Subgroups for Qu-16

[Andreas]

What precision do you need??? Musically a mSec is nothing to care about. If you want to mix the loopback with the original signal, you’d need to delay the original in terms of samples not msec to avoid comb filtering. The specified 1.2mSec and delay adjustments in tenths of a msec are both way too coarse. And since we’re going through analog you’d also need to take into account the internal processing of the DAC and ADC as well…
If you’re worried on a sub-mSec level, just don’t make external loops. Otherwise just ensure the delayed signal never meets the original and you’re fine.
Just a question of interest: Did you ever do analog mixing and also worried about the delays within your sidechain FX?

[Anonymous]

We’ll need to measure the length of our input cables soon 😉

[Andreas]

I’m already checking for matched pair stereo cables to reconnect keyboards…
…next nightmare’s topic probably is to find a time-matched multicore… 😉

[Anonymous]

Maybe A&H could implement a negative delay option?

[Andreas]

I couldn’t resist to add some tech-talk hopefully not feeding any confusion regarding latency within the Qu. The observed values are pretty good and could be ignored for normal use of the desk!

Just for the sake of curiosity I just measured what we’re talking about (and since the specified 1.2mSec did not make much sense to me in terms of efficient sample processing), check out the picture in the attachment.
Setup was rather easy:
Mic at Ch1, Ch1 routed to Mix1 and Mix2, External loopback from Mix1 to Ch2, Ch2 routed to Mix2 as well.
Four tracks were recorded, Ch1 input, Mix1 send (to outgear), Ch2 input (loopback return) and Mix2 send (mixdown).
From this simple test, external route takes exactly 64 samples resp. 1.33mSec (64/48, not expressible in mSec at all..).
Interestingly there is an additional internal processing delay of 23 samples between Ch1 and Mix1 sends, if anyone ever need to record the same channel via direct out and mix/group, this must be taken into account inside the DAW (moving either track)….

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[Anonymous]

23… I was expecting 32…

[Andreas]

yes, 23 is irritating me also. But you can easily count dots (=samples) from left selection up to the cursor, its 23.

[Arno]

I want to be as accurate as possible. Is that a bad thing?
The way I look at it there is always going to be certain margin of error. If I do everything I can to be as accurate as possible, the smaller the deviation when everything gets added together.

I understand small discrepancies don’t cause concern in you fellas. And that’s great. But I want to be able to get as close to perfection as possible without having to nudge and move around once everything is set.

[Andreas]

Nothing against perfectionism, I’m in that club as well. But discussing about a msec more or less isn’t really worth it. There are much more critical topics to check when it comes to live sound (we’re still talking about monitoring, right?).

[Arno]

To be frank, I’m also thinking about mixing. For live such small numbers are perfectly fine, but when it comes to recordings and using this board for creative mixes even small amounts start to make me weary.
The 16 was marketed not only for live use, but for installation & studio as well. I am trying to determine if despite of what A&H claims this board can be useful in a small studio.
So far I’m starting to think that for a “full featured” board one has to step up to the Qu24 or 32. No, I’m not saying that the Qu 16 is a child’s toy. It’s a great ‘little’ board, but there are limitations that bum me out. I feel I should have stepped up and gotten something less… lmitating. Turns out I use subgroups more than I thought.
My only hope is A&H decides to provide is with a patch that would allow for stereo mixes to be internally route-able to the LR.

Thanks for reading my rant. I’m sorry if I sound too needy.

[Andreas]

But for recording this latency is absolutely irrelevant and normal for any type of digital recording, sorry. Even with the lowest possible ASIO buffer sizes you’ll add much more latency to the overall system. But this may completely eliminated from latency compensation. Either this is done automatically, depending on your host system and DAW or should be able to be adjusted manually.
Of course you need to verify and optionally adjust your system once to obtain zero latency (playback signal through monitors, record that with a microphone and a second track and peaks should match). But this all makes sense only, if your musicians are “synchronized” to headphones, otherwise each foot (30cm) distance to your monitors will add another mSec delay. Much more that we have from the Qu (or any other digital system).

[Arno]

Okay.

[Author]

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