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
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
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.
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
Not a FLAC, but I tried it on this video by reencoding to an mp3 at 320 kbps, then subtracting the original, amplified it a bit, and got this. The song is definitely recognizable, but heavily distorted.
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)
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).
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.
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.
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.
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.
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
Audiophiles are just a victim of their own smugness. Human ears are pitiful to start with, but then the neural processing that goes on is even worse. We can't hear shit and what we hear we can't even all remember or recognize. And that's at a young age, at age 30 the hearing is already deteriorating. Hearing has never been a strong point for humans, when our fight or flight response kicks in, the processing of audio is the first thing to go. If we didn't use it for communication as much, we might have lost it even further. Even our sense of smell is better and compared to other animals our sense of smell is very weak.
Audiophiles consider themselves special because they "honed" their skills and can hear stuff others can't. But you can't hone what isn't there, there's no fixing crappy hardware. In a double blind experiment almost all of them would fail even identifying a regular old Apple Music AAC file all the normies listen to compared to a lossless version. And when they buy expensive shit, that distorts the music in a way they like, they convince themselves that is the true version and all other versions must be wrong.
But hey, on the spectrum of all the bad and or dumb shit humans do, being someone with too much money who enjoys music isn't half bad.
You can quite famously (and easily) fool any "audiophile" into thinking a given system sounds better than another -- or after some mysterious modification -- by doing nothing but turning the volume up one notch.
This is easily demonstrable, and repeatable. And a tactic often exploited in oldschool hi-fi shops, back in the days when you were expected to walk into a high street store and be greeted by a salesperson rather than just order whateverthehell off of the internet.
I went to school to be an audio engineer and audiophiles amuse me. While it is true that expensive speakers and FLAC and so on will make music sound better than it would on the cheapest stuff- we mix so it will sound decent on the cheapest stuff. We never mixed with you guys in mind. When I was doing it, we were keeping mp3 players in mind. These days, most music is mixed with streaming in mind.
My professor told us to take our mix out to our cars and drive around somewhere noisy and listen to it and then go and remix it after that based on what you heard.
Sure, there are exceptions. Not very many of them. Because companies want to make money from albums and they know most of the people listening to the music aren't going to be listening to lossless audio on $4000 speakers.
I find it especially amusing because, until the digital era, all the greatest music that was recorded since multitrack recording started in the 1960s was on bits of magnetic tape held together with bits of scotch tape and the engineer prayed that nothing would go wrong when it they were making the final two-track mix. It is highly unlikely that "what will this sound like on super expensive equipment?" was given consideration.
They really obsessed over something and need to feel superior about it. They're harmless at least.
Unless of course you're googling about speakers for a TV, in which case you're about to get some terrible advice from some middle aged dude who's really pissed about soundbars existing.
Christopher Nolan certainly does not mix his movies for the cheap stuff...
I think people get a little silly about it when you get above maybe 192kbps, but there 1000% is a huge difference between a 128kbps mp3 and a 192kbps mp3, and I would take a blind test every day of the week to prove it.
128kbps mp3s sound like aural garbage. It's like when you go to a wedding, and you can tell that the DJ just downloaded "Pachelbel's Canon" from KaZaa because when played over the PA, it sounds like someone farting into a microphone.
Are we talking about movies or music? Movies are mixed to sound good in theatres and then they are later remixed to sound good on at least cheap surround systems, but, again, they aren't generally doing it thinking about the people who spent $4000 on their system. And, again, the chief concern outside of the theater these days is audio for streaming.
I am not denying that a $4000 home audio system will sound better than a $100 one just by virtue of at least some of the components not being cheap Chinese crap, but I doubt even Christopher Nolan is ensuring his Blu-ray releases (or whatever) sound best on expensive audiophile systems. There's a point of diminishing returns here.
When I was in a band, we had our albums professionally recorded, mixed, and mastered, but we had a pretty decent set-up in the studio. After every practice, I'd do some rough mixing and burn us each a CD to listen to in our own cars and email MP3s for those of us who used devices. We'd take that and decide what needed to be fuller, what was getting lost, etc. and change any arrangement as necessary. Of course we might do more layers in the album itself than we could do live (well, without sampling machines going constantly and whatnot), but we still wanted to make sure we had at least the basics of where we thought people would listen to us.
Actually, you see, it is not the original bits, yeah? They get compressed, and that removes bits, and then they are uncompressed, and bits are added. Those are RE-CONS-TI-TUTED bits. It's like reconstituted tomato juice, the taste of the original water is gone forever! And you can hear that. With music, I mean, not with the tomato juice. Like, who says it's even the same kind of bits, the same quality? You can so hear the difference. You want a double blind study? Well that's just silly, if it's double blind, it means its not blind, because the two blinds cancel each others out. Basic science, duh!
As an audio connoisseur, I will not settle for anything less than a private, live showing by the band without any digital assistance like microphones.
You see, when the audio goes into a wav file, it gets converted into BITS. That's not audio, that's food! And those bits aren't even used immediately--they're saved for later! They go STALE in the hard drive only to be used later to create synthetic bits, losing both quality and purity during the process. To make matters worse, those synthetic bits are used to make synthetic audio waves, which get turned into electrons and sent down a wire to make synthetic pressure waves. Nothing is REAL with digital audio. It's fake music made from fake sounds made from fake waves made from fake bits made from a fake copy of real, honest-to-goodness music.
It may come at a premium multi-million cost for a single album, but gosh dang-it, I'm listening to music as it was made to be!
Toslink is a bad example for your point. It is the same S/PDIF digital signal that is sent over fiber and it isn't even using laser but a standard diode, so won't even work long distance (shorter than if you would use the normal RCA cable with S/PDIF).
The most ridiculous thing I saw was the gold plated toslink cable.
Every time I see audiophile stuff it makes my expetactions of them dig a deeper and deeper hole.
Why is it always some snublord jerking themselves off over they 25k setup, like their ears are blessed by Zeus themselves.
I once realized there are audiophile speaker cables that cost hundreds of euros per meter because they are "burnt in" with some kind of awesome machine that pushes a specific amperage through it for a specific amount of time. I'm sure they improve the sound quality tremendously.
I just read an article about a blind trial between a present day ~3000$ hifi-set and an equally expensive (value adjusted, of course ) and perfectly restored late 70's hifi-set. Among the listeners were a couple of audiophiles, musicians, journalists and one pro audio engineer.
They listened to 5 pre-selected songs in FLAC via a top-of-the-line DAC plus one song of their own choosing.
Everyone else gave 7 or 8 points to either set, but the audio engineer gave just 4 to each. Most of the time the audiophiles were unable to recognize which set was playing.
Afterwards they did an audio labratory sweep on the sets and found them basically equal in terms of sound quality, the only major difference was a drop in the 70's set mid-hi frequencies, which was theorized to be the result of reversed polarity in one of the tweeter elements. None of the participants mentioned noticing this directly, but the audio engineer did talk about "unclear higher frequencies" in some songs.
Technically, an issue with lossy formats is if they get saved, moved, and/or re-encoded then there is a risk of media degrading over time, over iterations. So you could potentially hear the difference.
But FLAC is lossless.
If the user likes the MP3 sound better then clearly they actually enjoy the lossy hum and buzz of compressed audio. I'm sure they would enjoy Vinyl.
What.cd was the greatest collection of obscure music the planet has probably ever seen. I dont even particularly care about lossless codecs, I was fine with 320kb/s mp3 as it was more convenient but even their mp3 rips were way better than other places, and you knew everything would be tagged and sorted correctly. And they had EVERYTHING you could think of, it was wild.
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.
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.
My point being that unlike the misunderstanding (or maybe just mis-explanation) of many here, even a digital audio format which is technically named "lossless" still has losses compared to the analog original and there is no way around it (you can reduce the losses with a higher sampling rate and more bits per sample, but never eliminate it because the conversion to digital is a quantization of an infinite precision input).
"Losslessness" in a digital audio stream is about the integrity of the digital data itself, not about the digital audio stream being a perfect reproduction of the original soundwaves. With my mobile phone I can produce at home a 16 bit PCM @ 44.7 kHz (same quality as a CD) recording of the ambient sounds and if I store it as an uncompressed raw PCM file (or a Wav file, which is the same data plus some headers for ease of use) it's technically deemed "lossless" whilst being a shit reproduction of the ambient sounds at my place because the capture process distorted the signal (shitty shit small microphone) and lost information (the quantization by the ADC in the mobile phone, even if it's a good one, which is doubtful).
So maybe, just maybe, some "audiophiles" do notice the difference. I don't really know for sure but I certainly won't dismiss their point about the imperfect results of the end-to-process, with the argument that because after digitalization the digital audio data has been kept stored in a lossless format like FLAC or even raw PCM, then the whole thing is lossless.
One of my backgrounds is Digital Systems in Electronics Engineering, which means I also got to learn (way back in the days of CDs) how the whole process end to end works and why, so most of the comments here talking about the full end-to-end audio capture and reproduction process (which is what a non-techie "audiophile" would be commenting about) not being lossy because the digital audio data handling is "lossless", just sounds to me like the Dunning-Krugger Effect in action.
People here are being confidently incorrect about the confident incorrection of some guy on the Internet, which is pretty ironic.
PS: Note that with high enough sampling rates and bits per sample you can make it so precise that the quantization error is smaller that the actual noise in the original analog input, which de facto is equivalent to no losses in the amplitude domain and so far into the high frequencies in the time domain that no human could possibly hear it, and if the resulting data is stored in a lossless format you could claim that the end-to-end process is lossless (well, ish - the capture of the audio itself into an analog signal itself has distortions and introduces errors, as does the reproduction at the other end), but that's something quite different from claiming that merely because the audio data is stored in a "lossless" format it yields a signal as good as the original.