Yikes...
If it's dialog, it will be tough, but you want to look at physical modelling to get both sounds to sound like a common denominator. I would start with fine tuning the EQ so the wave forms match, then add some ambience, convolution or reverb to try to thicken the cam mic to match the real mics. You are going to need some NR as well. You will need super short reverb, and, even then, it may not work.
Camera mics use heavy compression, so you can experiment with overdriving the compression on the good mics to get them to sound a bit closer to the cam mics. You can also add a bit of noise as you are adding compression as cam mics drive up the noise floor. If you have the option to pan, try narrow panning the good mics as the cam mics will be narrow unless there is some weird surround thing going on, like with Sony. You can also invert one of the channels which is going to add some cheese, if you you want that, if you can't get it narrow enough.
A lot of convolution programs have "rooms" or spaces. You can put both types of audio into the same space, to try to make them sound more similar. Again, don't use a long tail, work with the density.

Can try https://www.izotope.com/en/products/mix/neutrino.html

I mean if you want something other than EQ.

@hardimpact Thanks. I'll give that a try, and I hope I can learn to understand it.

You shouldn't ever be using the tracks on the camera itself, they only exist as scratch reference to line up with your real audio from your Tascam recorder.

I know that with samplitude you can copy an FFT dynamics snapshot of one recording, then paste the filter onto another recording to help match up the dynamics

Add in some ambiance to blend in the cuts together.

I agree, compression should always be the last or second to last step. I put them in order of non destructive.

1. dc offset - offset power errors affects dynamic range so this goes first, a critical pre-alignment step
2. match volume - normalize 'peak amplitude' everything to -6db to see serious errors easier like clipping, offsets, phasing etc.
3. declip - run declip effect, remove buzzing with dynamics processing, notch filter the rest.
4. equalize - run parametric equalizer to reduce wind, hiss via freq cut, give vocal clarity 'D' curve
5. phase correct - once tracks layed, detect any phasing errors
6. noise removal - declick, de-reverb, remove hiss, wind, learn sound model/noise reduction, all else healing brush
7. studio reverb - to add vocal weight
8. RMS -23db audio - to roughly match vocals begin basic audio editing with levels(vocals center and 12db higher than music, etc.)
9. compression - multiband compress sfx: 4:1, vocals 2:1 db to give pleasing db range. always 2nd to last step because any other processes affect dynamic range. It should soft limit for you. hard limits are audible and not recommended.
10. quality control - match loudness ITU-R BS.1770-3 -23 LUFS film, -16LUFS youtube - for standard

most compression is destructive. the closest 'undo' you can do is audition's effect-dynamic processing acting as an expander/compander. The quality depends on if the audio was 16 or 24 bit, similar to 8 bit vs 10 bit video. Also, if the audio was hard limited, the quality had possibly been reduced as well.

@hardimpact I would not normalize the volume until you balance out the compression (using micro edits), otherwise you are just boosting the noise. Also, I would recommend using some sort of physical modelling to match the cam sound to the mic sound, or match both sounds to an intermediary sound space.
@Brian_Siano although you might be able to do what you want to do in Audition, to get a somewhat better result you probably would have to use a real DAW, or just pay someone to process the audio for you.

@spacewig Yes, that's exactly the situation: shotgun mikes for most of the project, camera mikes for the rest. They're stereo, too.

I'm going to try to use Audition to plow on through a rough sound cleanup. It won't be perfect, but it'll be better than what we currently have. Eventually, I'll have these skills nailed.

@hardimpact it might be that normalizing is lossless in Audition, but I've never tested it. For example, it may automatically add dither, I have no idea. I've never gotten good results from it, but I don't use it that often.
But the question is how to handle the basic problem with in-camera compression. You could use or refine a ducking limit, so some parts of the background noise would automatically be set below the threshold, but I would just take the time to go chunk by chunk, split each part into audio and background with a crossfade for each bit, and then smoothly set the crossfade for each part so it sounds natural, before introducing any processing. Each background chunk would then have a custom amplitude, set 2-6 dB lower, and then when you process the audio you mainly boost the "good" part, and any dither simply adds detail instead of adding noise. You may need some light dither to help with the compression and convolution, but you only want to add it once. If one tested the system and you could see that the normalization was lossless, sure, do the micro edits after boosting it, but even on a high end DAW I would trim the audio before processing. By using the microedits you are reversing what the camera did, which is to boost the low level signals. It won't effect any clipping or brick wall effects, and it won't sound as good as a regular recording, of course.

anyways, I was directly comparing izotope rx 6 to audition with noise removal. audition worked well with certain FFT sizes like 4096, but others not so well. I usually work backwards because its easier to hear if you're on the right track by ramping up the intensity of sound removal to max, then output noise only. this way, if I hear any dialogue, I am not on the right track. then I simply need to try other settings.

Anyway, the main reason I am posting is that RX 6 advanced had all these new dialogue filters and they all seemed very similar. Apparently, they were each machine learned by a different way; one by dialogue vs footsteps and traffic, another by random broadband noise like radio static, and one just for wind. As it turned out, I had one file that had all three problems.

So I decided to try them out. They all worked reasonably well, at least as good or slightly better than audition. but I was wondering. What if I tried all three. What order would they go in? Would it make it worse overall from distortion? What I found out is very interesting.

I've found that if you run dialogue isolate with a very low setting like 1.5 separate and -6db reduction. then do voice denoise, adaptive -6, then all that's left is a nice voice with no noise except the wind, then the de-wind works 10x better. Now you're left with clear dialogue with zero distortion. So, apparently you need all 3 in that specific order. I tried other orders but my instincts were correct that you use the least damaging filters first, then all the excess noise falls through the chain in a certain way.

This is also how I use audition and grading as well

Your thoughts?