Forum Replies Created

Viewing 15 posts - 31 through 45 (of 1,688 total)
  • Author
    Posts
  • #101597
    Profile photo of volounteer
    volounteer
    Participant

    Sounds like a DAW issue.

    could Ableton be allowing some extra headroom?

    Does it matter on the boot order or the initial Qu fader settings when you boot the DAW?
    Do you get different results with different trim settings?

    #101572
    Profile photo of volounteer
    volounteer
    Participant

    @salem874

    do nothing to the LR mix
    leave what is assigned there alone
    that is fine
    so just use it for the live service and now forget about it wrt streaming

    create a duplicate of that in mix78 as a starting point (just hit mix78 button)
    and then tweak it to suit the livestream

    I get confused about how AH PAFL means different things and when. I wish they would say pre and post and just mean those.
    you may or may not (I think I recall you would) need to use PAFL on mix78 when you listen on phones to check it
    do not think you need to do it otherwise. maybe one of the pro FOH guys will show up and clarify that.
    easy enough to play with that though just push pafl buttons and see what happens.

    I use OBS as a generic term since most churches actually do use OBS.
    It could be your DAW or some other streaming program. Maybe even to local radio/tv.
    Our video guys started with streamlabs but finally got smart and now they use OBS
    I recall there are a few very minor competitors but I would be afraid to use them as they might not survive in the long run
    and none of them seemed to have anything special to recommend choosing them. UK might be different.

    I do not do the assigning but I am sure you can send mix78 to any usb number you want but that would leave you with a signal that most people complain is too low at the PC. I have no idea if you can assign a mix to LR, but why ???? you already have LR on LR. most of us use the mix to feed an onstage monitor or other rooms.
    Also pretty sure that you cant assign LR to anything but the LR outputs that are hardwired to xlr connectors.

    Again please , please, do look at the block diagram and signal flow diagram and trace your signal path from in to out.
    that will help you understand a lot more about what is happening and how.

    always recall that you are doing two parallel mixes that are essentially the same but with minor tweaks to suit live or streaming
    focus on one or the other and do not conflate them at all. they are separate things for different usages.

    If you go to the bother to set up mix78 then just send it from the xlr outs labelled mix78 and into an audio interface where you can get plenty of gain into the PC to hand off to DAW OBS or some other program.

    unless you are going to mix in the PC there is no point in using usb to send a stereo pair. then you would send all the channels over usb so you could mix them in the PC. but why? you already did it on mix78 which is so much easier than adding another program in the livestream chain and adding more work to the video operators on that side of your service support.

    #101569
    Profile photo of volounteer
    volounteer
    Participant

    @salem874

    you do not assign the LR to mix78
    they are separate outputs with separate mixing for different purposes
    LR is to feed the amps live in the nave and mix78 feeds the PC with OBS for streaming

    The point is you do not send LR to the mix
    but you create a duplicate mix with tweaks to better fit OBS
    the sound and levels you want live are not what you want on the livestream
    live and streaming will need different mixes to be best for their purpose

    What you need to do is select mix78 button on right and then create the mix you want for the livestream.
    use the same faders as for LR
    they are now changing the mix not messing with LR

    In our case we have to add our audience mike to pick up the organ which is not in the LR
    otherwise the livestream would have no organ music
    we raise the piano mikes level for streaming as we dial it back live because it is already loud acoustically
    also we need to tweak the rf mikes for how loud the user is talking/singing. that might be done on LR and used for mix78
    and we custom set the preacherman’s mike level for his RF mike. that might also be done on LR and used for mix78
    and if there is a guest we adjust the level at the podium mike
    we had a standalone mike for the music director to use but that one went missing

    This set up should be done at sound check time.
    If you have the same people every week then you could leave it the same as was recalled in the scene that saved everything.

    Now our video guys are different from most people and they sometimes want more of this and less of that
    so when they ask we just push mix78 button and tweak the faders for whatever input they feel a need to change

    initially you need to push the mix button then use the faders to adjust the mix
    send the mix out the xlr pair for the mix to an audio interface and then into the PC and DAW/OBS
    they are already hardwired pre-‘mapped’ to those two outputs on the back

    groups are different you do not need them for this mix for livestream

    yes the documentation is meant for the engineers who built it not typical users
    you would make a lot of progress if you looked at the signal flow diagram and the block diagram
    and then traced the signal path from channel inputs on the left through everything including busses and out the mix78 on right

    I made a higher level summary view of the busses that showed where each can get signal from or send it to
    which I found useful to help my memory when I tried to do things like you are attempting

    #101567
    Profile photo of volounteer
    volounteer
    Participant

    @JohnOC

    you get 18dB headroom from the SQ
    you may not have recorded to the peak so get a few more dB – that could be your 20 value
    and you may get up to another 18dB from the PC/DAW

    but it is all digital and the DR is humongous as is the SNR
    so you can still raise the signal in the PC without problems with noise

    #101556
    Profile photo of volounteer
    volounteer
    Participant

    @Jon

    use audacity normalize function – that is more than adequate and easy to do for this purpose

    normalize to -18 dB while removing DC offset
    do any other editing tweaks fx yada yada that you want
    then normalize to -6dB for playback (you could go to -1 safely but some AD/DA chips have minor issues at their extreme limits)
    lower the gain on the SQ input or DAW to reach the dBSPL that you want to listen at, or raise it some if you should need more gain

    #101554
    Profile photo of volounteer
    volounteer
    Participant

    that is the way digital works

    SQ gives 18dB headroom to protect against clipping.
    Your DAW will give it another 18dB (maybe a little lesa)

    you need to normalize for peaks and remove the DC offset while amplifying the usb file to the level you want to use for playback by the PC

    #101551
    Profile photo of volounteer
    volounteer
    Participant

    @Europe

    Video might be a bit harder, but I have run two audio recorders and found their sync did not vary. That was long ago in analog.
    Now I would sync the start with the sound board and let it run.
    Then edit the sound length by tweaking the end of sound to also match the end of video.
    What is in the middle will be better than lip sync.

    If you do a lot of recordings then get the clock device when editing for sync becomes too much work.

    #101498
    Profile photo of volounteer
    volounteer
    Participant

    true
    my experience is with usa

    and toe-knee who started the thread is in the usa
    yet still his church may be one of the big megachurches where money is not a drawback to doing things as good as possible

    #101496
    Profile photo of volounteer
    volounteer
    Participant

    Don’t!

    Send mix78 to the presonus and avoid all the needless effort and potential problems.

    We looked at usb and the presonus and went with the audio i/f as being better.

    See messages here some folks trying the usb but complaining the levels are too low.
    Ours are plenty high thanks to the i/f so we have to lower the Qu levels we send.

    Make sure you have the right driver. Be sure you are sending to the output going to the presonus.
    Inputs should be as I recall on 1 & 2. Assuming you send stereo to presonus not just one channel.

    No output comes from it?? You trying to send back to the Qu ?? why ?
    If you mean no input coming in then check the driver and the other bizarro settings that need fixing in win 10.

    #101495
    Profile photo of volounteer
    volounteer
    Participant

    If it is a church then I suspect it would be low cost and functional, and not super expensive high end like possibly a Bway theatre. If it is stereo at all, I would expect two speakers on either side with perhaps a center fill if the LR are very wide.
    I do not know any churches that have stereo. But I guess there might be some of those somewhere.

    #101481
    Profile photo of volounteer
    volounteer
    Participant

    Again, tell us the big picture and ask general questions to get general help.
    Tell us the details of what you did and the problem you then have and get more specific help.

    One problem about online help may be the time zones. Many users here are in UK and EU or farther away from US
    and we have 3 main zones plus AK and Hi that are farther from the east coast.

    As I noted there are videos on things like routing.
    At least look at them then you will understand the answers better if you ask or do get zoom help.

    What I did was write down a detailed step by step process for using the board at our church.
    Then I do not worry about my memory. If I forget I look at the notes and quickly see what I need to do.
    After a while you wont need the notes but can feel ‘safer’ by just by having them.

    As to loss, my wife died almost 3 years ago. I try to stay busy doing things that she would have liked to see me do.
    Perhaps you should consider focusing on finishing up the mixing of the tracks you two worked together on and learn the Qu more slowly.

    If you know the DAW that should be easy to do. If you have to learn the DAW too then definitely put off learning the Qu for now.

    #101473
    Profile photo of volounteer
    volounteer
    Participant

    If you ask specific questions people will help you.
    Otherwise you may get lucky and get a PM from someone who will tutor you on zoom.

    Also if you know what you need to learn then there are plenty of tutorials on utoob. Could be faster than waiting for responses on a forum.
    And when all else fails read the fine manual. The signal flow and block diagram are very helpful reference to guide you.

    It would help us to help you if we knew exactly what you are doing. And what you need help with.

    If you have tracks on a DAW and are mixing them then you do not need the Qu16. Use the DAW.

    If you need to record tracks more tracks then ask specifically about what that problem is.
    What do you need scenes and routing for?
    Scenes are good for saving time by recalling settings you want to use again.
    If you could sign in then go to scenes tab and select the scene you want.
    Or save a new scene after you create it.

    As to routing, what do you want to route to where? It depends but is not that hard.

    As to outputs: Plug in the headphones select what you want to hear and turn up the volume knob.
    If that fails then tell us what else you were doing that might stop you from hearing.
    Usually it would be as simple as pushing PAFL, or using the faders on another level.

    Sorry for your loss. But he was lucky not to linger.
    My brother in law got the diagnosis of stage 3 but suffered for 8 more months.
    I want to go immediately not stay around in pain and be dependent on others for care.

    #101430
    Profile photo of volounteer
    volounteer
    Participant

    We tried both matrix and mix for our livestream. We settled on the mix78 stereo out.
    Adding the usb may seem to be easier. Making it work right could be another problem you dont need to fix.

    The matrix lets you do more things to the sound like having another mixer in the chain to the PC for streaming.
    Look at the block diagram and see the pros/cons versus a mix.

    As I noted, we use mix78 with one stereo pair to an audio interface (we use presonus) and there is plenty of gain that we have to lower it to keep the stream level from clipping. We also considered usb, but abandoned that as there are just too many people having too many problems with usb, and we do not want to fight problems we can avoid.

    AFAIK mix/matrix would work fine, but the A1 just found that using mix78 was easier to do when the livestream folks ask for changes, which their golden ears often do. We humor them even though the audience does not notice any of it outside of perhaps the lufs levels changing. We have asked them to use the QuYou ap and do it for themselves but they are stubborn and ignore everything we suggest they do. Well at least for many months as they did finally just go to real OBS which we had noted was THE standard for churches doing streaming. Maybe by 2022 they will change their mind on the ap too.

    In theory you can also put a DAW in front of OBS and raise the gain there even more, and then send the signal to obs and then on to the CDN. But still better to just just use the presonus or other audio i/f.

    #101421
    Profile photo of volounteer
    volounteer
    Participant

    Unless somebody is playing games with mike sensitivity ratings the R10 is not a problem. Nor is the SQ.
    Look at all the SM57s and 58s that are in use. They work just fine with the SQ.
    There are bigger questions to consider like response and polar pattern. Durability, and other factors too.

    from the manual at https://royerlabs.com/pdf/manuals/R-10manual.pdf
    Sensitivity: -54 dBv Ref 1 v/pa and
    Recommended Load Impedance: 700 Ohms (or greater) (from their web site at https://royerlabs.com/r-10/ )
    SQ Input Impedance: >5kΩ is just fine.

    And from Shure web site:
    The 1 Pascal the SM57 is using a 94 dB SPL input level. That is, 1 Pascal is equal to a sound pressure level of 94 dB.

    The SM58 spec that somebody said is using a 74 dB SPL input. That is, saying “0 dB = 1 v/ìbar” is equivalent to saying “74 dB SPL input”. Those two phrases are exactly the same.

    So, if you do the conversion, the SM57 and SM58 have the same input sensitivity.

    The last bit 1 Pa = 94dB SPL is simply stating the test criteria. That is, they are using the standard of 94dB SPL is equal to one pascal. This is the normal standard used these days by most manufacturers (an older standard used one pascal as 74dB SPL).

    There is more to a good mic than its sensitivity. We must also look at its frequency response, its tone, its maximum SPL without distortion, its directionality and its handling noise among other factors. It is also important to note that mic sensitivity is not necessarily telling us it is a good mic or not. It all depends on what the mic is being used for. Most folks wouldn’t want to use a Videomic as the vocal mic for a rock band singer, any more than we would want to use a SM58 for distance recording. But using them for their intended use is normally ideal.

    Much more on the R-10
    http://recordinghacks.com/microphones/Royer-Labs/R-10
    Being less sensitive, the R-10 will need more gain from the mic preamp, especially on quiet sources.
    (So how quiet is your intended source?)

    The wind protection of the 3-layer mesh also dampens proximity effect.

    Three new features … also make it more rugged for live use. Royer states in their Sweetwater video that some artists didn’t want to take their R-121s on the road because of durability (and therefore cost), so the R-10 was designed with this in mind.

    Got just over a grand to blow on a matched pair and the time to play with it?
    Then give it a try.

    #101418
    Profile photo of volounteer
    volounteer
    Participant

    Why not?
    Cost and effort.
    If the audience does not care why do it? It would be to make yourself happy thinking you made it ‘better’.

    Sp just how do you define ‘better’?
    Better is VERY subjective and hard to get everybody to agree on it.
    Assuming they can actually hear any difference at all.
    ABCX testing shows that ‘better’ result is usually not proven with a test audience.

    What is ‘fat’ sound?
    I would suspect you could get an effect that would do it as well and also cheaper easier.

    So it looks like it you care and are willing to spend the time and money then it should be tried.
    Or even if you just want to do it and can afford to try,
    otherwise reconsider what you will do if anything.

Viewing 15 posts - 31 through 45 (of 1,688 total)