Video Transcript
AEC & Q-SYS Conferencing System (Part 4)
10m 4s
00:08
Welcome back.
00:09
So let’s say we have a system with proper signal flow and AEC reference.
00:14
We can use a simple process to get the best performance from the AEC processing.
00:19
Step one is all about gain structure! Calibrate each microphone input to -20dBFS nominal.
00:28
This applies to ALL microphones regardless of whether they come in on analog inputs
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or via other transports such as Dante or AES67.
00:38
With analog mic inputs, simply used the analog preamp gain control to achieve this.
00:43
In external microphone systems, you’ll most likely want to log into that system
00:47
and calibrate the microphones to send the appropriate nominal level.
00:51
Remember that ALL system inputs should be calibrated to the same -20dBFS nominal.
00:58
This includes program feeds and signals from the far end.
01:01
Now that the conferencing mics have proper gain structure,
01:04
we’ll next want to check the signal to noise ratio of each microphone.
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It’s fairly easy, as we’ll simply want to be quiet for a few moments
01:13
and let the mic sense only the noise floor of the room.
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The only other consideration is that we want to do this while any air conditioning,
01:22
projector fans or other noise sources in the room are running.
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If the noise floor of the room as sensed by the microphone is below -35dBFS,
01:32
it meets the SNR requirements for a conferencing system.
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If not, external methods of noise control are recommended
01:38
to get the required 15dB of signal to noise at each microphone.
01:43
We might think we simply need to turn on the noise reduction in the algorithm,
01:47
but it will most likely not function well in a poor environment.
01:51
Once the microphones are set to nominal level and we know the signal to noise meets the standard,
01:56
we should be able to leave them at a unity gain throughout the signal chain.
02:00
The next step is to calibrate the near end side of the system.
02:04
For this step we’ll want to first turn the amplifiers all the way down.
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If there are no amplifier controls,
02:11
reduce the Max RMS setting of the Q-SYS output block all the way down to -40dBu.
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Now start playing a program source and calibrate it to nominal level of -20dBFS at the Q-SYS input block.
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Leave this program source playing with unity gain through the entire signal path to the output.
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Using an SPL meter,
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bring the amplifier controls or Max RMS setting up until the SPL reaches a comfortable listening level.
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The standard for conferencing systems is 70 to 75dBSPL.
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Finally, we’re ready to make a test call.
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You’ll want to have a trusted person on the other end
02:51
who can give you qualified feedback about the level and audio quality at the far end.
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Adjust your caller’s far end audio source to reach the -20dBFS nominal level.
03:02
With unity gain through the signal chain, you should
03:05
have the same 70-75dBSPL in the room you measured with the program source.
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If the far end caller tells you there is noise at the far end, enable the noise reduction and level.
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Keep in mind that the noise reduction algorithm is designed to remove constant,
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steady state noise, not things such as the noise of passing traffic, talkers in the hallway, etc.
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Engage the noise reduction with the NR enable button
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and apply only the amount of noise reduction needed to eliminate the noise.
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Applying too much noise reduction could adversely affect the quality of the mic signal at the far end.
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If noise reduction is required you’ll want to make sure it’s applied to all the microphones in the room.
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Now we’d like to check the performance of the AEC algorithm when only the far end is talking.
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Looking at each conferencing mic, we’ll want to confirm that the RMLR meter is showing green as in the diagram.
04:01
The target is 0,
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meaning we want the reference signal level to match the level of resulting signal at the microphone.
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In most rooms this will require that some attenuation is applied at the reference signal,
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which can be done in the AEC block itself.
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As previously stated,
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it’s a good idea to mute any conferencing mic signals in the DSP after the AEC processing block.
04:24
That would be the first choice, but sometimes the AV designer doesn’t have that choice.
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The microphone audio could come from a Dante enabled microphone for example,
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without a way to get a mute signal back to Q-SYS.
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The mic mute button would then most likely completely mute the Dante audio feed to Q-SYS from that mic.
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Q-SYS has at least one option to accommodate this scenario.
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The ‘hold if mic level below’ setting holds the AEC convergence
04:52
when the mic audio level goes below the setting shown.
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If this is set to be just a few dB below the noise floor of the mic when unmuted,
05:01
the algorithm will hold there through the mute state and have less work to do when it’s unmuted again.
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Starting in Q-SYS version 8.1 there’s also a setting
05:12
to hold the AEC convergence if the reference level goes below a certain level.
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This is useful if the far end signal must be muted before it hits the AEC reference block.
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If there’s still some residual echo to your far end test-caller,
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then it’s time to engage residual echo suppression, or RES.
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First use the button to enable the feature, then engage only enough to remove the residual echo.
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After that, it’s time to test the system in the double-talk condition.
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Have far end and local talkers speaking at once and see if echo reaches the far end.
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If so, increase the RES percentage to just enough to eliminate any echo.
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You’ll want to be sparing with this,
05:55
as conference participants may start to feel like the system is half-duplex as it’s increased.
06:01
As an aside here, note that one mic in a room might be harder to calibrate than others due to placement, etc.
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As a first pass, it might make sense to apply the same settings to each mic,
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but if the test caller hears echo it may get more complicated.
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You’ll want to isolate exactly which mics in the room
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are returning the echo and mute those that aren’t problematic.
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From there you can fine-tune each problem microphone independently
06:29
of the others until you have the performance perfectly dialed in.
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Of course many rooms are required to use multiple methods of conferencing.
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Sometimes they must be used simultaneously.
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If this is your room, first do a test call with each conferencing mode independently
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and then finally all methods at once making all necessary adjustments.
06:50
So what if you’ve tried the steps as outlined and still have echo?
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Let’s think about a few troubleshooting strategies.
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One common problem is that the ‘echo’ heard by the far end isn’t echo at all.
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A quick way to check this it to mute all the conferencing mics and see if the far end still hears themselves.
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If so, the conferencing receive signals are accidentally looped or misrouted.
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If there are any large matrix mixers in your design,
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carefully examine the crosspoints to make sure that the far end receive signals
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aren’t accidentally being sent directly back to the transmits.
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This routing can be tricky when multiple conferencing modes are required, so keep that in mind.
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If an external video codec is being used, the codec itself can be a source of such a loop.
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Many codecs have a ‘reinforcement mode’ or reinforcement output
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that includes the conferencing mic signals and the far end.
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The intent of such a thing is to plug directly into an amplifier for voicelift and conferencing in the room.
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If this type of codec output is plugged into Q-SYS as a conferencing input,
08:01
you can hear some very interesting echo like effects.
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In addition, the microphones will be sent to their own AEC reference signals.
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This usually results in some modulated sounds of the mics to the far end.
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The Q-SYS hover monitor is an easy way to check for this.
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Hover over the input node and if you hear the local microphones, you know this is a potential problem.
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Other sources can cause similar issues, so an easy way to find the culprit is systematically muting each source
08:30
and microphone until you find the one that muting makes the echo go away.
08:35
You then know exactly where to focus your problem-solving skills.
08:40
If you know all routings and gain structure are correct
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and you’ve done everything you can to calibrate AEC with no success,
08:48
it’s possible you need to extend the tail length of the AEC algorithm.
08:52
The tail length required should depend on the size, shape and reflectivity of the room.
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If you’re using the 100ms tail length, it’s very possible you’ll need to extend to 200ms.
09:05
The 200ms algorithm can handle about 85% of conference rooms,
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but if the room is very large and very reflective it might require 300 or 400ms tail lengths.
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Remember the longer the tail length, the fewer algorithms can fit in a given core.
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The eternal struggle here is that architects and
09:25
end users love elaborate room shapes made of very reflective surfaces.
09:31
Glass, hardwoods, and ceramic tile are not the greatest environment for conferencing,
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but it’s what we get to work with as AV engineers and technicians.
09:40
In some cases it’s necessary to propose acoustic treatment
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be used to absorb some reflections to provide the best in intelligibility and response.
09:49
Something like this should do the trick!
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In all seriousness, there are many tasteful,
09:54
low profile acoustic treatments available that can help maximize performance.
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That's it! Thanks for watching.
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AEC & Q-SYS Conferencing System (Part 4)
10m 4s