Hello, community,
As the Qu-16 is my preferred desk for conference jobs because of its small footprint, I would highly appreciate the possibility of subgroups (at least one stereo group) with dedicated GEQ and PEQ for Headsets, Lapel Mics, etc. so I can have different EQing for eliminating feedbacks for only those microphone channels, without affecting the whole EQing of the desk, which is preferrably meant to eliminate room resonances, etc. As the Qu-16 does not provide any additional faders for subgroups, maybe this could be an option for the user defined layer. Another possibility couls be to enable at least one stereo mix to be routed to LR (similar to the fixed busses in the Yamaha consoles) and the channels to be set to 0dB fix and being routed off the LR master.
Yes, being able to route mixes to LR would be really nice.
It is easy to do a mix on one of the stereo auxouts, such as ⅞, or 9/10, and route it back in to ST1 or ST2.You could have two stereo submixes that way. Or go in one of the mono channels at line level.
Hi, I have a question on the subject.
Let’s say I were to route a Mix back into a Stereo Input. What kind of latency are we talking about? There is a set of DA/AD conversion the audio signal must go through, there has to be some sort of delay in signal. Answer in milliseconds would be much appreciated (vs a “should be very small” answer).
Another one is - is it possible that an Internal routing of a Mix to the LR be added in a future firmware update? Or is this a limitation that cannot be overcome via a software upgrade?
small single digit milliseconds (as reported elsewhere on this forum), it’s not a significant problem to patch out through the analogue domain.
Should be easy to work out by recording a signal through the desk, you’ll get two time synced wav files, so count the samples between the features.
According to the manual internal processing takes 1.2mSec (XLR in to XLR out), and since we’re adding a full processing path that should be the number of interest.
i.e one foot of air…
So then effectively it’s double that…?
After all the signal coming from the microphone goes throgh the board once, goes out and then come back in for second time into stereo in. Aren’t we looking at 2.4
Sure, total delay with two paths through the desk would be 2.4msec. One step towards the stage perfectly compensates that…
As long channel routes are completely separate (channel feeding either LR or the virtual subgroup) you’re fine, otherwise you’ll get a nice notch filter…
…tried that once using 8 loopbacks in parallel, sound got really spacey… 
And of course the sound your comparing with has gone through once, so the extra delay from the analogue loop is back down to 1.2ms and because imperial measurements are sane that’s 1.2feet (close enough). Unless you’re measuring your speaker positions that closely…
less than 3 feet!
Try it . you wont hear the delay although some people may?
I tried this subgrouping wiring system last couple of days and it works well.
it was covered here in this post Submixing - Qu Feature Suggestions - Allen & Heath Digital Community Forums
AN old trick when doing “LOUD” monitors was to delay the horn sections monitor feed (to themselves) so they could hear themselves.
And then they stood back from the mic as it was too loud!? sometimes you just cant win!
That helps guys, thanks for all your inputs!
I understand and agree with what you are saying. The reason I’m asking about total delay is in cases when a “grouped” signal is present in monitor wedges. As we know sometimes musicians can be very sensitive to delay in their own signal, and their baseline is their instrument.
Basically, my concern is not having performers being weirded out by the side effects of this work-around. That’s all. The rest ain’t a big deal.
If I want to group process something and bring it back in (parallel compressing f.e.) I guess the delay fiction at the preamps of each channel could come in handy. Suppose one could delay all non participating channels to avoid comb filtering.
Anyone in front of a Qu right now? Can you check if the Qu would let you dial in very precise delay values, like decimals and stuff?
Delaytime is adjustable in ~0.3ms steps. Interestingly sometimes a 0.2msec step in (0.0, 0.2, 0.5, 0.8, 1.1, 1.4mSec etc…), maybe times are based on some internal sample cluster processing (~14 Samples @48kHz).
Well, in this case the delay compensation will never be right on. It’ll be almost on…
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?
We’ll need to measure the length of our input cables soon
I’m already checking for matched pair stereo cables to reconnect keyboards…
…next nightmare’s topic probably is to find a time-matched multicore…
Maybe A&H could implement a negative delay option?
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)…
