Insufficient gain

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This topic contains 20 replies, has 5 voices, and was last updated by Profile photo of volounteer volounteer 1 month, 3 weeks ago.

Viewing 15 posts - 1 through 15 (of 21 total)
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  • #96391
    Profile photo of volounteer
    volounteer
    Participant

    when we only have one mike input
    and set it low enough so that it does not clip
    we can barely see any output signal on the LR lights in upper right corner

    the power amp can give us the SPL level we want
    but it would be better if AH ADDED AN AMPLIFIER IN THE LR OUTPUT STRIP
    so we could normalize the output to 0dBFS and then reset our amplifier to the SPL level we want

    we could then control the LR sound levels using the LR fader
    as it is the sound is so low that we must keep the fader way up

    if we had 32 channels (or even only 8) feeding the LR then they would add up to a louder signal
    and we could lower the fader if needed to control the output level

    looks like AH cut corners expecting most users to be bands sending many mikes to the LR all the time
    which overlooks the small users and those who are interested in amplifying a voice not a choir

    PLEASE ADD AN AMPLIFIER TO THE START OF THE LR OUTPUT SO WE CAN ADJUST THE SOUND LEVEL UP ENOUGH
    and when needed can lower it via the LR fader , which for us now is essentially useless

    #96394
    Profile photo of SteffenR
    SteffenR
    Participant

    we already told you that you need the right gain staging

    #96398
    Profile photo of Mike C
    Mike C
    Participant

    If you really feel the need to add make up gain to the output compressor/limiter.

    That said I have NEVER even thought I need something like and I do sound for some events
    that are very quite with only a couple mics.

    #96399
    Profile photo of volounteer
    volounteer
    Participant

    SteffenR

    I know gain staging.
    And I am trying to get them to do it right, now that the acoustic fixes are in and we are redoing the Qu set up.

    What I am finding is that there is insufficient gain in the Qu to do it right.
    In the past I would set preamps as loud as possible without any clipping.
    Then set the output so that 0dB would give the desired loudness when we adjust the power amps.
    Finally I would link the channel strip to the output via an amplifier that would ensure we got 0dB out when the channel strip was at max signal.

    A&H saw fit to omit the linking amplifier hoping we would be using enough channels to add up to 0dBFS at the output.
    And if more we could lower it. My problem is that I need to raise it.

    #96400
    Profile photo of volounteer
    volounteer
    Participant

    @mikec

    I considered make up gain. I will need to do some computations to see if it could be enough.
    And then ensure we dont run out of range and start feedback going.

    I am hopeful we might succeed as we are going to reEQ the input mikes after we do one final ringing out the room.

    Our problem is that some speakers, including the main pastor, have way too wide a DR.

    The pastor goes up slowly over the sermon so we could ride the faders on him.
    But too many others go from whisper to shout for some reason.
    Perhaps they think it is theatrical or something, but it really causes us problems.

    And it makes the livestream audio way too low to hear as they cannot allow it to get too high or it distorts before transmission. We may need to add the compressor in the SLOBS software as well as do more on our end with the Qu.

    #96402
    Profile photo of Mike C
    Mike C
    Participant

    WOW!!!! Once again you are blaming the hardware for problems
    that no one else seems to have!!!!!!!!!!!!!!!!!!!!!

    #96403
    Profile photo of volounteer
    volounteer
    Participant

    @mikec

    ***comment removed***

    If you have a solution for the super wide DR of our speakers who I have no control over pls let us know;
    complicated by the havc noise level which I also have zero control over.

    If we had gain in the groups and LR and matrix we could probably get the compression to work well with NY style parallel compression.

    But we really need upwards compression not just downwards compression.
    Trivial to do in software if I had a way to program the Qu.

    #96409
    Profile photo of KeithJ A&H
    KeithJ A&H
    Moderator

    @volounteer – There is no cutting corners, there are no standard/expected gain stages that have been omitted here, and your suggestion that any part of mixer development is ‘trivial’ displays a lack of awareness and just comes across as rude.

    The fader of any channel or mix controls a gain stage, so there’s already one there, it’s just not referred to as gain to avoid confusion.
    There is also usually gain/attenuation at the amp and as you know, 0dB on the mix meter equates to a line level +4dBu at the output to the professional line level inputs of other equipment.

    You definitely don’t want the signal hanging around near 0dBFS at any point (internally or as an output… +22dBu!) – where’s your headroom?!

    There are up to 4 compression stages from input to output for you to achieve reduction of the dynamic range.
    But rather than trying to boost the quieter sections, why is it not possible to simply increase the gain at the amp then attenuate louder sections at the mixer?

    Cheers,
    Keith.

    #96414
    Profile photo of MarkPAman
    MarkPAman
    Participant

    @volounteer

    What sort of mic do you use for your pastor, and the others. How far from it are they when speaking, and does the distance change much while they’re speaking?

    #96415
    Profile photo of volounteer
    volounteer
    Participant

    @markpaman

    It is a shure wireless. We got a special mike for the transmitter from shure as the standard one was omni and tended to more feedback issues. The new mike works fine. Feedbacks issues are gone; although acoustic panels did help more but the mike had fixed it first.

    #96416
    Profile photo of SteffenR
    SteffenR
    Participant

    It is a shure wireless. We got a special mike for the transmitter from shure as the standard one was omni and tended to more feedback issues. The new mike works fine. Feedbacks issues are gone; although acoustic panels did help more but the mike had fixed it first.

    well… what Mike said many times before

    #96417
    Profile photo of volounteer
    volounteer
    Participant

    @KeithJ A&H

    If you did not cut corners then we have different standards for a proper design. Remember I started with analog in the 60s and another amplifier in the gainstaging was normal. I do know digital quite well but from DAW use. The Qu is the first digital mixer I had to use other than a couple of very small ones in my home which were much simpler.

    So I expect a gain stage between the channel strip and the LR output so I could adjust LR to hit 0dBFS when the channel strip has peaked. The average signal would of course be lower. And this lets us see the room SPL using the Qu leds well enough to keep different operators providing similar loudness. And could lower the LR fader if it did get too hot by easily observing the yellow leds.

    Faders remove gain. I need gain added. Is there really gain somewhere that the block diagram does not show?

    Yes I know +4dBu = 0dBFS and that is what we feed into the power amp for setting our max room SPL level at a safe value.
    We are not some theatre or rock band that likes 100+ average:) I am trying for 85dB SPL max but would settle for 90. And of course if somebody sent a higher signal it would go higher although not much when we use the final limiter on LR.

    I see no gain at LR except to raise the fader to max which could be a problem when a channel strip is low when a different orator is talking.

    Looking at the block diagram I see no way to increase the signal going to the LR out.

    No I do not want the signal hanging at 0dBFS but I do want to reach that on peaks while keeping the average signal between -20 and -10 dBFS (65 -75 dB SPL) which is needed for intelligibility at the HVAC level BG noise levels.
    I note again that we have zero control over the settings on the HVAC and can not turn it off as some suggest.
    And our goal is maximum intelligibility for a mostly older audience not to make the music sound different somehow.

    Now, we can reach that level now but the output lights toggle between signal detect and the minimum value. On extreme peaks the leds might go to -12 which is the loudest I have ever seen and is rare. I suspect it may have gone higher but the leds dont light fast enough to see it.

    /You said/
    There are up to 4 compression stages from input to output for you to achieve reduction of the dynamic range.
    But rather than trying to boost the quieter sections, why is it not possible to simply increase the gain at the amp then attenuate louder sections at the mixer?

    I am considering using the compression to add gain. We seem to have feedback problems if we add enough in the fx and also we tend to have clipping if we do enough anywhere in the input channel strip. Remember we are trying to fix orators with extremely wide DR. And I note again we have no control over these people.

    We already have the gain up in the power amp and do attenuate the signal in the mixer which is the problem I would like to change so we can better see the dBFS levels and correlate them to the final SPL max we want.

    #96418
    Profile photo of MarkPAman
    MarkPAman
    Participant

    Sorry – Post removed.

    There’s no point. 🙁

    #96419
    Profile photo of KeithJ A&H
    KeithJ A&H
    Moderator

    @volounteer – I can’t speak for the 60’s, but the basic gain stages in our consoles have been pretty much the same for 50 years.

    ‘Gain staging’ in a mixer relates to balancing up the signal level throughout the console, a ‘gain stage’ is used to refer to something that is only concerned with altering level.
    When you move a fader, you are adjusting a gain stage. On the Qu, fader gain goes down to -80dB(ish) before -inf, and up to +10dB.
    When you use a DCA you are also adjusting a gain stage.

    Send a test signal to an input, set the preamp so the meter shows 0dB at the preamp, set all other gain stages to 0dB (PEQ flat, no make-up gain in the compressor, fader at 0dB) and assign to a mix. Set that mix master fader to 0dB also and you will have 0dB on the output meter and +4dBu at the output.

    Yes I know +4dBu = 0dBFS

    except +4dBu != 0dBFS
    dBFS is also a measurement of digital signals, so as you are using analogue inputs and outputs, it plays no part in this discussion. In fact – that’s exactly the reason the meters on our digital consoles work like the analogue ones, so you don’t have to think about dBFS or digital clipping.
    In Qu: 0dB on meters = -18dBFS = +4dBu on the output socket.

    Relating dBFS to SPL is not possible without providing all the information about your system and venue.

    However rude you have been, I still don’t want this to come across as arrogant or patronising in any way – but as others have suggested, your posts suggest you would benefit from getting a reputable professional in to sort everything out. They will be able to set everything up as best as it possibly can be set up, which will then give you a base level, allowing you to see where you could improve things and where equipment (or mixer features) could help.

    At the moment, you’re talking about higher end or more niche concepts one minute and then questioning audio engineering fundamentals the next.
    Unfortunately this makes it very difficult to take forward or even consider any of your suggestions.

    Thanks,
    Keith.

    #96422
    Profile photo of volounteer
    volounteer
    Participant

    @KeithJ A&H

    I really hope that you are using all digital inside between the ADC and the final DAC.
    And I know you allow 18dB headroom. That is the value I always set for my digital work with DAWs.
    So, Congratulations! That is the correct value in spite of all the folks who think they can get by with much less.
    Or archaic analog folks who are afraid of ‘noise’ by using a large headroom but do not fear clipping.

    Relating dBFS to SPL is quite easy.
    We set a steady signal of 0dBFS on the LR and then move the power amp until we measure the 85dB SPL in the room.
    Then when the green leds show 0 we know we are at our maximum.
    And when they bounce between -20 and -10 we know we are in the good range for optimum intelligibility of speech.

    I apologise if you think I have been rude. I never intended to be rude to you. You have always been polite and helpful.
    What I do see, from my perspective, is that a couple of others have been way beyond rude in their replies to me.
    I will admit that when attacked by bullies I will stand up for myself and might reply in kind to *them*.

    We are having your dealer come in once more to tweak things.
    They have made significant progress, but our MD and A1 only seem to address one issue at a time.
    I was hoping to make this the last time we paid people to fix things that should never have been wrong.

    To be fair we all inherited the set up and the MD was not a digital person, and not really an acoustic one either although he has excellent ears for music and how it sounds, when the dealer first replaced our old large analogue AH mixer with the somewhat smaller digital one, so he (MD) did not insist on everything being set. To be fair with the acoustic issues with a reverby echo, which the dealer has now fixed with special panels, we could not have done much useful back then, but now we need to do it all right and make it final.

    We could live without seeing the SPL on the Qu leds. We often measure it by hand now, but that is just a one time value when we really need a LUFS readout on the Qu. LUFS meter on the mix would be a plus so we can feed a proper signal to the video team doing the livestream with our sound mix.

    I appreciate your assistance.
    I would hope you could grok both high end and other concepts that are lower at the same time.
    I am not questioning audio engineering fundamentals just your choice to ignore a basic one that I always had available in the past, admittedly now a somewhat far past to adjust the channel strip to the output with another amplifier.

    It really seems y’all find it too easy to add gain by shifting bits that you over look the need for enough gain.
    I wonder if it is because most users do not use one channel with someone talking and most of the time it is 8++ channels with music that would add up to a humongous signal.

    I will have to look at DCAs again and better understand them. I do know digital (far better than most folks as I had that in grad school in the 60s) and DAWs but the AH mixer is far different than the analog ones so there are many new concepts , almost too many to do until needed, to learn. I admit that since they dont seem to exist except as a concept that I did not make them a priority.

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