Q-SYS Corner

[Steve Guttag]

Separate Surround Settings for Different Formats
While we’re exploring possible ways to modify the Sample 7.1 Design, and, in particular, the Surrounds, what about having separate level and EQs for the Surrounds when you have mono, 5.1 and 7.1 surround formats?

In theory, what we’ve been doing since the days of the CP650 and Surround-EX, should work. That is, calibrate the surrounds, using our established reference pink noise to 82dBc, per quadrant, referencing the RLP (2/3rds back from the screen and slightly off-center). Then, presume that when Left-Side and Left-Rear sum, acoustically, to 85dBc (which is why we have the offset to bring it back down to 82dBc). How well does that work, in practice? How about the tuning? Does the tuning change when you have two quadrants playing at the same time? How about when you have all surrounds playing at the same time (mono surrounds)?

Setting your EQ up to have three different sets of EQs/Levels will allow you to exactly match the conditions of a mono/stereo or 7.1 surround system.

How hard would that be to implement within Q-SYS?\ Here is an example:

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The first thing you should have noticed is that I moved the levels, delays and High Pass filters over past the router that selects which set of Surround EQ to use. The High Pass filter, if needed at all, will be the same regardless of surround format as it is there to protect the speaker(s). Likewise, if there is any needed delay on the surrounds, by zone, they will be the same regardless of format.

By setting the levels for 7.1, we will better balance out the room. Odds are, if your theatre is long and skinny, the rear surrounds will be set louder than a squarer room.

I’ve put the surround level offset with their respective modes (-3dB) so we should omit any other Surround Offset if this scheme is used. The output levels for 5.1 and Mono Surrounds are really just offsets to take into account any imperfections in how the speakers sum, particularly after the various configurations are tuned with their pairings.

The output of the three modes feed into an Audio Router with 3 inputs and 1 output (set to multichannel with 4-channels). You will need to tune 7.1 first. Tune for 82dBc per quadrant.

Then select 5.1 and tune for 5.1 and set the 5.1 offset levels for 82dBc/side.

Finally, select Mono and tune the Mono Surrounds and set them to 85dBc.

Be sure to place the Surround Mode router in the Format Snapshot bank and configure/save the Snapshots such that it is in the correct mode on each format.

Again, this is an example and since it is not compatible with the unmodified Sample 7.1 Design, I am not leaving it in but it is something that you could implement into your designs, if so desired.

Clearing Past Settings
If you are copy/pasting your design to a different theatre, you will likely want to zero out your previous settings so you can start fresh. There are several techniques. For an EQ, select all of the levels (click and drag left-to-right, or shift-click each level control). Type the number 0 and they should all go to 0dB (no boost or cut).

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And then bypass them all. It is important to first zero them out. If you don’t and decide to use one of the bands, the moment you unmute it, you’ll apply, possibly, an undesirable boost/cut to your signal before you can correct it.

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If you find that you have a standard complement of components per channel, build up a channel with everything in your starting settings and save it into your User Components. Then, drag that user component onto your schematic. Use “Copy All from Component”/“Paste All to Component” component-by-component.

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If you find that your EQ group is always the same, save a copy of the whole thing in your User Components. Drag the whole thing into your design and then cut/paste the Signal Names and delete the ones that have the old settings.

Cut/Paste Signal names is not as complicated as it may sound. Select them all in one selection, “cut” <CTL+X>…then select the pins of your standard EQ group that you drug in from your User Components, and paste <CTL+V>.

Conclusions
Hopefully, this blog has given you some insight on how you can configure your design for tuning a theatre. Regardless of method (or methods), you should be able to tune your theater with Q-SYS as well or better than any other system out there. If you adopt using PEQs for your equalization, I think you’ll get a better/more uniform outcome.

You have the flexibility with Q-SYS to customize and optimize your sound processor to get the most of your theatre’s capabilities.
©2025 by Steve Guttag

[Blog-10, Page 4 of 4, End of Blog]

[Steve Guttag]

Q-SYS updated their Sample 7.1 design yesterday (3/18/25). the only changes were, effectively, cosmetic (component alignment) and they zeroed out the EQ left behind in Left Channel. Though they updated the version number to 4.3, I would have thought that 4.2.1.0 would have been more appropriate since there is no substantive change. They did use QDS 9.13.0 for the updated version but it should work just fine in 9.4.8 LTS.

[Steve Guttag]

Q-SYS For Cinema

Blog-11, QDS–Part-7, Sample 7.1 Part-6, Audio Flow Part-5, B-Chain Part-2

Current QDS Versions: 9.13.0 and 9.4.8 LTS.
Sample 7.1 design version: 4.3.0.0 Introduction

This blog will finish up the audio portion of the Sample 7.1 Design with the Outputs and Speakers sections. This blog is a direct continuation of my previous blog on the Equalization section.

Disclaimer

If any of the content in this blog happens to show up in a Q-SYS exam, it is not my intention to provide an answer-sheet beyond the discussion of good practice. I have not seen any form of the cinema final exam (my Level-1 was before there was a cinema version).

Disclosure

I do not, in any way, work for QSC/Q-SYS. These thoughts are my own based on my own interactions with the product(s) and implementing Q-SYS within actual cinema environments. I do work for a dealer that has sold QSC products since the 1980s, including Q-SYS and its predecessors. For the purposes of this blog, I represent only myself and not my employer(s) or any other company.

Amplifiers and Speakers (Outputs)

Really, these two go together. Since this particular blog is about working with QDS rather than sizing/choosing amplifiers, I’m going to keep the discussion to how you work with them once the amps and speakers are chosen.

Planning for Minimizing Failure Impact.

Unlike with conventional cinema processors, with Q-SYS, we can strategically plan out how the system fails to minimize how it can affect the presentation. We have all of the controls over the signal path.

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In this design, there are just two amplifiers and they are 8-channels each. This can be a very economical way to handle a typical 7.1 sound system with bi-amplified screen channels. However, you are putting a lot of “eggs” in one “basket.” Planning on the day that one of those amplifiers fail should guide you on how you choose what channels to place on each amplifier.

The number one thing to remember is to never put any part of the Center channel on the same amplifier as Left and/or Right. Your entire bypass scheme relies on either Left and Right working when Center is not or vice-versa. The Sample 7.1 Design honors this concept. Left and Right are on Amplifier-1 while Center is on Amplifier-2.

What else? You should try to balance out your amplifier so if one fails, the system will remain balanced. That is, the sound stage shouldn’t feel like it shifts all to the Left or Right. While it is not ideal to have a failed component, how you lay out your design can minimize its impact so patrons won’t notice it as much. If you look at how they loaded their amplifiers in the design, you’ll see this. If Amplifier-1 fails, you will lose Left, Right, half of your side surrounds and half of your subwoofers. By selecting the appropriate Bypass mode in your design (covered in Blog-9), no critical audio will be completely lost; it just will not be optimal.

If you chose to have a second side surround channel due to quantity of speakers and the loading to the amplifier, consider wiring them in an alternating fashion (if you number your speakers around the horseshoe pattern around the room with 1-8 on the left side wall, consider wiring 1, 3, 5, 7 on one amplifier and 2, 4, 6, 8 on the other) so that with an amplifier failed, the theatre remains, somewhat, covered, just less so.

If Amplifier-2 fails, you lose Center, half of your side surrounds, half of your subwoofers and all of the rear wall surrounds. Once again, the system fails in a balanced manner and there should be a Bypass mode to redirect Center to Left and Right. If you want, you could include the rear surrounds in your Bypass scheme so that the rear-surround information is mixed into their respective side surround. It’s not ideal, but you are trying to make the best of a bad situation. And this condition only lasts until you replace/repair the amplifier(s).

It is worth looking at the rear of a DPAQ/CXQ amplifier to see what it takes to swap one of them out. Everything is “plug-in,” with the speaker outputs having the option of securing with a couple of (captivated) screws.

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The swap can be very brisk. All that would need to be done, once physically swapped, is to configure the replacement amplifier’s IP address and name. The Core will immediately update the amplifier to be on the same firmware as the rest of the system and then load the design in.

Alternately, if you set up your system to utilize dynamic pairing, you can make it so once the new amplifier (or any other part of the Q-SYS ecosystem) is plugged in, it self-configures and starts working as soon as the firmware matches and the design is loaded (automatically). The system will take care of the naming/IP portion so no remote technician involvement is required.

Here is a link to a Dynamic Pairing video. It does not discuss, specifically, how to set up for amplifier swaps. It does discuss moving devices from room-to-room (or theatre-to-theatre, for us). So, if you have a lectern for rentals/Power Point/Zoom…etc., it is easy to see how Dynamic Pairing could be a handy tool.

Video On Dynamic Pairing

However, the same concepts can be used for swaps of any equipment, including amplifiers. The key points are:

• Have a DHCP server on the Q-LAN network(s) so when the amplifier is plugged in, its IPs can be set by the DHCP server.
• Have LLDP (Link Layer Discovery Protocol) active on your network switches used for QLAN.
• Configure the Dynamic Pairing (in the Administrator Program that is almost never used in cinema) to use the “Switch Port” method of pairing. This will have the system decide that if a proper device is plugged into a configured switch port, then that device is to be dynamically paired. So, if a CX-Q2K4 is set up on port 3 of the QLAN switch in the sound rack, if a different CX-Q2K4 is plugged in, it will configure it and put it on line. In fact, even if you plugged a different model amplifier that had sufficient specifications (4-channels like a CX-Q4K4) it should work with it.

Another strategy one can use is to increase the number of amplifiers to lessen the impact of any one amplifier failure. This could be very appealing if you split up your amplifiers and locate the screen/subwoofer amplifiers behind the screen, with the speakers (shorter/cheaper speaker runs) and have the surround amplifiers in the booth. A popular configuration for a 7.1 theatre is to use three 4-channel amplifiers (Left/Right on one, Center/Subwoofer on another and Surrounds on the 3^rd amplifier).

You are not compelled to use Q-SYS amplifiers. If you are using either conventional analog/linear amplifiers or digital amplifiers that use Dante or AES67, instead of amplifier components, you’ll need to use the appropriate output components and that will be the end of Q-SYS signal flow as that will be the “off-ramp” to the system.

[Blog 11, Page 1 of 2]

[Steve Guttag]

Speakers

If you are using Q-SYS amplifiers, you’ll need speaker components. If they are QSC speakers, there should be components in the Inventory (LSP) and they will have Q-SYS’ “Intrinsic Correction” as well as the various protection information (limiters, crossovers, high pass filters…etc.). It’s all canned in the one component.

Key Point: Breaking with tradition, Q-SYS makes the top pin on the speaker components as the highest frequency of a multiway system. All of the previous crossovers used with cinema (including QSC) have had the lower-channel be the lower-frequency and one worked their way up (e.g. XC-1, XC-2, XC-3, the entire DCM series, the DCP series, and even going all of the way back in the Series 1 days of the Model 1400). If you use Signal Names, you can easily keep the standard convention. If you use wires, they will cross each other if you keep with standard convention. The Sample 7.1 Design adopted the inverse as it is native to Q-SYS. Just beware when you are first firing up your system that it is very easy, if you’ve been putting in systems for a while, to get an HF/LF being transposed. Keep your levels down until you can verify that everything is connected properly for your design.



As covered in a previous blog, you can create your own speaker components (in the Inventory as a “Custom Speaker” and “Custom Voicing”) and assign the HF/LF as you desire.

DCIO Output



Your ADA channels (HI and VI-N) exit the system (off-ramp) here (DCIO-H Analog Out component). Your booth monitor exits here too. You have the option of a line output for an amplified monitor or a speaker level output to a conventional passive speaker. The DCIO’s speaker output is only a 10W amplifier with no headroom.

If you look over to the speaker group, at the bottom, is an AD-S4T (surface mount utility speaker with a 4-inch woofer, so not too big) that is being used for the Booth Monitor. If you reference back to a previous blog when we covered the Processing group, we have a source selector to feed the MON-PRE-A1 Signal Name. So, from the speaker component, we take the unconventional step of sending the signal back to the left to send it out of the DCIO-H. While the AD-S4T is indeed a speaker, when the speaker is being used as an “in-line” component, it is clearer to have it in-line with the audio flow. That component is representing the tuning of the speaker rather than the speaker itself. So, I’d move it to before the DCIO-H output.



Based on experience, I’d probably drag the “Clip” LED out of the Analog Output for the speaker out.





This way, you’ll know if you are clipping or not if the monitor starts to sound distorted. You may also find that you’ll want to add in a compressor and possibly a limiter (it all depends on the speaker you are using and how the monitor is being used; in many theatres the monitor is almost never used).

As we’ll discover in a future blog on the control of the Sample 7.1 Design, the method controlling the booth monitor volume is by controlling the DCIO-H speaker gain (internal to the “Analog Out DCIO-A1” component). Why do I think that is a bad idea?

• As we’ve discovered in previous blog(s), any component that comes from the Left-Side Pane cannot be copy/pasted to other designs. So, in order to copy this design, at the very least, after adding in the DCIO-H to the new auditorium, you’ll need to open it up and then remap a control to the Monitor fader. If we had an external Gain component, that whole aspect would be eliminated.
• We have no control over the range of the internal Gain. So, depending on the speaker/environment, if we want to restrict the range to prevent clipping, that is not possible.

I’m not going to leave them in the design because I want the control aspect of the design to match for the unmodified versions but this is how I might modify the Booth Monitor if I were to deploy it (I would also clean it up to avoid the diagonal wires and the overall “stacked” appearance):



This sort of scheme eliminates the problems identified.

Tip: If you are using the speaker output for the booth monitor, the line output is a regular balanced line output that you can use for anything else you may need. Just because they labeled it as monitor does not force you to use it as such.

Conclusions

Here we are. All of the audio channels have now exited from the system. You should be able to tune your theater with Q-SYS as well or better than any other system out there. The amplifier and speaker components, be they Q-SYS or your own custom components should provide superior protection to other methods as the limits of the speakers/amplifiers are included into the component.

Hopefully, you will have seen just how flexible the system can be and how much control you have over it. Furthermore, you can make changes, as your needs change, without asking/begging a manufacturer to implement a feature you may want that others may not find as necessary.

And, as complicated as it may seem at first, you can be up and running on a system rather quickly. As we will see in a later blog, cloning a design can be quick for fast deployment.

The next set of blogs will go over the control aspect of the Sample 7.1 Design.

©2025 by Steve Guttag

[Blog 11, Page 2 of 2, End of Blog]

[Steve Guttag]

Correction to Blog-8

https://www.film-tech.com/vbb/forum/...5669#post45669

The Preshow Match pin should be pin 1, not pin 4

image.png

[Steve Guttag]

Q-SYS For Cinema

Blog-12, QDS–Part-8, Sample 7.1 Part-7, Logic and Control Part-1, Status

Current QDS Versions: 9.13.0 and 9.4.8 LTS.
Sample 7.1 design version: 4.3.0.0

Introduction

This blog will start the analysis of the Sample 7.1 Design from the viewpoint of the logic and control section, which will include the UCI (User Control Interface). As I did with the audio portion, I’ll try to work from left-to-right (though, by its nature, control tends to jump about). So, we’ll start with the Status section.

Disclaimer

If any of the content in this blog happens to show up in a Q-SYS exam, it is not my intention to provide an answer-sheet beyond the discussion of good practice. I have not seen any form of the cinema final exam (my Level-1 was before there was a cinema version).

Disclosure

I do not, in any way, work for QSC/Q-SYS. These thoughts are my own based on my own interactions with the product(s) and implementing Q-SYS within actual cinema environments. I do work for a dealer that has sold QSC products since the 1980s, including Q-SYS and its predecessors. For the purposes of this blog, I represent only myself and not my employer(s) or any other company.

Status
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Having a good method of conveying the status of key parts of your design will go a long way in terms of troubleshooting as well as providing a means to let the user know that things are “OK” or in need of attention. As mentioned, in a previous blog, I’ve changed the color of the Status components so that they would contrast with the status LEDs.
In this design, there are 5 status components for the 5 pieces of hardware plus two Status Combiners. Status Combiners are key to not overwhelming the user with information that they will not understand (it isn’t reasonable to think that the typical user of the sound/control system (i.e. the Manager) will be intimately familiar with the minutia of the design) yet let them know that everything is okay or that there is a problem. While there are only two amplifiers in this design, you could have a lot of amplifiers in a Dolby Atmos system. Is it practical to present say 15 amplifier’s statuses to the user? If so, where? However, if you send all of those statuses to a combiner, then you have a single status that encapsulates all of the devices that are attached to it. Looking in the Status Combiner that has the amplifiers connected we have this:

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We know that everything is okay and if there was a problem with either (or both) amplifiers, the Status would change to reflect the status and the labels presented in the Status are the Labels you provided. In this case, it is “AMP1” and “AMP2.”
In case you are wondering, yes, you can feed Status Combiners into other Status Combiners (as we will see a bit later) to create a “Tree” of statuses. However, the status for the original source is conveyed.
The other status combiner drives the point home a bit more. It is for the speakers. We have 11 speakers. It is impractical to bring all 11 statuses to a UCI except via a Status Combiner. Since its status is green, we know, at a glance, all speakers are ok. Again, by using the labels, if there was a problem with a speaker (missing, shorted…etc.), the label you use will be put in the status.

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As a technician, the LEDs on the components in the schematic are sufficient to let you know that all is well. For the end-user, this information needs to make it to the UCI. If you look at the UCI for this auditorium, we can see that they chose to not bring the speaker status out and did bring each amplifier status out.

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My guess is that adding the results of the Status Combiners are a test question. It isn’t hard to imagine that using discrete status LEDs does not scale well. Just providing the LEDs also doesn’t aid the user in troubleshooting. If it isn’t green, what does that mean? What is wrong with it? Q-SYS is capable of conveying much more information than a simple go/no-go LED to an end user. Perhaps a pop-up button that has the Status boxes would help?

[Blog-12, Page 1 of 2]

[Steve Guttag]

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So, why do you think that they used independent Amplifier Status versus the combined status? And, for that matter, why didn’t they add any speaker status?
My guess is that the Status Combiner does not present a single status LED in addition to the text box. They present “traffic light” LEDs for the three general levels of status: “OK”, “Compromised”, and “Fault.”

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So, if you just drag out the “OK” LED. When it is off, it just changes into a less intense green LED. However, we are certainly capable of changing that in the properties of the LED. Drag out the OK LED.

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With the LED selected, go over to the properties, change it to a bright green when on and red when off. You will have to “unlink” the on/off colors.

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Size them to match the other LEDs on the UCI and then you will have the Status Combiner “OK” LED mimicking a status LED.

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What do you think? In my opinion, it is probably still too much information for an end-user. Why take up real estate on the UCI for 4 different statuses? Really, the end-user (manager) just needs a good/bad notification. So how about a single status? Let’s create a Status Combiner that covers it all.

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Everything the user needs to know in one simple indicator with the ability to drill down, if a problem arises. How is it done?

• Add a new Status Combiner with 5-inputs.
• Reveal the Status pins of the components to be monitored (Core, Touchscreen, DCIO, Amps and Speakers).
• Add Signal Names to the Status output pins of the various status components and to the new Status Combiner.
• Name the inputs of the new Status Combiner and move the OK LED out to the UCI and change its LED properties as described previously.
• Place that LED on the UCI in lieu of the other previous ones.

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This is a continuation of a theme with my philosophy of system design. Know your audience. The needs of a technician that has to troubleshoot a system quickly are quite different than the needs of the user that just needs to know if it is working or not (with respect to status). This is why I brought the traffic light LEDs out and placed them on their respective components. It lets the tech to see where the problem is quickly. If anything is RED, they know to look inside to see where the problem is. For the end-user, one global status is sufficient and more efficient use of space. Status Combiners survive a copy/paste of one design to another for duplicating auditoriums.

Conclusions

Having/creating a good status system will go a long way to quickly troubleshoot a problem, if/when one comes up. As such, the work put into setting them up will pay off with each use of your design. They also “scale” well (grow/shrink to the size you need, for the components in use).
Also, with respect to cinema, keep with the standards of the industry. Green = OK and the show should be able to run properly. Red = problem that can/likely will interfere with a show. Make sure that your status colors reflect that.

©2025 by Steve Guttag

[Blog12, Page 2 of 2, End of Blog]

[Steve Guttag]

The next big set of changes to Q-SYS is set to be announced at their next "Activate" webinar. Part of the announcement includes QDS 10.0.0 and new Cores.

One can register for the event with the link.

https://www.qsys.com/campaigns/activ...e_qsysinaction

[Leo Enticknap]

For those of you who use IP2IRs to control consumer devices that can't be driven by RS232 or IP from within Q-Sys, there is an update for the plugin just out:

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[Steve Guttag]

Q-SYS For Cinema
Blog-13, QDS–Part-9, Sample 7.1 Part-8, Logic and Control Part-2, Format Selection

Current QDS Versions: 9.13.0 and 9.4.8 LTS.
Sample 7.1 design version: 4.3.0.0

Introduction
This blog will continue the analysis of the Sample 7.1 Design from the Control (and UCI) perspective. While I’ll try to move, mostly left-to-right, the very nature of control has one jump around since where the design execute controls, including UCIs may be in different locations from the component that they control.
Most of the actual Control Components are on the far-right of the bottom section so it might be best, for this discussion to, temporarily, rearrange the design for clarity.
So, picture it looking something like this:

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I’ll do likewise for the Test and Measurement (essentially put things back).

Disclaimer
If any of the content in this blog happens to show up in a Q-SYS exam, it is not my intention to provide an answer-sheet beyond the discussion of good practice. I have not seen any form of the cinema final exam (my Level-1 was before there was a cinema version).

Disclosure
I do not, in any way, work for QSC/Q-SYS. These thoughts are my own based on my own interactions with the product(s) and implementing Q-SYS within actual cinema environments. I do work for a dealer that has sold QSC products since the 1980s, including Q-SYS and its predecessors. For the purposes of this blog, I represent only myself and not my employer(s) or any other company.

Manual Control
The first thing we have in the Manual Control section are the “AUDIO PRESETS.”

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One thing to figure out is…what are they? And what is with those grey dots in the lower-right corner?

Key Point: <CTL-F> should be your friend when working with controls. Controls, Signal Names (wire-tags) Signal Snakes…etc. can be buried in containers and on different pages. Rather than hunting all over a design, let <CTL-F> “find” the source (or even another copy) of a control.

So, what happens when we select one of them, say FEATURE 7.1 and press <CTL-F>? This box pops up:

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This will let us find the source of the control as well as we now know that there are other copies in the design. But first, let’s find where this one starts:

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It is the Load-4 button in the Snapshot Controller called “FORMAT-A1.” Check out that the Find command also lets us know about the External Control “F71.” More on that later.

Snapshots

Snapshots record the state of things that are in their bank. That is, whatever setting it might be set to, at that time the preset was “Saved.” So, that could include a fader, a button or a value…any or all.
Snapshots are a Left-Side Pane (LSP) which means that they don’t copy/paste. So, if you fill your design up with Snapshots and plan on copying the design from theatre-to-theatre, you will have the overhead of also recreating the Snapshot configurations. As such, I try to minimize my use of them.
Snapshots are quite handy for when you want multiple things to change at once. Let’s look at the Snapshot for the “FORMAT-A1”:

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It recalls the:

• Presets in the ROUTING (chooses which input is active).
• MASTER FADER (so it will set preset levels).
• SURR.OFFSET GAIN (applies the appropriate level-shift between 5.1 and 7.1 formats).
• Digital In DCIO-A1 (most likely to choose between HDMI and AES 9-16).

Note, they took the easy method of dropping entire components into the Snapshot bank (All Controls). This is not required. You can just drop in the controls of interest. Sometimes, you may not want all of the controls within a component to follow a Snapshot preset.

If you need a review on how to use Snapshots, here is a link from the Level-1 training videos (you have taken that course…right?)

Snapshot Training Video
(Note, QSC is going through some web-site updates so the links may be broken either temporarily or permanent.)

Another thing to note from our “Find” search was that it landed us on an actual button…the Load 4. That means that this button

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and this button

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are one and the same thing! To create the Format buttons, all that was done is to drag out the Load buttons and rename them and give them a color consistent with the theming of the design. Pretty easy, huh?

[Blog 13, Page 1 of 3]

[Steve Guttag]

Format Design

Using Snapshots for format selection is going to be pretty standard in most cinema designs. I certainly use them for that in my designs.
However, let’s look a bit deeper into the architecture of this Audio Preset/Format Selection.

They have separate buttons/presets for Trailer 5.1 and Feature 5.1. Why? My guess is that it was an outgrowth of a simpler time and a desire to mimic a feature popular on an earlier QSC processor, the DCP line of processors (DCP100, DCP200, DCP300). One could create up to 16 formats and dedicate some to being Trailer formats and some Feature formats while having some for non-DCP formats.

The simpler time was when trailers just came as 5.1. What do you do if the client uses some 7.1 trailers? Can we add that? Sure, this is Q-SYS. They have an 8 Snapshot bank and 8 buttons so we’ll need to expand that a bit.

• Change the Snapshot’s properties to having a count of 9.
• Copy an existing Format Button (e.g. Trailer 5.1) so we get the same size and color properties and paste it to the design.
• Rename it TRAILER 7.1 (click on the new button and press your backspace key…that will let you rename it…so backspace over 5.1 and type 7.1 and then click off of the button).
• Drag the Load 9 button over to our new button (you should see a dialog box telling you to press and hold the Control key) but DON’T RELEASE the mouse just yet.
• Press and hold the “Control” key on your keyboard and release the mouse…you should have a selection that allows you to “Remap” the Control

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Great, now we have a button. But it doesn’t do anything. We need to get it to apply the Trailer volume level but with the 7.1 audio formatting.

• Start Emulating (F6)
• Select TRAILER 5.1 and note what its Fader level is (-6dB).
• Select FEATURE 7.1 so the audio path configures itself correctly.
• Adjust the Volume down to -6dB.
• Click on “Save 9” in the Snapshot bank.

We now have our new format:

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Pretty easy. But we’re not done. Stop emulating (F7) and you’ll notice that our new button does not have a grey dot in the lower-right corner. Why?

The grey dot represents that the control is a “Named Control.” Name Controls can be controlled by outside/3^rd party devices, like cinema servers. If we want cues to be able to select TRAILER 7.1, we’ll need to add it to the Named Controls bank.

On the Left-Side Pane, select the Named Controls. You’ll see all of the current Named Controls:

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Grab our freshly minted TRAILER 7.1 button (either from the Snapshot bank or from the “AUDIO PRESETS”). Drag it over to the Named Controls and release the mouse. It will automatically be given a name of what the control is:

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That doesn’t match the other named controls so let’s rename it to T71, to be consistent.

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Are we done yet? Nope.

We need to expose the “Match 9” pin of the Snapshot Controller and change the Logic OR component to a 9 input so that TRAILER 7.1 is also capable of turning off the Pink Noise:

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We’re still not done, however.
We haven’t given the user a means to select it. We need to go to the UCI and provide a Trailer 7.1 button. In much the same way as we made the button in the schematic, let’s copy/paste an existing button (Trailer 5.1) so we get all of its properties.

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Rename it as you did with the schematic button. (Select, backspace, adjust the text, click off of the button).

Then, either go split-screen or click and drag (and keep holding your mouse button) the Load 9 button up to the “Auditorium 1 UCI tab at the top. The program should switch to the UCI tab and you can then continue with your drag down to the new button (keep holding). Press the Control key on your keyboard and release. Select Remap Control.

Now what? The UCI needs fixing so it looks intentional. You can make the buttons all smaller to fit the new one, or rearrange them. One possible solution is to do something like this so all of the 7.1 buttons are above their 5.1 counterparts:

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We have 9 buttons so you could put them in a 3x3 grid. The possibilities are almost endless. However, you’ll want to create a design that isn’t confusing (just an array of buttons that all look alike, at first glance can be confusing). The UCI should help the user quickly accomplish whatever they are using the UCI for. How you arrange your buttons, adding Group Boxes and such can go a long way to helping the user find what they are looking for more quickly.

For those that don’t wish to modify their Sample 7.1 design, I’ll leave the original buttons where they are so my screen shots will match (more or less).

One of the things I don’t like about this form of handling sound formats is that it doesn’t scale well. We now have Trailer and Feature versions of 5.1 and 7.1. What if an immersive sound format is added? We need to add more buttons. How about ads? It spirals out pretty quick when you have all of the permutations.

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[Steve Guttag]

The Problems of Convolving Volume with Formats

Okay, let’s explore one of issues of convolving volumes with formats. Put your Sample 7.1 design into emulate mode (F6).

Select any of the format buttons on the UCI (or in the schematic). I’ll select HDMI 7.1 because it is closer to the Volume control.

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So far, so good. What happens when you adjust the volume? Let’s set it to 0dB. What happens? The HDMI 7.1 button dims to about ½ brightness because we no longer match the Snapshot, which included the volume level. That is a little sloppy.

Let’s repeat the experiment with Feature 5.1. Press Feature 5.1. Now, lower the volume to -6dB. What happened?

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So, are they in the Feature or Trailer? Trailer 5.1 is certainly brighter but Feature 5.1 is also lit at ½ brightness. Confusing? Imagine the user that is faced with this. A UCI should reduce confusion, not add to it.

Here is another issue…how do you change the preset volume levels for the various formats?
The procedure would be:

• Select the format you want to change so it loads that snapshot bank.
• Note, which Snapshot is loaded.
• Change the Volume as desired.
• Press Save for that Snapshot number.

It’s not hard but it isn’t something you can’t really leave for the end-user. They would need access to QDS and the design itself. You certainly couldn’t put it on a UCI or they might make other changes and save to the wrong preset, overwriting another preset (for everything, not just the volume).

How might you modify the design for allowing adjustable Trailer, Feature and even Ad volumes?

Well, the first thing that I would want to get rid of is any chance of more than one format showing as active, even partially. To do that, we need to remove the volume from the Snapshot. We also want the preset levels to be user-assignable without any need to enter QDS and alter the design.

As with most things within Q-SYS, there are multiple ways of attacking a problem. Here is my quick solution to the problem(s).

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To make it easier to work with, I transferred (or copied) the relevant parts to another schematic page to work with them.

I added a Custom Controls with enough level controls to have one for each preset. I then added a Control Router so we can choose which of the level controls should be loaded into the Master Fader upon a format selection. It is the Control Router that we need to place in the FORMAT-A1 Snapshot bank in place of the Master Fader.

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I then added another Control Router (2x1). This allows us to inject the new fader level at a format change but allow for the full fader feedback to work at all other times. So, the “GAIN-A1” Signal Name is moved to here. I added a new Signal Name (GainPreselect) to complete that control signal path back to the Master Fader.

Now, for the control. The Snapshot has Load pins that we can monitor to know if a Preset has been Called or Pushed (by any source). We run all of those into a Trigger Combiner (that is in the Control Functions group, for some reason). For this part of the control, we don’t really care which format was selected (that is handled by the Snapshot), we just want to know that some format was chosen.

The Trigger Combiner triggers an LFO (Low Frequency Oscillator) configured to “one-shot” by un-ticking the “free run.” The wave form is set to “Square” and the time isn’t too critical but I have it set to 100msec.

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The idea is, when a format is chosen, we pulse slightly after the snapshot recalls the preset (which sets up our new control router with the fader preselect that corresponds to format chosen), and pulses the 2x1 control router with the preselect fader, which updates everything downstream (e.g. the DCIO’s fader) so when we release the 2x1 control router, all faders in the chain already match.

Now, you might want to merely provide separate Ad, Preview, Feature, HDMI and Mic levels and not have separate 5.1 and 7.1 volume levels. So, you could combine things.

Or you might want to give the user the option of not having a preset fader level for some formats.

I’m going to leave this in the design by moving it all to a container and dropping it back to where the Snapshot controller was:

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Conclusions
If it were me, I would keep formats as formats and volume cues as volume cues but if you like this type of design with convolved volumes, you should continue to use them. What works for one theatre or chain may not be as suitable for another. Q-SYS is flexible enough that you can make it provide whatever features you desire.

But whatever designs you come up with, they should help the user and not add to the confusion. My suggested changes are in that light. If you are going to implement such a strategy, make it such that it is user-friendly.

Also note, it may be part of the design to deliberately keep the end-user from altering a “corporate” volume setting for the various portions of their shows. There is no one right answer. It all depends on what the design criteria are.

My next blog will continue through the control/UCI section of the Sample 7.1 and concentrate on the Master Fader itself and how it is presented.
©2025 by Steve Guttag

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[Steve Guttag]

Q-SYS For Cinema
Blog-14, QDS–Part-10, Sample 7.1 Part-9, Logic and Control Part-3, Linear Fader vs dB Fader
_5/10/25

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Current QDS Versions: 9.13.0 and 9.4.8 LTS.
Sample 7.1 design version: 4.3.0.0

Introduction

We’re at the point in our analysis of the Sample 7.1 Design where we should discuss the Master Fader from the perspective of control. This seems as good a place as any to have “the talk.” That is, whether to use a dB style fader (-100dB – +20dB), that is native to Q-SYS, or to use a more traditional linear fader (0-10).

For the most part, in my blogs, I’ve taken the position that one of the great things about Q-SYS is that it allows the designer (you) to create a system that suits the needs of the space it is being used in. That position remains true in this blog. However, I will make the case as to why a linear fader, in a typical cinema environment, for use as the Master Fader (Volume Control), should be the preferred method as well as why a dB type fader is a poor choice.

So, perhaps you should buckle-up or, if you prefer, grab a tub of some nice hot buttered popcorn and let’s get into it.

Disclaimer
If any of the content in this blog happens to show up in a Q-SYS exam, it is not my intention to provide an answer-sheet beyond the discussion of good practice. I have not seen any form of the cinema final exam (my Level-1 was before there was a cinema version).

Disclosure
I do not, in any way, work for QSC/Q-SYS. These thoughts are my own based on my own interactions with the product(s) and implementing Q-SYS within actual cinema environments. I do work for a dealer that has sold QSC products since the 1980s, including Q-SYS and its predecessors. For the purposes of this blog, I represent only myself and not my employer(s) or any other company.

The Case for a Linear Fader over a dB Fader for a Volume Control

dB Fader Pros and Cons
It will probably be easier to start with the pros and cons of the dB fader before discussing the linear fader.

Those that are proponents of the dB fader will point out that 0dB is the reference. Additionally, while adjusting volume, you “know” how far off reference you are. And that’s it. Those are the big “sells” to a dB fader. It is not that there are no other benefits but typically those are the justification that is used. In a screening room or post-house environment, there could be other benefits to a dB scale as well.

Now, internally, to the design, working in dB notation is helpful since you are always working in dB. So, if you need 2dB more level to reach 85dBc, while balancing the system, dB is a handy scale for that.

If you are mixing a live event and want to know where you stand relative to maximum amplifier power, again, a dB scale can be beneficial. If you see your amplifier is getting up to within 6dB of clipping, you know you can’t go up more than a few dB, regardless if you think it needs to be louder.

However, none of that applies to the cinema employee that just needs to respond to the complaint of “it’s too loud” or the exceedingly rare “it’s too quiet.” If your fader is set to -3.3dB and the complaint is that it is too loud, where do you turn it to from there? -3.4dB? -4.3dB? -6.6dB? How many managers, if asked this question would come up with the same answer? How many technicians asked this question would give you the same answers as each other?

dB (decibels) are, by their design, a comparative scale. That is, we take a reference, compare another value to that reference and apply a logarithm to it (oh yeah, you have to be “into” logarithms to work with the dB scale). And then use the appropriate multiplier (which one to use also changes based on what you are comparing due to how logarithms work with units raised to a power).

Are you with me so far? And, with respect to a “Master Fader”, since 0dB is the reference, most of the time, since often the movie is too loud (for normal humans), we will be working with negative numbers! So, you are asking typical theatre employees to work in a non-linear numbering system, that uses logarithmic scale and perpetually with negative numbers. As such, ‑6.6dB is lower than -3.3dB. Honestly, you’d have to try to come up with a more unfriendly system for people to work with. And, for an added bonus, it is a sliding scale due to how humans hear things. Changes around a normal (comfortable) level will sound like a greater change than changes further down the scale. So, a 10dB change around -20dB will not sound like as much of a change as a 10dB change around 0dB (presuming your system is calibrated to industry standards).

What is the maximum level? Why? What is the minimum level? Why?

It necessarily compresses the most used portion of the volume control’s range into a very small portion of the scale (-10dB – 0dB), as we’ll see later. When you have a fader with a scale from ‑100dB to +20dB…10dB is a rather small window.

So, the cons boil down to two big ones:

• It isn’t user-friendly.
• It compresses the useful the range in an unfavorable way for its intended use.

Linear Fader Pros and Cons

The cons are limited. The biggest one thrown about is that, in cinema, 7.0 (or 70%) is the reference on a 0-10 scale. Rather than be self-apparent, like 0dB, one has to be told that 7.0 is the reference (consider yourself, now, told/informed; if you find this too tricky, by all means, add text to your UCI to the effect of “Ref = 7.0”).

Note, even on a dB scale, it is not a “given” that 0dB is the reference volume level. 0dB references a -20dBFS (dB Full-Scale) reference signal (pink noise with a 12dB crest factor) that will play at 85dBc at the reference listening position (see SMPTE ST202 or most any Dolby cinema processor manual). One could have used -20dB as the reference for volume too (that is what is used, internally, for the signal path).

So, here are the pros:

• It is human friendly (and this is the biggest reason, really). If your fader is at 0, it is off. If it is at 10, it is all of the way up (what are the limits on a dB fader again?).
• If 7.0 is a little too loud, most will turn it down to 6.5 or perhaps 6.0. And, if they find they went too far, or not far enough, they can quickly get to the level that minimizes complaints.
• It puts the sweet spot of the fader’s range in the most used areas so you have more granularity in setting the level. 4.0 = -10dB and 10 = +10dB. If you think about where you’ve set a theatre fader, if you are like most people, it is between 4.0 and 7.0 (particularly due to ads and trailers being even more loud than the features). As I’ll show later, this has greater implications on the UCI front.
• It is very well established. All Dolby processors (since the mid 1970s) have used the 0-10 scale (so we’re going on 50-years). Most sound systems prior to Dolby also used a linear scale on their volume controls…so we’re over 90-years with a linear scale.
• All of the logarithm stuff is still in there but it is baked into linear fader. The user isn’t burdened with the technicalities of it. They just need to raise/lower the fader as desired without the math or negative numbers.

Audio Taper

Let’s look at the audio taper (aka log taper) of a potentiometer. This is how the level increases. It is rapid, at first, but it gets more gradual as you reach the top of its range. This coincides with how we hear. As you move towards quieter levels, you need to move the level progressively more for it to appear to be going down as fast as it was at the upper end of the range.

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Dolby Fader Characteristic

It was established, by Dolby, with their first “Digital” sound processor, the CP500, in the 1990s, to approximate the audio taper with a two-slope approximation such that from 4-10 it uses a gradual slope and from 0-4 to use a steeper slope.

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It does not matter that it does not precisely track what a physical audio taper potentiometer would do since the only reference point is still 7.0. All other points are, by definition, not standard. What the dual-slope method establishes is a means to easily represent the essentials of the taper (it is gradual towards the top of range and more aggressive towards the bottom end of the range).

For values between 0 and 4.0, the step amount is 20dB. So, if 4 = -10dB, then 3 = -30dB. Heading up the other way, each step is 3-1/3dB. So, 5 = -6.67dB. Great. Fractions (but for the user, that is all hidden away and baked into the linear fader)!

Note, you could come up with your own approximation to the log taper, so long as you keep 7.0 (or 70%) as your reference (it is well established and called out in numerous studio documents as the reference level) and your users should be able to work with it with ease. The advantage of using Dolby’s published characteristic is that it is:

• Figured out (e.g. the Classic Cinema Fader plugin).
• Your system will have a direct correlation with other systems with respect to audio level in the theatre versus the level on the fader. I have some Q-SYS systems sitting side-by-side to existing sound systems using the same volume cues so there is a benefit to having some uniformity to the relative levels. Now, if you are mixing a Q-SYS system in a site with QSC’s DCP line (or DPM), then you’d want to continue with a dB scale for uniformity within the site. Or, better yet, switch it all over to Q-SYS let them all have a linear fader!

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[Steve Guttag]

Linear versus dB Fader on the UCI

Okay. Let’s drive the point home with the UCI. I’ve added a linear fader next to the original dB fader to the UCI.

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At first glance, there doesn’t appear to be much of a difference beyond the scale. But let’s look at how the scale affects the user experience.

Most would agree that the bulk of fader settings are between 4.0 and 7.0 in a typical cinema for the various content types that include any preshow. Let’s lower the fader to 4.0.

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Look at how little the dB fader moved for that drastic a change in fader level. You compress the area of importance into a very small portion of the fader’s range. Visually, it appears that you haven’t made much change at all when, in fact, 10dB is a HUGE change in level in the theatre. The dB fader has about 80% of its range left for minute detail in an area that is rarely if ever used.

The linear fader better represents the relative volume because it has the logarithmic taper cooked into it for the user but presents itself in a manner that is consistent with how loudness is perceived by people. You also don’t give up anything for it. You can still lower the level all of the way. But, with the linear fader, it automatically lowers the gain at a higher rate below 4.0 to better simulate how people will perceive the level change.

So, lowering the fader from 4.0 to 3.0, lowers the dB fader from -10dB to -30dB. In the next image, ask yourself, does 3.0 (or 30%) better represent the level in the theatre or the -30dB that looks like you are at about 60% up?

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How to add a Linear Fader

For this blog, I used a plugin called the “Classic Cinema Fader” by Jay Wyatt. At the moment, it is not in Asset Manager but it is part of the “Q-SYS Level Two Cinema – Student Assets-2023” zip file.

Student Assets-2023.zip

I believe Q-SYS will provide this plugin, upon request too. Using a plugin (any) requires a scripting license to be active on the Core running the design (but you don’t need one when merely emulating on your computer). It is also available on the Facebook Group “QSC Q-SYS Programming Super Group” in the files section.

Classic Cinema Fader Link in Facebook

In a later blog, on control/logic, I will discuss how to make such a component, including with just logic blocks. However, for now, we’ll use the plugin.
The Classic Cinema Fader replaces the Master Fader (which is just a Gain component).

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Double-clicking on it one can see how it is different than just a Gain component.

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In the properties, I enabled the ramp controls (the Increase and Decrease buttons) and exposed the Gain, and the Master Mute pins, as well as the Dolby Level pins. It has both faders and it updates the “other” fader, regardless of which is used. So, if you put a linear fader on your UCI but the DCIO uses the regular dB scale, everything still works and the appropriate conversions are applied.

Here are the steps needed to make the change in the Sample 7.1 Design.

I copy/pasted the “Dolby Level” of Classic Cinema Fader onto the existing faders in both the UCI as well as the “Master Fader” controls within the schematic:

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Since the + and – buttons are also gain controls (string buttons), those too had to have their properties remapped.

String Buttons

String buttons were one of the tools that was (more than they are now) used before ramp controls showed up on Gain components. They allow you to apply a fixed change to a numerical value. If you click on the plus/raise button and look at its properties.

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It is created by dragging the “Knob” out of the component. Then in properties, you present it as a “Button” and the Push Action is “String.” Then, every time that button is pushed, the string is executed. In the example above “+=1” is interpreted as “Increase by 1.” Note, it is unitless. It just takes the numerical value that is represented by the knob, and adds 1 to it. It could be 1dB or 1 to the volume level.

So, in the case of a linear fader, increasing (or decreasing) by 1 is a bit too course (you’d only be able to adjust it in whole numbers. If you want to change it to 0.5 steps, change the value to .5. If you want it to move in 0.1 increments, change it to .1. Here is the change to .5:

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The same applies to the minus/lower button:

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For the linear fader on the UCI, I copy/pasted the existing fader and then remapped its level/string buttons using the “Dolby Level” from the plugin. Now, I could have also switched them to the ramping buttons from the plugin, with a few extra steps. We might explore what is involved with that on a later blog.

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[Steve Guttag]

I also had to put the plugin’s “Gain” into the Named Controls bank so the server or other automation can control the level. Which fader to use? If you want to be consistent with the default Sample 7.1 design, you’d need to use the dB fader and call it “GAIN” as the original. We can also bring the linear fader into the Named Controls bank too. I did and called it “FADER” though you could call it whatever makes sense for you. In my designs, the dB fader is not used outside of schematic of the design so “FADER” is sufficient.

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But we’re not done yet. If you recall back to the previous blog, I/we changed the Preset levels. If we’re moving to the linear scale, those should be updated as well. You’ll need to open the container (if you copied what I did) and make the changes in there:

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The “Custom Controls” need to be changed from dB knobs to Float Knobs. Change the range of the Float knobs (in their properties) so that minimum is 0 and maximum is 10 (or 11 if you want to do that).

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The blue text boxes need to be remapped to our new Float knobs so they are now float boxes.

The Signal Names actually gets easier because all of the normal dB Signal Names can go where they did on the default Sample 7.1 Design. The “GainPreselect-A1” that we added now gets attached to the “Dolby Level” pin.

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Changes to the Dolby Level pin via the presets will update the dB fader, automatically when our LFO “one-shot” fires on a format change to load the preset level.

Conclusions

Hopefully, I’ve made a sufficient case for you to use a linear fader over a dB fader on your UCIs.

There is also nothing stopping you from using one and providing an indication of the other on the same display but I’d argue what the point of the dB notation is for the typical cinema user. In other applications, I can see having both notations showing.

Note, I don’t agree with referring to the linear fader as “Dolby” since using linear faders predates Dolby entering into the cinema market. Most volume controls are on a 0-10 or 0-100 scale. Of course, Marshall guitar amplifiers go to 11. And, if you wanted to make your linear fader also go to 11, you are welcomed to do so too!

Like with all designs, which fader/display you use should fit your intended user and their environment. With Q-SYS, you have the flexibility to customize a system to suit the needs of the space and user.


Appendix-A: Why 7.0?

So, why 7.0? What is so magical about that value? I wasn’t in the industry when 7.0 was established, back when the Dolby CP100 came out. It used 7.0 as the reference level.

I can speculate…

First, you will want to have some range above reference to handle quieter movies, particularly back in the big barn type theatres of yesteryear (think 1000-5000 seats). Plus, back in the day, movies were not mixed so loud as they are today.

The other realities of yesteryear (1950s) are…historically, multi-channel volume controls entailed multi-gang (1-gang for each channel) stepped attenuators (they are made with precision resistors to get their values to match). As you can imagine, this sort of construction is both large and quite expensive.

Dolby used Voltage Controlled Amplifiers (VCAs) to make a multi-channel volume control using just a single simple potentiometer to control the VCA.

Early VCAs were made with discrete components, including ICs that had a collection of discrete transistors for better matching/thermal coupling. However, that helps you, primarily, on one channel. One of the tricks in a multi-channel system is getting everything to match so that as you adjust the volume, all of the channels should track together. You don’t want the sound to shift to the left or right as you raise/lower the volume.

By locating the reference towards the top end of the range, it puts the control in the sweet spot where small changes on the control still result in small changes to the output. And thus, I think, 7.0 was born as our reference. At least, that is how I believe it came to be.

©2025 by Steve Guttag

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