Does sampling ~> Nyquist frequency capture harmonics? In a debate on the value of 24b/192KHz consumer audio (using a good brick wall filter, minimal jitter), a friend posed the argument:


  Nyquist never properly dealt with harmonics, which change sound
  pressures by riding a higher frequency on a lower frequency "carrier",
  increasing the propagation of the higher frequency and lending some
  extra punch to the lower frequency.


A post from ComputerAudioPhile.com poses the same:


  So let's say you have a flute that produces a tone at 14kHz. It may
  then produce overtones or harmonics at 28kHz, 42kHz, etc. If say the
  28kHz overtone is sufficiently strong, and you truncate at redbook
  sampling frequency (corresponding to a 22kHz signal), then you might
  get an artifact at 22kHz - (28-22kHz) = 16kHz, which, if you are young
  enough, might be audible.


My Nyquestions are: How does this actually work? Which papers should I read?
This is called "aliasing" and it is simply a result of periodicity of sine.

$\sin ak$ = $\sin (a+2b\pi)k$ (for integers $k$,$b$ - $k$ being the sample index)

So, if you maybe have $a=1.3\pi$, you could also see it as a signal at $-0.7\pi$ (a signal of negative phase at $0.7\pi$), or rather: you will not see any difference in the samples.