[cornelius78Participant]

I don’t think anyone not from A&H will be able to confirm without seeing the documentation (unless it was mentioned in the Webinar, idk, I missed it,) but given that it’s a 48 channels console with potentially only 16x local preamps (SQ5,) and how the audio racks work with GLD\Qu, my guess is that it’ll be even more flexible than the X32 series: you’ll be able to use a mixture of local and remote preamps, AND there won’t be any “blocks-of-8” limitation like there is with the X32.

[cornelius78Participant]

Thanks for that Keith. Higher track count at lower sample rate would make sense, especially if people are going to use AR\AB stageboxes, which AFAIK are clocked at <96kHz anyway.

[cornelius78Participant]

Hi Keith.

Can you confirm whether the SQ-Drive (onboard multitrack, similar to Qu-Drive, on the top of the console near the phones socket, not the 32×32 CoreAudio compliant interface on the rear of the console) is limited to 18 tracks like Qu-Drive, or can it do more?

[cornelius78Participant]

Brilliant! Thanks for the reply.

Wasn’t sure if that was possible using only SLink sockets between consoles or if SLink could only be used to interface with stageboxes\gigaAce.

It means if 24\14 is enough IO for the stage (not counting TRS connections) and the patching is that flexible, then you can do a FOH/Monitor split without needing expansion cards.

I also didn’t realise that there would be a SLink expansion option, I thought it would only be the usual suspects (Dante, WSG, etc.) That means, as you said, that if you did need more IO (presumably on stage, as even an SQ5 with 16/12 @ FOH should be enough local IO for bgm,) you can use an SLink expansion card in the monitor console, rather than use onboard Slink to stageboxes, then need something like Dante to connect to FOH, and therefore also need an expansion card in the FOH console to RX the Dante signal. That said I’d probably get a Dante card anyway just because it’s so prolific in the rest of the industry, and interfacing the console with other Dante-enabled systems @96k would be nice. I suppose it will depend on cost.

PS I assume you can force a console to clock off its SLink socket instead of its own clock generator so you don’t have multiple clock masters on the one system.

Final question (I’m comparing to the X32 ecosystem.) Can a console relinquish control of its own local\remotely connected preamps (gain and +48v) to another console connected via SLink (eg if someonce decides they want the FOH operator to control preamp gain on the Monitor console (a questionable idea for all sorts of reasons, but I know some operators who’d like the option.) Given that SLink is able to connect and control AR\AB boxes it must be able to send some OOBD for preamp control along with the actual audio…

[cornelius78Participant]

Using gigaACE on the SLink port means you can indeed link to another SQ (or dLive), though this would be for audio as opposed to any control data if that’s the kind of link you mean.

@KeithJ A&H

Does this mean you could have an SQ6 on stage using its local preamps for inputs, and its own DSP for mixing monitors, and it could pass its preamp signals to another SQ5/6 @ FOH, which would use its own DSP for mixing FOH, and the two consoles would be connected via the SLink sockets on each (no need for expansion cards?) The FOH mixer would then pass its LRM mixes back down to the monitor console over SLink, those LRM mixes would effectively appear as inputs on the monitor console, and DOs from those inputs be patched to the monitor console’s XLR outputs alongside the monitor mixes?

[cornelius78Participant]

It’s normal. When an FX engine is configured as an insert you don’t get normal send/return faders to work with. Instead (most, if not all IIRC) the FX have a dry/wet control. If you want your faders back you’re going to have to change the FX routing from insert back to mix>return.

[cornelius78Participant]

As above. If the channel source is a local preamp, then it’s routed 1:1, ie if ch5 is set to source a signal from a local preamp, it will automatically choose local preamp#5, and this cannot be changed. Any instances of ch5 on the custom layer will all have the same processing.

However if you tell ch5 to source a signal from a remote preamp (ie dSnake,) then you can choose which remote preamp you’d like it to take signal from. This allows you to have the same remote preamp feeding multiple channels.

So to sum up, in reference to your original question: if using local preamps, then no. If using remote preamps (dSnake,) then yes.
If don’t have a dSnake stagebox, then just do what Dick said and use a y-split cable between the mic and the console to get the one mic signal to 2 separate local preamps. Then you get separate gain too.

[cornelius78Participant]

I’ll just add that IIRC the line inputs on the Qu Series have a 10db pad c/f the XLR inputs, which will further diminish the level you get.

Re micing the amp: it’ll certainly sound different, as if you go bass>di>mixer then you’re getting the signal straight off the bass guitar. If you have a head with an output at an appropriate level and record off that, the head will colour the sound a bit. If you mic the cab, then you’ve got the sound of the cab + room, along with the mic’s own characteristics, in addition to the bass head, to colour the sound.

It’s not unheard of to use multiple channels for a single instrument, even live. eg use a DI for the bottom end in one channel and mic the cab (even with a non-bass mic, like your ’57) for the top, then mix both channels together.

[cornelius78Participant]

@Bramasole
That only applies up to a point, and then you get diminishing returns. Don’t forget the quality of the source material (if the musician is off key, out of time, has bad mic technique, a poorly tuned instrument etc there’s only so much you can do,) the room acoustics, the importance of having a decent power supply, speaker and mic placement, proper gain structure, having an engineer who knows what he’s doing, and is ideally familiar with the music etc. Just sinking money into more expensive transducers and expecting quality to jump from amateur to professional won’t necessarily work if the above considerations haven’t been addressed.

[cornelius78Participant]

Speaker management\amps\recorders that can take an AES3 input: keep the signal digital as long as possible.

[cornelius78Participant]

I think what the OP wants is a “Solo in place” function: ie the signal coming out of LR is replaced by whatever is soloed. If nothing is soloed, you get the full LR mix. If something is soloed, the LR mix is replaced by that soloed signal. It’s useful in the studio (especially if you don’t have CR outputs,) but quite dangerous live: you can end up with a floor-tom solo in FOH mid-gig if you’re not careful.

Easiest way I can see to do it is to run LR from the Alt-Out sockets (they’re balanced,) and route the PAFL signal to them. In the PAFL settings, additive mode off, input AFL off, output AFL on, LR to PAFL on. Perhaps a softkey to clear solo could be useful too.

[cornelius78Participant]

You can bring the signal from the external pre into the Qu using the line inputs (above the preamps.) From what I’ve read, your Manley will be outputting a balanced signal over XLR, and the line-ins on the Qu are balanced, so an XLR-TRS cable would be best. NB that the line-ins on the Qu have a 10dB pad c/f their associated XLR inputs, so you might have to run the output of the Manley hotter, or increase the trim on the line ins to compensate.

Once the signal is in the Qu, it can be passed out to your DAW via the USB-B socket. Setup>IO Patch>USB Audio to ensure that channel is being fed to the DAW (p75 of the manual.) I recommend using the “insert point” send option to keep as much of the Qu’s processing out of the DAW as possible.

Then it’s just a matter of the DAW seeing a multi-channel interface, and assigning a track to an input of that interface.

[cornelius78Participant]

All the mixers Qu series have 4x FX engines, each with their own dedicated stereo return. They can be used as send-returns, channel-returns, or as inserts (individually configurable.) However, while the Qu24 and Qu32 (and Pac\SB) have a dedicated send for each FX engine when you want to use them as send-returns, the Qu16 only has dedicated sends for the first 2x FX engines (the blue FX1 and FX2 buttons above the Mix1 button on the RHS of the console.) In order to use FX3 and FX4 in a send-return configuration, you have to use another bus (anything from mix1-10 that you’re not already using for monitors etc.) To configure this you’d press the FX button next to the screen, select the relevant FX engine, and press the “Back Panel” button. Here you can change where that FX engine is sourcing its input signal.

Now re using an external FX processor: you need to run the output of the bus into the input of the external FX, then the output of the FX engine back into a spare channel (or 2, the stereo channels are useful for this.) The output sockets on the rear of the Qu16 are fixed, so pick a spare bus (eg mix9+10,) run cables from those XLR outs on the rear of the console to your FX processor, then the outputs of the FX processor to a spare pair of channels. Now sending channels to that FX unit is just the same as building any other mix: press the blue mix9+10 button on the RHS of the console and push the faders up. PS, ensure the mix9+10 master is up, the return channel is up in LR, and you’ve got the right pre/post settings for the channels feeding mix9+10.

If you’re using a dsnake you have more flexibility with output patching, but the process is pretty much the same.

[cornelius78Participant]

I might be thinking of gld (or perhaps something non a&h entirely, it’s been a long day,) but iirc a scene is a file with a bunch a settings re faders, eqs etc, a show is a collection of scenes.

[cornelius78Participant]

Analyzing a short burst might help flatten the speakers, but unless there’s a longer analysis there won’t be time to analyze room acoustics. IME these days newer speakers (especially active ones) are reasonably flat, it’s more the room issues you have to watch out for, as as Andreas pointed out, the response measured will change around the room. Also (and unless you’re using a decent array,) most point-source systems don’t have a very wide field of HF coverage. If you measure in the wrong spot, the auto-eq algorithm could decide to add +15dB to the top end, and that could be bad for the tweeters and bad for people who are actually in the tweeters’ coverage. Also, if it did do a long analysis and found hole at 100Hz, how would the system know if that hole is there because of speakers’ frequency response curve, and therefore it should boost, or if that hole is there because of a room cancellation issue, and so boosting +15dB @100Hz won’t do anything, in fact it could damage the speaker? It could also be there because of an XO issue. A lot of the subs I’ve seen, even something as big as dual 18s, generally only have decent frequency response down to 30-35Hz. If you’re analyzing 20Hz-20kHz and your algorithm decides there’s not enough 20-35Hz it’ll boost that range, which the subs aren’t designed for. At best it’ll mean no change, because the subs will be protected from such freqs, but if there’s no protection (no speaker management, amps with no DSP, passive subs) you could end up distorting the low end, or even breaking something.

And peq in stead of geq all the way. Sweepable and adjustable Q: makes so much more sense than a geq (scalpel vs hatchet.)

[Author]

Posts