Nowadays there exists audio files recorded in up to 96 kHz with 24 bit samples.
Nowadays there exists audio files recorded in up to 96 kHz with 24 bit samples.
So sharp low-pass filters are extremely difficult to implement as analog filters with reproducible characteristics.
Even when they are implemented as digital filters, as in most modern equipment, i.e. the audio ADC uses a much higher sampling frequency and its output signal is interpolated and decimated digitally to 44.1 kHz, it is still difficult to design the very sharp low-pass filter so that it will not introduce any audible artifacts on transient signals (the sampling theorem is based on ideal non-realizable filters).
Raising the sampling frequency to be distant from the 40 kHz minimum value removes all difficulties with the design of the low-pass filters making it easy to guarantee that the digital signal reproduces exactly the analog input.
When the audio CD format has been designed, the digital technology was much less advanced and storing a lot of bits was a far more difficult task than later.
So they have made the trade-off of requiring very expensive analog low-pass filters in order to minimize the amount of bits stored on the disc and transmitted over the digital interfaces, because the digital processing and storage was even more expensive anyway.
Nowadays, it is much more convenient to just use higher sampling frequencies for audio signals.
You will however need to accept delay from the lowpass filter, which is fine for music reproduction.
So while these effects are measurable with very expensive equipment, they are in all metrics orders of magnitude better than any analog equipment.
24/192 is becoming common, and you can even find 384 kHz files for sale (transcoded from DSD recordings.)