ATEM Television Studio HD Output Volume is Low

I am pushing audio from a Presonus StudioLive 24.4.2ai into an ATEM TV Studio HD then out through an SDI distribution. It's going to several TVs in the building as well as a VidiuPro for the live stream feed. But I've noticed that the output volume from the ATEM is really, really low. Is there any way to boost the volume from the switcher?

The levels on the sound mixer are really high, too high really. I have the faders maxed out and am having to compensate with gain adjustments (this is a live streaming mixer only, so no worries about affecting in-house volume). I'm running WAY too high on the mixer side just to get sufficient volume out from the switcher.

What am I missing here?

Known problem with the Atems! They require a +4dBu audio signal. This will max out your mixer.

Known problem with the Atems! They require a +4dBu audio signal. This will max out your mixer.

Pardon the ignorance of this question (audio is not my specialty), but is there any risk of blowing something internal on the mixer running it that high? Will it send a clipped audio signal?

If the audio mixer will clipped signal is fully up to the audio mixer..

But normally it would kind of work.. But yes the volumes are low!! And could be much higher..

There is no way of seeing if the signal clips.. (Yes the red led's come on.. but that does not say it is clipping or not.. But when you are hearing pops and other distortion.. Then it is clipping!!

anyone has a solution to this? We are doing live band and volume is really low streaming in Youtube live. Pls help, I appreciate it thanks.

The levels on the sound mixer are really high, too high really. I have the faders maxed out and am having to compensate with gain adjustments (this is a live streaming mixer only, so no worries about affecting in-house volume). I'm running WAY too high on the mixer side just to get sufficient volume out from the switcher.

Assuming you have the Main L/R Presonus outputs connected to the ATEM L/R XLR inputs...you levels leaving the Presonus should be lighting all of the green LEDs on the Presonus Main meter, and occasionally 1 yellow and less occasionally 2 yellow LEDs. Mix accordingly. Also, there is a level knob next to the XLR outputs on the rear of the console, make sure it is fully clockwise to the "0" setting-anything less is attenuating what leaves these connectors (bad).

On the ATEM, make sure your master fader and the XLR input faders are pushed up to the "0-" for starters. Best way to match up levels between these 2 devices is to send tone from your Presonus at "0", the highest green LED. With any luck you see this tone level on the XLR meter on the ATEM audio screen and it should align with the -20 mark about 1/3 up from the bottom. "0" on the presonus meter should equal -20 in the ATEM. If it's at least close to this proceed on. Adjust the XLR fader until the Master meter is also reading -20. Leave these settings, and maybe select "Save startup state" on the ATEM file menu to lock them in even after a power reset.

As your show is happening, your Presonus meters should be peaking up to the 1st or 2nd yellow LED, and this should correspond to the ATEM Master meter right around the top of the yellow, just barely touching into the red. If the volume downstream still seems too low you will have to look into Dynamics (compression) which will enable you to get a more dense signal that "sounds" louder without raising the meter levels.

Pardon the ignorance of this question (audio is not my specialty), but is there any risk of blowing something internal on the mixer running it that high? Will it send a clipped audio signal?

While running highly distorted audio into large PA speakers *can* cause speaker damage, high levels and distortion will not damage your mixer or ATEM. It does hurt your ears to listen to this mess tho.

I've been through this many times. The simplest solution is to buy another mixer (I personally use a Yamaha MGO6). I like that mixer because it has XLR and 1/4 inputs (channels 1&2) and also XLR outputs. There are several small mixers from other manufacturers that will work well however often you will need adapters.

I've fought the battle of trying to take the output off the main console directly into the switcher and always had issues. This way the Audio Engineer can mix the signal the way they want to and I just get a L/R feed and I can boost the level into the ATEM to meet my needs.

I can walk into any environment and just say give me your L/R feed and take it from there. This way it preserves the integrity of the original sound environment.

Hope this helps!

Best,

David

anyone has a solution to this? We are doing live band and volume is really low streaming in Youtube live. Pls help, I appreciate it thanks.

I have at rack small behringer XR12 mixer .. I love all X/M-32 and X/M Air concoles for Automix and Precision limiter .. Im getting L/R from FOH audio mix .. Im adjust my gain to be somewhere up to -10dBFS .. On ouput I have Precision Limiter with +10 dB gain .. and limiting to -3 dB .. MASTER outs from XR12 is going to ATEM TVS HD .. It geetting signal limited to -12dBFS .. So in atem audio mixer i have volume on XLR inputs + 6db .. So my signal into stream is limited to -6dBFS .. It enough for me ..

Hello, Is there a Blackmagic fix for this, other than using a second mixer with levels boosted?

We are using the Production Studio 4K and don't have access to the advanced audio controls, due to it's old age. In Blackmagic Software Control, we are boosting our channel +6db, and our master +6db.

Our loudness on YouTube is still around -7db.

Same answer as I posted 2 years ago. Buy an inexpensive mixer and put it inline. See my post above.

Best,

David

I seriously think BMD has made a mistake here. The +4dBu is the signal level used on professional equipment. No matter what (I consider) professional equipment I connect to it, the volume is always very low.

Not sure if all ATEM models support it but on the Constellation HD for example I can switch the audio input level from +4dBu to -10dBV. This lets the ATEM think that the audio input is at the "comsumer audio equipment" level of -10 dBV (eventhough I'm pretty sure my equipment works at +4 dBu). This effectively boosts the signal by almost 12 dB and the signals are now a lot more comparable.

@Xtreemtec do you know if this was fixed in the Constellation? (or anyone in the know).

Thanks!

David

Known problem with the Atems! They require a +4dBu audio signal. This will max out your mixer.

On the Constellation 8K i still have to push far over 0 at the audio desk to get -4 on the Atem metering.

That will result in -14Lufs on the SDI embedded output which is the level you want to send to Youtube to be "normal" level

On the Constellation 8K i still have to push far over 0 at the audio desk to get -4 on the Atem metering.

That will result in -14Lufs on the SDI embedded output which is the level you want to send to Youtube to be "normal" level

Just bought the Audio to SDI converter. Going to see how that works. I can use an ATEM Camera converter and run my mixer into that and get a decent signal chain. I have spare SDI inputs so if that can work as a dedicated audio input I'll use that. I'll report results back here. It sounds like they still have not fixed the issue on the Constellation mixers (which just saves me a bunch of money and worry about Constellation 2ME 4k).

Best,

David

ATEM meters shows peaks, not loudness. To get high LUFS levels, one must be agressive with dynamics, starting with clean gain & good S/N ratios. It's good to have loudness meter (a la Clarity M or similar).
Also, broadcast sound is much more "open" (-23 or -24 LUFS) and basicly match with usual 0VU = +4dBu = -18 dBFS setup.

Maybe easiest way is to have mastering proccessor (fast limiter/multiband), and drive it hard enough.

Precision Limiter solve this ..

Precision Limiter solve this ..

Please explain how a precision limiter solves a 17db deficit on the XLR inputs. I've measured it.

If you have a 1ME or 2ME 4k you can buy a simple XLR to RCA cable and it's an instant 12db of gain. Everything is much more normal. On the Constellation HD you can select -10 or +4db gain so the same applies.

Sounds like at this point the only oddball out is the Constellation 8k.

Best,

David

Wait a minute.... You give +4 dBu 1k tone to atem XLR input and get -36 dBFS at meters? This is faulty mixer, then, and need replaced.

If You get -18 dBFS (ATEM's full green), this is perfectly normal.

For normal *streaming* loudness (-16 ... -14 LUFS, this is BTW 7..9 LU hotter, then EBU broadcasting standard) you can and must stay below -10 dBFS readings at ATEM meters (peaks at -3 dBFS TP). And levels and loudness are different things.

Thing is, all consumer audio is too hot and this is source of evil

example- (one setup we use)- A&H GL2800 with BMD ProductionStudio 4k. I use Behringer Ultra-Curve Pro in group insert as master limiter and send signal out through matrix output (little gain added there, maybe 2 dB). Also i have channel compressors ofcourse. What i need is ~6 dB gain lift for working "against" -10 dBFS in ATEM, i push little bit group faders (4 dB or so) and thats it.


Best!

I get -18db when sending tone not -36db.

This is a known issue as shown by the responses to the thread. If I'm sending the BM 0db tone I should get 0db in the BM not -18.

The BM XLR inputs require you to run the mixing desk at very high levels (to the point of almost clipping) and then you need to run the BM faders almost max to get normal output on the SDI to feed an encoder out to youtube or other platform.

0db tone should be 0db tone on all pro gear, no?

David

Wait a minute.... You give +4 dBu 1k tone to atem XLR input and get -36 dBFS at meters? This is faulty mixer, then, and need replaced.

If You get -18 dBFS (ATEM's full green), this is perfectly normal.

For normal *streaming* loudness (-16 ... -14 LUFS, this is BTW 7..9 LU hotter, then EBU broadcasting standard) you can and must stay below -10 dBFS readings at ATEM meters (peaks at -3 dBFS TP). And levels and loudness are different things.

Thing is, all consumer audio is too hot and this is source of evil

example- (one setup we use)- A&H GL2800 with BMD ProductionStudio 4k. I use Behringer Ultra-Curve Pro in group insert as master limiter and send signal out through matrix output (little gain added there, maybe 2 dB). Also i have channel compressors ofcourse. What i need is ~6 dB gain lift for working "against" -10 dBFS in ATEM, i push little bit group faders (4 dB or so) and thats it.


Best!

0db tone should be 0db tone on all pro gear, no?

No. dB's in themselves are meaningless and just a ratio.

The Atem measures in dBFS where 0dBFS is the absolute maximum value possible. Positive dBFS values are not possible - the signal has been clipped and lost.

Most professional sound mixers measure in dBu, where 0dBU is a reference to a specific analogue voltage (about 0.7746V RMS) level. Some other audio gear uses dBV, where 0dBV is a relative value compared to 1V. Both standards allow positive values where the signal level is higher than the reference value.

The EBU recommends that 0dBu is -18dBFS. SMPTE recommends 0dbu is -20dBFS.

There is a good description of all the terms and confusion here - https://ips.org.uk/encyclopedia/audio-levels/

Fair enough. The issue is still that if you run a mixer at 0db you are at -18 on the BM. This does not give enough level for normalized sound while recording or streaming. Everyone here has to push the BM in order to get good levels for streaming.

David

0db tone should be 0db tone on all pro gear, no?

No. dB's in themselves are meaningless and just a ratio.

The Atem measures in dBFS where 0dBFS is the absolute maximum value possible. Positive dBFS values are not possible - the signal has been clipped and lost.

Most professional sound mixers measure in dBu, where 0dBU is a reference to a specific analogue voltage (about 0.7746V RMS) level. Some other audio gear uses dBV, where 0dBV is a relative value compared to 1V. Both standards allow positive values where the signal level is higher than the reference value.

The EBU recommends that 0dBu is -18dBFS. SMPTE recommends 0dbu is -20dBFS.

There is a good description of all the terms and confusion here - https://ips.org.uk/encyclopedia/audio-levels/

In mixer You have 0VU as reference zero. It outputs... something, usually +4dBu, but there's no real fixed standard. It equals usually with -18 dBFS in digital domain. This is also not always so.

Problem is, consumer audio levels (and streaming) is broken big ways and have some kind of offset (as much as 12 dB) compared to pro audio. ATEM levels makes perfect sense for broadcasting by the way

Anyway, what You need, is dynamic range ~10 dB or less (thats why you need compress and limit your signal) and peaks -3 dBFS TP max. Then You can simply push output little bit to get needed loudness.

And no, you can not "normalize" live sound, never ever. But you can set up brickwall limiter at -3 dbFS in the end, you can use channel/bus/master compressors, you can use multiband, you can use hi-pass filters & EQs etc.
And also you have faders and You can choose, how hard you want drive mix bus, if you have some processing there.



Best!

If you don't push the ATEM levels the output is too low compared with normalized volume coming from most youtube channels. I've done hundreds of livestreams and always had to work around BM signal level by compensating somewhere else. Simple workaround is use RCA.

https://f.io/zvTW7vxq

Just play this clip while playing a youtube channel and you will see that when I switch from XLR to RCA the volume is more comparable to a normal youtube video.

Part of the issue is that the older ATEM's that don't have the fairlight mixer don't have any gain controls for the inputs.

Best,

David

In mixer You have 0VU as reference zero. It outputs... something, usually +4dBu, but there's no real fixed standard. It equals usually with -18 dBFS in digital domain. This is also not always so.

Problem is, consumer audio levels (and streaming) is broken big ways and have some kind of offset (as much as 12 dB) compared to pro audio. ATEM levels makes perfect sense for broadcasting by the way

Anyway, what You need, is dynamic range ~10 dB or less (thats why you need compress and limit your signal) and peaks -3 dBFS TP max. Then You can simply push output little bit to get needed loudness.

And no, you can not "normalize" live sound, never ever. But you can set up brickwall limiter at -3 dbFS in the end, you can use channel/bus/master compressors, you can use multiband, you can use hi-pass filters & EQs etc.
And also you have faders and You can choose, how hard you want drive mix bus, if you have some processing there.



Best!

Distortion happens, when analog inputs near to clipping. It's safe to go up 10 dB (e.g. +14 dBu in XLR / 0 dBu in RCA), but not much more, especally with ATEM's analog ins. This is why right way to handle situation is watch your dynamics & keep levels right.

Please read https://en.wikipedia.org/wiki/EBU_R_128

Targets for streaming are -16 ... -14 LUFS (NB! LU means Loudness Unit and to be able read this you need loudness meter somewhere in signal path), safe peak level is -3 dBTP
Something like this-

If you can handle dynamic range right, all you need is keep levels against -10 dBFS.

BTW, the end is still very quiet

Peak -3.9 Clips max 0 LUFS-M max -17.6 LUFS-S -19.7 LUFS-I -20.4 LRA 3.3

Please post a video of your setup including the metering on the Mixing Desk, BM Audio Mixer and Encoder. I'd love to see how you did that.

Best,

David

Distortion happens, when analog inputs near to clipping. It's safe to go up 10 dB (e.g. +14 dBu in XLR / 0 dBu in RCA), but not much more, especally with ATEM's analog ins. This is why right way to handle situation is watch your dynamics & keep levels right.

Please read https://en.wikipedia.org/wiki/EBU_R_128

Targets for streaming are -16 ... -14 LUFS (NB! LU means Loudness Unit and to be able read this you need loudness meter somewhere in signal path), safe peak level is -3 dBTP
Something like this-

If you can handle dynamic range right, all you need is keep levels against -10 dBFS.

BTW, the end is still very quiet

Peak -3.9 Clips max 0 LUFS-M max -17.6 LUFS-S -19.7 LUFS-I -20.4 LRA 3.3

ATEM (PS4K) is pushed 6dB, "encoder" is HyperDeck Studio. -10dBFS is used as (kind of) reference zero for input.

Source audio is compressed using multiple stages (this is piece of audio guide) and used simply as playback.

You have to push ATEM to 6db (no headroom left) to still get -10dbfs.

https://f.io/_03nAm2W

Use RCA input and you don't have to push anything. Since tone is consistent - I'm using tone in this example. This really is no different than on new Constellation HD where you can select +4 or -10 for the input.

The conclusion is that if you use the XLR inputs on any of the switchers you have to PUSH the signals to get normalized sound out of the SDI feed to an encoder. If you have a switcher that has RCA inputs you can use normal mixing desk signals and not have to push signal 6db.

Best,

David

ATEM (PS4K) is pushed 6dB, "encoder" is HyperDeck Studio. -10dBFS is used as (kind of) reference zero for input.

Source audio is compressed using multiple stages (this is piece of audio guide) and used simply as playback.

The conclusion is that if you use the XLR inputs on any of the switchers you have to PUSH the signals to get normalized sound out of the SDI feed to an encoder.

The conclusion should be that to get levels higher than 0dBu, you need to run push your levels higher than 0dBu. If you're aiming for -10dBFS for your Youtube stream, that's around 10dBu, not 0dBu.

You must not use analog inputs this way. There is difference using top of analog headroom and trimming in digital domain after A/D
I personally do NOT want use last 10 dB before input or output clip. There is thing called gain structure in pro audio.

But, whatever & good luck!

P.S. please read this- viewtopic.php?f=4&t=67511 and this- viewtopic.php?f=4&t=65308

Both XLR and RCA inputs are analog, no?

https://f.io/ofSKoJ1l

Same device, same settings, just change XLR to RCA and back.

At the end of the day it solves a problem for me to not have to run mixing desk at very high level or push Blackmagic to very high level. I have headroom on both units.

Best,

David

You must not use analog inputs this way. There is difference using top of analog headroom and trimming in digital domain after A/D
I personally do NOT want use last 10 dB before input or output clip. There is thing called gain structure in pro audio.

But, whatever & good luck!

P.S. please read this- viewtopic.php?f=4&t=67511 and this- viewtopic.php?f=4&t=65308

No, it does not solve Your real problem, but brings in new

Analog domain does not work similar way as digital, but have nonlinearity. E.g. you can not use all headroom. For ATEM clean area is up to +14dBu (= -10 dBFS) a.k.a. +10 dB VU. Read, what Gary says.

It is safe push up level *inside* ATEM. As long as you stay below -1 dBTP i's all O.K., because you are after A/D.
And btw, Constellation has Fairlight onboard (e.g. dynamics in every channel with make up gain), i have no clue what you talking about. If using older mixer like Production Studio (without eq/dynamics) one must use outboard gear, simple as that.


And real problem is too open sound e.g. too wide dynamic range. Peaks up to +10dB VU is not "too hot".

Not going to debate. It solves my problem and doesn't bring in any new problems for me. I now don't have to run my mixing desk at very high levels and I have headroom in the BM. Simple as that. XLR and RCA inputs are both analog.

I don't have a constellation so I don't have fairlight.

For others looking my video is clear proof that you can use the RCA inputs and not have to run everything at extreme levels.

Best,

David

No, it does not solve Your real problem, but brings in new

Analog domain does not work similar way as digital, but have nonlinearity. E.g. you can not use all headroom. For ATEM clean area is up to +14dBu (= -10 dBFS) a.k.a. +10 dB VU. Read, what Gary says.

It is safe push up level *inside* ATEM. As long as you stay below -1 dBTP i's all O.K., because you are after A/D.
And btw, Constellation has Fairlight onboard (e.g. dynamics in every channel with make up gain), i have no clue what you talking about. If using older mixer like Production Studio (without eq/dynamics) one must use outboard gear, simple as that.


And real problem is too open sound e.g. too wide dynamic range. Peaks up to +10dB VU is not "too hot".

You simply overdrive RCA input and thats it. But, whatever, anyway.


viewtopic.php?f=4&t=23057



For others & please correct me, if i missed something or get it wrong

There is 3 options in real life for streaming audio somewhat "correctly". For start- reason for all this mess is 10...12 dB offset between pro & consumer audio levels, also reduced or missing headroom. (Somewhat reminds me, how DJ's dealing with levels )


a) mix in digital domain, get needed LUFS levels & use brickwall limiter @ -3dBTP, feed ATEM thru SDI or HDMI line (if MADI or AES/EBU not present)

By the way, did some testing today using old Ultrastudio Express- software mixing and using separate device for input works 100% with Logic Pro! So (somewhat) low budget full digital workflow can be done this way absolutely.


or (using analog in, no Fairlight present)

b) mix in analog or digital and set brickwall limiter @ +10 dB VU, feed ATEM so peaks hits -9 dBFS, rise level inside ATEM 6 dB. You get needed loudness and stay below -3 dBFS, also avoid analog input overdrive. For more comfort, feed console thru matrix/group out and trim it up there, if available. Also it is possible to use inline mastering processor (a la ultracurve pro or similar) and set brickwall to +10 dB VU here.

or, if Fairlight present (minis, TVS pro4k, Constellations etc), both digital/analog input

c) mix as normal, e.g. using nominal levels etc, in ATEM set brickwall limiter @-3 dBTP and use makeup gain.


Anyway, you need limiter somewhere in the end of signal chain in any case.

I posted audio and video clips where there is no distortion and I'm not overdriving the signal to the RCA input.

Unlike you, I posted photos and videos of REAL levels and REAL audio. My solution requires no special equipment or processing.

https://f.io/gm28fvMb - Video of setup
https://f.io/zvTW7vxq - Actual Wirecast recording. Please tell me when you hear RCA overdrive I don't hear it.

I don't need to run BM Faders at +6. Please post screenshots and video of your setup.

David

You simply overdrive RCA input and thats it. But, whatever, anyway.


viewtopic.php?f=4&t=23057



For others & please correct me, if i missed something or get it wrong

There is 3 options in real life for streaming audio somewhat "correctly". For start- reason for all this mess is 10...12 dB offset between pro & consumer audio levels, also reduced or missing headroom. (Somewhat reminds me, how DJ's dealing with levels )


a) mix in digital domain, get needed LUFS levels & use brickwall limiter @ -3dBTP, feed ATEM thru SDI or HDMI line (if MADI or AES/EBU not present)

By the way, did some testing today using old Ultrastudio Express- software mixing and using separate device for input works 100% with Logic Pro! So (somewhat) low budget full digital workflow can be done this way absolutely.


or (using analog in, no Fairlight present)

b) mix in analog or digital and set brickwall limiter @ +10 dB VU, feed ATEM so peaks hits -9 dBFS, rise level inside ATEM 6 dB. You get needed loudness and stay below -3 dBFS, also avoid analog input overdrive. For more comfort, feed console thru matrix/group out and trim it up there, if available. Also it is possible to use inline mastering processor (a la ultracurve pro or similar) and set brickwall to +10 dB VU here.

or, if Fairlight present (minis, TVS pro4k, Constellations etc), both digital/analog input

c) mix as normal, e.g. using nominal levels etc, in ATEM set brickwall limiter @-3 dBTP and use makeup gain.


Anyway, you need limiter somewhere in the end of signal chain in any case.

Clarity shows digital domain LU level (playback). Signal is trimmed before analog in. Console is flat. Both channel fader & group faders are pushed little bit to get +10 dB VU peaks, also matrix out is rised one dB from UG.

As you see, desk peaks at +14 dBu at the output and this is *****perfectly****** normal.



And please understand, this console does not have line drivers upgrade kit installed and matrix output level is -2 dBu e.g. i have 6dB low ride at outputs other then LRM.


...and group insert settings-

group insert 1

20220830_134926.jpg (274.11 KiB) Viewed 3973 times

group insert 2

20220830_134937.jpg (238.54 KiB) Viewed 3973 times

And yes, i can hear the distortion. But in most part audible distortion is caused not by RCA overdrive (which also exists). (examples- WireCAST, NOW etc). I do not know, what is it. Some dynamics processing?

There is no audible distortion on RCA and you made my point. You have to push mixer high and BM output master to FULL +6 and you have no headroom. Do as you wish and I will do the same.

I will no longer respond to this thread.

https://youtu.be/OFToI2NY6qg

Clarity shows digital domain LU level (playback). Signal is trimmed before analog in. Console is flat. Both channel fader & group faders are pushed little bit to get +10 dB VU peaks, also matrix out is rised one dB from UG.

As you see, desk peaks at +14 dBu at the output and this is *****perfectly****** normal.



And please understand, this console does not have line drivers upgrade kit installed and matrix output level is -2 dBu e.g. i have 6dB low ride at outputs other then LRM.


...and group insert settings-

20220830_134926.jpg

20220830_134937.jpg

And yes, i can hear the distortion. But in most part audible distortion is caused not by RCA overdrive (which also exists). (examples- WireCAST, NOW etc). I do not know, what is it. Some dynamics processing?

Well, GL2800 real *headroom* is-

in channels +20 dB
in mix +22 dB

Line levels-

L, R, M outputs XLR balanced pin 2 hot <75 ohm, +4dBu, +26dBu max
Group (aux) out XLR balanced pin 2 hot <75 ohm, +4dBu, +26dBu max
Aux 1-10 output TRS impedance balanced <75 ohm, -2dBu, +20dBu max
Direct out TRS impedance balanced <75 ohm, 0dBu, +20dBu max
Matrix out TRS impedance balanced <75 ohm, -2dBu, +20dBu max <- This is what i use in peak i have 6 dB headroom left! On *output*. And in ATEM i have 10 dB analog headroom before clip. Period.

On mix bus i have in any moment at least 12 dB headroom. Business as usual.


... at the same time-

"plenty" of analog headroom ;)

IMG_5018.png (285.78 KiB) Viewed 3927 times

no comments. 6 dB LESS headroom in analog input (and ~4 dB before clip!). And, by the way, these inputs do not have gain control.

Analog and digital domains behave and can be driven differently. This is my point and I have no interest in further discussion either.

We have the same problem as most. Analog out of a Stereo Aux Send 9/10 on an AH GL3800 results in low audio to a Television Studio Pro 4K. All volumes are maxed out, and it's still low. We inserted a Powered mixer in-line, but it seemed to introduce a little bit of noise.

The Pro 4K does not have RCA inputs, just XLR. Any other ideas?

Hello,

Pro4K has Fairlight, use master limiter and markup gain and go high in digital. Do NOT run ATEM inputs hot, keep 10 dB headroom (peaks @ -10 dBFS)!

if drivers upgrade not present (console is in factory settings), GL3800's stereo auxes running 6 dB low (@-2 dBu)- if not using reverse (e.g. MAIN XLR outputs).

You can patch auxes back and route them out through group XLR outs. Then you can also use group inserts for dynamic processing. Do not do it in one stage, that sounds bad. Best practice is channel compressors + bus compressor + master limiter.

Check levels in digital domain, using LUFS meter. Keep peak level no more -2 dBTP, i'd keep spoken voice @ -16 LUFS integrated for streaming.

Best!

Thank you. Can I schedule time to walk through this with you? I am not the full time sound technician, but am trying to help. I was reading the GL3800 manual today, and recall some of what you referenced, but can I scheduled time with you, to make the changes live, or take notes while I reference the console?

I think best solution for you is get good sound guy I.R.L. and let him set it up.

We are having trouble finding a sound engineer. I will pay for your time, if you are willing to assist.