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[-] Carnelian@lemmy.world 130 points 10 months ago

lol. They can’t hear the difference even with the most expensive equipment. The resultant signal from decompressing a FLAC phase cancels with the original signal if you invert it. Meaning they are indeed 100% identical. Lossless, dare I say.

Literally all it does as a file format is merge data that is identical in the left and right channel, so as not to store that information twice. You can see this for yourself by trying to compress tracks that have totally different/identical L and R channels, and seeing how much they compress if at all

[-] Darkassassin07@lemmy.ca 54 points 10 months ago

This is like trying to explain to a SovCit, why they need to have a license.

You're wasting your time.

[-] Carnelian@lemmy.world 98 points 10 months ago

No, it’s like explaining FLAC to anyone who happens to be curious about it after seeing this screen shot and wondering how something can be both compressed and lossless at the same time. Many people appreciate this type of information being accessible easily in the comments

[-] murmelade@lemmy.ml 12 points 10 months ago

Certainly do. I learned something neat, thank you!

[-] TheDannysaur@lemmy.world 1 points 10 months ago
[-] circuitfarmer@lemmy.sdf.org 2 points 10 months ago

Interesting. It must do more than that though -- for example, FLAC offers different compression "levels", which you choose when encoding. To my knowledge all of them are lossless, but what do the levels do if it is only merging identical channel data?

[-] Carnelian@lemmy.world 6 points 10 months ago

You’re absolutely right about that. My use of “literally all it does” was employed poorly, and is a pretty extreme oversimplification

There’s a whole mathematical thing happening with FLAC generally, regardless of L/R channels, where it replaces your original waveform with a polynomial approximation of it + the differences between that approximation and the actual. When played back together, those two things always result in a perfect recreation of the original.

The various compression levels you can choose from essentially control presets relating to how sophisticated those approximations can be, thus cutting down on the amount of differences that need to be stored.

The reason you may want to play with these settings is somewhat outdated now. But a higher level takes more time to encode, results in a slightly smaller file size, and also takes slightly more processing power to decode. Any modern piece of equipment can handle the maximum setting with no issues.

But yeah, as a result of these processes (rather than as the prime goal explicitly, if that makes sense), it does joint-encoding and merges anything from the L and R channels that can be merged. This enables it to pull “identical” sounds from L and R even when the data itself is totally different, which is actually more common than not in music due to the use of multi-channel effects such as reverb.

In the end, a massive amount of the space saved as a result of the compression in typical music comes from removing duplicate information from the stereo field. But all sorts of funky stuff would happen if you opened up a DAW and started contriving different situations for it to compress

[-] circuitfarmer@lemmy.sdf.org 1 points 10 months ago

Thanks for the detailed explanation!

[-] AdrianTheFrog@lemmy.world 1 points 10 months ago

polynomial approximation seems like a weird choice for audio, is it really more efficient than a frequency based encoding?

also, it seems like audio compression formats have seen a lot less development in recent years than images have. I want to try encoding audio as a lossless jpeg xl now just to see how it does, I think it should be possible as jpeg xl supports extremely large image dimensions

[-] Carnelian@lemmy.world 1 points 10 months ago

It’s fairly well optimized for audio. Waveforms are usually continuous and relatively repetitive. The other really important aspect is how easily it can be decoded, so that it remains a usable audio file on potentially underpowered equipment.

Although I wonder if there exist some cases where other formats might do a better job

[-] LodeMike@lemmy.today 0 points 10 months ago

Flac is literally lossless in the mathematical sense.

FLAC still cuts out part of the signal. It's limited to 20khz.

Bhat's typically well above the limit of an adults hearing, especially someone old enough with enough money and equipment to be considered an audiophile.

[-] LodeMike@lemmy.today 0 points 10 months ago

Even uncompressed audio cuts out frequencies. With digital audio capture it is impossible to capture everything. There will always be a floor and a ceiling. In the case of flac it’s typically 20-24hkz.

Audiophiles have moved onto “high res lossless” because regular lossless wasn’t good enough for them.

[-] moody@lemmings.world 0 points 10 months ago

FLAC is totally lossless. You can rip a CD to 44kHz WAV, compress it to FLAC, and then decompress it and get a bit-perfect copy of the original WAV.

Lossless converting a CD to FLAC. But that CD was recorded at 44hkz sampling rate, which gives you a maximum frequency of 22khz. You have lost audio above 22khz. Children can theoretically hear frequencies higher than this, but typically adults cannot.

https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem#%3A%7E%3Atext=If+the+essential%2CNyquist+interval.

[-] moody@lemmings.world 0 points 10 months ago

FLAC doesn't cut anything out though. Whatever input you use, FLAC compresses losslessly. You can use 96kHz 24bit recordings and the resulting FLAC file can be decompressed back into a bit-perfect copy of the original.

In the OP, the messages in red are correct. FLAC is like a ZIP file designed to be more effective at compressing audio files. And just like a ZIP file, you can reconstitute the original file exactly. There's no data lost in compression.

Yes if you're transcoding a CD to FLAC it's lossless. That's not what I'm talking about. I'm talking about the process of digitally recording the audio in the first place.

Nevermind the fact that nobody seems to have paid any attention to the original joke which is that the boomers who can afford high end stuff can't even hear the difference.

[-] uranibaba@lemmy.world 0 points 10 months ago* (last edited 10 months ago)

You began this by saying

FLAC still cuts out part of the signal. It’s limited to 20khz.

Recording from analog to digital is lossy, in the same way as previously described about images. But this has nothing to do with FLAC.

Recording from analog to digital is lossy

That's the entire yoke.

this post was submitted on 15 Dec 2024
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