My first post incorrectly referred to software brickwall filter. I didnt proof read sufficiently. I meant brickwall analogue filter which is what a steep low pass filter is. I wrote it when I was with a friend for a lunchtime "few" beersYes you're right about that first gen CD players had awful D/A converters and by 88 they'd got a lot better.
The example about the sinewave isn't totally correct though - bare in mind that the signal is already digitised on the CD. It's already been captured - it can't be made "smoother" afterwards, thats like trying to zoom in on a digital image. What you're talking about is oversampling at the end of the DAC chain, the sinewave example is talking about capturing. Most records aren't even tracked at higher frequency rates, the industry decided it was basically a waste of resources and that you could get the same benefit with plugins individually by super sampling within the signal chain (similar to your DAC example). This didn't change the file tracked at 44.1khz but the theory is that it would capture the processing of the plugin effect itself in more detail instead. In same cases it offers other undesirable results though, it's not universally declared to be a good thing.
It seems like you did a bit of a Google inbetween this post and the last one because you were talkings some crazy stuff about CD players having software brickwall limiters in them 🤦♂️ this time it makes a lot more sense.
Even though this is vastly off topic from original point. I still know you can't do any of those blind tests posted earlier with any DAC you want to use in the world and get better than 50% (eg a guess) because I know you can't hear the different, like me, or anyone else in the world.
You might be mixing the process of DACs and ADCs by my reference to "sampling". Sampling related to DACs is the process of converting a sampled and stored data stream back into analogue.
The example of the 5k5 signal. The data points for this with their associated amplitudes are stored as just 8 samples @44K1 per cycle on a CD. When the DAC converts these to an analogue wave converting @44k1, a stepped waveform occurs which absolutely needs to be "smoothed". Obviously, people arn't listening to pure sinewaves, but all waveforms can be resolved to the summation of sinewaves.
I did Google to find a decent representation of the distortion of waveforms with insufficient sampled points. I do understand the basics of CDs as I am a retired electronics engineer
Also, TV was must watch today as the UK Supreme Court hammered Boris Johnston