converts mp3 to flac
Kalm
::: print vinyl from flac :::
This is why I only listen to audio formats that add information to the music, not degrade it by taking away.
Don’t. Don’t give them ideas. TruMotion like frame interpolation is fucked up enough.
ChatGP3 Ultra Lossless
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.
Audiophiles are flat earthers for music.
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.
What’s the equivalent of people that buy ultra high-end stuff to trick themselves into getting high and orgasming from “binaural beats”?
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.
Aside from the Christopher Nolan thing, I was referring to music.
What I am saying is there’s a point of diminishing returns. That point might be a 192kbps mp3, but there is still a point where 99% of people or more will not know the difference and there’s no money in marketing to that 1% who will.
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.If your pre amplifier isn’t laid on cones carved from the purest quartz inlaid with gold, are you even really an audiophile ?
Speak brother
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.
Zeus can hear a duck ovulating 100km away
He needs that ability, ok?
He obviously is right! I have a mechanical keyboard because it transmits better key presses. Aviator connector helps reduce the noise as well.
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.
Yes, transcoding. At least re-encoding, I’m not sure if simply moving the file degrades it…
All of this talk is making me miss what.cd. You’d get the boot if you uploaded a transcode
In the context that automated systems will compress files for transfer to save bandwidth then it could potentially result in loss.
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.
That’s not really correct.
If you re-encode (at the same level) you could not lose some of the signals, but maybe you will.
If you lose data when “saving or moving” then both the formats are at risk.
I like to witness this grey area in between misconceptions that comes up with a hybridation of absurd takes and obvious truth.
It’s a file, if you get it fucked by copying it will just break, not “degrade” in sound quality.
If you reencode a lossy encoded file you will turbofuck it, obviously.
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.
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.
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”.
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.
Not really infinite points since energy is quantized. In a crazy particle physics sense analogue data is effectively the same as digital, when resolutions match.
deleted by creator
Fake it 'till you make it is not applicable to scientific or technical discussions.
IVE HAD THIS GIF SAVED FOR OVER A YEAR AND I FINALLY GET TO USE IT
the 320kbps mp3 is for listening to, the flac is for storage / archival so I can convert it to opus or other formats
I don’t… I don’t get it
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
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!
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.
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
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.
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.
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?
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.
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.
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.
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.
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.
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
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.
The end result sounds awesome. I would totally listen to that on its own.
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).
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.
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.
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…
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.
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.
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.
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.
Ough or more loudness :-(
See the loudness wars.
Out of all shops that got nicked in the last 20 years by internet shopping, high-end audio stores with concierge salespeople are still doing great.
Yup, 100% all of this. It turns out that audiophiles can’t tell the difference between a “best-in-class” cable and a coat hanger: https://www.soundguys.com/cable-myths-reviving-the-coathanger-test-23553/
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!
/s
And it’s all bits now! The 12" reel-to-reel tape that can’t be played in any device made since 1980 had KIBBLES to go with the bits!
Saving music to a external memory thing also losses quality because it has to be copied twice
To keep the highest quality of the music, you need to listen to it on the device it was created
That’s why I’ve always said using Optical/Toslink etc. is a mistake. Sending music with light just means you’ll hear shade in your music.
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.
DIY Perks just did a video where he modded a toslink TX to use a laser so he can do surround speakers wirelessly using line of sight.
That’s cool, I didn’t know him before, but looks like he has some extensive knowledge about audio or someone behind scenes is working with him.
The speakers he builds seem cool, but unless one has extensive audio knowledge at best can just replicate the same thing (assuming they can find all the parts used)
Saw this one yesterday. Pretty neat!!
Gold plated TOSLINK cables crack me up.
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!
Einbildung ist auch eine Bildung!
Hahahaha, that’s a good one. I’ll be using that
