Thanks! I always help if I can. You can find my mixing tips all over the place within this forum. There’s not much I can say here. My basic advices for mixing are:
That’s it. Need more details? Or is there anything else I should mention?
@Jek Thanks for asking these questions! Thinking about them gave me some stimulating new ideas! 
The perceived left-right placement in a stereo setup can be manipulated in at least two ways:
When thinking about panning, one thing to notice is that a pan knob can work in a few different ways. As an example the pre- and post-panners in Renoise’s mixer/track DSP chains work differently. If memory serves me right, the pre-panner only controls the volumes of L and R channels, while the post-panner also includes crosstalk, so that in the extremities you essentially have a sum of the L and R panned to one channel. I have a collection of panning doofers with different panlaws that work similarly to the post-panner, since the post-panner is not automateable and I don’t know where I could set the panlaw. I don’t think there’s a setting for that in Renoise?
If you think about it, the “stereo field” is just an illusion of perception. It’s just two mono channels, no matter how you look at it. You can separately process left, right, sum (mono/mid) and difference (sides) of a stereo signal, and you can do some fancy spatial trickery with delays, reverbs, filters, convolution etc. But I really can’t think of a way to apply processing only to say material that’s panned >75% left. You would have to be able separate that somehow from the stereo signal. I’m not aware of any way to do that. As @slujr suggested, maybe some advanced mid-side wizardry could be used here? Or maybe someone could cook up some neural network that could pull it off in some way not humanely understandable? IDK, wouldn’t be the first time AI did something I didn’t think was possible. Or maybe some Dan Worral -type polarity flipping madness could be used here? Panning to the desired position and extracting the desired part of the signal with some polarity flipping trickery? That could very well be possible, and it’s probably the most fruitful direction to start poking at if you really want to do it. 
Now, while I don’t know if what you’re after is possible, you can still use parallel processing in some fancy ways to effectively pan you FXs separately:
I haven’t ever actually tried this though, it just came to mind while reading your question. But I don’t see why it wouldn’t work. Though maybe it’s just a more complicated version of having a simple separate 100% wet parallel processing channel that you can pan where you want to. 
So maybe not worth the effort 
Yes, the formula device can be used to compare inputs and make the output do what you want based on the comparison. Basic mathematical comparisons can be done as you’d expect:
< and > work as expected. “Equal to” is == (returns true or false),
<= and >= work too (I’m not 100% sure which symbols comes first though). If you’re looking into specifically comparisons of amplitudes of two different signals, you can hook two signal followers to the A and B inputs of the formula device. I’ve found that “if-then-else” structures are a true do-it-all workhorse when comparing signals to each other. Wrapping everything into a function and calling that function as the output of the formula device has always worked wonderfully for me.
I’ve been messing with some native dynamic convolution this past year and I’ve split my signal with these basic mathematical symbols and "if-then-else"s, based on either the actual amplitude of the signal (signal follower), or the note velocity (velocity tracker), basically just setting up a single macro to control which FX chain is fed and how much. So yes, you can absolutely split your signal to different FXchains, even different tracks based on volume/velocity/some other parameter or rule you make up yourself. Some kind of comparison between L and R channels is one possible output, and you can also write very specific rules about how the comparison controls your sends or dry/wet percentages or whatever in each channel. If you figure out a way to separate the portion of your signal to which you want to apply your effects, what you are searching for (if I understood it correctly) could be possible.
OK, here are my general principles for mixing psytrance, applies to many other dance oriented genres as well, with some exceptions
•mono kick and bass, at least below 200-300Hz. Sometimes I will have stereo fx and width on my bass above this range. make sure your kick and bass aren’t competing with each other in time or frequency content by sidechaining/volume shaping/using syncopation/phase remediation - Mfreeformphase and an oscilloscope can help with this last part
•HP and LP (almost) everything except kickbass. I usually let synths play through from 200-500Hz on the bottom to around 9-11KHz on the high end, sometimes higher if I want the air. percussion like high hats get to ring through on the high end. this helps keep things crisp
•check the mix in mono vs stereo throughout. things can sound very different. make sure your mix sounds good in mono - especially if it’s meant for a big system. I have a dedicated midi trigger set to mono my master channel so I can check this easily and quickly
•anchor your most important elements and mix everything else up around them. I usually anchor my kick around -12 db and my bass around -13db, then snare against those, then lead synths against those… if you can get those 4 elements sounding good together, the mix is a long way towards being decent. mix other elements up to sit well within that framework. I use a lot of maximizers to control peaks at every stage. clippers and limiters can work well here, too. use your ears to make sure you’re not breaking up the sound too much, unless that’s what you’re after.
•minimize competition in the time and frequency domains. this can be done through arrangement, sidechaining, eq, volume automation, etc. be conscious of what parts of the frequency spectrum each element is dominating. eliminate mud and confusion as much as possible. still very much working on this one, lol
• it’s much easier (for me) to focus on a solid mix from the beginning of composition and make improvements as I go, rather than separating out composition from mixing, Other people like to write first and mix later.
•watch your peaks, use the spectrum analyzer religiously. compress, limit, eq, maximize, but keep it clean - unless you want it raw/crunchy/dirty - in which case, things become quite a bit easier 
•mix at low volume, checking the mix on all your options for monitoring - headphones, monitors, auxiliary speakers, etc. different problems will become apparent with different listening setups. aim for balance, with the mix sounding as good as it can across different setups. if your mix sounds good at low volume in mono, it will likely sound great at high volume in stereo 
I’m sure there’s much more to add, but this is a decent place to start, imo. like any rules (or guidelines/suggestions, in this case) know when to break them creatively 
Yeah, I’ve made some channel strip doofers in the past, but prefer to have the devices more modular in each track these days
You nailed it with your tips. I would claim that this fits to any genre. 
Shortlist of my tips, which are “surprisingly” pretty much the same than yours:
That’s the obvious stuff, right?!
Some of my personal rules:
I’m somewhere in the middle. If the beat is mixed well right from the beginning, all the rest is a “piece of cake” and will be mixed mostly automatically the right way without thinking about it. So I check the beat first (or at least kick and bass), then I focus on composing (headphones all the way) and at the end I switch to speakers and start doing the whole mix by mono mixing first. But sometimes I start composing through headphones and do the mix afterwards when the song is finished. But in this case it could happen that the low and high ends completely suck through speakers. In my opinion mixing through headphones only is fucking impossible, no matter what all the pros like Andrew Scheps are saying. There’s a big difference between headphones and speakers.
when you do parallel processing do you use eq or filters?
I usually use the crossover filters in the multiband send itself, dialing in crossover frequencies and send amplitudes to color the sound prior to further processing, where I might use filters or eqs, but more likely spatializing effects like chorus, phasers, and delays in conjunction with compression and distortion, depending on what I’m after. Oftentimes, the parallel compression is the thing I want
here’s what I mean by using the multiband send to color the sound:
you can choose different filter types in the “type” field.
obviously, each send would be going to its own fx chain…
I use them everywhere in mids and highs, lol
but if mono sounds good, I don’t sweat it
everything is a tradeoff, afaict
but yes, taming such dsp is an ongoing effort
Not sure if this goes here but needs to be talked about more:
(I mean, mixing starts at good sound selection)
I felt stupid finally answering my question as to why things like a saw wave sampled at C-4 loses the top end when pitched down, why aliasing happens, and why my instruments sucked. I was obsessed at first with sampling just certain notes of plugins (C-4, C-1, and C-0) when building instruments which is extremely restrictive as you are just sampling a certain limit, and if your sample has envelopes or a lot of harmonic movement, it may not end up with the right timing/env at the target keyboard section you wanna play your instrument.
So, audio with fewer harmonics can be sampled at higher notes than fuller audio without losing information when pitched down or introducing artifacts when pitched up
A few wee things, nothing major;
If you have weighty guitars in your mix, putting a signal follower on your bass and using that to drive a dynamic EQ on the guitar bus (as a gentle HPF) will tighten up low end in busy sections while letting the guitars sound full-bodied on sections without bass.
A single vox track often has more impact than double tracked vox.
If you have a car, you have a vocal booth.
Zensphere and TNT have provided brilliant advice above already. Particularly agree with mixing in mono. I think I saw a video years ago about mixing in Reason and the dude demonstrated mixing in mono using only HPFs, LPFs and the faders. Seems like a good way to start a mixing session in my opinion.
Finally want the ultimatively clean, defined reverb in a mix? Then I think there is no other way than isolating the wet sound of the reverb and to process it separately from the dry signal.
How to do this? I like to use groups for this. Put your tracks in a groups to be reverberated together. Make on each track a send to a dummy channel that will output to master. This is the dry signal without reverb. Make it a “keep source” send. You can put a gainer behind it to be able to control the send amount to the reverb, and at this place you can also later add EQs or filters to preprocess each track signal that would go to the same group reverb. Then on the group, put your reverb/convolver/whatever ambience you want to process in a wet isolated mode. Just make it 0%dry and XX%wet, the wet amount being the one you would have used when just using the reverb in a track passing through the dry signal.
So you will now see that the dry signal and the wet of the reverb add together on the master, and the sound would be like if you used it normally in a track. But now you can put an EQ or filters or any other effects on the group track to ONLY affect the reverb. You can heavily EQ JUST the reverb, to make it fit in the mix, i.e. removing all frequency parts of a reverb that make the mix muddy, or accentuate ranges that are desireable to “stick out”. You can use the stereo expander to control the width only for the reverb. You can put modulation effects like phaser or chorus before or after the reverb to add some depth and modulation. You can stack any amount of delay or reverb stages and have their effect isolated, just of course the last stage must pass through 0% of the dry signal. You can add exciters or other distortions to make the reverb more rich and stick out or shine, or to let the bass rumble and to extend that effect with harmonics. You can make a dedicated sidechain send channel just for the reverbs, and send the 0% dry reverb signals all from the group tracks to there, to make a sidechaing affect ONLY the reverb, but not the dry signal. Phew…lots of ways to shape an ambience so it fits just nicely into the mix.
Of course, if you do not like the group track solutions, you can just make an extra send channel for the 0% dry reverb, and send with “keep source” mode to there from your tracks. This is probably the more efficient solution if you want to reuse a reverb device from different tracks in your song, and if you do not want to use so many devices. I really like the freedom to use a custom tuned reverb device for every group of instruments that play together like a voice or in similar context, and really fine-tune it so each kind of sounds have their defined shape in the mix, and so that their ambience fits right in without obstructing other sounds too much.
Another pretty simple but useful tip I would give is to apply EQ on specific resonances on most of our instruments (mostly beats and particularly cymbals) both to clean out the sound, and leave space to other instruments that may have better sounding things at those frequencies.
To do it it’s very simple, just use a very-sharp EQ curve and put it all the way up, then browse every frequency level until you find that “harsh” resonance sound. Reducing it from only a few dBs is okay most of the time.
If you have trouble understanding what I’m talking about, just take any cymbal sample and try it out, you’ll find the specific resonance pretty easily. Then try cutting it out and see the difference. (Even better if part of a complete track to hear the space it leaves to other instruments/percs)
Spectacular advice. I’m curious @TNT or @slujr if you could give any insight on taking the step from having mixed in mono to adding stereo interest.
Panning seems fairly intuitive (to the same degree as getting your anchors leveled correctly, say), but adding width is where I struggle. I generally only add width to a few elements, and I don’t think I ever set the surround on the stereo expander to more than 50% (unless I’m doing specific mid-side processing).
I also a/b compare to mono to make sure the sound I’m widening can “take the hit” in mono mode.
are there any more advanced / purposeful methods that people take with “adding stereo” to a working mono mix?
Absolutely! But not only the selection of sounds matters. The sound quality is pretty important, too.
Yup. All you need is a stereo expander adjusted to mono in your master, then you just have to adjust the volumes of your instruments while checking through speakers at low volume. At least that’s how I start to mix.
Yes, this is the best solution in case you want to have a same-sounding reverb on several instruments. According to professionals this reverb should always be 100% wet. Personally I prefer an amount I like, it doesn’t need to be 100% all the time, even though it mostly is. But I have to say that I usually use this method to gain some “air” in the background. I prefer to have different reverbs on several instruments.
Yup, this is recommended if there are some disturbing frequencies in the sound. But I would only do it if the sound really doesn’t sound as expected and intended. But the native Renoise EQ isn’t the chosen one for such adjustments. I would recommend the Melda EQ, which is free.
Hm, not sure what you’re after, there is not much I can tell. All I do is to put a Stereo Expander in mono in my master track and activate it. Then I do all the mixing stuff as described. When everything sounds good in mono I simply deactivate the Stereo Expander and everything is stereo again. Last thing to do ist to finetune the mix in stereo. That’s all. Just like you I only “add stereo” (expand up tp 50%, surround max 15%) to specific instruments, primarily pads (gaining a “sound carpet”). When it comes to short synths or percussions I tend to use stereo delay effects and not a reverb. Is this what you wanted to know?
A lot of the stereo interest is on a per instrument basis in my music. I do a lot of randomized panning on rhythmic sounds so that they move across the field, which doesn’t negatively impact the mono signal. One thing I’ve experimented with is doing a mid/side process of a finished mix and using a stereo expander on the side signal’s mid and high frequencies. I’ve liked the result, but ymmv
stereo fx like flanger, phaser, delay (as long as the left and right delay times aren’t identical), and chorus could also be interesting to experiment with adding to side mids/highs, but I haven’t tried it. might be trash
bumping eq here also makes the side image pop, but you probably already know that
Why exactly do you think the Renoise EQ is not right to do such things ?
I’m not sure what you mean, my question wasn’t regarding EQing, and not as far as I know :^)
Exactly, same here. Panning on percussions, short synths, fx like delay output or reverb, no problem at all. The volume remains the same. The only problem I’ve noticed is when you want to expand the stereo signal. The more you expand the louder it gets in stereo, but it doesn’t affect the mono signal. I always bear that in mind while mixing.
I didn’t say “not the right one”, I said “not the chosen one”. You can do that with the Renoise EQ, too. But the Renoise EQ only has got a small window and is not as accurate as the Melda EQ, that’s why I prefer the Melda EQ for surgical interventions.