[TerryParticipant]

With the AR/AB series stage boxes you are limited to 40 channels from the stage boxes – this is a limitation of the d-SNAKE protocol used by these boxes. The good news is you can plug the AB168 into the AR2412’s expansion post and have access to all of he inputs and outputs. You can also use the local inputs on each desk, and/or the USB, in order to get to 48 total channels.

Connect the stage boxes to the S-Link port on the monitor desk and connect the S-Link expansion card to the S-Link port in the FOH desk. Use tie-lines to patch the AR2412 and AB168 sockets to the FOH desk’s input channels. Make pre-amp settings adjustments on the monitor desk and use trim at FOH, if needed. Note that you don’t have to use these specific ports, but by following a standard set up you’ll reduce problems and make troubleshooting easier.

[TerryParticipant]

Hi Nivra.

It sounds like you are coming from an analog setup, where what you describe is quite common.

It’s also possible you’re on an older digital console, as some of them did not have dedicated FX routing. The SQ does have dedicated FX routing, which means you do not have to use up Auxes (Mixes on SQ) to drive the reverb and delay, and you don’t have to use up input channels to get the signal into the PA.

There are four dedicated FX buses, accessed through the top 4 blue buttons on the right side of the desk, which are routed through FX rack slots 1-4. These buses are in addition to the 12 Mix/Group buses the desk has. Each FX bus also has a corresponding stereo return channel available in the SQ (fader Layer C by default IIRC) which do not count against the 48-channel processing limit – in effect giving the SQ 56 channels.

The FX rack has four empty slots in it as well, which can be used to create additional FX buses, but it’s more common to use these as bus or channel inserts (multi-band compression, dynamic EQ, etc) because you would need to give up channels otherwise.

[TerryParticipant]

+1 for more colors at a minimum. Better would be a fully mixable color palette.

Interestingly this subject came up this morning when I was reminded that the (very expensive) console I’m working on today has non-changeable colors based on what the fader is doing. My comment was that on the SQ you could at least change the color being used.

[TerryParticipant]

I recently put together a fiber system for my SQ5 and AR2412. As Keith says, it required a Fast Ethernet media converter on each end in order to make it work, but once I got there it’s been fine. Currently it’s only 50m, because I wanted to be certain it worked, but I’ll be adding a 250m fiber soon, which will cover just about every situation I’m in.

My media converters are TP-Link MC110CS that I got from B&H Photo. They’re SC/APC connectors, based on the advice of a friend who does a lot of work in broadcast video, but LC connectors seem to be more readily available in pre-made TAC fiber.

[TerryParticipant]

A Dante card in the Avantis, if you can get one, would allow you to connect to your Pro Tools rig (assuming it has Dante Virtual Soundcard installed). You can set the direct out point of the input signal to variety of different places in the signal path – this is a global setting so all channels will be the same.

You cannot connect two AR2412 stage boxes with any A&H digital console setup. They were introduced with the GLD and operate at 48K using the dSNAKE protocol which can handle a maximum of 40 input channels so it won’t meet your stated need for a minimum 48 inputs.

Your best setup option would be a GX4816 connected to the Avantis sLink port and a DX168 connected to the DX port on the GX4816. You will have 64 inputs, 24 outputs, and the ability to add additional DX boxes and/or an ME system in the future.

[TerryParticipant]

Reese, to expand on what Mike C said, if you have a stage box for the SQ, one side of the analog splitter would go into the stage box inputs and the other side would go directly to the SI Expression inputs. A friend has been using a similar setup for several years to run a GLD at FOH and Qu-32 at monitors.

I have about 3/4 of the pieces I need to build my own version of the CBI splitter, but currently don’t have many shows where it’s really needed. When finished, the splitter will be on the back side of same rack as my AR2412 with a short set of tails going to directly to the stage box and the “monitor” side on a pair of 56-pin EDAC connectors; I have several 16-pair cables with the other end of the EDAC already on them from previous projects.

[TerryParticipant]

Tyler, you don’t really lose an input because the Talkback socket can be routed like any other local input, while also retaining it’s specific TB functions. In combination with a foot switch and/or soft key, it’s a very useful tool.

[TerryParticipant]

I don’t sing, but have a talkback system set up similar to what you are looking for. My TB mic goes into a local socket and is routed to two channels. One channel goes to the stage mix(es) so the musicians can hear me. The second channel goes to a powered speaker at the A2/SM station (we have no coms system) so I can speak to the folks backstage. The channel going to the stage is muted using a foot switch (Boss FS6) so I can turn it on/off without being on the same layer as the fader – very useful on an SQ5. The A2/SM has a mic going to a powered speaker at FOH so I can hear her.

The second side of the foot switch is most often set to mute the FX sends DCA.

[TerryParticipant]

The connection limit is per console. In order to do that you would have to have some of the IEMs mixes built on the FOH or Broadcast console, which kind of defeats the idea of having a Monitor desk . . .

[TerryParticipant]

I remember my transition from analog to digital. Vaguely, since it was a long time ago, and I’m old now, but I remember it . . .

You are going to become very familiar with ALL the ways the SQ (or almost any digital mixer) can make this process both easier and harder at the same time. The main thing to keep in mind is that there are multiple ways to patch both inputs and outputs on the SQ and it’s easy to get tripped up. Each physical output port (A&H calls them sockets) can only have one source assigned to it. Usually this is a Mix of some kind (L/R, Aux, Matrix, etc.) but it can also be a single processing channel via Direct out or Insert out.

Assuming you only have 12 sources that need to go into the Aviom, there are a couple of ways to handle this.

The first is to use Direct Outs for those channels, and assign each one to an output socket. (Note that you do not have to use outputs 11&12 as the L/R, that’s just the A&H default because the SQ5 only has 12 sockets.) As you stated this will require a bunch of XLRF->TRS cables. By using a Direct out you have the choice of where in the signal path you want it to come from – this can be anywhere from immediately after the Pre-amp to right after the channel delay, and can be set up as either Pre or Post Fader, and Pre or Post Mute.

The second way is the way you were doing it on the analog desk and tapping the channel insert. I’m not at my SQ5 right not, and cannot see how to set this up using the MixPad app, so cannot give you a lot of detail. I will say that this method will ONLY work if you are using a digital stage box, because the desk has no dedicated insert points, unlike the ZED 428. You will need to assign the channel insert out to an output socket, then bring the signal back in via an input socket. Without knowing exactly what the Aviom analog I/O is, I can’t tell you what cabling you’ll need.

Either of the above methods will need a bunch of extra cabling.

If you need to get more than 12 individual channels into the Aviom, you’ll need to be a little more creative. You can still have the key channels that everyone wants to hear a little differently set up individually, while also grouping other channels. Creating a Drums group or a Keyboards group comes to mind. For this you can use either a Group, or an Aux and there are good and bad points to using either. I would use groups and NOT send those groups to your main L/R mix (I know you run in mono, more about that below). The desk ships with 8 Auxes and 4 Groups, but you can reconfigure it in any combination of 12 total buses. Unless you are running a variety of stage mixes in addition to the Avioms, you really won’t need that many.

Something to look at moving forward is the A&H ME system. It’s basically their answer to the Aviom, but the ME-1 can take up to 40 input “channels”, while the ME-500 can handle 16. Channel routing is built into the SQ I/O page, and there is no special hardware interface required. For larger systems you can use most off the shelf POE/POE+ data switches, unless you need the ability to access Dante or MADI audio sources.

Since you are running your PA in mono you might want to create a mono Matrix and send your L/R mix to it. That Matrix would then be send to the output socket currently getting your L channel. For your subs, use either a second mono Matrix (driven by the L/R mix) or a post-fade mono Aux with only those channels producing low frequencies assigned to it. This will clean up your signal flow and give you the ability to EQ each part separately.

Cheers.

[Author]

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