Digital Music Sounds Flat !

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And distorted! Eh. I try Spotify and I could spend hours there, but I just can't get over how bad the sound quality is. I like to listen to my record collection on my turntable when I have the time. I guess mp3s are the McDonald's version of music listening , but it just sounds awful.

Yeah, MP3s can sound very "trash-can". It's possible to fix them up somewhat with EQ. iTunes has (or at least had) a range of options and presets that fixed many a track for me, but they still can't compete with LPs and real EQ, except for not having to get up to switch out albums.

Get a DAC for your computer. I have the Geek Out 450 by LHLabs. It makes ANY music coming from my computer sound amazing. Youtube, CD, Mp3 etc.
Just don't use a DAC with iTunes, it compresses everything it loads. JRiver media center or Foobar2000 are the best and easiest to use and sound amazing.

Several reasons as to why consumer Digital Music sounds so bad:

1.) The 44.1Khz sample rate and 16 Bit wordlength developed by the RIAA in the 1970's for digital music was deemed sufficient to pass signal from 20Hz to 20Khz. Condenser Microphones and high end Studio consoles from the 1970's were capable of tracking frequencies higher and lower than that. --Neve and SSL mixing consoles could amplify signals from 0Hz to 100Khz. Tape decks used for recording had to be Bias adjusted to the tape formulation being used for the recording, and Tape heads had to be alighed correctly. Still though Analog tape decks did NOT record flat frequency response. Studer decks had a bump in the Low frequencies when recording at 15 IPS, and had extended HF response and a corresponding LF loss while recording at 30 IPS. Furthermore, Analog tape was similar to a tube amp when tracking hot signals as it would compress the dynamic range of the signal. Digital, OTOH does none of this, but because of the low RIAA specified sample rate loses fidelity when tracking higher frequencies which makes the HF sound harsh and cold compared to analog. --That particular RIAA spec is simply insufficient to capture all data being recorded.

2.) Furthermore, until the 1990's digital recordings were recorded at the same RIAA sample/bitrate as the CD/DAT playback mediums. What further degraded the quality was that the sample clocks on all digital equipment that passed information were not syncronised togeather which caused further data loss. This data loss from uneven clock frequencies is referred to as 'jitter', which not only caused the high frequencies to sound harsh, but also produced strange harmonics all throughout the frequency spectrum which made digital recordings sound lifeless and artificial. Modern recordings are made with between a 96Khz sample rate and 24 bit resolution for semi pro gear, and 192Khz sample rate at 32 bit resolution for professional studio recordings with all gear that passes Digital signal sycronised to a master clock.

3.) SACD was a format that came into being @Y2K which was essentially 2 channel audio recorded on a DVD at 192Khz/32 Bit format as in a recording studio. This format was finally sufficient to track exactly what was being seen from the recording console, however the format essentially died because the market shift was away from packaged music to downloadable formats such as MP3. MP3 is a 'lossy' data compression format that leaves only the minumum required audio data. While this provides a smaller file size than the PCM data of a CD, it leaves audible artifacts from the data compression that further degrades HF fidelity and adds it's own noise footprint on the audio which can be fatiguing to listen to at bitrates below 192Khz. .FLAC data, and high resolution .WAV data formats are much better than MP3, and of the two, the latter provides the same resolution as an SACD, as .WAV files are what are now utilised in studios, and nonlinear editing software, however, nobody wants to download and store 4GB of data for an album, when they can get what they percieve to be the same as a CD in a 120MB package.