The meme is literally correct. when you make an audio file into an mp3, it takes some of the sonic information that our brains are bad at noticing and just…completely removes it(to save storage space). if it’s a high quallity mp3 its essentially completely unnoticable. That’s why it’s psychoacoustic, it uses psychology to fool your brain into thinking it sounds better than it does, litterally.
Psycoachoustic is basically just a big word for taking into account how our brains deal with what we hear.
Also I’m not saying mp3 is bad, in fact I think the opposite is true. I think it’s good to think of it like being the .JPG of audio, you’re not getting the original quallity and that’s the point and unless you need to do manipulate that audio file and its a reasonable quallity then you likely won’t notice the difference.
Ogg vorbis (.ogg) is a better codec though IMO.
*edited for spelling and grammers
You successfully managed to ignore the joke entirely.
actually I found the joke to be funnier because it was litterally technically correct.
YOU’RE literally technically correct.
deleted by creator
somebody explaining a component of the joke doesn’t mean that they didn’t get it.
That’s an internet thing.
You often can’t easily distinguish irony/sarcasm from stupidity, written language (i.e., no cues like tone), language barriers in general (as a non-native I might not get subtleties or might use them wrong), and the high density of people on the spectrum doesn’t exactly help.
Understanding people is hard.
- literally* correct
- it’s* essentially
- That’s why it’s psychoacoustic*, it uses psychology* to fool your brain into thinking it sounds better than it does, literally*.
- that’s* the point
- it’s* a reasonable quality*
In general I don’t believe you can tell any difference between MP3 and FLAC if you listen to the audio at the intended sample rate.
Meaning that @44100hz with 8 bit samples; you can’t tell.
Listening at higher sample rates with higher bits per sample; sure…there’s lots of room for unwanted and even audible error. Audio interpolation algorithms are not miracles, not smart, and not even close to being finely psychoacoustically tuned to your ears in most cases.
If you say you can hear a difference…you are lying or you are cheating by playing back the MP3 over an audio pipeline with a higher sample rate and bits per sample. Anyone could hear the difference when cheating like that. Human hearing can span all the way up to 128Khz; but oftentimes most people can’t notice a credible difference even at 96Khz.
But if you listen to a 44.1Khz signal via a 96Khz set of equipment; you’ll pick out exactly when the audio output shifts between being 96Khz and 44.1Khz.
This is how you can tell when audio is a recording at a lower sample rate. Most hardware is capable of outputting 96Khz so long as you don’t put older things in your audio chain (The pipeline from file to your ears, and yes this includes software and your operating system as well).
The problem usually arises when something is upsampled. Going from 44.1Khz to 96Khz is noticeable when you “Compress” the audio signal to boost apparent loudness. Most low-end equipment and unaware software will do this sort of operation automatically when upsampling your audio to make sure the process does not render your audio too quiet to hear. Your ears can hear frequencies being clipped or limited to a certain volume as well; which can also happen a lot to prevent certain issues. Because most people are unable to regulate this hidden software aspect of their playback chain; you can sometimes hear it.
Luckily if you spend some time with proper DSP software and/or hardware, you’ll be able to unmuddle/unmix these mistakes in your chain. It does take time and patience; and you’ll need a large blend of HQ audio (like FLACs or MQTTs) as well as your standard “downsampled” audio (like MP3s and other lossy tracks), and you’ll be able to tweak things so that everything sounds good.
Software packages like Viper4Windows or Viper4Android are good starting points and are often easy to figure out how to use and offer a very wide and diverse range of controls you can use to adjust the audio to your needs and liking.
Because everyone’s ears are different; there’s also plenty of tools that claim to adjust for your individual ears…and those can be helpful as well in chasing your perfect flat audio response curve and equalizing things to your preferences.
There’s literally an entire industry of bullshit cables and devices designed to “improve” sound quality that demonstrably does fuck all. That’s enough to tell me that most people saying they can tell the difference are probably full of shit.
Most of the time I can’t tell a difference but with orchestra / classical music I can.
Also most of the time I listen to music when I’m in a factory with 75db-80db noise floor so it hardly matters how good headphones and source I’m listening to.
It’s just at home where I can fully enjoy my flacs with HD 650… Not that I bother listening to them too often anyways.
I’ll take good mp3 256kbps master over bad flac master any day though.
As someone who owns a similar set; I can estimate you’re probably dealing with the upsampling issue due to an OS configuration issue. You should try listening to MP3s and FLACs at 44100hz sample rate for your comparison. Not 48000, not 96000.
Not just mp3, all lossy audio formats use psychoacoustic analysis. That’s how they figure out which data to throw out.
i feel insulted
They shouldn’t be giving you so much flac.
I stopped bothering with loseless to save space. Guess what? The drive is as full as it was before.
Does anyone know of any massive double blind study where they see what actual mp3 bitrate where people stop being able to tell the difference in quality?
I’ve tried and can’t tell the difference between any 320kbps and lossless.
There was one where the guy behind it went to massive lengths so people couldn’t easily distinguish the example files by other means than audio quality. Verdict was that people with more expensive equipment even preferred the sound of the MP3s (320kbps CBR). I think it was this one (Links to Parts 2 and 3 at the bottom.).
Somewhere else I’ve read that - for most humans - 256 kbps MP3s encoded with VBR-ABR using a high-end encoder are basically indistinguishable from the lossless original. Even at 192 kbps it’s still more hit&miss than it should be. But I don’t remember where I’ve read that.
Did you try on Bluetooth headphones? If so you won’t notice any difference because Bluetooth has very limited bandwidth and can’t really handle anything more than a standard MP3 (unless you have Sony headphones which use LDAC).
Maybe slightly related, but does anyone know any good codec (beside FLAC) for ultrasonic recording?
ogg vorbis, already mentioned above.
Alright, will check it out
They all ended up corrupted over time.
Sounds like a storage medium issue. Bit rot.
