Consider starting playing an instrument. Learning to play other people's tracks is a really easy way to build your "chops" for writing your own stuff. Also, check out Mussar's ear training podcast (https://soundcloud.com/mussarmusic/sets/the-mussar-method-bi-weekly) - this stuff can help you immensely, because it will give you the tools to analyze other people's music. At this stage of your journey, basic musical skills are MUCH more important to learn than how to make wavetables with Serum.

And accept that 7 months is nothing. Getting good will probably take you years, if you started from zero musically.

Fanu wrote a nice post yesterday, I highly recommend reading it (especially the parts "Love it" and "Don't try to handle all at once" are relevant): http://fanumusic.com/advice-for-producers/

Short answer: it makes no difference. Try it, you shouldn't be able to hear any difference whichever way you choose.

The reason is that sidechain compression just adjusts the gain of the input, independent of what the input is. EQ and gain adjustment (which is just multiplication) happen to be operations in which the order doesn't matter. (note: this doesn't apply to analog modeling EQs). For the same reason it doesn't matter in which order you chain EQs - have you ever asked in which order the different filters in your EQ are applied?

What Marrow said is true and very important, but it doesn't matter in this particular case. If you'd asked the same question for regular compression, the answer would be different.

E: now everyone can see how slow I'm at replying 

Mussar. Just give the guy a piano! https://www.youtube.com/watch?v=JCguq3hTC2M

If you don't have a spare piano, a kazoo should get him started. If he's got what it takes to be successful.

For some reason, your audio interface and DAW were probably working at different sample rates. That is, your DAW sample rate was 48000 while the 2i2 was working at 44100. So one second's worth of audio in your DAW (=48000 samples) would take more than one second (48000/44100 = 1.088 seconds) in the output.

There may be separate settings for these in your DAW's audio settings and the 2i2 driver settings, so you should make sure they match.

But yeah, like NP said there's not much reason to use 48000. Higher sample rates are needed in some processing (to avoid aliasing: https://en.wikipedia.org/wiki/Aliasing), but those are achieved by oversampling inside the plugins.

Try Softube Saturation Knob. It's a great freebie.

In.

All I was trying to say was as a response to this:

it does bring things out of balance, because you're adding volume to a particular frequency bandwidth.
that's why it's better to start out with subtractive EQ than to just go for additive EQ.

You could give the exact same argument FOR additive EQ.