Is It Wav or MP3

Sadly this ^^^^^^^^^^^^^^
MP3 was developed in a time when storage was a premium & content delivery was primitive (anyone remember dial-up?)
Today this isn't really the case, but the format is still used to cut costs for bandwidth usage & storage & those were its primary concerns at the outset & not the quality of the resulting audio.
It's kind of perverse to think that a codec was developed around an antiquated infrastructure & yet today most major tech companies design their audio infrastructure around the codec. That's not progress. Just laziness & greed.
Back on point though, the perceived quality of the resulting audio is largely dependant on the equipment you use to listen to it on. Sound cards in pooters, cheapo Chinese mp3 players, state of the art amplifier circuitry, car radios & so on will all have an impact on what you actually hear. Not to mention the state of your own hearing which naturally deteriorates as you get older.
If you're going to load your phone up with music to listen to either on the inbuilt speakers or some cr@ppy earbuds, then it makes no sense to fill that storage with 24bit hi resolution wav (or other lossless) files. You simply won't hear any difference.
If you take that same audio file set that you loaded into your phone/ mobile player etc. & play it on a decent mid to high end audio system at home, then it will sound tragic.
The thing is, not many people actually do comparisons. They'll just play a source file & assume that's how it should be.
Try it yourself. Get a good quality, lossless audio file of something with a good array of instrumentation al the way from the bottom to top end of the frequency spectrum.
Make a copy & convert it to a lossy file format.
Then play them one after the other on your home system without other things to disturb you (washing machines, vacuum cleaners & so on)
Play the lossy one first & then play the original after. It will actually sound fractionally slower. Not because it is, but because there is so much more information in the lossless audio that your brain takes a tiny bit more time to process it. It's like putting on an audio equivalent of a pair of glasses.
I found a really good comparison of a short guitar sequence that makes the difference so clear .
First sample is uncompressed WAV. Second is compressed vorbis. Third is compressed mp3
Pay attention the the ringing of the strings. On the compressed samples they a obviously degraded & sound quite "splashy" compared to the original WAV sample which gives a clean ring as no information has been stripped from it.
The algorithm used for creating mp3s makes a decision for "us" as to what we can & cannot hear & arbitrarily removes those sounds that an artist &/or recording engineer has gone to great lengths to include in the original source.
Fun fact!! The development of the algorithm for mp3 audio used Suzanne Vega's Tom's Diner & in 2015 a composer called Rym Maguire actually released a track called moDernisT (an anagram of Tom's Diner) which is just the audio signals that were stripped out of the original song

Ok steve I listened on my monitors to that file. There is a head slapping obvious difference and the wav is of course the best. The vorbis is wobbly and the MP3 is muffled. BUT is the MP3 320 kb/s ? Because that is where he is saying you can't tell the difference. Because there is 128 kb/s etc... and 24 bit vs 16 bit yeah?

BTW all of my songs I'm working on are in 44.1 Wav Right now.

I believe it's a very low rate, but as a comparison it's alarmingly obvious isn't it?
Naturally, the higher the bit rate used, the less obvious the difference, but it's still there.
As for 24 vs 16bit, that's something entirely different really.
It refers to the number of bits of information sampled. The higher the number, the better the resolution as you typically have a much clearer picture of the dynamic range (the difference between the quietest & loudest signals in the audio track).
Lossy compression has virtually no bearing on that. It only serves to strip "unwanted" information to facilitate a smaller storage requirement & better streaming performance for lower bandwidths.
If you work on your masters at as high a resolution as you can handle (24/48 or 24/96) then you have much more information to work with as you tinker with settings & effects etc. After you've done playing & are happy with the finished item, you can dither it back to 16/44.1 which will be more compatible with playback gear. !! Even working at 24/44.1 is going to give you a good base from which to work from. Effectively it means your sampling 24bits of data 44,100 times per second as opposed to just 16bits in the same timeframe.
But beware. The high res files will take up a lot more storage space.

Ultimately, it comes down to personal preference though. Your ears are yours & nobody else's.
But recording engineers/ artists try to cater for everyone's ears & the myriad systems the work will be reproduced on.

There are 2 popular cure shows that are frequently shared as fake flac. Almost no one ever complains. They were originally uploaded by David McQuay (pale amber glow) at about 2003 on his website as 160kbps mp3 files. The shows are

01.05.1982 London - Hammersmith Odeon

30.05.1984 's-Hertogenbosch (complete AUD)

The latter was pitch fixed but AFAIK the source is the same fake flac also.

David himself of course didn't share these shows as Flac - someone downloaded them and started to trade (?) them as the real thing. I have the original mp3's of 30-05-1984 floating around somewhere if someone wants proof.

Very clear explanation. I would add another variable: recording/mastering/remastering process of the track/music itself. At some point, every late remastered music (including The Cure of course) is really damaged in frequencies. So, in this point it would no really big differences in WAV or MP3. (I hope my english is good enough...)