[cornelius78Participant]

Depends on the console.

A Qu16 is limited to 16x mono +3x stereo channels. Sources for those 16x mono +3x stereo channels can come from your choice of local preamps, Qu drive or the USB interface, but regardless of where they come from, the console can only process 16+3 at any one time. By default they come from the local preamps. Choosing to have ch5 source its signal from a USB input comes at the expense of ch5 sourcing its signal from the local preamp#5, ie the mic plugged into preamp#5 is no longer part of the mix: you’re still limited to 16+3. Adding a digital snake (not sure where you’re getting 18 from, it’s either 24 with the AR2412, 16 with the AB16, or 8 with the AR84) just adds another location (remote preamps connected via ethernet) from which the 16+3 can source their signals. It doesn’t add a whole lot of power to the console to be able to process an EXTRA 16 (or 18?) signals. In theory you could have 16(local preamps) + 3(local line ins) + 18(qu drive returns) + 24(USB returns) + 24(AR2412) + 16(AB16) = 104 input signals connect to the Qu16 at once… however you’ll only be able to mix 16+3 of them at a time.

In a similar way, the Qu24 is limited to 24+3, but it’s up to you where you source those 24+3 signals: local preamps, remote preamps, Qu-drive or USB interface. Whatever combination you choose, the console can only process 24+3 of them at a time. Qu32 is the same but limited to 32+3.

Where it gets interesting is the QuPac and QuSB options. These consoles have the DSP for 32+3 (like the Qu32,) however unlike the Qu32 they don’t have 32x local preamps. In this way adding a stagebox adds to the console’s input pool like in every other scenario, but also seemingly “unlocks” an extra 16 channels of processing. In fact these extra 16 channels were there to start with, they just weren’t being fed with preamps. It’s important to think of “inputs”\”sources” and “channels” as separate things.

[cornelius78Participant]

I think it was mentioned in the webinar that SQ would re-sample the incoming audio stream coming in from AR\AB stageboxes (and 48kHz clocked exp card) up to 96kHz to allow the SQ console to always operate at its native 96kHz. Then, if audio is leaving the console digitally (as opposed to out of the console’s own analogue XLR sockets) to a stagebox (or exp card) that can only run at 48kHz (eg AR\AB as opposed to a DX\DM rack runing @ 96kHz) the console will down-sample the outgoing audio stream to 48kHz before sending it down to the stagebox. The stagebox then does its normal DAC operation on that 48kHZ stream and outputs the (now analogue) signal via its own XLR sockets.

[cornelius78Participant]

On the overview page the XCVI core runs “variable bit-depth for ultimate precision and noise performance,…” I suppose that a variable bit-rate could allow for some more separation between noise floor and useful content. TBH I don’t know enough about it.. except to say that people I’ve spoken to who have used dLive are impressed at how high they can crank things and it still sounds good, without needing to resort to eq\compression.

I read that all internal processing is 96kHz, and each “accumulator” (output) bus is 96-bits, so that’s a lot of headroom for summing signals without clipping internally.

Then if outputting (digitally) to something running a lower sample rate\bit-depth, (eg AR\AB stageboxes with the SLink socket running in dSnake mode, or a Dante expansion card clocked lower to interface with other (older) Dante hardware that can only run at 48kHz,) the console will re-sample the digital output to the appropriate sample-rate\bit-depth for that digital connection. Of course if using DX\gigaAce, (or Dante @96kHz) no need for re-sampling the digital output.

[cornelius78Participant]

That’s my understanding too. From what I’ve read users who register their console and download the v1.1 firmware will receive a DEEP preamp (guessing a tube\valve emulation?) model for free. Being a DEEP plugin, it can be inserted across as many channels as you want without affecting the console’s latency, so everything remains coherant. This is a big deal compared to some other consoles that allow for certain processing emulations to be inserted across channels, but are limited in that doing burns the console’s FX racks, and it messes with the latency, and coherancy is lost.

I imagine that A&H are doing this to give users who’ve not experienced dLive a (free) introduction to what DEEP processing models can do for their sound. I imagine that A&H will then gauge demand and set up individual\packages of preamp\compressor\geqs etc ported over from dLive.*

*All just conjecture on my part.

[cornelius78Participant]

8x DCAs, 8x mute groups from what I’ve read (more than Qu, not as much as GLD.)

AFAIK all 6 layers are customizable with a drag-n-drop interface similar to GLD.

[cornelius78Participant]

Technically neither. AFAIK you build your mix in 96k on the console, then if the SLink socket is operating in dSnake mode, it’ll downsample the outgoing audio stream down to 48kHz before it gets sent to the AB168, so when it hits the AB168 it’s already at 48kHz, the AB168 will do DAC on that 48KHz signal and you’ll feed your PA the analogue signal.

[cornelius78Participant]

As above. If it’s the channels feeding FX2, they’ll still feed it when the group is muted, because the channels themselves remain un-muted. You can either run the returns into the group (and un-assign them from LR) so although the drum channels still feed the FX engine, the returns get muted when you mute the group, so you don’t hear the returns in LR, or you can forego the group completely and just use a DCA for those 6 drum mics. You could also explore the option of using FX2 inserted across the group, but you lose some control of how much verb is added to each drum: it’ll be proportional to how much each drum is up in the group.

[cornelius78Participant]

Setup>Audio>PAFL Additive mode: On

PS, if you turn this on, it’s probably worth adding a PAFL clear function to one of the softkeys on the surface (Setup>Control>Softkeys Select a softtkey, Function:PAFL clear) so you don’t have to go hunting through layers to find that one channel\bus that’s still PAFLed.

PPS, if you’re wanting to listen to a whole lot of BVs together and have their levels and FOH panning in the cans (if applicable), Setup>Audio>PAFL Input AFL: On. Of course the downside to this is that the AFL setting applies to all your inputs, which means you can only PAFL inputs that have their LR fader up.

[cornelius78Participant]

Extrapolating what I’ve heard about how SLink works when running dSnake into 48KHz stageboxes, the console will do SRC on the incoming Dante streams to bring them up to 96k before they hit the rest of the console’s processing. The console itself stays at 96KHz. Similarly, the console will do SRC on the outgoing Dante stream down to 48\44.1KHz (if that’s what the Dante card is clocked at.)

[cornelius78Participant]

You can mix and match from the expansion card (Dante\WSG) and the USB 32×32 interface at the same time (and expansion card and SQ-Drive.) What you can’t mix and match from is the USB 32×32 and the SQ-drive at the same time. Just looking at the block diagram for that, haven’t read anything official.

[cornelius78Participant]

Not stored on the SQ itself. However you can plug a USB stick into the SQ-Drive and it can play back mono\stereo wav files 44.1, 48, and 96kHz at 16 or 24-bit. It can also do a 16-channel multitrack playback of 96kHz wav files @ 24-bit. There’s also the 32×32 channel USB\expansion card if you bring a laptop with all the tracks on it, and of course if you want to go analogue there’s a 3.5mm stereo TRS socket.

[cornelius78Participant]

The vents in the front and rear mentioned by Keith are shown on the sq6 technical datasheet. The “gaps on the sides” are just voids to allow for air intake (I assume the fan is sucking rather than blowing.) It’s not mentioned to “keep clear for airflow” or similar, but you’d hope a case manufacturer would question how the fan is going to work if they enclosed it by covering those voids. Biggest problem I see is that due to the positioning of the rear vents, a dogbox might not sit flush with the top edge if the mixer.

[cornelius78Participant]

Originally saw a post on PSW, seems to have been taken from p5 of this on A&H’s own site…. and now reading through the rest of that document, (instead of just the screen grab of the block diagram on PSW,) some of my questions are being answered. Still want a proper manual though…

[cornelius78Participant]

Edit: Ignore the last question re mixing USB sources. Just saw a block diagram indicating you can’t. Annoying if you’ve got a DAW running backing tracks, and have bgm for breaks on a USB stick, but I suppose you wouldn’t be using them both at the same time anyway. And if you were to connect to the DAW via Dante instead of the 32×32 interface, it becomes a moot point.

[cornelius78Participant]

8x FX engines, 4x of them with their own dedicated send buses, the remaining 4x FX engines would need to be used as inserts or channel>returns, or you’d need to sacrifice the other stereo buses to feed them ala Qu16.

Given that there are only 12x physical buttons on the console surface that provide access to the mixes, I have a feeling it’ll be fixed as 12x stereo mixes (unless you can do it with the softkeys, but there are only 8x of those on an SQ5, so not enough to access 24x mono mixes, or if there was a “shift+mix” button to press to access the odd\even side.)

Hoping some manuals are available soon.

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