The paper that you gave to me was about blind testing 16/44.1 and upper and now I'm asking about 25years old resampler. Now a day we use digital resampling and in digital resampling if you resample in to different factors it losses part of kbps and someone said in 2019 there is no problem about resampling in to different factors and no losses on kbps and i want sources for that i don't care how you guys criticize me because i know there is many non educated people's and i understand that so now i want sources about resampling has NO LOSSES
there is never no losses. playing the original file at the original sample and bit depth will not be perfect. because nothing is. at least in the analog domain. even between 2 so called bit perfect streams, you can usually measure differences at the lowest level. not that it's very relevant when noises and distortions will surely swamp the all thing with much bigger differences(which is why we so easily disregard small errors). the idea of perfection in music playback is a fantasy. which is why people usually prefer to care about what's audible or not. which is something we demonstrate with listening tests.
now resampling can turn out to make some audible difference on some gears, because the resampling solution really sucks, or because maybe the extra high frequency content destabilized the playback rig a little. or maybe because the DAC's filter for 44kHz is rolling off the trebles a lot more than at other freqs. none of the effective cases really have anything to do with your idea that we're causing errors or loss in the redrawing of the analog sine waves. because those errors simply are so tiny in the first place that you shouldn't expect them to be audible.
in real life, anything is possible, but possible exceptions shouldn't be taken as the norm. as gregorio told you, the album you're playing has content that was most likely resampled several times(because many audio plugins work best at a specific sample rate, so they convert the signal to that rate before doing what they do. what you can deduce from that very common plugin behavior(and we're talking pro stuff used in studios), is that the guys making them considered that the resulting sound with resampling is going to be better than without. that should give you some idea about how destructive they consider resampling to be.
now that example is usually about taking a sample rate and increasing it. so we don't discard any information. taking a high sample rate file and reducing it to 44 or 48kHz means objectively discarding information and that is a fact. mainly we can treat that as discarding the frequencies at sample/2 and above. which begs the question, are you going to miss ultrasounds? most controlled listening tests suggest that you won't. but of course the best answer relies on taking your own gear, your own ears, and test it out under controlled conditions. then you have a reliable answer about your circumstances(which is probably all you really care about, and you wouldn't be wrong^_^).
your last question about something at 44kHz being turned into 48kHz, as pretty much any increase in sample rate it is probably not something to be concerned about. by going to 48kHz you now have a little extra room for the final low pass filter in the DAC, so even if it's some extremely cheap extremely badly implemented solution, by pushing it a little farther in the ultrasounds we reduce even more the chances to get some audible impact. so I'd argue that overall, it could be seen as an improvement because what you may lose, is outweighed by what you might gain(which is really the entire reasoning behind all resampling).
as for something at 48kHz being converted to 44 by a cellphone, again, the 2 concerns are:
the ultrasound range you discard. will you miss it? will you hear it? probably not.
and also the DAC's filter if it's one that is known to be audibly different at 44kHz(relatively rare nowadays, but could still happen).
now about resampling in general, any DAC worth something will apply massive oversampling to the signal before going to analog(for various good reasons). it has become very hard to act paranoid about resampling because of it, as it now kinds of equals being against digital audio. we still find that "a
small village of indomitable Gauls still holds out against the invaders" in audio forums and keep purchasing NOS DACs(mainly for the wrong reasons). but the rest of the world has moved on about this matter.
as I was saying before, this in no way means that you shouldn't go for hires files and playback that respects the file's sample rate if that's what you want to have. objectively it's not a mistake, and some DACs will measure better when fed with a hires file(or a normal 16/44 converted to 24/96 before being sent to the DAC. my old ODAC is like that. I can't tell the difference by ear, but I can measure it. what to do of such an information is the user's prerogative. I'm surprisingly not interested in measured fidelity and always pick what's more practical over what measures best. but that's me. someone else may have other priorities. but you have to keep in mind how that desire for fidelity, and the notion of audible consequences, might not agree all the time(or if your gears do a proper job, ever).
