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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
This is like trying to explain to a SovCit, why they need to have a license.
You're wasting your time.
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
Certainly do. I learned something neat, thank you!
how something can be both compressed and lossless at the same time
I assume most people who've used a computer are familiar with lossless compression formats like ZIP files.
Of course, though, it doesn't matter how familiar an audiophile is with digital formats. They'll still believe that more expensive cables sound better, and they'll keep on believing that even if you show to them that they can't tell an expensive cable from a bargain bin one in a blind test.
Imagine zip being lossy.
Like me!
Flac is literally lossless in the mathematical sense.
Yeah but my ears don't care about mathematics.
Some audiophile, probably...
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.
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.
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.
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.
I dont think you understand the difference between a lossless file format/encoding algorithm and "losless" recording/storing of signals. If anyone ever speaks of a lossless encoding algorith theyy mean that avter encoding and decoding the input and output will be the same e.g nithing was lost. Why would the recording have annything to do with the lossyness of the encoding algorithm? If the music was made digitally there would be no loss in any sense since the output of your daw or midi file etc is already digital. Btw in general you just cant record any arbitrary analog signal but you can record a lot of it. You will also never in no media be able to store the exact signal. There is always noise always some variation. Even if you store your signal analog there is only so much variance of the magnetic field in a tape and only so many atoms of height difference in the groove of your vynil. The thought of lossless recording is just dumb if you think about it because you change the signal by measuring it annyway so what even is the "original" signal?
No, it doesn't. Digital PCM audio, as a concept, can only represent frequencies up to the sample rate used. Which can be anything. Typically 44kHz.
Going above that is pointless as humans are unable to perceive the ultrasonic frequencues that would unnecessarily include.
Lossless doesn't mean "perfect recording". By that logic lossless images or videos aren't lossless, because they don't include an infinite amount of pixels between every pixel, representing every photon that was captured.
Lossless refers to data-retention, not reality retention.
You can encode at higher bit depths and sample rates. I have music I’ve bought at 24bit 48Khz. (I know I won’t ever be able to hear the difference between that and the more common 16bit 44.1Khz.) I think you can go up to 96Khz, although I’m not sure I’ve actually seen it before.
I've seen, and I have 24bit 96khz files.
They're less common than your average 16bit 42khz, but they do exist.
Isn't 44.1 KHz more common?
Yeah I fixed it idk why I thought 42
No its not lol
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.
And this is because audiophiles don't understand why the audio master is 96 kHz or more often 192 kHz. You can actually easily hear the difference between 48, 96 and 192 kHz signals, but not in the way people usually think, and not after the audio has been recorded – because the main difference is latency when recording and editing. Digital sound processing works in terms of samples, and a certain amount of them have to be buffered to be able to transform the signal between time and frequency. The higher the sample rate, the shorter the buffer, and if there's one thing humans are good at hearing (relatively speaking) it's latency.
Digital instruments start being usable after 96 kHz as the latency with 256 samples buffered gets short enough that there's no distracting delay from key press to sound. 192 gives you more room to add effects and such to make the pipeline longer. Higher sample rate also makes changing frequencies, like bringing the pitch down, simpler as there's more to work with.
But after the editing is done, there's absolutely no reason to not cut the published recording to 48 or 44.1 kHz. Human ears can't hear the difference, and whatever equipment you're using will probably refuse to play anything higher than 25 kHz anyways, as e.g. the speaker coils aren't designed to let higher frequency signals through. It's not like visual information where equipment still can't match the dynamic range of the eye, and we're just starting to get to a pixel density where we can no longer see a difference between DPIs.
If that's happening you need to fix your transcoder settings.
The "high res lossless" you're referring to, is still FLAC. FLAC has no downside. Whatever PCM audio you want, it can represent perfectly, while using less storage.
FLAC doesn't "limit" or "cut out" anything unless you or the software you're using is reducing the bit depth or samplerate of the source PCM waveform.
Which is something you might want to do, since it will reduce file size significantly to not use a higher samplerate than necessary. But FLAC itself doesn't do or require that.
On new formats, you might be thinking of MQA, which supposedly encodes the contents of a higher samplerate PCM waveform into a lower samplerate file, but it has been proven to be largely snake oil, and lossy as hell in terms of bit integrity.
LOL... FLAC happily handles 192kHz
And at 327,675hz (the maximum for FLAC) you can still be missing out anything 327,676hz and greater. But that won't stop the audiophiles.
Based
do you know if anyone has tried this with a flac and an mp3 file? Theoretically all that should be left is the "loss" right? what would that sound like?
eta: I'd try myself but I'm not an audiophile and wouldn't even know where to get a flac file (legally) and doubt my crappy $20 in ears would be capable of playing it back if I did
you wouldn't need a flac file, you can use any wav file, the audio of both is identical.
regadring your question, you can think mp3 as the jpeg of music. both mp3 and jpeg use fourier transforms*. so, to image what mp3 is doing to the audio, you can see what jpeg does to images (spoiler alert, unless you are aggressively compressing it, it is not noticable)
(*jpeg actually use discrete cosine transform instead of fft, but it is similiar enough)
This is why I always use PNG
Another place is bandcamp. When you buy music from there you can choose the encoding.
I generally download FLACs when I can; after building an mp3 library, then adding oggs, and most recently opus, I value having a source that I can transcode into whatever new, improved codec takes the lead every few years. However, you have to be prepared for the size requirements. FLACs are still pretty big: I recently bought Heilung's "Drif", and the FLAC archive was nearly 650MB. Granted, it's bigger than usual; the average album comes in around 400MB, but still... you have to commit to find sizeable long-term storage if you keep those sources, and off-site, cloud backup can get pricey. Or, you can trust that where you buy it from will provide downloads of your purchases indefinitely.
If you want free, legal FLAC files just to play with, this Zelda fan music album is free and legal to download in FLAC format (you do need to torrent the FLAC version, yes legal torrents exist).
It also has some good tracks in it IMO.
Legal torrents, such as Linux ISOs ;)
If you own the music, you are allowed to own a backup of it in FLAC (or any format).
HDTracks.com sells DRM free albums in FLAC format
Bandcamp too sometimes
I've tried myself, and the "loss" is really not that much. You can see it if you zoom, but if you listen to it you can't make out the track it comes from. It sounds more like noise. That was at least on the track I tried this with, maybe in a less compressable track there is more of a difference.
Did you mess around with compressing it yourself at all? Like, if you "deep fried" it, would the difference be recognizable?
If any youtube/peertube creators are reading, I'd click that video...
I did it once a few years ago (IIRC with a copy of Falling Down by Muse, not for any particular reason), and compared V0 320 with FLAC.
After amplifying the tiny, tiny wiggle of a sound that was left, I was left with very slight thin echoes, mostly well above 10k.
The sort of stuff you really wouldn't worry about, unless you 100% wanted bit-perfect reproduction, or wanted to justify a £2000 pair of headphones.Funnily enough, that was the point I stopped bothering to load FLAC onto my DAC, and just mirror everything into V0 for portable use.
Yeah, I think the difference between a FLAC and v0/320kbps is negligible.
However, the difference between a 128kbps mp3 and a v0/320kbps mp3 is massive and absolutely noticeable (and yes, it becomes more noticeable on higher quality equipment). Anything under 192kbps (or maybe 160), and you start to get noticeable degredation imo.
If anyone wants to claim that one cannot tell the difference between 128kbps and 320kbps, I'd take a blind listening test right now.
Theoretically all that should be left is the "loss" right? what would that sound like?
Like noise, more or less, but at frequencies that are hard to hear.
wouldn't even know where to get a flac file (legally)
BandCamp offers FLAC downloads. There are some other sites that do too, like Quobuz and I think some Japanese ones. Soundtracks I’ve bought via Steam sometimes come in FLAC too.
You can also rip a CD.
The easiest free way I know to get a FLAC file legally is to go to your local library, borrow a CD, and rip it to your home PC direct to FLAC. You'll have to deal with the fact that your ODD might introduce some noise, but it'll be the same noise as playing it from that same drive. Then rip the same disc to MP3.
Yes, WAV is in the middle both times, but that's how you can get a FLAC file to compare legally.
lol, I don't even own an optical drive anymore. I'm 100% streaming these days. It looks like from other comments there are places to buy FLAC files directly (which I'd hope would be decent quality)
It's all academic though, I'm not really interested in becoming an audiophile. Streaming quality is fine for my needs.
Fair enough! But at least you know there's a method in case it comes up. Also, I suggest you get a CD/DVD-RW drive, and BD-RW drive, just on principle - and use your local library for media! Your tax dollars pay for it, so you ought to get that value back!
The noise of the optical disc drive? I, erm, that's not how digital data works.
More to the point, the easiest way to get a FLAC file would be to record some audio in Audacity (or equivalent) and then output it as FLAC.
Fair, but the recording method comes down to microphone quality; I'm trying to go from a known good recording with something that can/will be lost in the MP3 transition.
The problem with your noise point is, I've used ODDs with less-than-impeccable lasers (either laser itself or the housing). I've had discs ripped with minor audio corruptions - I've always called that 'noise' because it's not the desired signal (and it can create literal random noise in the recording). Maybe there's a better term for it, but simply put, not all drives are perfect, not all lasers are perfect, and there is a possibility of imperfect copying. It's just a fact of life. Just like sometimes you might burn a frisbee, there's times you don't get a 100% clean rip.
Data corruption is one thing, but calling it "noise" is tremendously misleading because that's such an issue when digitising from an analogue source. I can't say I've ever experienced it due to the drive, but I have experienced it with scratched CDs. I've been using optical drives since the '90s and it's so rare that bringing it up is really muddying the waters.
With regards to sourcing audio, the emphasis was on "easiest". Most people haven't had optical drives in their computers in a long time. Ultimately they'd probably be better off finding something on Wikimedia in PCM as their "known good". Ripping audio isn't difficult for you and me but we're clearly nerdier than most!
Scratched discs are definitely a big problem, but I've had some bad drives in my time, where even good discs would get issues. I don't really have a better shorthand for the issue that's more descriptive than noise.
And you're right, I just tend to assume a very high level of nerdiness of anyone on lemmy/kbin/mbin.
Whilst it's a fair assumption usually, I think that the fact that they had to ask is indicative.
As for "noise", what's wrong with "data corruption"? A noisy recording and a corrupt recording sound nothing alike.
I think it is easier to download a test sample from a music label or any creative commons music released in flac. I can do it right now without standing up. For example.
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?
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
Thanks for the detailed explanation!
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
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