Why do 96 or 128khz sampled audio files sound better than 48khz ones? I blind tested and could always tell the difference between them, but not between 128 and 192
Typically, high sampling rate files are part of a different mastering process than what is published as a 44.1kHz cd audio or 48kHz dvd audio.
Also, you might possibly be sensitive to resampling artifacts if your output device runs at 44.1kHz and your file is 48kHz or vice versa.
Audio testing is hard, and testing on yourself is tricky... But if you have a sample that you're convinced sounds better at high rates than lower rates, I would urge you to put it through a tool to resample it down to lower rates and see if/when you can tell the difference. If the rate isn't an even multiple, it's worth using a tool that can dither; dithered resampling artifacts are less abrasive than undithered... I had some voice recordings to play over the phone, and everything needed to be 8kHz u-law; the 48kHz original recordings sounded better than 44.1kHz original recordings because one is even multiple and the other isn't, but either way, the waveforms looked worse than it sounded.
> If the rate isn't an even multiple, it's worth using a tool that can dither; dithered resampling artifacts are less abrasive than undithered...
This seems to be mixing up two things; proper interpolation and dithering.
If you have limited bit depth (in practice, 16 bits or worse), you should pretty much always dither, ideally also noise shape. This is independent of the interpolation you're using; having a rational relationship between the original and downsampled signal makes some of the implementation a bit easier, but even for something like 48000 -> 24000, you'll end up with effectively a float signal that you need to convert to your chosen bit depth somehow, and that should be done better than just truncating/rounding.
And even for interpolating between two prime rates, or even variable-rate interpolation, you can and should get great interpolation (typically by picking out polyphase filtering coefficients from a windowed sinc of some sort).
Oh come on. I have handheld recorders that do 192khz.
"Headroom"
And the idea that humans can't hear over 20khz is like "humans taste 'sweet' on the tip of the tongue, and 'bitter' on the sides"
As we get older the hairs in out ears break or whatever and our perception decreases, but I could hear the fly backs in my old monitors, I used to be able to see the flicker in 3khz pwm LEDs, and my induction hob drives my kids crazy but it's merely midly annoying to me.
Get a real soundcard and some young people and play square(pwm) and sine tones starting at 16khz and find out where they can't hear it anymore. I find studio monitors with tweeters that are not paper are the best.
If you think you can hear ultrasound, it's nearly always due to nonlinearities in your system producing non-ultrasound when you try to play it. Seriously. (You can sometimes hear above-20 if it's very loud and/or you are pressing the source against your skull. Above-40 would be completely insane.)
The extra headroom can indeed be useful for some kinds of processing, but you can safely discard it for actual listening.
ime 48/96/128kHz 16/24bit through a modern DAC and well warmed Class AB amp and barely more than okay headphones can be told apart in a double blind test
but you do need phile-enough gears(minus the gilded pebbles hot glued onto circuit breakers)
sampling theorem only applies to sine waves. the rate is bit like the order of (fourier)series expansion and so approximations deviate as rate reduces. how many orders is enough depends and is situational
