Ask an Audio geek

[Alex K]

Since it is very hard to build a filter circuit which effectively shuts off the signal at 22 khz while still leaving it untouched at 20khz, and the ones you can build from ordinary analogue circuitry have strange properties, it is better to use an AD-Converter which operates at a much higher sample frequency and use a more moderate conventional filter on the input side. When converting back down to only 44100 samples per second via number crunching in order to press a CD, one can use much more precise computer-based digital filtering operations which can cut frequency very steeply without producing weird phase shifts. That way, one can basically avoid aliasing without having to use a conventional filter circuit with insane steepness which ruins sound quality.

Aliasing, if you haven't heard it, is something you know from movies - when a spoke wheel turns fast enough on film that the next spoke replaces the last one from one frame to the next, it will look as though the wheel stands still. When it turns even faster, it will look as though it turns slowly backwards. The same happens with audio sampling. If you sample at 44100 times a second, a signal with 30000 Hertz will result in a fake recorded signal of 16000 Hertz, in the hearing range. This is why one needs to very strictly filter high frequencies out before sampling. If you don't do it, ultrasound gets converted to horrible noise, and if you do it badly, like in some early digital recordings, it sounds like crappe because the filters distort the signal in the hearing range. Again, with digital filter algorithms working with high precision numbers, maybe even floating point, this is not an issue any more.

The only objection one might have against a perfectly realized 44.1 digital recording, apart from quantization noise due to the 16 bits, is that it really brickwall-filters everything above 20khz or so, whereas any analogue equipment will roll off more smoothly. I don't know whether that might change perception of the sound, but I doubt it.
In any case, it is something one can arbitrarily improve by putting higher sample rates on the medium. If we go to 96k samples per second as is already done in many places, the hard cutoff is ideally at 48 kHz, so far away from the hearing range that I strongly doubt that there is a chance in hell to hear the difference between a slow rolloff starting near 20kHz or a hard cut near 48kHz.

But there definitely are enough things that can go and have gone wrong to let a CD sound objectively worse than vinyl in ways that are qualitatively different from the imperfections one gets from analogue recordings, and it needed a lot of sophisticated technology improvements and digital signal processing techniques like those I mention above, to really reach near the mathematical ideal.