[cornelius78Participant]

I also wish I was in the UK. I’ll be online most of today if you have specific questions you want to ask. The factory templates are a good starting point, and customising the desk in terms of fader layout, softkey and rotary function goes a long way to getting a workflow you’re comfortable with.

[cornelius78Participant]

But what if you’re using ipad#1 for foh, and someone else is using ipad#2 for monitors, and you both have different ideas about what you want on the custom layers?

[cornelius78Participant]

As above. By default the Qu-Drive will record ch1-16 on track1-16, and LR on track17+18: that’s the 18x channel limit of Qu-Drive. The channel>tracks are freely patchable though, it’s entirely possible for you to record ST1L and ST1R on track17+18 (of course at the expense of being able to record LR.)

If you plug a laptop in via the USB-B socket, that’s a whole extra interface (32×32 IIRC for the SB,) and you can freely patch whatever signals you want to record there, eg your 16x inputs, a couple of stereo channels for keys, a couple of FX returns and your main LR mix, all of which is separate to what’s going on with Qu-Drive.

[cornelius78Participant]

Most digital consoles are calibrated such that a meter reading of approx -18dBFS at the DAC will get +4dBu measured at the analogue output. Of course when working in dBFS, a meter reading of “0dBFS” means that the console has run out of bits: it can’t count any higher and you get digital clipping (ok, most consoles will show a clip light a dB or 2 before actually clipping, but you get the idea: a meter reading of “0” in dBFS is not the same a a reading of “0” in dBu.)

This caused confusion with people coming over from analogue, who were used to aiming to get signals\output levels hitting “0” on the meters, because their “0” was +4dBu, not 0dBFS, which could be closer to 22dBu, depending on the console.) If they tried to aim for 0dBFS, they’d clip a lot of the time.

A&H chose to address this by calibrating their meters on their digital consoles (it’s the case for Qu, GLD, iLive, and AFAIK dLive, I haven’t used one yet,) such that a meter reading of “0” was closer to -18dBFS, and still equated to +4dBu. This made things simpler for users coming over from analogue: they could aim for “0” on the meters before, and they can aim for “0” on the meters now, and be getting +4dBu in both cases.

PS, AFAIK the peak lights in the dlive are multi-point sensing, ie if there’s a clip anywhere in the signal chain they’ll illuminate, so it may not be the preamp that’s clipping, eg it could be the MUG in a compressor. That said, the dlive does have a ridiculously high internal bit-depth: it’s nigh on impossible to get them to clip in the digital domain without deliberately routing a feedback loop.) Maybe re-do your gain structure aiming to get “0” on the output meters: you might not need to run your preamps as hot.

[cornelius78Participant]

Re the qu-drive, yes, you can use the one drive for all those things. Once you format the drive for use you’ll end up with a folder structure on the drive that allows for all this. The only caveat is that you cannot record and playback simultaneously. If you need to record and playback simultaneously, you’ll have to hook up a laptop to the USB-b socket and use that for (at least) one of the operations.

[cornelius78Participant]

Setup>Control>Network>Unit name.

It’s on p68 of the manual.

[cornelius78Participant]

Yes. Doing this can be useful for keeping one on your foh mix, and the other on your fx. Or perhaps use one as a meter bridge. They have custom fader layouts too, which is handy.

Qu-pad requires ios7.1 or later. Will work on an old ipad2 (assuming the iPad is running 7.1 or later,) don’t think it’ll work on an old Ipad1.

Airport should be fine, once it’s configured.

[cornelius78Participant]

True the sd9 is an awesome console and it’s no surprise it can do it, but the X32 can also do it, and they’re leagues apart.

As dpdan said, 50Hz is just a starting point, it’s not set in stone. You can use a different freq if it sounds better. Using a sub harmonic synth as an insert could also work. I know it’s not on Qu, but it’s on gld.

[cornelius78Participant]

AFAIK no. Fattening a kick drum using the oscillator would require the gate to be able to key listen from a source other than “self,” (ideally with a post-fader tap) which the Qu series does not allow. You’d also have to be able to route the sig gen’s signal from the assigned bus to a channel (the buses don’t have gates,) which the Qu doesn’t allow internally: you’d have to go out of the desk and come back in.

[cornelius78Participant]

Not on a Qu, but still…

Walk up FOH+mons gig that involved band>speaker>band. Band did first set, all was fine, speaker started to make his way to the stage, I started muting band inputs for them to get off stage (GL2400, no mute groups, about 24 band inputs, so 24 mute buttons to press.) Speaker went fine, band got up again, I started un-muting inputs. Band played intro of song, all was good, went to start signing first verse: no lead vocal. Band looking at each other and me oddly, started playing intro again… I’d somehow missed un-muting the lead vox channel. Whoops.

Of course I’ve made the usual mistakes with double bussing, eqing like a mad man with the eq switched out, adjusting the wrong fader, routing channes>groups and not getting any level because the group master was down etc, plugging in insert cables backwards and wondering why I was getting no signal outboard (but to be fair, someone had labelled the cable backwards in that instance.)

[cornelius78Participant]

I think the author of the article is calling the geq fader flip function the 4th layer. I haven’t read of anyone using the geq fader flip as a method of controlling anything in a daw.

[cornelius78Participant]

The first part is about using the custom layer (accessed by pressing the two layer buttons simultaneously) as midi control. Instead of a layer of faders for the Qu’s normal channels, mixes and DCAs etc, you can assign midi controls to them and use them to control faders in a DAW.

The second part is about hitting the “GEQ” button in the superstrip whilst a mix master is selected. The faders change to represent the bands of the geq on that mix.

See p79 and p34 of the manual.

[cornelius78Participant]

If the signals were identical then yes, when summed you get approx +6dB (IIRC due to logs the number is actually +5.94db with a lot of decimal places.)

The more different the signals are from the L and R outputs of the keyboard the less coherently they’ll sum, resulting in less of a boost at the input of the M bus. In fact if they’re out of phase with each other you’ll notice a decrease in level when they’re summed. If they were so different that they were 180degress out of phase with each other they’d sum to nothing.

You might have to dig around a bit to find the exact numbers for the GLD, but often peak lights on consoles are configured to illuminate a couple of dB before anything clips (eg on an analogue console who’s outputs could do +26dBu before distorting, the peak light might come on at +22dBu, so despite the peak light, you weren’t actually clipping. Some digital mixers will show a clip light at -6dBFs, even though technically the user still has 6dB (1-bit) of headroom before actually clipping.

With only 2x vox and 1x gtr amp, no acoustic kit and no bass stack, are you sure you actually need to be running aux-fed subs? Sure if you had an acoustic kit a bass stack and you had LF from stage bleeding into all the 6x vocal mics and other drum mics it would make sense, or if your vocalists were wandering around in front of the subs it would make sense, but do you really need it for your setup?

[cornelius78Participant]

You could use an aux instead of the M bus, and it would give you more flexibility in terms of sending more of one channel to the subs (send level not fixed at unity.) Personally I don’t like that method, as it messes up the balance of the instrument when it’s routed to both subs and tops: to my mind a fixed send at unity makes more sense. To each his own. If you do decide to use LR+Aux, you can still tie LR+aux masters together on a DCA.

If L and R were the same signal (you were essentially running dual-mono) and you summed them to mono you’d get +6dB at the mono bus input. Depending on how close you were to 0dBFS, that would have caused the clip light to illuminate, even if the M bus fader was down.

[cornelius78Participant]

LR+M is what you want. Separate mixes for LR and M. LR for tops, M for subs. LRMsum simply sums L+R to a single bus for a mono output. LCR changes the panning to pan between left, centre and right, based on your channel assignments, if you’ve got that type of setup. If you’re just wanting to do aux fed subs, use LR+M. Channels can be assigned/unassigned from the M bus using the Mix buttons. By default everything is assigned (and the sends from channels are post-fader fixed at unity, essentially a subgroup,) which is probably why you thought it was just a copy of LR. Unassign the channels you don’t want in it, just leaving channels like kick, bass, keys, playback etc assigned. As for control, just assign LR and M to the same DCA. Have the “mains” DCA fader on the surface, you can bury the original LRM faders (or remove them from the surface completely) if you’re happy with their relative levels and assignments. You can put their Sel buttons on the softkeys if you need.

Edit: As for the peaking, the peak meters are multi-point sensing, ie they illuminate if the console senses a peak anywhere in the channel\bus’s signal chain, not necessarily just post-fader. The channel\bus the fader could be down at -inf, but if the preamp\input signal is peaking, or the gain on the eq, or the make-up gain on the compressor is peaking, the peak light will still illuminate.

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