Snash,
You are not the first person to think of subtracting to find-out what's lost with lossy compression, but alas,
it doesn't work. 
Yes, you
can find (and hear) the
mathematical difference, but this is not the
audio difference or the data that was "thrown-away"
QUOTE
I got to thinking about this when reading about joint stereo and how information from the channels is added and subtracted from each other.
FYI - The sum & difference files are created
before applying the lossy compression.
With lossy compression, the precise timing & phase are not preserved. As [JAZ] suggested, uncompressed audio is stored/represented the
time domain. For example, at 44.1kHz, you have one sample (one "sample point") 44,100 times per second. (See the
Audacity Introduction To Digital Audio.) Once you understand how digital audio is sampled, it's easy to see how
a very small time-shift can throw-off any addition or subtraction. It doesn't take much time-shift to introduce enough phase-shift that you end-up "subtracting a negative number", which of course results in
addition!
In fact, by time-shifting a waveform and adding it to (or subtracting from) the original, you create a
comb filter. If you listen to the original and time-delayed file, they
sound identical (because the actual audio
is identical), but when you subtract the two files, something "funny" happens! (This is not
exactly the same thing that happens with lossy compression... You do not get a comb filter when you subtract the compressed & uncompressed files... It's just shows the
kind of things that can happen with time/phase shifts.)
Another example... If you invert the phase of a file it will sound identical to the original. But then if you subtract it from the original, you are "subtracting a negative" and you will double signal level, and perhaps get clipping (distortion). What you are hearing is the
mathematical difference (assuming no clipping), which doesn't tell you anything about the
audio difference.
You can actually play around with
M/S coding using Audacity (or other audio editor)... You can open a WAV or MP3 file, create the sum & difference channels, save that file,, then re-open it and add & subtract again to re-create the left & right channels. (If you're comfortable with algebra, you should be able to figure-out the details.)
There is one thing... When you add and subtract (mix*) the left & right, you need to divide by 2. This is because some samples will already be at, or near, 0dBFS (the digital maximum) and you can get clipping after mixing. Then, when you re-construct the left & right channels, you'll need to multiply by two to restore the original signal levels
You can also try
subtracting a file from an exact copy of itself... You will get absolute silence! And, you can try time-shifting before adding/subtracting to create a comb filter.
* Audio editors don't usually have "add" or "subtract" functions, but
mixing is done with addition. So, if you want to add two files, you use the
mix feature/function. If you want to subtract, you invert the phase of one file before mixing.