He obviously is right! I have a mechanical keyboard because it transmits better key presses. Aviator connector helps reduce the noise as well.

TBH FLAC can’t do 32bit floating point encoding, neither more than 8 channels per stream, so he’s not technically incorrect that it’s not zipping a wave file.

The first one concerns recorded source material (there are 32bit fp recorders for the last few years really helped loudest and quietest recordings) and not a single end product.

The second one limits FLAC to 7.1, so as a container it’s not suitable for theatrical purposes, neither for Dolby Atmos/DTS, nor for higher order ambisonics (used in vr).

But neither of these limits concerns end users, and if we’re talking about music, they can get fucked with whatever exorbitancy they prefer.

Strictly speaking, as soon as an analog signal is quantized into digital samples there is loss, both in the amplitude domain (a value of infinite precision is turned into a value that must fit in a specific number of bits, hence of finited precision) and on the time domain (digitalization samples the analog input at specific time intervals, whilst the analog input itself is a continuous wave).

That said, whether that is noticeable if the sampling rate and bits per sample are high enough is a whole different thing.

Ultra high frequency sounds might be missing or mangled at a 44.7 kHz sampling rather (a pretty standard one and used in CDs) but that should only be noticeable to people who can hear sounds above 22.35kHz (who are rare since people usually only hear sounds up to around 20kHz, the oldest the person the worse it gets) and maybe a sharp ear can spot the error in sampling at 24 bit, even though its miniscule (1/2^24 of the sampling range assuming the sampling has a linear distribution of values) but its quite unlikely.

That said, some kinds of trickery and processing used to make "more sound" (in the sense of how most people perceive the sound quality rather than strictly measured in Phsysics terms) fit in fewer bits or fewer samples per second in a way that most people don't notice might be noticeable for some people.

Remember most of what we use now is anchored in work done way back when every byte counted, so a lot of the choices were dictated by things like "fit an LP as unencoded audio files - quite luterallyplain PCM, same as in Wav files - on the available data space of a CD" so it's not going to be ultra high quality fit for the people at the upper ends of human sound perception.

All this to say that FLAC encoded audio files do have losses versus analog, not because of the encoding itself but because Analog to Digital conversion is by its own nature a process were precision is lost even if done without any extra audio or data handling process that might distort the audio samples even further, plus generally the whole thing is done at sampling rates and data precision's fit for the average human rather than people at the upper end of the sound perception range.

What's the L stand for again..?