When downsampling or recording, and when playing back 44.1 kHz audio, the data must be low-pass filtered to eliminate aliasing effects.
But filters aren't perfect. Even a decent low-pass filter (say 3rd order Butterworth) requires an order of magnitude of bandwidth to drop the output 60 dB (practically inaudible). This means that with a Nyquist limit of 22 kHz, you're either attenuating everything above 2.2 kHz (the "knee"), or you're letting some aliasing noise through.
With a 192 kHz sampling rate, the filter's knee can rise to 9.6 kHz, and the stuff between 9.6 kHz and 20 kHz won't be appreciably attenuated.
It's important to note also that this attenuation can't be fixed by simply boosting the high end -- filters are linear, so such an adjustment (with an equivalent order filter) would merely cancel out the low pass filtering and reintroduce aliasing noise.
(Edit: I am not an audio engineer but I have a strong signal theory background. So actual audio engineers please feel free to correct me.)
But filters aren't perfect. Even a decent low-pass filter (say 3rd order Butterworth) requires an order of magnitude of bandwidth to drop the output 60 dB (practically inaudible). This means that with a Nyquist limit of 22 kHz, you're either attenuating everything above 2.2 kHz (the "knee"), or you're letting some aliasing noise through.
With a 192 kHz sampling rate, the filter's knee can rise to 9.6 kHz, and the stuff between 9.6 kHz and 20 kHz won't be appreciably attenuated.
It's important to note also that this attenuation can't be fixed by simply boosting the high end -- filters are linear, so such an adjustment (with an equivalent order filter) would merely cancel out the low pass filtering and reintroduce aliasing noise.
(Edit: I am not an audio engineer but I have a strong signal theory background. So actual audio engineers please feel free to correct me.)