Where does 8ch get their mp3's ???

...

the human brain can't process more than 24 kbps

I get my mp3's from idope if it's older music and mp3 goo if its new music

In what world is this better than mp3?

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Then add -f opus

If it's on bandcamp you can easily take the mp3s by looking at the dom tree, just ctrl+f "audio".

You could aim to get everything in .flac first and then convert it to something else. Converting from compressed to compressed is generally not a good idea.
Opus is seriously amazing, it gets transparent for the average listener at pretty low bitrates. If you're more of a collector, have not chosen your preferred method of storage yet and there's enough space available - you may even want to keep your stuff in flac for now. The good thing about this is that you can still choose a different format if your knowledge about this has grown.
Amazing as well was your OP, it is astonishingly crap in every way possible. Rethink the post and post it on /g/ or maybe better on some board that is not even tech related, as everyone will just think you're an inbred cretin if you write like that.

If it's on bandcamp you can easily take the mp3s by looking at the dom tree, just ctrl+f "audio". This is all last resort of course, it'd be a lot easier to just get the full album at once off a torrent.

youtube-dl supports bandcamp.

amazon prime

useless bloat.
you should convert by yourself from lossless source to V0 at most (or better, find the transparency threshold for your ears by ABX testing some hard samples) if you have to use mp3, otherwise use a better format.
using about twice than necessary space for lossy is stupid.

FLAC

define "without being tracked"
there isn't such a thing as 100% anonymity you know

who said that you need 320 kbps?
why not 400 or 640 or …?
is it only because it's the maximum possible bitrate of a shit format (mp3)?
then do you know that youtube doesn't use mp3 for sound encoding?

m4a is a container, it can use a couple of different codecs, your comparison doesn't make sense

they are 128kbps CBR, and encoded with some crappy settings, even I can hear that they're shit. (not sure, maybe they fixed this recently)

true

ftfy

true.
it's about `--bitrate 128`.
it's not actually a bitrate setting, it's a quality setting with such a strangely named scale. it will actually spend more bytes on more complex music and less on easier. you will get wildly different average values depending on what do you listen, and that's totally fine. lossy encoding only makes sense with a quality goal and not a filesize goal, unless it's for streaming or VoIP.

true, + keep the flac for archive anyway

also go RTFM at hydrogenaud.io/

Damn. Y'all make this shit too difficult. I just got a browser extension that makes a nice "download mp3" button next to all youtube vids. Takes me off site, converts, downloads. Easy peezy lemon squeezy.

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why do you even need to convert?
this takes CPU power and makes quality even more shit.

is there any tracker that's filled in the gaping whole what.cd left?

The state of the board

rutracker, redacted

I use this, but with the addon. Works for me.

forgot link
convert2mp3.net/en/index.php

I prefer youtube-dl which also can integrate with mpv instead of some shady addon made by idk who

I use obsfucated Russian music download sites that don't rip a shitty mp3 from jewtube. Every mp3 is of superior quality.

i use aiff

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Flac on rutracker (or buy the CD on discogs if it's not here) then convert to opus in ogg.

AAC is wildly better than mp3, ignorant faggot.

rutracker

Use ogg or flac, faggot.

Why would I use ogg over mp3, the patent on mp3 is expired and more devices support mp3

I buy what I can from Bandcamp. The rest I rip from CDs that I buy. I don't want to fuck with streaming services like Spotify, so I don't.

Vorbis and Opus (especially Opus) blow the shit out of MP3 when it comes to size/quality ratio. For personal listening, there is no purpose at all in sticking with something shitty, old, and objectively inferior, patents aside. The only reason to use MP3 ever now is if you have some device or program that doesn't support better formats. I reencode my music into MP3 when I share with friends because they're all tech retarded, and though their machines will play ogg opus files, they get confused and don't understand how something can be a "music file" that's not an MP3.

How much of a lower bitrate can you use with AAC in average to keep about the same quality as a given MP3 bitrate? For example, would (let's say) 200kbps AAC give about the same quality as 320kbps MP3 in average?
Another thing: what about possibly increased power consumption (which translates to shorter battery life) by a more sophisticated decoding algorithm (which AAC would qualify as as compared to MP3)? Is that a factor, and if so are there any average figures to go by?
I'm asking because it would be great to have specific metrics to estimate the optimal bitrate to go with to minimize file size (and thus maximize available storage space on a mobile device) while keeping a reasonable quality and avoiding increased battery drain.

I use a utility similar to youtube-dl called bandcamp-dl. You your CLI you type bandcamp-dl BANDCAMPALBUMLINK and it will download it to a directory with album art and what not.

In your*

Encoding existing mp3 to opus is just going to lower the quality.

How low on an IQ curve they must be in order to be like that?
Fuck this shit, your post just made me angry

btw you might encode to Opus and rename the extension to .mp3, many players would be okay with that, and this tactic has some nice trolling potential

Go to a flea market.
Buy a shitload of CDs you want for cheap.
Rip and encode as flac or just wav.
Transcode whenever you want to throw it on some sort of device.

Resell or donate your CDs and repeat.
Old, early pressings are best before the studios started normalizing and amplifying everything, and are typically less expensive.

Get FLACs and recode to Vorbis yourself. No, Opus is not yet ready for prime time. At high bitrates it still suffers artifacts that aren't there with well-tuned AAC or Vorbis encoders.
t. audiophile

Any tips for such well-tuning?

how is it different from youtube-dl?
does it do something that youtube-dl doesn't?

>about the same quality as 320kbps MP3 in average
How would you measure the quality?
If you can't tell 320k MP3 from the original in ABX (blind) test (that is, if it's transparent to you), you cannot compare its "quality" to anything, it just falls into the area where you can't tell it from the original anymore.
By your broken/non-existent definition of "quality", MP3 would be better than MP3, because LAME (the recommended MP3 encoder) V0 (or even V2 or V3) is indistinguishable from the original as well as 320k, but it takes less space. (most people can't hear the difference between original and LAME V2)

The only thing that you could sensibly compare is the minimum average bitrate on your music collection when you encode it with some quality setting when you can't tell the difference in the blind test (ABX) no matter how hard you try.
For MP3, it will likely be somewhere in between V3 and V0, that's a lot lower than 320k for the vast majority of material and you didn't even switch the format yet.
For Opus, it's about what you get with --bitrate 128.
For AAC I didn't test, it depends on the encoder, the best encoder is proprietary (Apple's) which I never used because fuck vendor lock-in, second best is FDK AAC, you can set the bandwidth to your limit (depends on your age, you can test it) and pick the lowest VBR mode which gives transparency for you. It'll probably yield similar average file sizes as Opus or a bit more.

Now go and test it.

Difference would be small if any, unless you listen for days and with low loudness.
You can also use Musepack which supposedly is very simple to decode, actually more simple than MP3, yet still better than MP3 if you target transparency. But chances are they all won't make any big difference w.r.t. battery life.
Also hardware decoders are completely different beasts, you can only figure out by actually testing.

You don't estimate bitrate, you estimate quality setting, and the encoder decides where and how much bits it needs to fullfill your quality goal. CBR and ABR encoding will always be inferior, all else being equal.
And the only sensible way to do that is by ABX tests.

Proofs? (a problem sample and your ABX score)

Vorbis doesn't need any. Use default settings and quality level 6 (or greater, but usually nobody hears the difference at 6, so you are unlikely to hear it too).
For FDK AAC it's setting the appropriate bandwidth with VBR mode, as by default it cuts too much HF unless you use max quality level (5) which is overkill.
Can't say for other AAC encoders, but they are either inferior to FDK, or proprietary.

I do this with all .gifs to fuck with gif autist

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Oh fuck… Looks like I need to up the game and add filtering by content magic number to uBlock Origin.

Spectrograms maybe (to see how much of the frequency spectrum is being preserved at a given setting)?

It doesn't mean jack shit to your ears.
You can easily have a perfectly fine spectrum and shit quality.

wiki.hydrogenaud.io/index.php?title=Placebo_effect
wiki.hydrogenaud.io/index.php?title=ABX
wiki.hydrogenaud.io/index.php?title=Transparency
You should read these.

For example if you choose to encode with WavPack hybrid at the lowest possible quality.
You'll get audible differences, yet all the frequencies will be preserved.
That alone demonstrates that your proposed "solution" of quality assessment completely falls apart.

So there is literally no objective way to measure which of two different lossy encodings with the same bitrate is the "less lossy" one (and thus no way to tell which encoder is effectively more efficient)?

I'm not an expert on these matters, I just mentioned/proposed something which can be frequently seen in the wild (such as spectrogram images accompanying audio releases). It was more of a question than a statement.

If you like listening to music under water, it certainly is.

None that I know of. There was an attempt to develop an audio similarity metric (PEAQ and a few others), but IIRC they are all patented and hard to get to work.
That leaves us with ABX testing.
If you don't have time or nerve to go through them, you can pick what was found to be transparent in public listening tests and hope for the best, and it's gonna be fine if your ears are close to average (which is overwhelmingly likely).

they can be used only to detect that a specific kind of lossy compression was done (for some of them). that is, to tell if it's likely to be lossless, or not. other than that it's impossible to tell anything else from spectrogram images.

Citation needed.

also, I will reiterate, the more reasonable approach to compare codecs is not that, but
which of two different lossy encoders needs less bits to achieve transparent result if you use both of them in the optimal way (that is, the way recommended by the developers as they certainly know better)

by the way this also means that if you, for some weird reason, want to encode everything with a bitrate above ~200 kbps, then the choice of codec hardly ever matters — as long as you don't use some really terrible shit like WMA or ancient MP3 encoders, etc., you won't be able to tell the difference.

That actually pretty much was what I had been asking in the first place - specifically, how much less bits (e.g. by a factor of what, 1.5? 2? 2.5? etc.) the more efficient encoder roughly needs in average (in this particular case AAC vs MP3, but could just as well be another pair of encoders I guess). For instance, if someone used to use 192kbps MP3 on his mobile player and was fine with that quality/size compromise, how much could he go down with bitrate if switching to AAC (reencoding from lossless and not from the MP3s, needless to say I guess) to save storage space while roughly maintaining quality. Would a) going from 192kbps MP3 do 125kbps AAC be in the right ballpark, and b) would decoding the AAC files not reduce battery life due to more CPU usage, that were the questions basically.

It really depends on your ears and brain and which AAC encoder you pick, but from my experience the factor is between 1.5 and 2.
b) battery life should not change

Thanks a lot, that answers what I was asking (so my estimate of 125kbps AAC in place of 192kbps MP3 seems to have been in the right ballpark).

Is 64 kbps aac really viable for someone who would be fine with 128 kbps mp3 (who wasn't back in the winamp 2 days, most music was that quality)? It sounds very tempting if storage conservation is a primary goal, given that an hour long album would only be about 27 megabytes.

1) I'd say that 160 is transparent for a good AAC encoder (which are all proprietary, sadly)
2) AAC is incredibly fast to decode (just test with ffmpeg).

The artifacts aren't comparable. At 64kbps, only opus or he-aac don't sound like shit.

THE ABSOLUTE STATE OF Zig Forums

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You want me to cite my ears? Why don't you sage your reply to show how ass mad you are right now?

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I'll keep finding more ways to bypass your blocks, there is no stopping me.

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That's okay, m8. Nobody really fell for your retardation.

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do people actually think the difference in quality between 320k mp3 and 1000k flac is negligible? i have tinnitus like a motherfucker and still can hear the difference like night and day

I remember reading somewhere youngfags actually prefer the shitty versions full of bad encoding artifacts because it's what they're used to. Maybe with good encoding for some songs you can get a mp3 with no noticeable difference from FLAC, otherwise I agree it's there unless you're playing on your shitty laptop speakers or something. If I remember mp3 right the problem is if you have quick changes in the wave

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wew

I can see cheap parents playing pirated mp3s to their children, never taking them to live performances, so their brains never grow to be capable of hearing the difference. Some schools don't even have band anymore. It's entirely possible this is the root cause of autism.

Then prove it. Sample and ABX log would be a good start.

For most songs.
LAME is a really good encoder given the limitations of the format.
If you can afford bitrates produced with V0 or even V2, it's extremely unlikely that you will hear the difference if you exclude placebo.
Other lossy codecs just require even less bits to reach transparency.

Nobody's really mentioned anything about WAV yet.
I use uncompressed WAV whenever I can which isn't very often.

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Do you save all your images as BMP too?

Actually yes, sometimes
Of course, I also have the excuse of owning music players that actually support WAV files.

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while not supporting FLAC?

Nope. I have two Dell DJs, first and second generation.
The first generation supports WAV, MP3, and nothing else. The second generation supports those and WMA.
I know it's a bit weird, but that's just how Dell was back then (these were made in 2004 and 2005). They're actually pretty damn good players.

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Rockbox is also unavailable for them?
That sucks.

That's also right. I'm sure it's possible for a third party to make hacked firmware for them, but if they aren't supported now then they probably won't ever be. Weirdly enough, they don't support Zunes either, which are the other players I like.
But the quality's so good that I stick with them anyways. Audio's fantastic (which you'd expect from something that can play WAV files) and the build quality is superhuman.
Seriously, you could drop one of these things from a ten-story building and the only thing cracking would be the sidewalk.

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So it is in almost any device nowadays, as long as it has sufficient max output voltage and is properly screened from EM interference.
Even the cheapest DACs are pretty much "overkill" for our ears if used correctly.

That is true, however not all players can support lossless formats, even with rockbox. Plus it has a good equalizer. I don't know if that's impressive or not anymore, but I don't have a single player newer than ten years old.
Maybe I'll give it a shot though, if I can track down a decent player that's supported.

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And you do hear the difference?

Depends on how well the lossy stuff is encoded, and the speakers I'm using.
I'm not even much of an audiophile, I just really like having complete things.
If lossy's my only option though, then I won't complain.

...

Don't put words in my mouth.
1: There's no such format as "64kbps mp3", there's just "mp3", and
2: I didn't say that.
That said,
at this point it's very easy to prove that mp3 at 64kbps is not transparent.
I can tell it in blind test easily, in fact I did it some time ago just for fun.
At 128, however, you will need to try your best for most music. For a few kinds of music it will be already pretty much impossible, if you used latest stable LAME. For example if it's actually a mono record to begin with (same data in both channels). Some records were actually released like that.

If you have enough size to not have to sacrifice your collection size, that's a reasonable approach.

...

I can't tell if you're trolling or if you really need to go back to niggertits /b/.

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I'm pretty sure it's not since WAV tagging and replay gain is a major pain or impossible.

WAV supports tags. Some software may not, though.

WMA does have a lossless codec. I have to use it in my car that doesn't support flac.