The Nyquist theorem says that sampling at twice the highest frequency in the source will reproduce it perfectly. So 44.1 kHz will get to 22 kHz in principle. But it is critical that there be NO signal above half the sample rate, or it is aliased below the 22 kHz into the audio band as bad distortion. So, players must have a sharp low-pass filter in the stream. The problem with this is that if the amplitude response has a sharp cutoff, the phase response oscillates wildly. The phase is equivalent to delay, and this can affect imaging. So, if the sample rate is, say 96 kHz, you can make a nice smooth (e.g., Gaussian) filter that has a smooth amplitude and frequency response. But it doubles the file size. IMHO, the DVD standard of 48 kHz should be sufficient for flat response to 20 kHz. The number of bits, to my ear does make a difference, especially on loud congested music, for example a symphony playing many parts loudly and at the same time (ives Symphony No. 3). 16 bits gets congested. It is hard to have the instruments maintain their unique place in the soundstage. In principle, by doing some slights of hand (interpolating -randomly- between bit levels) CDs claim to be able to get 19 bits, which might be sufficient. And I have heard some very good sounding CDs. But not that many. Remember that with 16 bits, there are only 65,000 levels (half negative), so there is a inherent 1/325 % distortion due to imperfect representation of the sample height. I care more about 24 bits than 96 kHz, since I am old and am lucky to hear above 15 kHz.
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