Music, sound card and correct sample rate for it

Question

Audio

The question is rather narrow-minded. I thought about what the correct sound output setting is all the same. I don’t have a special quality of hearing and acoustics, but still, before buying something, I want to know how it works and what you need to buy in general. Interest and problems arose at the stage when I tried to organize what is sometimes called the correct sound output to the Senheiser HD202 headphones, and now also the Microlab Solo 1 acoustics. So far I have settled only on music and only on two-channel. Under Windows, one of the best players is foobar2000, somewhere on Habré there was an article on how to restore normal sound in Windows 7. Then I came across an article, which describes the setup process (using WASPI exclusive and SoX resample). Then I tried to do something similar on Linux, and I think I managed to disable resample altogether.

I have a Lenovo B560 laptop, which has built-in speakers and two (three - if you count HDMI) input-output (microphone and speakers). The audio card is a standard Intel HDA, based on the ALC269 chip. So, judging by the description on the Realtek website, this chip, or rather its 2-channel DAC (and there are apparently 2 of them, independent) supports frequencies 44.1, 48, 96, 192 kHz at 16/24 bits. Then the thought arises that the best option would be to output all formats, whatever they are, as they are (for example, flac 192kHz 24 bit or flac 96 kHz 24 bit without any sample rate conversion, and not load the processor on the conversion). I don’t understand how to organize this in Windows, since in the sound card settings you still need to set the sampling rate, and there is also an item 192 kHz 24 bits. On Linux, as I understand it, you can generally disable this resampling and then, if the card supports it, it will output it as it is. I want to note that it was all configured in deadbeef. I removed DSP Resampling, chose the sound output through Alsa, and not through Pulse (which is not at all clear how it works and why it is needed, but this is just my lack of interest in dealing with this because it is useless). Next, the option was selected to output sound without all software conversions. And in the Alsa plugin, the use of its resampling is also disabled. Next, the option was selected to output sound without all software conversions. And in the Alsa plugin, the use of its resampling is also disabled. Next, the option was selected to output sound without all software conversions. And in the Alsa plugin, the use of its resampling is also disabled.

Can such a setting in Linux be considered the best, and is it really possible to output sound without software fixes, conversions, and other things? After all, logically, changing the sampling rate introduces distortions, and the option without changing is the most honest. I looked at the output frequency in alsa like this, and each time it corresponded to the one that the file had (in this example, the FLAC 96 kHz 24 bit file):

$ sudo cat /proc/asound/card0/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 96000 (96000/1)
period_size: 1024
buffer_size: 8192

And can the manufacturer be trusted that everything is honestly organized there?