I use GarageBand to write music but in theory this question could apply to any composition and recording workflow.
A problem I often run into is that different audio players seem to have their own "flavor" of output. For example, whatever I write obviously sounds like what I intended in GarageBand. When I export it to MP3 format and listen in QuickTime Player, same result.
When I import the same file in iTunes, it seems to apply some sort of treble booster that makes the whole track sound tinny. Ditto on my iPhone (where the default music player is basically a subset of iTunes). In practice, I "fix" this with a bass booster or treble reducer in the Music -> Equalizer menu. I'd expect the default media players on Windows, Android, and Linux to have their own specific quirks.
My current approach is to test on a small selection of hardware and software that I have at home, although obviously it's impossible for me to test on every possible piece of equipment my music could potentially run on.
It gets even more difficult if I have to account for different types of headphones and audio equipment. Some headphones and speakers are well-balanced, some might be bass-boosted, and others might have low-quality tinny audio. The amount and type of padding between the ears and speaker holes on a pair of headphones will obviously cause some sort of frequency-based distortion or bias. I'd like to be able to hear bass frequencies on my music loud and clear on my headphones without throwing an equalizer into the mix. I'd also like other people, some of whom might use heavy bass-enhanced or noise-canceling headphones, to be able to listen to the same music without voiding the warranty on their eardrums in the 20-100Hz range. It's not as simple as "make all the bass instruments louder".
Different headphones have different audio leakage profiles as well - for example, the ones I use leak a lot of noise through from the outside. I consider this to be a feature rather than a disadvantage since it keeps me aware of my surroundings. Meanwhile, someone else might be using a noise-cancelling headset that blocks everything except the music.
On top of that, I'd like to build my music so it can be reasonably listened to in highly variable environments. Indoor or outdoor, and also on mass transit with a lot of possible engine noise polluting the lower audible frequency spectrum. Do NOT assume the listener is using a noise-canceler. My experience so far is that a bass booster or treble reducer is a must for listening on a noisy train, for example. Any way to fix this without causing the reverse problem when there isn't a noisy train engine humming loudly in the background?
The main question: Is there an industry-standard way to calculate frequency distribution and instrument volumes for the optimized audio quality across a wide range of hardware, software, and environments? Preferably a workflow built for a DAW but anything reasonable should be adaptable.
EDIT: Writing music is a hobby for me. My production budget is only the time I spend on it, the cost of my computer and associated hardware (to be fair, I use that computer for a ton of other stuff; GarageBand comes free with a Mac), and the electricity it runs on. I definitely don't have the budget allocation to bring a recording engineer on board, except as a volunteer.