this post was submitted on 15 Dec 2024
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Confidently Incorrect

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When people are way too smug about their wrong answer.

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[–] [email protected] 3 points 4 months ago (4 children)

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

[–] [email protected] 1 points 4 months ago (1 children)

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

You're wasting your time.

[–] [email protected] 2 points 4 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

[–] [email protected] 0 points 4 months ago (1 children)

Flac is literally lossless in the mathematical sense.

[–] [email protected] 0 points 4 months ago (3 children)

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.

[–] [email protected] 1 points 4 months ago

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.

[–] [email protected] 0 points 4 months ago (1 children)
[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] 1 points 4 months ago

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.

[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] -1 points 4 months ago (1 children)

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.

[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] 1 points 4 months ago* (last edited 4 months ago) (1 children)

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.

[–] [email protected] -1 points 4 months ago

Recording from analog to digital is lossy

That's the entire yoke.

[–] [email protected] 0 points 4 months ago (2 children)

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

[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] 1 points 4 months ago* (last edited 4 months ago)

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.

[–] [email protected] -2 points 4 months ago (2 children)

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.

[–] [email protected] 1 points 4 months ago (1 children)

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.

[–] [email protected] -2 points 4 months ago (1 children)

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.

[–] [email protected] 0 points 4 months ago* (last edited 4 months ago) (1 children)

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!

[–] [email protected] -1 points 4 months ago

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.

[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] -1 points 4 months ago

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!

[–] [email protected] 0 points 4 months ago (1 children)

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?

[–] [email protected] 1 points 4 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

[–] [email protected] 1 points 4 months ago

converts mp3 to flac

Kalm

::: print vinyl from flac :::

[–] [email protected] 1 points 4 months ago (1 children)

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.

[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] 1 points 4 months ago* (last edited 4 months ago) (1 children)

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.

[–] [email protected] 0 points 4 months ago (1 children)

What I meant is yeah, you are right about that, but no, lossless formats aren't called lossless because they don't lose anything to the original, they're called lossless because, after compressing and decompressing, you get the exact same file that you initially compressed.

Another commenter on this post explained it really well.

[–] [email protected] 0 points 4 months ago* (last edited 4 months ago) (1 children)

They're deemed "lossless" because there are no data losses - the word actually comes from the broader domain of data handling, specifically Compression were for certain things - like images, audio and video - there are compression algorithms that lose some information (lossy) and those which don't (lossless), for example JPEG vs PNG.

However data integrity is not at all what your average "audiophile" would be talking about when they say there are audio losses, so when commenting on what an non-techie "audiophile" wrote people here used that "losslessness" from the data domain to make claims in a context which is broader that merelly the area were the problem of data integrity applies and were it's insuficient to disprove the claims of said "audiophile".

[–] [email protected] 0 points 4 months ago (1 children)

By your definition, PNG isn't lossless because it's not an exact representation of every single photon of a picture that was taken. You'd need infinity pixels in order to be completely faithful to the "analog" thing that you're trying to picture, in the same way you'd need infinity points to completely translate an analog wave to digital.

When you compress anything with FLAC, you will get the exact same thing you compressed out, so there is no data loss.

Of course, that wave which you compress will not be faithful to the analog thing, but that's just a limitation of digital computers.

[–] [email protected] -1 points 4 months ago* (last edited 4 months ago) (1 children)

A PNG is indeed an imperfect representation of reality. Are you claiming that the lossness in the data domain of the compression algorithm in a PNG means its contents are a perfect representation of reality?!

(Funnilly enough, the imperfections in the data contained on a PNG are noticeable for some and the lower the "sampling rate" - i.e. number of pixels, bits per pixel - the easier it is to spot, same as audio)

As I've been trying to explain in my last posts, a non-Techie "audophile" when they claim FLAC is not lossless aren't likely to be talking about it's technical characteristics in the data domain (i.e. that data that you take out of a FLAC file is exactly the same as it goes in) but that its contents don't sound the same as the original performance (or, most likely, a recording made via an entirelly analog pathway, such as in an LP).

Is it really that hard to grasp the concept that the word "lossless" means different things for a Technical person with a background in digital audio processing and a non-Technical person who simply compares the results of a full analog recording and reproduction pathway with those of a digital one which include a FLAC file and spots the differences?

This feels like me trying to explain to Junior Developers that the Users are indeed right and so are the Developers - they're just reading different meanings for the same word and, no, you can't expect non-Techie people to know the ins and outs of Technical terms and no they're not lusers because of it. Maybe the "audiphile" was indeed wrong and hence "Confidently Incorrect", but maybe he was just using lossless in a broader sense of "nothing lost" like a normal person does, whilst the other one was using the technical meaning of it (no data loss) so they were talking past each other - that snippet is too short to make a call on that.

So yeah, I stand by my point that this is the kind of Dunning-Krugger shit junior techies put out before they learn that most people don't have the very same strictly defined technical terms on their minds as the junior techies do.

[–] [email protected] 1 points 4 months ago

You really do not understand the subject you're bloviating about. Consider stopping now.

[–] [email protected] 1 points 4 months ago* (last edited 4 months ago)
[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] 0 points 4 months ago* (last edited 4 months ago) (1 children)

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.

[–] [email protected] 0 points 4 months ago (1 children)

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.

[–] [email protected] 1 points 4 months ago

Aside from the Christopher Nolan thing, I was referring to music.