Subgroups for Qu-16

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This topic contains 26 replies, has 7 voices, and was last updated by Profile photo of Arno Arno 4 years, 8 months ago.

Viewing 15 posts - 1 through 15 (of 27 total)
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  • #39371
    Profile photo of tonzwerg
    tonzwerg
    Participant

    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.

    #39372
    Profile photo of
    Anonymous

    Yes, being able to route mixes to LR would be really nice.

    #39378
    Profile photo of Lou
    Lou
    Participant

    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.

    #45193
    Profile photo of Arno
    Arno
    Participant

    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?

    #45195
    Profile photo of [XAP]Bob
    [XAP]Bob
    Participant

    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.

    #45198
    Profile photo of Andreas
    Andreas
    Moderator

    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.

    #45200
    Profile photo of [XAP]Bob
    [XAP]Bob
    Participant

    i.e one foot of air….

    #45201
    Profile photo of Arno
    Arno
    Participant

    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

    #45202
    Profile photo of Andreas
    Andreas
    Moderator

    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… 😉

    #45203
    Profile photo of [XAP]Bob
    [XAP]Bob
    Participant

    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…

    #45204
    Profile photo of
    Anonymous

    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 http://community.allen-heath.com/forums/topic/submixing

    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!

    #45206
    Profile photo of Arno
    Arno
    Participant

    That helps guys, thanks for all your inputs!

    #45210
    Profile photo of Arno
    Arno
    Participant

    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?

    #45212
    Profile photo of Andreas
    Andreas
    Moderator

    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).

    #45266
    Profile photo of Arno
    Arno
    Participant

    Well, in this case the delay compensation will never be right on. It’ll be almost on..

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