converts mp3 to flac
Kalm
::: print vinyl from flac :::
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.
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.
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.
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".
He obviously is right! I have a mechanical keyboard because it transmits better key presses. Aviator connector helps reduce the noise as well.
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?
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
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.
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.
IVE HAD THIS GIF SAVED FOR OVER A YEAR AND I FINALLY GET TO USE IT
I don't.... I don't get it
When people are way too smug about their wrong answer.
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