assign Decoder” button. This will open the
   Select Video Decoder window – click “Browse Dante Network” to select one of
   the decoders from the Dante network. Do the same for the remaining decoders.

1. To control which source is currently playing on the wall of decoders, it is
   necessary to create a preset for each encoder source. Click “Add…” to open the
   Add Video Wall Preset window. Select the Preset number, then click “Browse
   Dante Network” to open the Locate Hardware window.

5. Locate the encoder on the Dante network, then click “Sync to Hardware”

The Host IP and Stream IP should now be automatically populated for you. Click “OK”, then repeat the process for additional encoder sources.

1. Presets can be triggered and previewed from within the Video Wall Wizard by clicking “Test”. To view a preview of the selected encoder source, be sure “Show Thumbnails” is checked. Also be aware that each decoder in the array will still show the full-picture from the encoder chosen as the source…in reality, Visionary Solutions will indeed break up the single encoder’s source evenly across the
   multiple decoders. These presets are simply part of the 1000 presets available in Composer,
   meaning they can be triggered by ARCs, the T-5, and any other controller

Working with the E4100 Encoder and D4100 Decoder (Non-Dante)

Despite the E4100 and D4100 units not having Dante capability, Symetrix is still able to control certain aspects of these devices – namely source select for each decoder, as well as the creation of video walls. Both options are controllable via the Wall Wizard option in the Tools menu of Composer. But first – a small bit about switch settings. Switch Requirements and Settings:

 

The same switch requirements and settings mentioned above apply. Do note that the second Dante-only VLAN is only necessary if there are other devices utilizing Dante audio. Otherwise, if the install doesn’t require Dante, a single VLAN that has the video and control on it will suffice. Visionary Solutions Web Admin: Follow the same steps as above for the Dante-enabled units. Make note of the Host IPs of all encoders and decoders that will be part of this set up. These will need to be manually entered in the next couple of steps. Composer Setup – Creating Source Selection for Single Decoders:

1. To set up source select for a single D4100 Decoder, first drag any Composer-based DSP into the Site View. Then click the Tools menu and select “Wall Wizard
   (VSI Video)”.

1. Click the “Add” button create a new Video Wall. As this is a single decoder, be sure to make it a 1×1 video wall. Give the video wall a specific name (e.g. the location of the encoder). Hit OK.

1. With the new video wall selected on the left, click on “Assign Decoder…”. Manually enter the Host IP of this specific decoder from the Web Admin. Click OK.

1. Now to set up source select! We will need to create a unique preset for each encoder that will be available to this decoder. First click the “Add…” button, and choose a unique preset (one that is un-used in the site file). Then manually enter the Host IP and Stream IP from the encoder’s Web Admin. Click OK. Repeat this step for additional encoders, making sure to choose a unique preset for each new encoder.

1. As with the Dante-enabled units, Presets can be triggered and previewed fromwithin the Video Wall Wizard. Click the “Show Thumbnails” checkbox to preview, then highlight a preset and click the “Test” button to see that encoder route to the decoder.
   Again, these presets are part of the 1000 presets available in Composer, and can be triggered by type of remote control.

1. Once all presets have been created and tested for one decoder, either add a new decoder by starting over at step 2, or click OK in the lower right to exit the Wall Wizard.

Composer Setup – Creating Source Selection for a Video Wall:

Creating a Video Wall build with non-Dante decoders is, for the most part, the same as the processes we’ve seen above.

1. First open the Video Wall Wizard from the Tools menu. Click the “Add…” button
   in the lower left of the Video Wall Wizard. Create a name for the video wall, and
   select the desired size. Click OK.

1. Highlight one of the Unassigned decoders, and click “Assign Decoder…”. Manually type in the Host IP of the decoder from the unit’s Web Admin, then click OK.

Repeat for the rest of the Unassigned decoders.

1. Now to set up the source select. As before, you will need to create a unique preset for each encoder that will be available to this video wall. First click the “Add…” button, and choose a unique preset (one that is un-used in the site file). Then manually enter the Host IP and Stream IP from the encoder’s Web Admin. Click OK.
   Repeat this step for additional encoders, making sure to choose a unique preset for each new encoder.

1. As with the Dante-enabled units, Presets can be triggered and previewed from within the Video Wall Wizard. Click the “Show Thumbnails” checkbox to preview, then highlight a preset and click the “Test” button to see that encoder route to the decoder.
   Again, these presets are part of the 1000 presets available in Composer, and can be triggered by type of remote control.

Additional Features for non-Dante Encoders/Decoders:
As you now know, with the Dante versions, there are modules with built-in GUI elements to work with in Site View. This makes it very easy to simply copy over the Decoder’s source select buttons, and the encoder/decoder video stream preview windows to a SymVue control screen. With the non-Dante versions it is still possible to get these controls over on SymVue control screens. But first, let’s build a convenient way to open the Web Admin of each Visionary Solutions device from within Composer.
Adding Command Buttons to Access Web Admin:

1. From the Toolkit, drag in a Command Button. This will open up the Command
   Button Properties window.
2. Enter a Label – this will be the name that shows up on the button, so be specific,
   e.g. Encoder 1.
3. Select the Web Page option.
4. Type in the Host IP of the encoder or decoder.
5. Hit OK.

The Command Button will now be in your site file. Simply double-click the button to launch the Web Admin in your default browser. 

Repeat this process for each encoder or decoder you want to access the settings of. Note, this is best used only within Composer to assist with system integration – you probably won’t want to give the end-user access to these settings

Adding Video Stream Previews to SymVue:

1. To add a video stream preview window to a SymVue control screen, first navigate
   to the “Device” page of the Web Admin. Click the “Monitor Button” to show the
   preview image. This image updates every second with a frame showing the video
   currently playing on the device.

2. Right-click the preview image and select “Copy Image Location”.

1. Back in Composer, create (or open) a control screen. From the Toolkit, hold down the Control key on your keyboard while clicking on “Picture” and dragging it into your control screen. Holding the Control key creates a different sort of image than the typical – this type can be linked to a web URL.
   At this point, the new image you’ve dragged in should say “Offline”.

1. Double-click the image to open the Properties tab. In the URL field, paste in the Image Location you copied back in step 2. Hit Enter and the image should now update to show the preview. Also note that you can manually enter the Host IP address of the Visionary Solutions device appended with “/thumb.jpg” as well. (E.g. 192.168.1.121/thumb.jpg)

5. Repeat the process for more encoders and decoders as necessary:

Adding Video Source Select controls to SymVue:
By following one of the two “Creating Source Selection” processes above, you should have some presets created that will handle the source selection for either single decoders or video walls. In order to trigger these presets from a SymVue screen it’s a matter of using Preset Recall Buttons. In fact, we can take these preset recall buttons, make them invisible, and layer them on top of the encoder video stream preview – that way the end-user can simply press the video source they want to see, and it will trigger the preset to show that source on the decoder.

1. Make sure there are some presets created for source select as done in the above steps:

1. Open the Control Screen in Composer. From the Toolkit, expand the “Preset Recall Button” option, then drag the preset buttons into the control screen. Place each preset button on top of its corresponding video stream preview.

1. Control-click both Preset Trigger Buttons so they’re both highlighted in red. Resize them to completely cover the video stream previews by holding the Shift key and using the arrow keys on your keyboard. Alternatively, highlight both, and manually enter the Width and Height in the Properties sheet.

1. Again highlighting both buttons, change “Use Name of Preset” to “False” in the Properties sheet. This will remove the text from the Preset Recall Buttons

1. Finally, change the “Transparent” field in the Properties sheet to “True”. This will make the Preset Recall Buttons 100% transparent so the video stream preview can be seen below the button. However the button is still active on the top layer, therefore if the end-user touches the preview, the preset will be triggered and the video source will change.

The purpose of this Tech Tip is to provide instruction on assigning multiple different sources to a single Network I/O Receive Module.

 

Composer versions prior to 6.0 used Dante ‘Receive Flows’ to receive audio from the Dante network, which were limited to a single source of up to 8 channels. Composer 6.0 and later versions now utilize Network I/O Modules. These modules allow the configuration of busses up to 64 channels from multiple sources, as well as supporting AES67.

 

Here are the steps for creating a receive module with multiple sources:

1. From the Toolkit, add a Symetrix Dante-enabled DSP to the Site View page.
2. Open the Design View page by double-clicking the DSP.
3. From the Toolkit, expand Network I/O Modules, then expand Receive Modules.
4. Double-click or drag in a New Network Receive Module.

The Network Receive Module Properties window will open automatically.

1. Select “Multiple Sources”.

The Network Receive Module Properties window will expand to show the channels and sources available.

1. Expand the DSP in the Available Sources section.

1. Next, expand Browse Network Devices. This will list all the available sources this DSP can receive audio from.

1. To add all channels of a source that is broadcasting, select the source and either double-click or click the “<<” button.

Individual channels are also assignable.

1. Select the next channel to be assigned.

10. Expanded the source to display the list of available channels.

11. Add the desired channels to the receive module.

12. Repeat these steps until all desired channels are assigned.

13. Click OK and the receive module will be added to the site file.

An exciting feature of Composer is the native support of both Shure and Audio-Technica Dante-enabled microphones. One new tool we have created here at Symetrix is a new super-module called the Button Processor.

This tool makes it extremely easy to integrate these microphone’s push-to-talk switches into your DSP. Four different modes are available per mic switch; Push to talk, Push to Mute, Toggle, and Disabled. This super-module can also be used with standard momentary (non-latching) analog switches as well. 1-button, 4-button and 8-button versions are included in Composer 3.0’s super-module library.

Begin by importing a 1-Button Processor super-module into the design:

Dante-enabled Audio-Technica and Shure Mics:
The process for both Audio-Technica and Shure microphones is the same unless otherwise noted.

1 Drag in a 1-Button Momentary module from the toolkit, and wire to the
“Button 1” input on the super-module.

2. Double-click the 1-Button Momentary module to bring up its GUI. Right click
directly on the “On” button, then click “Set Up to Remote Control” and select the relevant Audio-Technica or Shure device from the “Remote Control Device” dropdown menu.

3. From Control Modules->Control Outputs, drag in a “Remote Logic Output” module. In the Remote Logic Output Properties window, choose the Audio-Technica or Shure device, as well as the Green LED option.

4. Wire the super-module output labeled “1 On/G” to the input of the Remote Logic Output from step

5. From Control Modules->Control Outputs, drag in a second “Remote Logic Output” module. In the Remote Logic Output Properties window, choose the Audio-Technica or Shure device, and the Red LED option.

6. Wire the super-module output labeled “1 Off/R” to the input of the second Remote Logic Output from step 7. Assuming you’ve set up the receive flow to bring the Dante mic’s audio into the DSP, your site file should now look something like this

7. Navigate to the Mute button for the mic channel you’re planning to control. Right-click it, select “Set Up Remote Control” and choose “Control Signal Assignment”. Click the “Select” button, and click the plus sign next to “1-Button Processor”. Highlight “1 Off/R”, then click OK.

8. Open the super-module user interface and select the preferred switch mode. Go online and test the switch while watching the super-module GUI. The Input LED will light when the switch is closed, and the On/Mute LEDs will respond accordingly.

For momentary analog switches (connected to an External Control Input on the DSP):

9. Drag in a 1-Button Momentary module from the toolkit, and wire to the “Button 1” input on the super-module.

10. Double-click the 1-Button Momentary module to bring up its GUI. Right-click directly on the “On” button, then click “Set Up to Remote Control”. Choose “Local Analog Input”, then select the physical External Control Input number the analog switch is wired into.

11. Go to the Mute button that is to be controlled by the switch. Right-click it, select “Set Up Remote Control” and choose “Control Signal Assignment”. Click the “Select” button, and click the plus sign next to “4-button Processor”. Highlight “1 Off/R”, then click OK.

12. Open the super-module user interface and select the preferred switch mode. Go online and test the switch while watching the super-module GUI. The Input LED will light when the switch is closed, and the On/Mute LEDs will respond accordingly.

13. Repeat the process for more momentary switches.

1) Open Composer to a blank site file or a site file in progress. Connecting to Dante capable Stewart Audio hardware requires a DSP such as the Edge, Radius NX, or Prism.

2) Locate a DSP by clicking the brown box in the lower left hand corner of the DSP icon.

3) Select the DSP in the list of “Available Units on Network” by double clicking it. A green check indicates the DSP has been located and can now be used to connect to a Stewart Audio Dante endpoint.

4) To receive Dante audio in a supported Stewart Audio end point (shown in yellow), drag the supported unit into the design and then click the brown box in the lower left hand corner to locate it.

5) Composer will search the currently located DSP’s Dante network for the Stewart Audio device. Double click on its Name field to select it. A green check will indicate if the unit is connected successfully.

6) Enter the selected DSP’s design and be sure a Dante transmit flow already exists or create a new one to send audio to the Stewart Audio device. If a transmit flow already exists, skip to step 10.

7) To create a new transmit flow, from the Toolkit under Dante Transmit and Receive Flows, drag out New Transmit and Receive flow.

8) Name it, set its channel count to 2, set it to transmit, label the channels if desired, and click OK.

9) Wire the Dante transmit flow into the signal path as desired, then close the DSPs design screen to return to the site view.

10) Double click on the Stewart Audio device to access its properties window. Under the Dante Audio Reception window, double click on channel 1 or 2.

11) Select the appropriate Dante flow and the starting channel. Click OK.

12) Dante Audio Reception should now indicate subscriptions to the transmit flows channels (shown in red). Click OK to close the Stewart Audio unit properties screen.

13) Push the site file to Composer. When completed, Dante audio should be passing from the DSP to the Stewart Audio device.

Note: The Stewart Audio Net AV I/O 2×2 has a pair of remote line level inputs. These inputs can be received in a DSP using the following steps.

14) Follow Steps 1 – 5 of this document, then skip to and complete steps 15-18.

15) Enter the design screen of a DSP by double clicking its icon in the Composer site view.

16) From the Toolkit under Dante Transmit and Receive Flows, expand the Receive Flow Modules for Existing Flows. Then drag the auto-generated Receive Flow from the Stewart Audio Net AV I/O 2×2 unit into the design. The default name of the Flow will include the Stewart Audio Dante network name, for example “Untitled Net AV I/O 2×2 Flow #1”. This is where the incoming audio from the Attero Tech will be received.

17) Wire the receive flow into the signal path as desired.

18) Push the site file to Composer. When completed, Dante audio should be passing from the Stewart Audio device to the DSP.

1) Open Composer and create a blank Site File or open an existing one.

Note: Connecting to Dante capable Attero Tech endpoint requires a DSP such as the Edge, Radius NX, or Prism.

2) If starting with a blank Site File, first drag a DSP into the Configuration from the Toolkit.

3) Locate a DSP by clicking the brown box in the lower left hand corner of its unit icon

4) Select the DSP in the list of “Available Units on Network” by double-clicking it.

5) A green check indicates the DSP has been located and can now be used to connect to a Dante capable Attero Tech endpoint.

6) Drag a supported Attero Tech device into the Configuration from the Toolkit.

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6) Drag a supported Attero Tech device into the Configuration from the Toolkit.

7) Click the brown box in the lower left hand corner of its unit icon to locate it.

8) Composer will search the currently located DSP’s Dante network for the Attero Tech model in the configuration. Double-click on its Name field to select it.

9) A green check will indicate if the unit is connected successfully.

10 ) To receive audio from the Attero Tech endpoint, enter the located DSP’s design by double clicking its unit icon.

11) From the Toolkit under Dante Transmit and Receive Flows, expand the Receive Flow Modules for Existing Flows. Then drag the auto-generated Receive Flow from the Attero Tech unit into the design. The default name of the Flow will include the Attero Tech Dante network name, for example “Untitled unDIO2x2 Flow #1”. This is where the incoming audio from the Attero Tech will be received.

12) Wire the Receive Flow into the audio signal path as desired.

13) To send audio to the Attero Tech endpoint’s outputs be sure a Dante Transmit Flow already exists in the design, or create a new one.

 

Note: If a Transmit Flow already exists in the design, skip to step 15.

14) To create a new Transmit Flow, from the Toolkit under Dante Transmit and Receive Flows, drag a New Transmit and Receive Flow into the design. In the resulting dialog, name it, set its Channels in Flow to 2, set it to Transmit, label the Channel Names if desired, and click OK.

15) Wire the Transmit Flow into the audio signal path as desired.

16) Close the DSPs Design View to return to the Site View.

17) Double-click on the Attero Tech unit to access its Unit Properties dialog.

18) Under Dante Channel Reception, double-click on channel 1 or 2. Select the appropriate Dante Flow and the starting channel, typically channel 1, and click OK.

19) The Dante Audio Reception section should now indicate subscriptions to the Transmit Flow’s channels, shown in yellow below:

20) Next, click the Configure Attero Tech I/O button and set the inputs and outputs to the desired levels. Be sure the “Set Default Power on Settings” option is checked.

21) Click OK to close the Attero Tech properties screen.

22) Push the Site File. When completed, Dante audio should be passing bi-directionally between the DSP and the Attero Tech endpoint.

This tech tip will provide instructions for recording audio from your Dante bus in Composer to your computer via the Dante Virtual Sound Card or Dante Via with ASIO or WDM supported recording software.

Potential Applications:

• House of Worship – Recording sermons and choirs
• Theatre/Live Audio – Recording live shows and performances
• Courtrooms – Recording court proceedings
• Conference Room – Recording conference or troubleshooting AEC

What you will need:

• Symetrix Composer
• Symetrix Dante enabled DSP (Prism, Radius and/or Edge)
• Dante Controller (www.audinate.com)
• Dante Virtual Soundcard (DVS) or Dante Via (www.audinate.com)
• An ASIO or WDM capable program such as Cubase, Logic, Sound Forge, ProTools, Reaper or Audacity

Creating your Dante Bus:
1) Connect your computer to your Symetrix Dante enabled DSP via the Ethernet port.
2) Open Composer.
3) From the Toolkit add the DSP to the site view page.
4) Double click the DSP and open the design view page.
5) Create a Dante transmit flow. Toolkit>Dante Transmit and Receive Flows>New Transmit/Receive Flow…
6) Enter your Dante Flow name, select the amount of channels you need, make sure Transmit is selected and label the channels.
7) Select OK.
8) Wire your audio source into your Dante Bus.

 

Creating your Dante Bus:
1) Connect your computer to your Symetrix Dante enabled DSP via the Ethernet port.
2) Open Composer.
3) From the Toolkit add the DSP to the site view page.
4) Double click the DSP and open the design view page.
5) Create a Dante transmit flow. Toolkit>Dante Transmit and Receive Flows>New Transmit/Receive Flow…
6) Enter your Dante Flow name, select the amount of channels you need, make sure Transmit is selected and label the channels.
7) Select OK.
8) Wire your audio source into your Dante Bus.
9) Go on-line with your DSP

Now that you have the site file archived in your DSP, we can connect your computer to your Dante network.
1) Connect your computer to the Dante Network.

2) Open and enable your Dante Virtual Sound Card or Dante Via.

3) Open Dante Controller and create the subscriptions between the DSP and your computer.

We can now open your ASIO or WDM capable program.

This example will be utilizing Reaper:
1) Open Reaper.

2) Select Options>Preferences.
3) Select Devices in the Audio menu.
4) Select DVS Receive 1-2 (Dante Virtual Soundcard) for the input device.
5) Select your Sample Rate (48000), Bit Depth (24 bit), and Channels and then Select OK.
6) Select Track>Insert New Track.
7) Select input source. Assign Input:Mono>Left to this track

8) Insert another track. Track>Insert New Track

9) Select input source. Assign Input:Mono>Right to this track.
10) Select and Arm each track. You should now see your audio metering in the Record window.
11) Select Record.
12) You can now record the audio from the Dante Bus to your recording software.

Note: Most recording software use WDM for 2 channels/tracks only (left and right), while ASIO is used for multi-channel/track recording. So, if you only need to record in stereo or 2 channels/tracks WDM will be fine. If you need to record 3 or more channels/tracks use ASIO.

This tech tip will walk through the necessary steps required to receive audio in an Edge, Prism, or Radius NX DSP from the Dante Virtual Soundcard running on a PC or MAC laptop.

The Dante Virtual Soundcard software allows a PC or Mac to connect to a Dante audio network. Dante Virtual Soundcard uses the Ethernet port on the computer to communicate with a network of other Dante enabled devices. No special hardware is required other than installing Dante Virtual Soundcard on a conventional PC or laptop. Audio applications use the Dante Virtual Soundcard as they would any standard ASIO or Core Audio sound card. Sending audio from your laptop to the DSP using Dante has many benefits including but not limited to: testing the Dante network, sending test tones or pink noise to the DSP outputs, and tuning the speakers with known audio content. Another application might be to play recorded content in an audio installation, such as intermission messages or sound effect playback in theaters. There are certainly many other useful applications so be creative.
What you will need:

• Composer
• Edge, Prism, or Radius NX
• Dante Controller (www.audinate.com)
• Dante Virtual Soundcard (DVSC) (www.audinate.com)
• An ASIO capable program such as Cubase, Logic, Sound Forge, Winamp

In this example Winamp will be utilized as it is a free download available on the web. From the Winamp website the ASIO Output Plugin will also need to be downloaded.

1) Open Winamp and go to Options->Preferences (Ctrl + P).

2) Next, click on Output section of “Plug-ins” and choose the “ASIO Output Plugin [out_asio.dll]” to select the ASIO driver for Winamp.

3) The Config ASIO dialog will pop up, and the Dante Virtual Soundcard will need to be selected.

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3) The Config ASIO dialog will pop up, and the Dante Virtual Soundcard will need to be selected.

4) Launch the Dante Virtual Soundcard by clicking the Control Panel button.

5) Turn on the Dante Virtual Soundcard by clicking the Power button. It will turn green when active.

6) Open Dante Controller located at Start->All Programs->Audinate->Dante C controller.

7) The Dante Device Network Name of the PC or MAC running the Dante Virtual Soundcard (DVSC) should be visible on the Routing page. In this example the name of the Dante network device is rcurtright-lap1. Write this name down for a later step.

8) Next, click on the Device Status tab, and then double click on the device name. In this case it is rcurtright-lap1. This will launch the Dante Controller Device View.
9) Click on the Transmit tab and then label all channels which you would like to receive in the Edge or Radius.

Since Winamp is being used, only 2 channels are needed to carry a stereo signal which has been named bgm1-L and bgm1-R in this example.

10) Now, on the Routing tab, expand the device in the upper area of Dante Transmitters and confirm that the two named channels are now listed.

11) Next, open Composer, locate hardware (Ctrl+Shift+L), and then enter the design view by double clicking on the Edge or Radius DSP icon.

12) In the Toolkit expand “Dante Transmit and Receive Flows” and drag a New Transmit/Receive Flow into the design.

13) A new Dante flow will be created and Dante Flow Module Properties will pop-up.

• Name for new Dante Flow: can be anything and is only for organization in C composer.
• Channels in Flow: can be 1-8 channels, although this examples uses 2 for stereo content from Winamp.
• Place Dante Flow Module: set to receive.
• Source: check the box for External Dante Device Network Name and enter the network device name from step 7. It must be typed exactly as displayed including any special characters or spaces in the name.
• Type: unicast.
• Channel names: name both channels with exactly the same names given in step 9 using Dante Controller.

14)Wire the Dante modules outputs into any module input or analog output. In this example Dante is wired into a stereo matrix mixer.

15) Push the site file to hardware.

16) In Dante Controller on the Routing Tab with Dante Receives and Dante Transmitters expanded the Edge, Prism, or Radius NX DSP should now show a connection between the DVSC channels.

17) In Composer opening the GUI for the Dante Flow should show audio on the meters, as long as a song is currently playing in Winamp.

Note: setting Winamp to repeat a song or to playlist is suggested for continuous audio.

Note: Dante network audio is 24bit / 48khz audio. This means that playing a mp3 in Winamp which is 16bit / 44.1khz audio will cause it to be pitch shifted due to the 44.1khz audio being played at 48khz by the device. For true testing purposes use software that can play 24bit / 48khz audio, a common example being Sound Forge.

The purpose of the Tech Tip is to assist with the implementation of the CLOCKAUDIO CDT100 into Dante enabled Composer based audio systems.

The CLOCKAUDIO Dante Transporter (CDT100) is a Dante communication product which combines audio and control transport using network based Dante protocol. By taking advantage of the advanced Dante technology and based on the high performance and pre-configurable hardware platform, this product transfers audio and control signals via Ethernet and is able to achieve extremely high sound quality.

The CDT100 has a variety of uses including but not limited to boardrooms, conference rooms, video conference rooms and any place else where multiple table microphones and control devices are connected back to the AV equipment cabinet.

One of the features of Composer is Third-party Dante Device integration. The CLOCKAUDIO CDT100 is among the list of available Third-party Dante devices.

Here are the steps to create the CDT100 Dante audio flows within Composer:

1. Make sure all the devices (DSP, xI/O, ect) in the site file are running matching Composer firmware (must be Composer version 4.1 or newer).

CDT 1

2. Drag a CDT100 from the Toolkit to the site design (Third-party Dante Devices>CLOCKAUDIO).

CDT 2

3. Locate all units in site file.

CDT 2

4. Right-click on the CDT100 and select “Configure CLOCKAUDIO Inputs…”

CDT 3

5. Check the box for Phantom if the microphone on that channel requires phantom power

6. Double click on DSP and open site design
 

CDT 4

7. From Toolkit drag in a new receive flow for the CDT100 (Dante Transmit and Receive Flows>Receive Flow Modules for Existing Flows>Untitled CDT100 Flow).

CDT 5

8. Wire the system design for the desired signal processing and routing.

9. Close the design view and return to the site view.

10. Push the site file and go on-line.

Here are the steps to create the CDT100 Dante control features within Composer:

1. After adding, locating and creating the Dante audio flow for the CDT100, the next step is to configure the control transport.

CDT 7

2. Drag in a 4-Button Momentary module from the Toolkit (Control Modules>Control Inputs).

CDT 8

4. Drag in a 4-Button Processor Super-module (Super-module Library>Import super-module).

• With ARM-C use the CDT100 Processor or CDT100 Processor w/Unmute Delay from 3rd Party Control folder.
• Without ARM-C use the 4-Button Processor from Tools folder.

CDT 9

5. Wire the output of the 4-Button Momentary to the input of the 4-Button Processor.

CDT 10

6. Drag in a Remote Logic Output (Control Modules>Control Outputs).

CDT11

7. Select the CDT100 for Remote Unit.

8. Select the Green LED #1.

CDT 12

9. Drag in a Remote Logic Output (Control Modules>Control Outputs).

CDT 1

10. Select the CDT100 for Remote Unit.
11. Select the Red LED #1.

CDT 14

12. Wire the output of the 4-Button Processor (1 On/G) to the G Remote Logic Output.

13. Wire the output of the 4-Button Processor (1 Off/R) to the R Remote Logic Output.

14. Repeat steps 5-12 for each remaining button.

15. When using ARM-C, select ARM-C for the Logic Output in the Remote Logic Output Properties.

CDT 15

16. Wire the output of the 4-Button Processor (ARM-C) to the Remote Logic Output.

CDT 16

17. Double click and open either a Gain Module or Automixer used in the signal processing and routing of the microphones.

CDT 16

18. Right-click on the Channel 1 Mute button.

19. Select “Set Up Remote Control”.

CDT 17

20. Select “Control Signal Assignment” for the Remote Control Device

21. Then click “Select”.

CDT 18

22. Expand 4-Button Processor.

23. Select 1 Off/R.

24. Click OK.

The mute button of the Automixer will now be controlled by the state of the red LED of the super-module. When the red LED is active, that channel will be muted. When the red LED is inactive, the channel will be unmuted. Repeat steps 17-24 for each microphone channel. Assigning the Mute button of each microphone channel to its corresponding x Off/R output of the 4-Button Processor.

CDT 19

25. Double click and open the 4-Button Momentary module.

CDT 20

26. Right-click the “On” button for Button 1.

27. Select “Set Up Remote Control”.

CDT 21

28. Select “Remote Mic Switch-‘CDT100’” for the Remote Control Device.

CDT 22

29. Select “Switch #1” for the Select Analog Control.

30. Click OK.

CDT 23

31. Repeat steps 26-30 for each button of the 4-Button Momentary module. Assigning each On button with its corresponding Mic Switch.

CDT 24

32. Double click the 4-Button Processor to open the Super-module control screen

CDT 26

33. Select the desired microphone switch operation (Push to Talk, Push to Mute, Toggle, or Disabled) ARM-C has Radio, Toggle, and Latching controls.

34. Push the site file and go on-line.

Basic Remote Terminal Commands

In the troubleshooting or information gathering process there may be times that certain commands become apparently beneficial to have at the ready. Here is a basic list of commands that can be used in Composer’s Remote Terminal (Tools > Launch Remote Terminal). Remember, the correct unit IP address must be specified in the top left “IP Address” field.

IMPORTANT NOTE: Some of these commands will require Remote Terminal be in Debug Mode (Ctrl+D or Options > Debug Mode). Use these commands exactly as shown and explained below. Remote Terminal sending/receiving false or incorrect command data can produce unintended negative results. Be aware of any currently passing signal that may be affected by any reboots or changes in parameter values as this could transfer to downstream equipment.

Save time and focus on system design with this updated Composer shortcuts list. Download and print the attached 1-page version for offline and desk use.

File Menu

Edit Menu

View Menu

Hardware Menu

Tools Menu

Help Menu

Basic Remote Terminal Commands

In the troubleshooting or information gathering process there may be times that certain commands become apparently beneficial to have at the ready. Here is a basic list of commands that can be used in Composer’s Remote Terminal (Tools > Launch Remote Terminal). Remember, the correct unit IP address must be specified in the top left “IP Address” field.

IMPORTANT NOTE: Some of these commands will require Remote Terminal be in Debug Mode (Ctrl+D or Options > Debug Mode). Use these commands exactly as shown and explained below. Remote Terminal sending/receiving false or incorrect command data can produce unintended negative results. Be aware of any currently passing signal that may be affected by any reboots or changes in parameter values as this could transfer to downstream equipment.

Comparing Symetrix Composer Site Files

Overview

This walk through explains the step-by-step process to compare Symetrix Composer Site Files using the “Generate Report” feature of Symetrix Composer software.

Comparing two site files is helpful in troubleshooting when changes in a signal path have occurred, comparing current settings with past settings of a system, or determining what is different between two Site Files.

Once a report is generated from Symetrix Composer software and then compared using WinMerge software, the side-by-side text and highlighted changes make it simple to identify the differences between the two Site File settings.

For this example, a sample room combine Site File is used. Changes will be made to the room select, room volume, and BGM volume then compared with the original settings.

Step-by-Step

1. Make sure to have a program to compare the generated reports. Symetrix recommends WinMerge which can be downloaded here: http://winmerge.org/downloads/

2. Create a baseline report from the Site File in Symetrix Composer. To create the report, press “Ctrl-R”. This will open a window to configure the report. There are three main sections: Report Contents, Options, and Output.

Comparing Site Files

Check or uncheck the desired content for the report. Having every box checked will provide the most detailed report. Selecting individual content will generate a more focused report containing only the selected content.

3. Click “Browse” in the Output section then select the filename and location to save the report. In this example the file name will be “room combine report tt”, and will be saved on the Desktop.

4. Make modifications to the Site File (or open a second Symetrix Composer
file) then create a new report. It is not necessary to save the Site File before running the report again. In
this example the new file name will be “room combine report tt new”, and will also be saved on the Desktop.

5. Open WinMerge and select the two report files to be compared. Go to File > Open, then choose the first report for Left and the new report for Right. Click OK.

6. Compare the two reports. There are three main sections: Location Pane, Reports, and Diff Pane. The differences between the reports will automatically be highlighted in the Location Pane and also in the reports. Scrolling between the differences is made simple by clicking on the desired section of the Location Pane.

Pressing ALT+Down Arrow will scroll down to the next difference or ALT+Up Arrow will scroll up to the next difference. Double click on any highlighted section of the reports will also select the difference. Once the difference is selected it will show in the Diff Pane.

The Automix Matrix 780 was one of the most popular stand-alone DSPs in the Symetrix line of DSP hardware. Many Integrators relied on the simple and straight forward graphic user interface (GUI) for setting up and controlling the Automix Matrix 780.

The Radius NX 12×8 can be used as an improved, feature-rich, direct successor to the Automix Matrix 780. Simply download the site files for Composer, “push” it to the Radius NX hardware using Composer, and setup the Automix system using the familiar GUI.

Here is a picture of the original 780 GUI side by side with the new Radius NX 780 GUI control screen. Notice that except for a few small cosmetic differences, the two are for all intents and purposes identical, both in function and operation.

Additionally, all controller number assignments used in the Automix Matrix 780 have also been assigned to the Radius NX Automix files. This means that any third-party control system integrated with the 780 will work instantly with the Radius NX after the automix file has been pushed to the hardware.

Below are the steps and considerations for using the automix files. It is very important that you read this document in its entirety, as some steps are crucial in commissioning and/or servicing the system.

Steps for Using the Automix Files

1. Complete Composer Online Training. Since the automix site files use Composer to communicate and setup the Radius hardware, it is imperative to have an understanding of how to use Composer. Before contacting Symetrix Technical Support, plan to complete the Composer Online Training, becoming certified in Compose and earning CTS points in the process.
   
    
2. Download the automix site files here.
   
    
3. Open the file in Composer.
   
    
4. Make a connection to the Radius NX hardware using Composer.
   
    
5. Push the file to the Radius NX hardware.
   
    
6. Open the GUI by clicking the green button on the site view entitled “Gain-Sharing” or “Gating” respectively.
   
    
7. Setup and tune the Radius NX hardware using the GUI.
   
    
8. Set the file to archive within the Radius NX hardware. On the site view, right click the Radius NX and choose “Properties” and check “Archive Site File When Going Online.” If using SymVue it is extremely import that each object have only one controller number assignment.
    

Gating vs Gain-Sharing vs Signal Path Only

The automix site file GUI comes in three formats:

• Gain-Sharing Automix Matrix #1 App – the GUI only contains controls specific to Composer’s gain-sharing automixer.
  
   
• Gating Automix Matrix #1 App – the GUI only contains controls specific to Composer’s gating automixer.
  
   
• Template Automix Matrix #1 App – contains the 780 DSP signal path without the GUI.
   

All 3 files contain both the Gain-sharing and Gating automixer in the “Automixers” super-module. If needed the automixer type can be switched between the two algorithms with a radio button control module entitled “Automix Type Select” or by triggering Preset 1 (Gating) or Preset 2 (Gain-sharing) respectively. This could be handy in a case where the gates on the Gating automixer are “chattering” due to excessive room noise and it is desired to change to the Gain-Sharing Automixer without losing the settings of all DSP modules in the file. Settings specific to the newly selected automixer such as NOM, NOM Atten, Hold, Off Gain, Slope, Response, etc, will need to be made on the automixer DSP module. The exception is that all controls common to both automixers such as Auto, Mute and Gain will work from the 780 GUI regardless of which algorithm is selected.
 

Feedback Fighter

The “response presets” buttons on a Feedback Fighter are not exportable to SymVue. In order to make it possible for these controls to be exported to SymVue the response preset settings have been stored to standard Composer presets in the range of 100-174. Preset recall buttons have been used on the feedback fighter GUI for these controls. Be aware that deleting any of these presets (100-174) in the Preset Manager will cause the response preset buttons to become non-functional. It is suggested that these presets are not used, edited, or deleted.
 

Loudspeaker Manager

The filter slope (dB/octave) and type (Butterworth, Bessel, Linkwitz Riley) are drop-down selections. In SymVue the action is somewhat tricky. To use them in SymVue simply click in the drop down with the mouse, hit the enter key, then use the arrow keys on the keyboard to move between selections. Submix 1-8 Transmitted Over Dante: Submixes 1-8 can be routed to the outputs in any combination using the Submix Assign portion of the Matrix/ Submix page of the Radius 780 GUI. Additionally the Submix Assign is also routing these same submixes to their respective 8 channels of Dante digital audio, such that audio going to output 1 is transmitted via Dante Flow1-Ch1, output 2 is transmitted via Dante Flow1-Ch2, etc.
 

Submix 1-8 Transmitted Over Dante

Automix

Submixes 1-8 can be routed to the outputs in any combination using the Submix Assign portion of the Matrix/Submix page of the Radius NX 780 GUI. Additionally the Submix Assign is also routing these same submixes to their respective 8 channels of Dante digital audio, such that audio going to output 1 is transmitted via Dante Flow1-Ch1, output 2 is transmitted via Dante Flow1-Ch2, etc.

The purpose of this Tech Tip is to explain how to integrate the Earthworks IML & IMBL Microphone and LumiComm Touch Ring with a Symetrix DSP. The Earthworks LumiComm Touch Ring features a dual-color LED light ring and a touch sensor output. The light diffuser houses 10 LEDs providing side illumination (5 Green, 5 Red). The logic controlled LumiComm Touch Ring provides system integrators complete freedom
to assign functions and LED color.

The Earthworks IML & IMBL Microphones and the LumiComm Touch Ring can be supplied with either a 5 pin Phoenix connector or an 8 pin R-J45 connector.

The LumiComm Touch Rings current consumption is 85 mA with 5 LEDs lit and 170 mA with 10 LEDs lit, so an external power supply is needed. A “regulated” power supply from 8-28 VDC can be used. Always check your power supply polarity before connecting your supply to the LumiComm Touch Ring.

The wiring diagram below uses the Earthworks IMBL Phoenix connector in this
example. Each connection between the Phoenix connector and the Symetrix DSP is
explained below.
Pin 1) Ground – This connects to both the ground (-) connection of the external power supply, as well as the ground connection of the External Control Input or the Logic Output used on the Symetrix DSP.
Pin 2) 8-28 VDC power supply – This connects to the (+) connection of the external power supply.
Pin 3) Touch Sensor Output – This connects to the External Control Input used on the Symetrix DSP. In this example, CTRL input 1A is used.
Pin 4) Red LED – This connects to the Logic Output on the Symetrix DSP used to activate the red LED. In this example Logic Output 2 is used.
Pin 5) Green LED – This connects to the Logic Output on the Symetrix DSP used to activate the green LED. In this example Logic Output 1 is used.
Symetrix DSP’s are equipped with 3.3V pull up digital inputs, so a 10K resistor is not necessary as shown in Earthworks documentation.

To create the programming for the LEDs we recommend using our Button Processor Super-module, which is included in Composer software. 1-button, 4-button and 8-button versions are included in Composers Super-module library.

 

The Button Processor Super-module makes it extremely easy to integrate these microphone’s push-to-talk switches into your DSP. Four different modes are available per mic switch; Push to talk, Push to Mute, Toggle, and Disabled.

1. Start by importing a 1-Button Processor Super-module into the design:

1. Drag in a 1-Button Momentary module from the toolkit, and wire to the “Button 1” input on the super-module.

1. Double-click the 1-Button Momentary module to bring up its GUI. Right-click directly on the “On” button, then click “Set Up to Remote Control” and select the Local Analog Input from the “Remote Control Device” dropdown menu. Then select which switch is being used under the “Select Analog Control” dropdown menu. Switch 1A is used in this example.

1. From Control Modules->Control Outputs, drag in the “Local Logic Output #1” module. Wire the On/G output from the Button Processor Super Module into the Local Logic Output 1 module.
2. From Control Modules->Control Outputs, drag in the “Local Logic Output #2” module. Wire the Off/R output from the Button Processor Super Module into the Local Logic Output 2 module.
3. Navigate to the Mute button for the mic channel you’re planning to control. Right-click it, select “Set Up to Remote Control” and choose “Control Signal Assignment”. Click the “Select” button, and click the plus sign next to “1-Button Processor”. Highlight “1 Off/R”, then click OK.

1. Open the Super-module user interface and select the preferred switch mode. In this example the Toggle mode is used. Go online and test the switch while watching the super-module GUI. The Input LED will light when the switch is closed, and the On/Mute LEDs will respond accordingly.

This tech tip will cover a variety of ways in which the “Input Logic Module” from
Control Modules->Control Logics can be used within a Composer system. Input Logic Modules are typically driven by External Control Inputs. As such the following section will cover the basics of Control Input modules.
 

Input Logic Modules:
The modules have two outputs labeled, True and False. The True output will be 100% when the logic function is True. The False output is always the complement of the True output (negative logic). Any input less than 50% is assumed to be 0, or false. Any input greater than or equal to 50% is assumed to be (1), or True. For each input, there is an enable button. When turned on, the corresponding input will affect the output. When turned off, the input will be ignored.

The Input Logic modules can perform one of (4) different logical operations: And, Or, Xor and On. The And function is True only if all enabled inputs are True. The Or function is True if any one of the enabled inputs is True. The Xor function is True if any odd number (1, 3, 5) of enabled inputs are True. Note: If all inputs are disabled, the Xor and Or functions will be false and And function will be True. The On function is always true regardless of the state of the inputs. (On may be useful to generate a constant 100% control signal). By using the False output, you can create the inverse function, i.e. Nand, Nor, XNor, or Off. Truth tables are show below for the 2- and 4- input versions to illustrate this.

The tables for the large input versions will follow the same logic as shown below.

Note: 0 = false or 0%, 1 = True or 100%. If the False output is used substitute 0 for 1 and vice versa for all outputs. The tables assume all inputs are enabled.

Controls:

• Off Level. Sets the control signal level that will be output when the button is off (0-100%). Adjust using the slider or click in the text entry box to specify a numerical value. Typically this valve would be 0%.
• On Level. Sets the control signal level that will be output when the button is on (0-100%). Adjust using the slider or click in the text entry box to specify a numerical value. Typically this value would be 100%.
  Note: to make the button act in reverse, set the off level to a value higher than the on level.
• On. Manually controls the button state. You can assign this to an external controller to provide an interface from an external controller to a control signal.

A space is provided to name each button. Click in the box adjacent to the On button and enter a name. The name entered will also appear on the modules output label(s).

Note: A button module can also be used to interface to an external on/off type controller (analog input
or RS-232/485). This allows processing and using an analog control or RS-232/485 controller as a
control signal. Assign the On button to an external controller. Then the module output will reflect the
state of the external controller and can be routed, processed, and connected as a control signal.

Example 1: Using “AND” Logic Operation

In this example, a Room Combine system with 3 rooms is configured such that, when separate, each room’s head table position is on the north side of the room and speaker delay is configured accordingly (Preset 2).

When all three rooms are combined into a single, large partition, the system needs to automatically reconfigure the head table position to the west side of the room and as such, reconfigure the subsequent speaker delays to support this new head table position (Preset 1).

To accomplish this, a 2 Button Latched module is used to mirror the two combine buttons of the room combine module. Notice the control number assignments on the 2 Button Latched Module match the control number assignments of the BGM Automix Combiner, using controllers #17 and #18 respectively. This effectively links the room combiner “Combine” buttons to the 2 button latched “On” buttons.

The 2 Input Logic module monitors the status of the 2 Button Latched, which mirrors the Room Combiner “Combine” buttons, and then triggers the appropriate preset based upon the combine status.

When none of the rooms are combined, both inputs of the 2 Input Logic module will be 0%. This means the output of the 2 Input Logic module will be False and Preset 2 will be triggered.

When only one pair of rooms are combined, either 1 & 2, or 2 & 3, the inputs of the 2 Input Logic module will be 0% and 100% respectively. This means the output of the 2 Input Logic module will be False since it is set to ‘AND’ and both inputs must be 100%. As such, Preset 2 will be triggered.

Once all 3 rooms are combined, then both inputs of the 2 Input Logic module will be 100%. Since both are at 100% and the 2 Input Logic module is set to ‘AND’, this means the output of the 2 Input Logic module will be True and Preset 1 will be triggered.

Example 2: Using “OR” Logic Operation

In this example a 2 Input Logic Module set to ‘OR’ provides a Fire Alarm Mute/Unmute all function.

A 1 Button latched feeds a single input of a 2 Input Logic module set to OR.
Preset 1 or Preset 2 will be triggered based upon the output of the 1 Button Latched module.

Typically in an application like this, the 1 Button Latched “On” button will be assigned to an External Control Input that is connected to a fire alarm relay. Once the fire alarm relay is engaged (stays engaged until the relay is reset), the 1 Button Latched “On” button is pressed and will output 100% to the 2 Input Logic module input. The 2 Input Logic module’s output will be True since it is set to OR and Preset 1 will be triggered (i.e. Mute All).

When the fire alarm relay is reset, the 1 Button Latched “On” button will turn off and the output will be 0% to the 2 input Logic module input. The 2 Input Logic module’s output will be False and Preset 2 will be triggered (i.e. Unmute All).

Example 3: Using “XOR” Logic Operation

In this example, a conference room has three modes of operation, where two of the modes require a projector screen to be lowered in the conference room. A 4 Radio-Button Module is used to select the conference room mode and trigger the respective preset that configures the audio inputs accordingly, while a 4 Input Logic module set to XOR is used to control the Local Logic Output that raises and lowers the projector screen.
 

Note: The Xor function is True if any odd number (1or 3) of enabled inputs are True. If all inputs are disabled the Xor functions will be false.

When radio button 1 is selected for Video conference mode, Preset Trigger-1 is activated to configure the audio inputs and the 2 Input Logic module set to XOR will output TRUE triggering the Local Logic Output which will lower a projector screen for the video conference.

When radio button 2 is selected for Audio conference mode, Preset Trigger-2 is activated to configure the audio inputs and the 2 Input Logic module set to XOR will output False so that the projection screen will raise.

When button 3 is selected for Presentation mode, Preset Trigger-3 is activated to configure the audio inputs for the presentation and the 2 Input Logic module set to XOR will output TRUE triggering the Local Logic Output which will lower a projector screen for the presentation/PC VGA output.

During daily operation end users may have access to volumes, mutes, combine states, presets and routing, and as such it may be required to reset these user accessible parameters to a default state; daily, weekly, or monthly. In many cases, certain venues or applications may require the DSP to be on-line 24 hours, 7 days a week. When the Symetrix DSP cannot be powered down, setting the DSP to a “power on state” or preset to reset the user accessible parameters to a default state is not an option. In these cases it may be necessary to reset a Symetrix DSP without taking the unit off-line or power cycling. This is known as a “soft reset”.

Here are 2 different methods that can be used to schedule and trigger an automatic soft reset of the site-file configuration.

Method 1: Create presets and schedule them in the Event Scheduler.

This method is compatible with Jupiter, 761, Solus, and SymNet Composer DSP hardware (Edge, Radius NX, Prism). It is best used when there is a limited amount of user accessible controls that need to be reset (i.e. Gain or Mixer) and easiest when these controls are located in a single DSP or SymVue page. This method will work with a large amount of parameter changes in multiple DSPs but can be time consuming. The steps for creating presets between SymNet Designer, SymNet Composer, Zone Mix 761, and Jupiter vary slightly, but are essentially the same.

For clarification, check the associated help files or online training for more info on “creating presets”.
1. Set everything in the site file to the desired “default” state.
2. Save those settings as a preset.
3. Rename the preset; in this example it will be named “default site file”
4. Go to Tools>Launch Event Scheduler
5. Click “Add Event…”
6. Title the event “default site file”
7. Select the “default site file” preset from the Preset to Schedule pull down menu
8. Select Recurring Event
9. Set the days and start time the reset event needs be to triggered
10. Click “OK”
11. Push the site file to the DSP to save the reset event.

Method 2: Use a scheduled event to trigger the “Global Load Configuration”

Super-Module to reset the system. This method is only compatible with SymNet Composer DSP hardware (Edge, RadiusNX, Prism). It is best used when there are many user accessible controls spread across two or more DSPs that need to be reset.

It is extremely easy and quick to set up. Preset 500 and 501 are auto-generated when this method is used, so 500 and 501 should be empty before proceeding. Additionally, all units must have a network connection that allows them to communicate with unit that initiates the soft reset. Inside the Super-Module are 1 Button Latched, Network String Output, Delay Logic, and Preset Trigger Modules. Preset #500 will turn on the 1 Button Latched Module and send a control signal to all connected Network String Output Modules. Network String Output module contains the command string “lc 1 r”. This command will load and start running the specified configuration on the addressed units. Delay logic will then sent a delayed control signal to the Preset Trigger Module. Preset Trigger Module will trigger preset #501 and turns off the button in the 1 Button Latched Module.

1. Go to Tools>Super-Module Library Manager
2. Click “Import”
3. Select “Global Load Configuration”
4. Click “Import”
5. Drag the super-module from the toolbox to the site file

reset 1

6. Double click the super-module and type in the IP addresses of the units that need to be reset by the event. ***

6. Double click the super-module and type in the IP addresses of the units that need to be reset by the event. ***

7. Go to Tools>Launch Event Scheduler

8. Click “Add Event…”
9. Title the event “Global Load Configuration”
10. Select preset#500 from the Preset to Schedule pull down menu
11. Select Recurring Event
12. Set the days and start time the event needs be to triggered
13. Click “OK”
14. Push the site file to the DSP’s to save the event.

***Note: The Global Load Configuration super-module was designed to reset 4 units. By default the IP address in the 1st field is set to 127.0.0.1 which is a local loopback IP address. This is helpful in one-box installs and you don’t need to know the IP address.

If it is required to rest 5 or more units, duplicate the super-module, or right click on the super-module and select “Open Design”. Then click on one of the Network String Output modules and press “Ctrl-D” to duplicate it. Duplicate it once for each additional DSP over 4.

Wire the new copy to the network string output daisy chain.

Next open the new (copies) of the network string output module and copy the IP address field to the super-module control screen.

Exit the super-module design and double click the super-module to open its user interface. Type in the IP address of each additional unit that needs to be reset into the respective IP field contained on the Global Load Configuration super-module control screen.

The Edge, Radius NX,, and Prism are configured using Composer software and are scalable from one to many DSP’s. System designers have the option to use or modify DSP design templates for basic projects, or, to create unique designs entirely from scratch.

To use example DSP design files with Composer:
➢ Click on “File” then select “New (based on template file)…”
➢ Follow this address to access folder containing design files C:\Program Files\Symetrix\SymNet Composer 2.0\Designs
➢ Example files are broken up into the following categories:

• Automixing
• Conferencing
• Distribution and Routing
• Loudspeaker Management
• Paging and Zone Mix
• Room Combine
• Special Purpose

➢ Select the desired example file to use or modify

 

Note: Another option to access the folder containing design files would be to click “File” then select “Open” and follow this address C:\Program Files\ Symetrix\SymNet Composer 2.0\Designs

If for some reason the folder containing the example files is difficult to locate Composer has an option that will place the folder on the Desktop.

 

These are the steps to restore example files to the Desktop.

1. Click “Help”
2. Select “Support”
3. Select “Restore and Update Example Files…”
4. A window will open showing the location the design files will be copied to, click “OK”

SymVue is a real-time user control panel application that displays control screens exported from Composer functioning as a multi-user, multipoint control environment for Composer systems.

SymVue runs on any Windows compatible device (XP, Vista, W7, W8, and any embedded version), including touch screen enabled PCs and tablets. The computer communicates directly with Composer hardware over Ethernet. The desired user control interface is created in Composer as a Control Screen and then exported to SymVue where it may be launched from one or many Windows devices for tailored, simultaneous operation.

When creating control screens within Composer to be exported to SymVue and opened on an end users Windows laptop, touch panel PC, tablet, or Surface Pro, it may be desired to create the control screen at a larger set of dimensions than the PC running SymNet Designer. Currently, Composer only supports control screen dimensions that will fit on the PC that the control screens are being designed on. There is however a method for creating a larger control screen dimension, locking the larger dimension which then creates scroll bars for navigation of the control screen being edited.

Here are the steps to creating a control screen with larger dimensions than the designing PC’s screen resolution:

1) Open a matrix mixer module and copy its connect matrix to a new control screen. If no matrix mixer is in the current design, add one temporarily for this step.

2) Position the connect matrix so that it is upper left hand justified as close to the control screen border as possible.

3) On the connect matrix on the control screen, double click to open its properties, or right click and choose “Properties – Matrix Button.”

4) Set the width and height to the desired control screen dimensions.

5) Grab the bottom right corner of the control screen and adjust the control screen window slightly in size. This should instantly create scroll bars.

6) Using the scroll bars navigate to the bottom right corner of the control screen and drop an empty text field into the lower right corner. The example text box does have text in it for clarification, but in the design the text box should be empty.

.
7) Right click the text box to lock its position.

8) Delete the connect matrix from the control screen. If a matrix was added for this process, the module can be deleted from the design now.

9) The control screen dimensions are now locked and scroll bars will always be on the control page for as long as the empty text box is locked in the bottom right hand corner.

10) Populate the control screen with the desired controls and position them.

11) Once completed designing the control screen, “Export to SymVue” and then transfer the .svlx file to the Windows device that will run SymVue control system.

SymVue is our custom control screen software that comes packaged with Composer. Oftentimes, an installer will set up a Windows tablet to execute a SymVue file upon boot up, thus making things simple for the end-user and creating a fully immersive control screen environment. In this type of scenario
however, it is best to give the end-user a means to shut down Windows properly, rather than doing a “hard” shutdown of their device at the end of the night.
This can be accomplished by the simple process of creating a .BAT file that resides on the device. This .BAT file can then be executed via a Command Button placed directly on the SymVue control screen.
Create the .BAT file:

1. Open the Notepad application on a Windows PC.
2. Type in the following command:
   c:\windows\system32\shutdown -s -f -t 00

What the parameters mean:
-s: Tells the computer to shutdown
-f: Forces running applications to close.
-t: How long the device waits before shutdown is initiated (in this case, 0 seconds)

1. Go the File menu and click Save As. From the “Save as Type” dropdown menu, choose “All Files”. For file name, type in “shutdown.bat”. Make sure to include the .bat file extension.

1. Drop the .BAT file into a directory on the tablet or PC that will be running SymVue. Make note of this file path. For example “C:\Users\ndanielson\Desktop\shutdown.bat”.

Create the Command Button:

1. In Composer, bring up the control screen to which you want to add a shutdown option. From the Toolkit, double-click or drag in a Command Button.

1. The Command Button Properties windows will appear – In the Label field name it “Shutdown”. In the Command field type in the file path from step four above. Again, this needs to be the file path on the device that will be running SymVue.

Other button parameters such as color and size may also be edited in this screen.

1. Position the Command Button on the control screen in the choice position. Of course, care must be taken on where to place the command Button, as you don’t want the end user to accidentally trigger the button and shut down their tablet until they actually mean to do it. Therefore, it may be best to create an isolated control page just for the shutdown button.

Use your best judgement.

A Symetrix system’s I/O can be comprised of a combination of hardware, from DSPs such as D100, Edge, Radius NX, or Prism to the I/O expanders such as the xIn12, xOut12, and xControl. Pushing the site file programming from the host PC running Composer Software to the system can take anywhere from a few seconds to a few minutes depending on the amount of hardware that must receive programming.

Often times during commissioning, pushing the file to the system will be performed many times over as changes are made to the signal flow, remote controllers, presets, or parameters, and of which it is desirable to listen to these changes and/or to save these changes permanently in the system. By speeding up the Push process, it is possible to shave many minutes off of the overall commissioning time of the system.

The first thing to note is that all control and routing is truly performed in the DSP units. No processing or control is actually performed in the xIn 12, xOut 12, or the xControl, as once programmed these devices simply send audio or control to the d100, Edge, Radius NX, or Prism.

What this means is that once the xIO have had their programming pushed into them, then changes to the site file signal path, DSP modules, or control will not typically include changes to the xIO devices. As such, disabling them from the Push process will eliminate needlessly reloading the same programming into these devices whose settings/programming is not changing between each subsequent push.

Take this example site file. It includes Edge, Radius NX, xIn 12, xOut 12, and xControl hardware. Before disabling the xIO units from push, first locate all hardware and push the design (F4) to program all hardware, including the xIO devices.

Then right click the xIO hardware and choose “Unit Properties”. When the Unit Properties window pops up, uncheck the enable box (highlighted in red below). Doing this will disable the unit from each subsequent push of the site file. Disabling a unit does not affect the unit’s functionality. To repeat, the
disabled units will continue to operate normally and communicate to the DSP hardware, they will simply be ignored by Composer software during the push process.

Once all xIO units are disabled, the push process will now update only the programming on the Edge, Radius NX, Prism, and D100. And as this document explains, over the course of the commissioning process, eliminating unnecessary units from being reprogrammed over and over will shave many minutes off the commissioning process.

It should also be noted that at any time these disabled xIO units can have their configuration edited by simply checking “enable” in the Unit Properties, making the necessary changes, and pushing the file.

Introduction
This tech-tip describes how to control a Symetrix Radius NX and telephony interface using a Crestron Pro2-style controller. A complete Symetrix Radius configuration file, the Crestron Simpl application file and custom module for dialing the telephony interface, and the Crestron Vision Tools Pro-e touch screen design for an Apple iPad are included and described. Although this example uses network (UDP) control, it can be modified to support serial (RS¬232) control.

This example supports the standard telephony features of dialing, do not disturb, onhook/offhook, redial, and mute control. In addition the telephone line status is displayed including ready, connected, busy, dialing, fault, and ringing. Features not supported in this example are storing or using speed dials, receiving caller id information, audio volume control, and audio meter information.

While this tech-tip is based on the Crestron Pro2 controller and uses the Crestron iPad application as the touch controller, it can be easily customized to support other Crestron controllers and touch screens including the Crestron XPanel PC application generator.

This tech-tip assumes the programmer is familiar with how to use Composer and has some Crestron Control programming experience controlling other products.

Getting Started
This tech tip starts with an overview of the Composer programming for the Radius AEC, continues with the Crestron Simpl program and customer dialer module, and finishes with the touch screen design.

Composer
The first step is to create the Composer site file for your conferencing application. The included file, v1.2 Simple ATI Card Demo.symx, is a working example of a single line analog telephony audio conferencing project with a single Radius AEC and analog telephony interface (ATI) card.

To aid in understanding how to control the ATI, this application focuses on the core telephony functions to create a fully functional dialer with output mute control. This application does not use speed dial functions, and does not control ancillary tone gains.

This Crestron dialer application works by translating key presses on the Crestron touch screen interface to Symetrix API commands and takes the Symetrix command acknowledgments and updates the touch screen as necessary.

Controller Numbers
In order for this Crestron control application to work, the Composer site file must have controller assignments defined for the telephony features as shown in the following figure. In this figure notice that the controller numbers have been assigned to the ATI user interface and range from 121 for the DTMF digit 1 to 145 for the fault indicator. While in your site file these controller numbers can be different from the ones predefined in the demo site file, it is very important that the controller number assignments used in the Composer file match the controller numbers that the Crestron controller will be using. More on this later when the Crestron Simpl program is introduced.

Figure 1. The Composer settings for the Analog Telephony Interface.

In addition, this project has control number 155 assigned to the mute button of the Telephony Output #1 as shown in the following figure. While this controller assignment can be any value between 1 and 10000, it is important that it match the controller number used by the Crestron Simpl module.

Figure 2. The telephony output signal mute controller.

Quiet Mode
For this demo application, it does not matter whether quiet mode is enabled or not. Symetrix recommends leaving quiet mode in the on state (factory default setting). Turning quiet mode off is only required when there are string commands to be interpreted (speed dial names and digits, and caller id information). In this example, only button pushes, and their respective acknowledgments, are used by the Crestron controller. In more advanced dialing applications that utilize speed dials, retrieve the last digits to have been dialed, or receive caller ID information, quiet mode will need to be disabled to support parsing detailed string acknowledgements returned by the Symetrix processor.

Push Enabling
Once the controller numbers have been defined, it is necessary to ensure the controller numbers in the following table are set to ‘Push’ so that changes in the state of the Symetrix device are automatically sent to the Crestron control system.

To enable Push, within Composer navigate to Tools-> Remote Control Manager,
and select each of the parameters in the following table and select Enable Push.

Table 1. The controller number assignments that need to be set to Push Enable.

The resulting settings should match the following figure.

Figure 3. Enabling push for the Symetrix controller parameters.

Once the Composer project is ready, push it to the Symetrix processor and then start with the Crestron Simpl programming outlined below.

Push Interval
The default value of the push interval (100msec) is recommended to ensure timely feedback as the state of the Symetrix processor changes. Changes to the push interval can be made using the PUI API command.

Note:

• This project supports the standard telephony functions including an output mute.
• The controller numbers used in the Composer project must match those used in the Crestron Simpl project.
• Enable push for the Symetrix controller numbers
• This system is compatible with quiet mode (factory default setting) on

Crestron Simpl Project
The example project is designed for a Crestron Pro2 controller using Simpl v4.0.2.20. If using a different Crestron control processor, click the Configure icon in Simpl, select the correct Crestron controller and drag it into the design. Be sure to add UDP communications and the touch screen of choice when
changing control processors to something other than what is used in the demo configuration.

Once completed the configure screen should look similar to the following figure.

Figure 4. The Crestron control system configure screen.

Next, set the IP address of the Radius AEC that Crestron will be controlling by double clicking on the UPD/IP communication block and selecting IP Net Address. In this example, the Radius AEC has the IP address of 192.168.100.105.

Figure 5. Configuring the IP address for the Symetrix device to be controlled.

When using a control system to control the Radius AEC, it is recommended that the Radius AEC have a static IP address to ensure the control system can always communicate with the device regardless of DHCP server status.

In this example there is also a Crestron Mobile device that is the control module for the Apple iPad controller configured using Crestron’s Vision Tools Pro-e software. Other touch panel devices could also be used. Using an Apple iPad with Vision Tools Pro-e requires installing on the target iPad the Crestron iPad application (currently $99) from the Apple ITunes store. Although the price seems high for this application, it makes the iPad an inexpensive touch screen for the Crestron processor. In the Program mode of Simpl (pressing the Program button on the menu), the Simpl file will look like the figure below.

Figure 6. The Crestron control example code.

The main parts of the program are:

• The SymNet ATI Simple Dialer v1.1 is the custom user module described in this tech-tip to control the Radius AEC and ATI card.
• The Preset example shows how a preset may be executed. This code is not used in this application.
• The Com section is there for applications where serial (RS-232) communication is used instead of network (UDP) Control. This section is commented out in the current project. When using serial control instead of network control, it is necessary to configure the serial porton the “Configure” screen in Simpl and uncomment this Com section. Network (UDP) control is recommended.
• The Global System Initialization section is used to start the program.

Simple-Telco-Example.smw
While the simple dialer module supports one phone line, it would be possible to have two of these modules in the Crestron Simpl program to control both analog telephony interfaces on an ATI card. The second phone line would require unique controller numbers and be configured as the single phone line was described previously. The input and output signals of the dialer module are shown in the figure below. The input signals (L1_1_press, etc.) originate from button presses from the Crestron Mobile symbol.

Figure 7. The inputs and outputs of the SymNet ATI Simple Dialer module.

The input and output signals are defined in the following table.

Table 2. The description of the input and output parameters of the ATI Simple Dialer module.

In addition to the parameters defined, there are the arguments that are passed into the dialing module that define the control numbers used by the Crestron program. The arguments are the same control numbers assigned in the Composer site file as shown in the following figure.

Figure 8. Control numbers for the Simple ATI dialer module (left) and assigned in the Composer project (right). These numbers must match!

The controller numbers should be entered with leading 0’s as shown because the controller numbers will be used to form the API commands that are sent to the Symetrix device and also are used match against the acknowledgments received for the controllers that have values “pushed” back to the control system.

ATI Simple Dialer v1.1 User Module
The ATI Simple Dialer module is defined in the file SymNet ATI Simple Dialer v1.1.umc and appears as shown in the following figure.

Figure 9. The inside of the Simple Dialer user module.

This module does the work of receiving the control signals and arguments and translating those signals into commands that are sent to the Symetrix device. In addition as acknowledgments are returned from the Symetrix processor, this module updates the user interface to ensure it reflects the state of the Symetrix processor.
 

To simplify keeping track of the digits that have been dialed, a queue of characters is created locally to store and then send the digits to the touch screen. The digits to be dialed are dialed individually as they are pressed on the touch screen.

The code for the button presses sends the appropriate commands using the assigned controller numbers to the Symetrix device. For example, pressing the 1 key on the touch screen will cause the L_1_press key to go high which in turn will send the command CS 00121 65535\x0D to the Symetrix processor where 00121 is the controller number assignment for the digit 1 that was supplied as an argument to the module. All commands are terminated with a carriage return, hence the \x0D after the command.

Figure 10. The commands that are sent to the Symetrix device.

The acknowledgements from the Symetrix device are monitored to determine the line status and whether the phone is ringing, etc. For example, if the control system receives the acknowledgment: #00141=65535

Then, as shown in the following figure, the Crestron code will parse that information and set the signal L_ringing_on to high to indicate the phone is ringing. This signal is processed and then user interface elements are updated and sent to the touch panel to inform the user that the phone is ringing. Once the line has been answered by sending the command: CS 00133 65535\0xD the Symetrix device will send the pushed acknowledgement: #00138=65535 to indicate the line is connected.

Figure 11. Parsing the acknowledgements is performed by matching particular strings within the data received from the Symetrix device.

The transmit mute state and the do not disturb buttons track their respective state from the Symetrix device. Changes to the mute or do not disturb buttons through some other way will be properly reflected in the user interface.
 

Note:

• The sample Crestron file and module are designed for one phone line.
• Set a static IP address of the Symetrix device to be controlled so the control system can always find the device.
• Set the IP address of the Symetrix device to be controlled in the UDP control settings.

Touch Screen Design
The Touch screen design was created in Visual Tools Pro-e v5.3.19. This example has two main screens – the dialer screen and the incoming call screen.

The controller numbers on the Crestron Touch Panel GUI match the control numbers on the Crestron Mobile controller in the Simpl program as shown in the following figure. For example the digit 1 on the touch screen has a controller number of 161 which becomes the L1_1_press control signal on the
signal press_161 which in turn is sent to the SymNet Simple ATI Dialer user module to indicate that the digit 1 has been pressed which in turn sends and API command to the Symetrix device to dial the digit 1.

Figure 13. The Crestron Mobile touch screen interface with the corresponding control signals to the Touch Panel design

Compile and upload this program and send it to the IP address of the iPad. See the next section for finding the IP address of the iPad.

Using the Crestron iPad application
Once the Crestron iPad application has been downloaded, launch the application on the iPad. As an alternative to using the Crestron iPad application, both the Vision Tools Pro-e project and the Crestron Simpl application can be modified to support XPanel or other Crestron touch panels.

Before configuring the iPad, note the IP address shown on the bottom of the iPad display as shown in the following figure. This is the address that the Vision Tools Pro-e touch panel design program use for upload of the user interface

Figure 14. iPad IP address at the bottom of the iPad display.

To configure the system, select Add System and enter the fields as shown in the following figure. Select Yes for Use Local File. By default the system will use Port 41790 for Part A and 41791 for Port B. While no password is required by default, a password must be entered. In this example, enter any password. Press save when done.

Figure 15. The configuration screen on the Crestron iPad application.

Next select the name of the system just created and press Connect. This should launch a screen like the figure below.

Figure 16. The main user interface of the Crestron iPad application.

When there is an incoming call, the Answer call window appears as shown in the following figure.

Figure 17. The user interface when there is an incoming telephone call.

To get to the configure screen again within the iPad application, press the gear wheel in the upper right hand corner. While not necessary, to re-initialize the Symetrix Crestron Program, press the Symetrix logo which will set the poll_dsp signal high and cause the ATI dialer to re-query the state of the line, do not
disturb, and transmit mute settings.

Note:

• Download the Crestron app for the Apple iPad (the $99 only hurts for a couple of minutes)
• Configure the iPad for “Use Local file”
• Upload the touch panel files to the Apple iPad

Troubleshooting
If the iPad touch screen does not seem to be working to control the Symetrix device then,

1. Check that the IP address of the Symetrix device was entered properly into the UDP control screen in Crestron’s Simpl configuration screen.
2. Check that the IP address of the Symetrix device hasn’t changed via the front panel LCD display. Remember to use a Static IP address for the Symetrix device.
3. Check that the controller numbers entered into the ATI dialer module match the controller numbers defined in the Symetrix configuration file.

In order to be effective, end user control systems need to be simple and intuitive. Some might even call the previous sentence an understatement.

In the audio world decibel (also known as dB) is the standard measurement of sound level and makes perfect sense when viewed on a fader or volume control on a control system. For the end user, reading a volume control’s current position in dB might be much like reading a foreign language, not making much sense unless they have received formal training on what the dB scale means.

Rather than providing training manuals to explain a dB scaled volume control, it may often times prove much easier to simply provide the volume control to the end user as a percentage, or % value instead.

When using Symetrix hardware this is easily accomplished with some creative programming in Composer. Here are the steps for creating volume menus in an ARC-2e or ARC-WEB that read in percent.

Step 1.

checked

First, be sure “Super-impose Assigned Controller Numbers” is checked under the Tools dropdown in Composer.

Step 2.

Next, assign controller numbers to the volume faders of a Gain, Mixer, Matrix, or Room Combiner in Composer using either:

a. “Auto-assign Next Controller Number” (see Figure 1.1)

b. “Set up to Remote Control… > Generic Controller Numbers
Assignment” (see Figure 1.2) to the dB faders in which the end user
will be given access to control with the ARC-2e or ARC-WEB. Do
not add these assignments to the ARC-2e or ARC-WEB at this time.

figure 1.1

Figure 1.1: Auto-assigning a controller number to a BGM Room Combiner volume fader.

Figure 1.2

Figure 1.2: Set up to Remote Control… > Generic Controller Numbers Assignment

Step 3.

step 3

After assigning volume faders a controller number, the assignment should be visible on the module’s user interface.

numbers

 See the green rectangles with control numbers.

Step 4.

Next, from the Composer toolkit, from Control Modules>Control Inputs drag out a “1 Fader” module into the design.

step 4

Next, from the Composer toolkit, from Control Modules>Control Inputs drag out a “1 Fader” module into the design.

Step 5.

Next, open the 1 Fader module and assign a controller number to the control fader. Notice the control fader reads in % instead of dB.

step 5

Next, open the 1 Fader module and assign a controller number to the control fader. Notice the control fader reads in % instead of dB.

Step 6.

Now open the Remote Control Manager under the Tools dropdown (or Ctrl+M). Notice the volume fader assignments and the 1 fader assignment in the Control Numbers tab.

step 6

Now open the Remote Control Manager under the Tools dropdown (or Ctrl+M). Notice the volume fader assignments and the 1 fader assignment in the Control Numbers tab.

Step 7.

Click on the 1 Fader control assignment to select it and click Set Up Remote Control…

step 7

Click on the 1 Fader control assignment to select it and click Set Up Remote Control.

Step 8.

Scroll to select the Remote Control Device of choice, an ARC-WEB or ARC-2e, and then click OK. This will add a 0-100% menu item into the selected remote control device.

Step 9.

Move to the ARCs tab and expand the ARC-WEB or ARC-2e associated with these controls and then double click on the Fader 1 menu, or click to highlight and hit the Edit… button near the bottom of the window to access the Edit ARC Menu.

step 9

Move to the ARCs tab and expand the ARC-WEB or ARC-2e associated with these controls and then double click on the Fader 1 menu, or click to highlight and hit the Edit… button near the bottom of the window to access the Edit ARC Menu.

Step 10.

Change the Menu Name and Controller Number to the desired dB fader assignment from Step 3. In this example Controller #1 was for Room 1 Volume. Note, the controller number must match the assignment but the Menu Name can be labeled anything. This menu name is what the end user will see on the ARC display.

step 10

Change the Menu Name and Controller Number to the desired dB fader assignment from Step 3. In this example Controller #1 was for Room 1 Volume. Note, the controller number must match the assignment but the Menu Name can be labeled anything. This menu name is what the end user will see on the ARC display.

Step 11.

step 11

Hit the Save Menu button.

Manager

Step 12.

Repeat steps 4-12 for all subsequent volume fader assignments that will read in % value, each BGM Combiner fader having its own 1 Fader control input module.

Step 13.

When completed with the % value ARC programming. Push the site file to the SymNet system and program the RS-485 network.

Step 14.

step 14

The end user will now see % values for volume controls rather than dB values.

Step 15.

Note: On a -72dB to +12dB volume fader, if it is desired to scale a volume control so that 0dB is the max level an end user can turn up the gain, set the High Limit to 84%.

The purpose of this document is to provide information and set by step instructions for the proper firmware upgrade process. With each new release of Composer also comes new firmware for the hardware. Firmware upgrades can include upgrades made to Composer (i.e. modules and tools), bug fixes, and can be a very useful tool when troubleshooting a problem. The firmware version in the hardware must be matched with the version of Composer software being used. The correct firmware for the version of Composer you are using is always distributed with Composer and installed on the hard drive by Composer’s installer. If there is ever any doubt, follow these instructions to check the firmware version or to upgrade the firmware.

IMPORTANT: Do not attempt to perform a firmware upgrade over Wi-Fi, as this may ‘brick’ the DSP and require repair. It is also recommended to disable Windows Defender Firewall and have the PC obtain IP addresses automatically.

Be sure to create a backup of the site file on a PC before attempting a firmware upgrade.

Part I. Initiate Firmware Update

1. Ensure the Symetrix Digital Signal Processor (DSP) is connected to the PC’s subnet. This can be accomplished one of two ways: Option A. Direct Connection: plugging the PC directly into the DSPs Control (Ethernet) Port. Option B. Network Switch: with both the host Computer (running Composer) and DSP connected to a Network switch. Set the PC’s Network Interface Card (NIC) to automatically obtain IP addresses.

1. To change the IP configuration in Windows OS: type “ncpa.cpl” in the Start Menu search bar which should query a Control Panel Item: Network Connections.

2. Right click on the “Ethernet” NIC, select “Properties”. Once in the Ethernet Properties, select “Internet Protocol Version 4 (TCP/IPv4)” from the connection list, and then click the “Properties” box beneath it.

3. Select “Obtain an IP address automatically” which utilizes Dynamic Host Control Protocol if that is available on the control network. In specific cases, a Static IP address must be set in the same subnet as the DSP. The DSP’s IP address can be found on the front panel display by navigating through the menus with the arrow buttons.

2. Locate the DSP in Composer. The unit in the site file will be successfully located when a green check mark appears on the bottom left corner of the unit module that has been dragged in from Composer’s Toolkit (see Figure 1).

3. Right click on the device module, select “Unit Properties…” (see Figure 2), and under the left middle section of the “[DSP Model] Properties” window (see Figure 3) labeled “Firmware”, select “Upgrade Firmware…”

4. This should bring up the “Upgrade Firmware/Hardware Settings” window. (see Figure 4)

5. Select the “Upgrade…” option (see Figure 4) under the “Composer Firmware File” option.

6. An open file dialog box will appear and by default it should direct to this folder: C:Program Files (x86)SymetrixComposer 8.3Upgrade. Select the appropriate CFW file and click “Open”. The update will stall at 25% momentarily as this is when data is transferring over File Transport Protocol (FTP) Port 20. However, it should last no more than 15-20 seconds and continue. Once complete, a dialog box will appear confirming success.

7. It is best practice to power-cycle the unit only once the upgrade is complete by unplugging/re-plugging the power cable on the back and re-pushing any site file programming to the DSP.

Note: The above steps are also used when upgrading T series touch panels and W series remotes.

Part II. Upgrading Dante Expanders

1. Right click the Dante expander module and select “Unit Properties”.

2. Select “Upgrade Firmware…”.

3. The “Upgrade xIn, xOut or xIO Firmware” window will appear. This window displays the currently installed User Firmware and Dante Kernel Firmware versions. New versions can be selected from the File Explorer by clicking either of the respective “Change Version…” buttons (see Figure 5).

4. Clicking the “Upgrade Firmware” button (see Figure 5) will upgrade both the User Firmware and the Dante Kernel Firmware to the versions selected in Step 3.

5. Once the firmware upgrade has completed, a dialog box will appear confirming success.

Note: A Composer-enabled DSP must be present and located in the site file and connected to the Dante Expander via the Dante network in order to upgrade Dante Expander firmware.

Part III. Upgrading Multiple Design Units Simultaneously

From the “Upgrade Firmware/Hardware Settings” window (see Figure 4), all design units in the site file except ARC and Dante Expander units can be upgraded at once using the top button: “Auto Upgrade All Design Units…”

Note: It is recommended to upgrade one design unit at a time in the event of a failure. Otherwise, none of the units in the site file will upgrade and there will not be a dialog box indicating the unit that is unable to complete the upgrade. Upgrading individually is also advised for troubleshooting as it completely reimages the firmware while Auto Upgrade will only update the files that it needs to.

Part IV. Common Issues/Troubleshooting

· Firmware update fails after reaching 25%: The issue is likely related to the network as the FTP requests are failing to reach the DSP.

Directly connect the PC into the DSP’s Control Port and attempt the upgrade without the presence of a network switch.

· Any dialog box indicating the firmware update failed:

In rare cases, the error is due to Composer losing communication with the DSP when it is rebooting and taking longer to come online than the timeout period Composer is expecting. Close the properties window and re-open it to verify if the firmware updated. Otherwise, attempt to power cycle the device. The update may have been successful before and may show as “current” in the “[DSP Model] Properties” window (see Figure 3) after the device reboots.

If the version is not “Current”, check to see if the desired firmware upgrade is older than one generation of the Composer version. It is best practice to install an older version of Composer and slowly upgrade the firmware to match it versus upgrading to the most current version (this may fail or ‘brick’ the device).

If the update continues to fail, and a local back-up of the Composer site file is available, factory reset the DSP using the following steps and then re-push the site file:

1. Pull power from unit.

2. Find the recessed button on the back of unit – for Composer based DSPs it will be by the RS-232 or Ethernet Ports; for Jupiter or Zone Mix, it will be by the ARC port; for xIO Bluetooth, it is on the left side of the unit.

3. Hold the button in with a pin or a pen.

4. Re-apply the power plug while continuing to hold button.

5. Count 20-25 seconds with the button pushed in while the unit boots up.

· Erase Unit Memories: The Erase Unit Memories function is provided in the Unit Properties (see Figure 3) for troubleshooting purposes. It provides a handy way to delete certain parameters stored in the hardware. The parameters that can be selected to erase either all or individually are: Stored Site File, Scheduled Events, Accessory Serial Port Settings, Analog Calibration Settings, and Firmware Log Settings (see Figure 6).

· If a Dante device will not accept a firmware upgrade, be sure the device has not been locked via Dante Controller. This is found in Dante Controller > Device View > Lock icon.

Figure 1. A green check mark confirms that the DSP is located.

Figure 2. “Unit Properties” menu option appears after right clicking the DSP module.

Figure 3. DSP Unit Properties – “Upgrade Firmware…” can be found here.

Upgrade Firmware/Hardware Settings – Includes “Auto Upgrade All Design Units…” and “Upgrade…” option for a single DSP.

Figure 5. Dante Expander Firmware Upgrade Window – Both the User Firmware and the Dante Kernel Firmware can be upgraded simultaneously here.

Figure 6. Erase Unit Memories – Can be used to troubleshoot hardware by restoring certain hardware properties to their factory state.

The Symetrix 2 Channel Analog Telephone Interface card (ATI card) provides a simple and intuitive solution for audio conferencing applications. Acting as a built-in telephone hybrid, the ATI card provides the means for a Edge or Radius AEC to interface directly with an analog telephone line from the local
telephone company or an analog port from a digital PBX. The graphic user interface for the ATI card within Composer hosts all user controls for dialing a phone number, speed dialing a number, as well as picking up or disconnecting a phone call. The telephony controls are usually accessed by the end user via a SymVue control system or a 3rd party control system.

Crestron and AMX dialer modules have been created and can be downloaded from the Symetrix website. The downloadable folder for each includes the 3rd party dialer module as well as an example Site File:
https://www.symetrix.co/products/audio-io-and-control-expansion/#2-lineanalog-telephone-interface-card
There are two different methods in which a phone number can be dialed using the 2 Channel ATI card

Dialing a Phone Number One Digit at a Time:

When programming a 3rd party control system to dial the ATI card, each digit of the telephone number can be triggered one digit at a time by assigning a controller number to each of the ATI dialer buttons and then triggering them using the CS command outlined in the 3rd party protocol available here: https://www.symetrix.co/?wpdmdl=8

Controller Set Command = CS <Controller Number> <Controller Position> <CR>

The advantage to this method is that no special module need be created to dial the phone number. Instead, each digit on the dialer is treated the same as controlling any button or Boolean control in SymNet with a 3rd party control system.

This means the same control code that turns on or off a mute button can also be used to dial a number button on the ATI card dialer GUI.
 

Below is a picture of the ATI card GUI, in Composer, showing the telephone dialer. This picture was taken from the example Site file included with the Crestron and AMX ATI dialer modules. Controller #121 through #133 has been assigned to 1-9,0, *, #, and the connect/disconnect button, respectively.

Best Practice:
In Composer, number the controller numbers on the Dialpad sequentially to make it easier to control
and debug the control strings. Controller numbers can be added by right clicking on the control in Composer and selecting
Edit Remote Control Assignment.
As an example, in order to dial the Symetrix phone number, 1-425-778-7728, the third party control system would send the following commands:
 

Analog Telephone Interface Dialer

CS 121 65535\r
CS 124 65535\r
CS 122 65535\r
CS 125 65535\r
CS 127 65535\r
CS 127 65535\r
CS 128 65535\r
CS 127 65535\r
CS 127 65535\r
CS 122 65535\r
CS 128 65535\r

After each digit is entered, the Symetrix device will respond with an “ACK” if the command was interpreted correctly or a “NAK” if the controller number does not exist.
 

After the phone number digits have been entered, the call can be triggered to dial by sending the connect/disconnect button the following command:
CS 133 65535\r
 

The phone call may be hung up (placed on-hook) by sending the same command again:
CS 133 65535\r
 

Note: the function of the connect/disconnect button changes depending on if you are connected or
not. Always send 65535 for this value and it will toggle the connect states.

Dial a Phone Number Using a Speed Dial:

Some 3rd party programmers may prefer to create a custom dialing module that sends the entire phone number to the ATI card using a single command string, at which point the phone number can be dialed using a second command that dials the speed dial. The basis of using this two command method is that an ATI card speed dial slot is used as a phone number loading dock, and once the phone number has been loaded into the speed dial location, it can be dialed with a single command.
 

The ATI card has 20 speed dial locations, so if this method is used one speed dial location must be dedicated to the control system and the end user will 19 remaining speed dial entries.

The command to load the telephone number is the (SSYSS) Set System String command. It is important to note that this command assigns a system string, such as a name or phone number, to one of the speed dial locations and when applicable executing this command will over-write any previous data in the specified speed dial location. Set System String Command =SSYSS <Unit>.<Resource>.<Enum>.<Card>.<Channel>=<Value>

<Unit>=In the site view of the Composer site file, above each unit icon is a number after the dash, e.g. “Edge-1” means =1. (See picture)
 

<Resource>= 1000 for speed dial number, 1001 for speed dial name.
Some 3rd party programmers may prefer to create a custom dialing module that sends the entire phone number to the ATI card using a single command string, at which point the phone number can be dialed using a second command that dials the speed dial. The basis of using this two command method is that an ATI card speed dial slot is used as a phone number loading dock, and once the phone number has been loaded into the speed dial location, it can be dialed with a single command.
 

The ATI card has 20 speed dial locations, so if this method is used one speed dial location must be dedicated to the control system and the end user will 19 remaining speed dial entries.
 

The command to load the telephone number is the (SSYSS) Set System String command. It is important to note that this command assigns a system string, such as a name or phone number, to one of the speed dial locations and when applicable executing this command will over-write any previous data in the specified speed dial location. Set System String Command =SSYSS <Unit>.<Resource>.<Enum>.<Card>.<Channel>=<Value>

<Unit>=In the site view of the Composer site file, above each unit icon is a number after the dash, e.g. “Edge-1” means =1. (See picture)

<Resource>= 1000 for speed dial number, 1001 for speed dial name.
<Enum>= 0 based count of 0-19, where 0-19 equals speed dial slots 1-20.
<Card>= 0 based count of 0-3 for card slots A-D. (A-D in Edge frame, D only in Radius AEC).
<Channel>=Not applicable for the SSYSS command since both ATI ch 1 and 2 share all 20 speed dial locations. Use a zero for this portion of the command.
<Value>= the phone number that should be assigned to the speed dial slot defined by <Enum><Card>
Example of SSYSS:
 

Notice the ATI card, “Telephone I/O”, in both the Radius AEC-2 and the Edge-5. If the intention is to store the Symetrix phone number into speed dial slot 20 on the Edge-5 unit’s ATI card in card slot C, the command would be determined as follows:

<Unit>=5
<Resource>= 1000
<Enum>= 19
<Card>= 2
<Channel>=0
<Value>= 425-778-7728
SSYSS 5.1000.19.2.0=425-778-7728

With Composer online with the hardware, the ATI card GUI would show 425-778-7728 in speed dial entry 20:

Then, once the phone number has been loaded into the speed dial entry, the ATI card can be triggered to dial the number by using the CS command to trigger the speed dial location. In the above example controller number 146 is assigned to speed dial #20 location. As such, CS 146 65535\r would dial speed entry 20
 

As another example, if the intention is to store the Symetrix phone number into speed dial entry 20 on the Radius AEC’s ATI card in card slot D, the commands would be as follows::

<Unit>= 2 (Device is Radius AEC-2)
<Resource>= 1000 (This is for the speed dial entry)
<Enum>= 19 (This is for speed dial 20)
<Card>= 3 (This is for slot D)
<Channel>= 0 (Channel is 0)
<Value>= 425-778-7728

To set the speed dial:
SSYSS 2.1000.19.3.0=425-778-7728\r
To dial the call:
CS 146 65535\r
In review, this two command method can be used to load a phone number into an ATI card speed dial location using the SSYSS command, and then the phone number can be dialed using a single CS command.

Testing the API commands
To help understand the command API, it can be helpful to manually type in commands to control the system. The easiest way to do this is with the built-in Remote Terminal application.

To send a command, type it into the command window as shown in the following figure and press enter to send the string to the device. The command acknowledgments will appear in the window below.