He obviously is right! I have a mechanical keyboard because it transmits better key presses. Aviator connector helps reduce the noise as well.
this post was submitted on 15 Dec 2024
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When people are way too smug about their wrong answer.
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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.
When we talk about lossless in the audio encoding world, we aren't comparing directly with the analog wave, as there will always be loss when storing an analog signal in a digital machine. Lossless formats are compared to pure PCM, which is the uncompressed way of representing a waveform in bits.
With audio, every step you take to transform it, capture it, move it or store it, even while working with the analog waveform, degrades it. Even by picking it up with a microphone you're already degrading the waveform. However, generally, the official release CDs or WebDLs are considered the original, lossless, master file. Everything that manages to keep that exact waveform is lossless (FLAC, AIFF, WAV, ALAC...), and everything that distorts it further is considered lossy (MP3, AAC, OPUS...).
Additionally, a "bad transcode" (which is a transcode that involves lossy formats somewhere that isn't the last step) is also considered lossy, for obvious reason. Transcoding FLAC to MP3 to WAV stores the exact same waveform that MP3 made, as it is the lowest common denominator, even though the audio is stored as WAV in its final form.
Transcoding between lossy formats also loses more data, even if the final lossy format can store more bits or is more accurate than the original. This is one of the main problems with lossy codecs. MP3 192kbps to MP3 320kbps will lose information, just like MP3 to AAC. That's why, normally, we use a lossless file and transcode it to every lossy format (FLAC to MP3, then FLAC to AAC...). This way you're not losing more than what the lossy format already loses.
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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.
What's the L stand for again..?
Lossy, of course!