Audio drops/clips during Live stream. NEED HELP

mrorlowski

New Member
I am a worship director at my church and we have been having an issue since upgrading our media that we experience this clipping or audio drop during the live streaming. We recently just upgraded everything to a 2022 Mac Studio (M1). We utilize ProPresenter 7 for slides and have it integrated with OBS to show the slides on the screen for Worship and the sermon. Audio is coming directly from a Behringer X32 (upgraded to latest firmware) out via USB and into the Mac Studio. We have the Audio input configured in OBS to recognize the USB audio, and have disabled all other Audio signals/inputs.
I have configured the advanced audio settings to increase the gain by 19db, (still not peaking in the red on the livestream) with a compressor and limiter
Compressor = Ratio=4.00:1, Threshhold=-5.00 db, Attack=100 ms, Release=94 ms, Output Gain=2.00 db, Sidechain/Ducking Source=None
Limiter = Threshold= -2.00 db, Release=60 ms

For the Stream settings, i have the Bitrate set to 4500, and the audio bitrate to 256.

We are using Ethernet (Not WiFi) and using Ookla speed test we are getting approximately 350 Mbps up and down. We Use Restream.io to stream to Facebook and YouTube, and the audio issue happens on both platforms. Please see the examples below, and the audio clipping happens every few seconds within the stream.

I have tried everything I can think of and nothing seems to help. Any input would be GREATLY appreciated.
 
I am a worship director at my church and we have been having an issue since upgrading our media that we experience this clipping or audio drop during the live streaming. We recently just upgraded everything to a 2022 Mac Studio (M1).
The problem is that the M1 may be great for some stuff, but isn't necessarily great at everything and many free, open-source software packages (including the ones OBS Studio is dependent on) are still coming to terms with whole new CPU platform... and Apple's whole lovely support for working outside their walled garden....
so that M1 may, or may not, have been an actual upgrade from what you had before

I have configured the advanced audio settings to increase the gain by 19db, (still not peaking in the red on the livestream)
that seems like a LOT of gain to me.. and many livestream audience members tend to use mobile device with terrible speakers, I like seeing a little into the Red during our services. Not sure of your specific scenario, but can you increase 'volume' on the mixer for the mix going to the Mac? I run a front-of-house mix, and a separate sub-mix (on mixer) for livestream (no need for in-house amplification of Pipe Organ, Choir, piano, etc but those mics need for livestream audience)

How is CPU utilization in System Monitor while this is going on?

For the Stream settings, i have the Bitrate set to 4500, and the audio bitrate to 256.
Why streaming at such a low bitrate? Are you streaming at 720p (in which case that number is fine)... but at 1080p I'd recommend closer to 7000kbps or maybe even higher... and let reastream cut down bitrate to Facebook if need be... but should be fine up to 10,000, iirc

Is the compressor local to OBS Studio (in which case, can be CPU intensive) or part of sub-mix on the Behringer (which is where I'd recommend doing Compression, and other audio filtering/effects.. assuming your mixer supports such)
@AaronD might have some more feedback on audio specific settings and approach.

We are using Ethernet (Not WiFi) and using Ookla speed test we are getting approximately 350 Mbps up and down. We Use Restream.io to stream to Facebook and YouTube, and the audio issue happens on both platforms.

Are you Recording locally?
- If yes, is the audio issue in the Recording as well?
Also, if you have the resources, I strongly recommend local recording as CDNs (YouTube, Facebook, etc) all HIGHLY compress their video. For marketing, snippets, or keepsakes (baptisms, weddings, etc), using the higher quality locally recorded video is strongly preferable
 
Is the compressor local to OBS Studio (in which case, can be CPU intensive) or part of sub-mix on the Behringer (which is where I'd recommend doing Compression, and other audio filtering/effects.. assuming your mixer supports such)
@AaronD might have some more feedback on audio specific settings and approach.
I agree. OBS should do as little as possible with audio. Dumb straight-wire pass-through. Do all of your audio processing in one place, elsewhere.

I run our stream mix as 5 post-fade aux busses in our X32:
  • Stereo main mix for the stream (not the FOH mix)
  • Stereo ambience mix
  • Mono speaking mix
The speaking mix gets a de-esser, inserted on the bus master. That's a more efficient use of resources than a parallel FX loop, as I can use the other channel of the same FX slot for the one singer who needs it too.

Those 5 busses feed 3 matrices:
  • Stereo stream (all 5)
  • Mono lobby (no ambience)
The stream matrix has a stereo LA-2A emulation inserted on it ("Leisure Compressor"), which works hard, and is followed by the standard channel compressor that is set as a very-soft-knee limiter. Ratio all the way up, instant attack, hold and release by ear, threshold to make it work a bit too ("ride the soft knee"), and makeup gain to work with the master fader that follows, so that the matrix master at +10dB (top of travel, against the end stop) produces *exactly* full-scale as seen by OBS. So it's impossible for the FOH operator to casually over-drive it.

Similar approach with the lobby, but with only the standard bus compressor, that is set even *more* aggressively, and no insert. I think the makeup gain is pegged on that one, and the threshold is set to produce the desired level from that, almost regardless of the input level. (small speakers, big space...)

Because we're using the aux outputs to feed a consumer line input, that full-scale in OBS is actually about -11dBFS or so on the console. That's a potential pitfall for the unwary. Our 32-track USB card is reserved for the sound guy's DAW. Virtual soundchecks, mixing practice at home, etc.

With our "typical" music, OBS's average is right on the border between green and yellow, and the peaks are on the border between yellow and red or slightly higher. Every once in a while, OBS fills the meter with red to tell me it's clipping, but it's actually not. I'm just using the top couple of values legitimately, and I know that because I've mathematically designed and checked the gain structure from the last bus compressor (the soft-knee limiter) all the way through to the line input with its lower full-scale.
 
I agree. OBS should do as little as possible with audio. Dumb straight-wire pass-through. Do all of your audio processing in one place, elsewhere.

I run our stream mix as 5 post-fade aux busses in our X32:
  • Stereo main mix for the stream (not the FOH mix)
  • Stereo ambience mix
  • Mono speaking mix
The speaking mix gets a de-esser, inserted on the bus master. That's a more efficient use of resources than a parallel FX loop, as I can use the other channel of the same FX slot for the one singer who needs it too.

Those 5 busses feed 3 matrices:
  • Stereo stream (all 5)
  • Mono lobby (no ambience)
The stream matrix has a stereo LA-2A emulation inserted on it ("Leisure Compressor"), which works hard, and is followed by the standard channel compressor that is set as a very-soft-knee limiter. Ratio all the way up, instant attack, hold and release by ear, threshold to make it work a bit too ("ride the soft knee"), and makeup gain to work with the master fader that follows, so that the matrix master at +10dB (top of travel, against the end stop) produces *exactly* full-scale as seen by OBS. So it's impossible for the FOH operator to casually over-drive it.

Similar approach with the lobby, but with only the standard bus compressor, that is set even *more* aggressively, and no insert. I think the makeup gain is pegged on that one, and the threshold is set to produce the desired level from that, almost regardless of the input level. (small speakers, big space...)

Because we're using the aux outputs to feed a consumer line input, that full-scale in OBS is actually about -11dBFS or so on the console. That's a potential pitfall for the unwary. Our 32-track USB card is reserved for the sound guy's DAW. Virtual soundchecks, mixing practice at home, etc.

With our "typical" music, OBS's average is right on the border between green and yellow, and the peaks are on the border between yellow and red or slightly higher. Every once in a while, OBS fills the meter with red to tell me it's clipping, but it's actually not. I'm just using the top couple of values legitimately, and I know that because I've mathematically designed and checked the gain structure from the last bus compressor (the soft-knee limiter) all the way through to the line input with its lower full-scale.
This is really awesome information, I just wish I understood all of it. I lead the worship team and I know how to do some very basic stuff with the Media, but I am not an audio engineer nor do I understand the Behringer X32 enough to assign busses or reconfigure the usb audio output.
I agree that I would love for the audio to be handled in the board and simply passed to OBS with little to no additional processing. The audio signal coming out is so low I have to compensate in OBS for anything to be audible in the stream.
I think I am going to try and hire a professional and then share these notes with him.
 
I know how to do some very basic stuff with the Media, but I am not an audio engineer nor do I understand the Behringer X32 enough...
It's not really that hard, once you sit down and overcome the intimidation. The "sea of knobs" on an analog board, or the "spaceship control panel" on a digital board, is what gets people more than the actual function.

It's not a monolithic block that you have to suddenly be intimate with all at once. It's a collection of parts, each of which does one specific thing, and you put them together to make an overall function that *you* design. Even with a digital board, that does a lot in one box, it's still that way.

Learn what a parametric EQ does, all by itself. Now you understand *all* of the channel and bus EQ's.

Learn what a compressor does, all by itself. Now you understand *all* of those too.

Learn what a mix or a bus really is. Now you understand not only the main mix but also all of the auxiliary mixes. And with just a one-line addition - it's a mix of mixes - you also have the matrices.

Learn to follow a signal through each bit of processing in turn, and add up their functions in the order that the signal encounters them. Then the concept of different tap points becomes obvious, like to feed each of the mix busses pre-fade or post-fade or pre-EQ or whatever.

Etc.

Take each concept separately, wrap your mind around each one separately, and maybe even play with them on an unused channel with a representative raw recording running through it. Once you have that, you can put them together, and now you know the whole thing!

YouTube is WONDERFUL for this! Lots of people explain the same things in different ways with different examples. Look at a bunch of them, see what "clicks" for you, and keep looking at multiples of those. Some know what they're talking about more than others, and there are several different philosophies for certain things, that each have their die-hard adherents. See if you can figure out WHY each of them insists that you do it their way, and don't just blindly do it. You want to understand the WHY, not the WHAT, so that you can rebuild a different WHAT when the WHY changes. ('cause it will)

A lot of tutorials use analog boards, which is perfectly fine. The traditional workflow was worked out and cemented with analog gear, much older than what is typically used in those tutorials, and digital is made to work that way too. And as someone who's had the education to know the math behind the scenes, even the fundamental rules for what can and can't be done and what the unavoidable side-effects are, are the same for analog and digital as well. I think it's fascinating that two completely different approaches - physical circuitry, and wacky-looking math - end up in *exactly* the same place.

So don't turn down a good tutorial because it uses a different board than yours, or because it's analog and you're digital, or whatever. The key concepts are exactly the same everywhere, for everyone.
 
The audio signal coming out is so low I have to compensate in OBS for anything to be audible in the stream.
For that specifically, I suspect what's happening is that you're sending a "live" mix out of the X32 to OBS, and the stream needs a "mastered" mix.

A live signal doesn't know what's coming, and so you have to leave enough headroom to allow for some "surprises" without clipping. And those "surprises" do come from time to time.

A mastered signal is ready to send out, on a rather hostile medium. No matter what it is - cassette tape, vinyl, broadcast radio, internet stream, whatever - it has deficiencies that mostly come from driving down the cost of that medium to what the average consumer is willing to pay for. So you turn yourself up to *exactly* fill the meter as it goes out, to drown out those deficiencies, without ever going over. And that's what every consumer volume knob is calibrated for.

That requires that you know *exactly* what's coming, so that you can turn it up by *exactly* that much. Except you *don't* know what's coming because you're live. That's where compression comes in, like my example of the vintage LA-2A emulation (a very old compressor that people still like the sound of) followed by the standard bus compressor. Those two together, both tame the peaks and put the signal at *exactly* full-scale to send out, in a way that doesn't wreck the sound in the process.
 
Happy to help you as a volunteer on this. Ping me on WhatsApp and I will troubleshoot these issues. my WhatsApp number - +971566989505
I am a worship director at my church and we have been having an issue since upgrading our media that we experience this clipping or audio drop during the live streaming. We recently just upgraded everything to a 2022 Mac Studio (M1). We utilize ProPresenter 7 for slides and have it integrated with OBS to show the slides on the screen for Worship and the sermon. Audio is coming directly from a Behringer X32 (upgraded to latest firmware) out via USB and into the Mac Studio. We have the Audio input configured in OBS to recognize the USB audio, and have disabled all other Audio signals/inputs.
I have configured the advanced audio settings to increase the gain by 19db, (still not peaking in the red on the livestream) with a compressor and limiter
Compressor = Ratio=4.00:1, Threshhold=-5.00 db, Attack=100 ms, Release=94 ms, Output Gain=2.00 db, Sidechain/Ducking Source=None
Limiter = Threshold= -2.00 db, Release=60 ms

For the Stream settings, i have the Bitrate set to 4500, and the audio bitrate to 256.

We are using Ethernet (Not WiFi) and using Ookla speed test we are getting approximately 350 Mbps up and down. We Use Restream.io to stream to Facebook and YouTube, and the audio issue happens on both platforms. Please see the examples below, and the audio clipping happens every few seconds within the stream.

I have tried everything I can think of and nothing seems to help. Any input would be GREATLY appreciated.
.
 
I am not an audio engineer nor do I understand the Behringer X32 enough to assign busses or reconfigure
I'm an IT guy, who got to learn livestreaming the hard way ... and once quarantine lockdown ended, had to learn interfacing with mic system.

As a non-audio person (and a bit of 'tin ear'), I found our audio mixer (Presonus AR12 USB) not all that complicated to learn and re-configure... I just get someone with musicians ears to help tweak final output levels, adjust balance, etc.

Coming from an IT background, I documented EVERYTHING... So my question, does anyone in your community understand the audio system you have? The challenge is inheriting something set up a while ago, with different people over time getting 'something' to work, and now no one has any idea what all the settings, outputs, etc are. *IF* that is your scenario, finding someone to document your setup, and possibly clean up prior 'band-aids' is probably warranted. Whether you need to pay for that or not - up to you to decide. You may have musicians that are interested and capable of figuring out the current setup. Depends on timing urgency and your budget
Once Behringer setup documented, adjusting to accommodate a streaming output probably not that big of a deal (hopefully)... even if full resetting and starting from scratch on settings/setup
 
The challenge is inheriting something set up a while ago, with different people over time getting 'something' to work, and now no one has any idea what all the settings, outputs, etc are. *IF* that is your scenario, finding someone to document your setup, and possibly clean up prior 'band-aids' is probably warranted.
...
Once Behringer setup documented, adjusting to accommodate a streaming output probably not that big of a deal (hopefully)... even if full resetting and starting from scratch on settings/setup
I'm presently helping a local church figure that out for themselves. Midas M32, which is literally a Behringer X32 with fancy A/D converters, a slightly rearranged user interface, and a higher price tag. The processing, capabilities, and ecosystem are *exactly* identical in every way, and cross-compatible.

They hired a professional to come in for what sounded to me like one evening, and "fix things up". From what I could tell, he may have done *something*, but without local knowledge to maintain it, it didn't actually help at all. And I probably undid a fair amount of what he did, in the name of not wrecking a signal that's not that bad to start with. Professionals are only called that because they're paid, not necessarily because they know something...

For the past several weeks, I've been trying to convince them to make three lists in preparation for a complete wipe and rebuild, because they really are a barely-working mess:
  • What physical inputs does everyone plug into? Mics, instruments, etc.
  • What physical outputs does everyone listen to? Monitor wedges, in-ears, PA, etc.
  • Where should each thing be on the board?
Those three lists are completely independent from each other. Board-order and snake-order are not connected to each other at all, because a digital board can rearrange all of that in its settings. And if you do rearrange it there, then it tells you what that arrangement is. If you rearrange an analog snake tail on the back of the board, it can't tell you that, which makes it harder to maintain.

So far, they haven't done that, and I'm going to stop just short of forcing it...but they do want to restart the live stream that they stopped doing because it was terrible. I think I've convinced them to replace their analog snake with a digital one (actually several small digital snake heads scattered around the stage), so maybe I can use that to say that they're rearranging their inputs and outputs anyway, and they need to do the rest as well...
 
I'm an IT guy, who got to learn livestreaming the hard way ... and once quarantine lockdown ended, had to learn interfacing with mic system.

As a non-audio person (and a bit of 'tin ear'), I found our audio mixer (Presonus AR12 USB) not all that complicated to learn and re-configure... I just get someone with musicians ears to help tweak final output levels, adjust balance, etc.

Coming from an IT background, I documented EVERYTHING... So my question, does anyone in your community understand the audio system you have? The challenge is inheriting something set up a while ago, with different people over time getting 'something' to work, and now no one has any idea what all the settings, outputs, etc are. *IF* that is your scenario, finding someone to document your setup, and possibly clean up prior 'band-aids' is probably warranted. Whether you need to pay for that or not - up to you to decide. You may have musicians that are interested and capable of figuring out the current setup. Depends on timing urgency and your budget
Once Behringer setup documented, adjusting to accommodate a streaming output probably not that big of a deal (hopefully)... even if full resetting and starting from scratch on settings/setup
I wish! The board was set up by a company many years ago. Since then the entire staff and team leaders have changed. At this point, I probably know the most, which is not much, and I am trying to learn more and more to be as useful as I can be. But during services and events I am usually leading or playing guitar in the band, so I try to troubleshoot in the off times, and when things are happening live, I can't do anything, but try to "fix" it later. I've been coming in on my days off to try and tweak settings and learn more, but having no luck with this issue.
 
I'm presently helping a local church figure that out for themselves. Midas M32, which is literally a Behringer X32 with fancy A/D converters, a slightly rearranged user interface, and a higher price tag. The processing, capabilities, and ecosystem are *exactly* identical in every way, and cross-compatible.

They hired a professional to come in for what sounded to me like one evening, and "fix things up". From what I could tell, he may have done *something*, but without local knowledge to maintain it, it didn't actually help at all. And I probably undid a fair amount of what he did, in the name of not wrecking a signal that's not that bad to start with. Professionals are only called that because they're paid, not necessarily because they know something...

For the past several weeks, I've been trying to convince them to make three lists in preparation for a complete wipe and rebuild, because they really are a barely-working mess:
  • What physical inputs does everyone plug into? Mics, instruments, etc.
  • What physical outputs does everyone listen to? Monitor wedges, in-ears, PA, etc.
  • Where should each thing be on the board?
Good to know we are a Little bit further along than where you are working now. I have these 3 lists. And especially since we had to upgrade the firmware on the board in order to get USB out for the new setup, I documented and took pictures of every screen and setting on the board, so if i had to rebuild it, i knew what the previous settings were.
Those three lists are completely independent from each other. Board-order and snake-order are not connected to each other at all, because a digital board can rearrange all of that in its settings. And if you do rearrange it there, then it tells you what that arrangement is. If you rearrange an analog snake tail on the back of the board, it can't tell you that, which makes it harder to maintain.

So far, they haven't done that, and I'm going to stop just short of forcing it...but they do want to restart the live stream that they stopped doing because it was terrible. I think I've convinced them to replace their analog snake with a digital one (actually several small digital snake heads scattered around the stage), so maybe I can use that to say that they're rearranging their inputs and outputs anyway, and they need to do the rest as well...
If you are local to NJ, I would love for you to take a look! haha. I am going to keep watching YouTube videos and learning more about the ins and outs of the board. I know enough to be dangerous, but I still consider myself a novice. I am a musician that knows a little bit about mixing and audio, but not enough to be the audio engineer or "sound guy", but for this instance, I need to learn it so I can then try to teach someone else who will then hopefully take it and run with it so I can focus on what I'm good at.
 
For that specifically, I suspect what's happening is that you're sending a "live" mix out of the X32 to OBS, and the stream needs a "mastered" mix.
YES! This is exactly what we are doing. Trying to just send the main mix to the Live Stream so what you hear in the sanctuary is the same as what you hear in the stream. We used to have a separate Laptop that ran OBS solely for the livestream but the audio was being sent by an extra Behringer P-16-M monitor. But depending on who was running the stream they would always mess with the mix in the monitor or turn it down so it sounded good in the headphones, but it sounded HORRIBLE on the stream. Sometimes they would accidentally mute channels, or solo one channel, and you had to turn the volume on your device to max in order to barely hear anything. So when we had to upgrade the Mac to run ProPresenter7 we decided to just do everything on the Mac and cut out the laptop and not have the audio issue... so we traded one problem for this one.
A live signal doesn't know what's coming, and so you have to leave enough headroom to allow for some "surprises" without clipping. And those "surprises" do come from time to time.

A mastered signal is ready to send out, on a rather hostile medium. No matter what it is - cassette tape, vinyl, broadcast radio, internet stream, whatever - it has deficiencies that mostly come from driving down the cost of that medium to what the average consumer is willing to pay for. So you turn yourself up to *exactly* fill the meter as it goes out, to drown out those deficiencies, without ever going over. And that's what every consumer volume knob is calibrated for.

That requires that you know *exactly* what's coming, so that you can turn it up by *exactly* that much. Except you *don't* know what's coming because you're live. That's where compression comes in, like my example of the vintage LA-2A emulation (a very old compressor that people still like the sound of) followed by the standard bus compressor. Those two together, both tame the peaks and put the signal at *exactly* full-scale to send out, in a way that doesn't wreck the sound in the process.
 
The board was set up by a company many years ago. Since then the entire staff and team leaders have changed.
Most professional installs leave a bad taste in my mouth. I understand why they lock things down - customers mess with it, and then blame the company for a "bad product" - I saw that myself when I was in Industrial Controls for a while - but it also prevents a new local legitimate expert from fixing anything or adapting to a new set of requirements.

Better, in my experience (at least with media gear that doesn't ruin a few million dollars worth of material when someone screws it up), to provide an easy way to get back to the installed settings, and then leave it wide open. If you don't like what your tinkering ended up with, re-load how it was.

In fact, I do that automatically for the last 3 rigs that I've built. Digital board, and a computer that's on 24/7/365 except for a periodic scripted update and reboot. (so far, I've done it with 2 Raspberry Pi's running the official Raspberry Pi OS, and 1 Mini PC running Lubuntu Linux, both of which are based on Debian Linux, which makes the scripting part easy) That computer also watches for when the board has been off for at least 2 hours (more than a momentary power glitch) and then comes back on, and pushes a scene file into it that I built for that rig. So no matter how someone left it, the next person gets that standard. If you want to start with something different, you need to have saved it before, and load it again after the reset.

I think it'd be great if manufacturers would include that function natively as an option, but so far, I haven't seen it.

At this point, I probably know the most, which is not much, and I am trying to learn more and more to be as useful as I can be.
Great! Keep working at it!

But during services and events I am usually leading or playing guitar in the band, so I try to troubleshoot in the off times, and when things are happening live, I can't do anything, but try to "fix" it later. I've been coming in on my days off to try and tweak settings and learn more, but having no luck with this issue.
Is there someone else that can lead, maybe once a month, so you can run the board live? You'll only catch about 1/4 of the recurring problems that way, but the fix-once-and-done ones will eventually come up while you're on. For the other 3/4 of recurring problems, you'll also have better knowledge to tell the person running it how to fix it.

Just be careful about mixing from stage. You have no idea from there, how it sounds in the audience. For that, just trust your FOH Engineer, no matter what their experience level. That's also a respect thing...

Good to know we are a Little bit further along than where you are working now. I have these 3 lists. And especially since we had to upgrade the firmware on the board in order to get USB out for the new setup, I documented and took pictures of every screen and setting on the board, so if i had to rebuild it, i knew what the previous settings were.
Good! A lot of people just run through a major reconfiguration like that, perhaps blindly following some poorly written instructions or advice, and then panic when they find that their stuff isn't preserved.

If you are local to NJ, I would love for you to take a look! haha.
Kansas City. Sorry.

I am going to keep watching YouTube videos and learning more about the ins and outs of the board. I know enough to be dangerous, but I still consider myself a novice. I am a musician that knows a little bit about mixing and audio, but not enough to be the audio engineer or "sound guy", but for this instance, I need to learn it so I can then try to teach someone else who will then hopefully take it and run with it so I can focus on what I'm good at.
That's more-or-less how I got started. I played piano and trombone growing up, and absolutely hated the theory part. It was never connected to the fun pieces that I just hammered into my memory so I could "just play them".

Then much later, after we moved and didn't find another teacher, my new church youth group had a hand-me-down system that nobody understood, including me, so I just went back there and figured it out. Mackie SR24: the classic, cheap, ubiquitous, and weird analog console.

In the years after that, during and after college, I ate up all of the articles and videos that I could find about how to run a generalized audio rig *well*. It wasn't until many years of that, that I realized that THAT'S THEORY! The very thing that I hated growing up because it was just pounded into me in isolation, I was now seeking out and devouring.

I still don't like music theory - maybe a mild traumatic thing - but I still read articles and watch videos about technical media. And my musical background has helped me many times to anticipate where the band is going, so I can push it slightly for an exciting bit, or pull it back to settle down, or highlight a solo, or whatever. The mixing desk is its own instrument, and the person that runs it is just as much a part of the band as the people on stage. "Set and forget" is like swapping out the drummer for a drum machine, or using a backing track for a fairly major part.

Trying to just send the main mix to the Live Stream so what you hear in the sanctuary is the same as what you hear in the stream.
The main mix, and thus what comes out of the audience PA, is *not* what you hear in the room! It's missing what spills off the stage all by itself, and it's unavoidably compensated for what the room does to it, which is unique to that room.

The stream needs its own mix, with studio monitors or headphones, and it needs to include more than what the house mix does. When I first set up our drum mics, the drummer was adamant that we didn't need them. He could play to the room just fine, with no need for amplification, which was entirely true. He really is that good. But I needed them for the stream. When I explained that, he was okay. They didn't go to the house at all...until we had a particularly energetic song one Sunday, and I boosted the kick a little bit. :-)

Plus that whole bit about "live" vs. "mastered" and what it takes to do that.

We used to have a separate Laptop that ran OBS solely for the livestream...So when we had to upgrade the Mac to run ProPresenter7 we decided to just do everything on the Mac and cut out the laptop...
If you have the spare capacity, and you're not doing very much, that works. But I've found that "separation of duties" works wonders! Dedicated machine for each task, even if you do have enough spare capacity in one machine to fit something else.

Dedicated controls with no switching, optimized settings for each job, if one crashes the others keep going, etc. Lots of benefits, and the extra expense of a (good!) physical capture device to get one machine's content into another, is not that much by comparison.

...the audio was being sent by an extra Behringer P-16-M monitor. But depending on who was running the stream they would always mess with the mix in the monitor or turn it down so it sounded good in the headphones, but it sounded HORRIBLE on the stream. Sometimes they would accidentally mute channels, or solo one channel, and you had to turn the volume on your device to max in order to barely hear anything.
I've heard of people using a P16 to run the stream mix. Sometimes it works, sometimes it doesn't. I think your problem was more that the operator didn't know the *actual* requirements to do it well, and/or didn't know what they were doing in the first place. Those problems are the same no matter what gear you use.

If they turn down *everyone's* volume to make their own personal headphones "right"...
 
I have no experience with the Behringer, so I'm not volunteering but food for thought
- there is the final (separate, as noted above) audio mix for in-house and for Recording/Streaming which requires actually hearing.
- BUT going over mixer basic setup? and config options to support streaming? in my experience, one doesn't need to be there in person for that
even the stream/recorded audio can be 'listened to' remotely.

FWIW - when COVID-19 lockdowns started, and HoWs having to figure out livestreaming, I helped out a number of churches with simple video calls to get them further along... so to get assistance on your learning curve with someone walking you thru settings, options, considerations, etc...
my recommendation is to NOT get hung up on someone local... easier (and usually quicker), yes, but not absolutely required for a portion of what you need.
- and the best person to learn & document system, settings, and operator guide, is someone who would actually operate mixer during service
Good luck on finding someone who isn't on stage during the service to learn to properly operate system ... (or at least have some guidelines on what sound system person during service is and is NOT allowed to do... hard to do with volunteers, but in my experience filtering out those that will cause problems is better in long run for those that remain to operate livestream)

And, local headphones are ABSOLUTELY NOT the right audio to listen to for stream audio check.
I personally prefer the Digital Usher approach, both for
1. comment moderation (typically deleting older/less techy member posting private details like ph# in public comment space) and
2. someone listening to livestream on mobile device is the correct place to check audio (with challenge of associated delay). Using headphone locally to get better feel for overall balance, sure. but there are plenty of audio aspects that have to be checked post-stream, and if that isn't being done... no surprise doesn't sound good.
 
@AaronD, Cannot thank you enough for the input and the patience. I have learned a TON over the last few days about the X32 and routing, assigning a separate mix to Busses, etc. So i have reconfigured the entire board settings (using a separate Scene) to assign a new mix specific to our livestream and assigned it to a Stereo Bus output, so we have the House Mix and a Streaming mix. I have removed all filters from OBS, and we will be doing a test stream tomorrow night to see if it changes anything.
At this point, if it doesn't change anything and the issue persists, I think we may need to look into another streaming platform, but hoping not to since OBS is open source.
 
At this point, if it doesn't change anything and the issue persists, I think we may need to look into another streaming platform, but hoping not to since OBS is open source.
Sounds like this will be a new setup... which means all the normal gotchas, oopses, and the rest. Don't give up too soon...
hopefully your audio drop outs will go away with just passing audio through OBS Studio ... assuming Mac isn't overloaded (watch during testing with System Monitor)... if OS/computer hardware resource utilization issue, changing away from OBS Studio may or may not help (depends on M-CPU optimization on other platforms using non-open-source code). if close, some OS optimization may be in order (as cheapest solution)

Then, once you know audio getting thru cleanly... then you'll probably want to adjust stream audio... but beware... there is no right answer.
- Adjust audio for home viewer watching with large screen TV and quality sound system (not TV speakers) and that will be one thing, but sound poor on mobile devices, and vice-versa. Personally, we focused more on the mobile device listening experience, and the others can make due with what that entails (especially as others tend to have better audio adjustment options available to them).
- Don't be surprised if that means running at a typically higher than usual DB level, and using audio compression
there are numerous videos and tutorials on House of Worship audio optimization for livestreaming. As a musician, you may be biased to focus on the more audio-perspective challenging music portion of the service, but I recommend getting spoken vocals solid (sounding great) first and foremost, then focus on the rest of your inputs.

Personally, I started by listening via headphones (balance, etc) off Mixer of stream mix, then I listened to Local Recordings from OBS Studio (which are much higher quality than the re-encoded livestream provider videos) ... Then, I listened to our streamed content for final tweaks
To Facebook, we stream 1080p30 at 7000kpbs (originally on DSL, now a fiber connection... so much better, but I digress). Facebook allows test streams (only Facebook group page admin will be able to view it). I recommend doing a test stream during rehearsal and check audio on mobile device (both with built-in speakers, and headphones, just so you have a point of reference)... Final audio level tweaking of audio means making small change, waiting 20-30 seconds (possibly faster of good ISP/WiFi than cellular) until audible on mobile device livestream, confirm/tweak some more, etc... tedious but doable if one is patient. Ideally you should be so much better off than you are now... then, you'll probably listen to livestream Recordings for a few weeks/month, and make additional small adjustments
be open to feedback (I recommend Digital Usher asking for the feedback) from your livestream audience members with less than ideal listening environments, the elderly, etc. You may easily find that the optimal streaming audio settings doesn't really work for a portion of your audience [though for some, hard of hearing, bad Internet &/or tech at home, they will always struggle... different issue... though unless you livestream audience is all listening in similar environment, you won't be able to create a really good audio experience for everyone at the same time with a single mix .. just not possible]

Good luck
 
If you guys have any issues with live sound, post-productions, live streaming, hybrid meetings with AV equipment, or TV media operations relating to codecs, feel free to ask anything on these topics.
 
At this point, if it doesn't change anything and the issue persists, I think we may need to look into another streaming platform, but hoping not to since OBS is open source.
OBS is probably not the problem here. And any replacement will probably do the same thing. It's either a bad input from the board (mixing, mastering, and/or wiring), or a bad network. OBS itself is fine.

...local headphones are ABSOLUTELY NOT the right audio to listen to for stream audio check.
Depends on the style and volume of the in-house music:
  • For loud bass-heavy live music, not a chance! Your stream mix in headphones in the same room will be sorely lacking in bass and probably not very accurate anyway, because you still feel the subs and the rest bleeds through the over-ear cans.
  • For quiet, "hearing-aid friendly" music with not much volume and hardly any bass, it's possible to turn the (sealed) headphones up enough to drown it out without hurting yourself, and then you can get away with it.
But yes, listen afterwards too, for a more accurate picture of what you actually did.
 
If you guys have any issues with live sound, post-productions, live streaming, hybrid meetings with AV equipment, or TV media operations relating to codecs, feel free to ask anything on these topics.
The questions are already here, for all to see. If you have answers, also post them here for all to see. That's how forums work. Continued attempts to take the conversation off-site to you personally, are likely to be flagged as spam.
 
UPDATE for anyone following...

So between last night and tonight I set up Busses on the Behringer X32 assigned the USB out, and created a separate mix specifically for Live Streaming that has compression and limiters set up on. I completely removed the Compression and Limiter from the OBS filters and i adjusted the gain down to +10 db (down from +20).
Additionally, we had both the Video capture (AJA U-TAP) and the X32 USB audio coming into the MAC on the same USB bus. So i moved the Video capture to a USB extender that comes in through Lightning (USB-C) to a completely separate USB bus, so that the USB audio input is the ONLY thing going into that USB bus.
I have confirmed that the audio signal coming out of the board is clean, no clipping or distortion.
With all the changes and new settings, we did 2 tests (1 before swapping the USB bus on the MAC itself,

BEFORE - https://www.youtube.com/watch?v=LDeXIQzkEd0
and the second was after
AFTER - https://www.youtube.com/watch?v=zY0VaZ0F4IA

The sound quality itself is MUCH better with the separate live stream mix, but the clipping/dropping issue still exists. I feel like it is a tad better AFTER swapping the Video capture to a separate USB bus, but I'm still getting Audio clipping/dropping.
Is it a problem with OBS or is it a Problem with the MAC?? This is the question i can't seem to figure out.
 
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