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Maybe something is summing your mix to mono before it gets to the speakers.

If you mute one of #2 or #3, does the sound come back? If so, can you confirm the one you hear is actually panned? It sounds to me like the two tracks aren't in fact panned. Are you recording the figure-8 side mic to a mono track?

People often misunderstand how attack and release work. They don't relate to the threshold, they relate to the target output level vs. the actual output level at any particular moment. The milliseconds marked on the attack and release controls aren't absolute times, they're the time it takes to...

Not quite. The way it works is that the first number represents the input level over the threshold and the second number represents the resulting (target) output level over the threshold. With a 3:1 ratio, if the signal is 9 dB over the threshold, the target output will be 3 dB over. I say...

Any audio editor or DAW can do that. You'll probably want to properly master it as well.

Free jazz, because no one would pay to hear it.

Correct pitch isn't necessarily the best pitch. A lot of the expression of singing is in manipulating pitch in ways that are technically incorrect but creatively superior. I don't think it's just robbing singers of the opportunity to grow, it's sanitizing their performances of expression.

You could try Equalizer APO, which operates in Windows, outside of your DAW. It apparently integrates with REW, importing correction files. It might not do that in ARC, but you could set the filters manually. I'm still of the opinion that acoustic issues should be fixed with treatment. Filters...

At best, you can correct for one point in the room, and no amount of eq will fix a deep null caused by 180⁰ out of phase reflections. Minor nonlinearities in system (i.e. speaker) response seem like valid targets for correction.

That's almost literally one of their songs. "And if the band you're in starts playing different tunes, I'll see you on the dark side of the moon."

I'd optimize levels in the console and skip the MicroLimiter. But if you use the Alesis, optimizing levels in the console will minimize any noise that the limiter adds.

Probably a basic 75 Ω RG59 coaxial cable terminated with BNC would be fine. Connect output to input. You may need use the Presonus app to set the receiving unit (slave) to sync to external word clock, specifically the BNC input (rather than the optical input or the internal clock).

Right, and the slave could also get it from the ADAT signal over the optical connection. But BNC is a more reliable and direct way to get it. I tend not to trust optical to be as rock solid as BNC. Also, it seems to be the case that Aggregate Device resamples the signals from the slave devices...

It's used on that unit to transmit and receive word clock, which seems like a more direct method than through the indirect path of USB.

Conglomerate device is a good solution. I might still clock them via the BNC.

The dual ADAT connections are for more channels or higher sample rates. I don't know that particular unit, but you most likely have to use the software to specify one as clock master and one as clock slave, and also specify which one is the signal source and which is receiving signal. You...

Many times when it says 30, it's actually 29.97. But if the OP is just doing audio and not doing anything that changes the play speed (like stretching and resampling), it should line up. I've noticed with mp3 files, the audio can be offset a few milliseconds, enough for me to notice.

29.97 fps is the standard frame rate for US video, has been since color was added to the standard. 30 fps is a rounded up approximation. Very often (but not always), if something says 30 it's actually 29.97.

Yep, it's basic resampling, no fancy pitch processing. Let the pitch follow the playback speed adjustment.

GIve it a try, it might be okay. If it drifts, it's fixable. It helps to have a clap or something at each end to line up.