Autistic Audiophile Formats

I can stop anytime i want

Rrright Sup Forums ??

Other urls found in this thread:

chiru.no
chiru.no/dl/
chiru.no:8081/highres.flac
linnrecords.com/linn-what-is-a-studio-master.aspx
en.wikipedia.org/wiki/Nyquist–Shannon_sampling_theorem
xiph.org/video/vid2.shtml
twitter.com/SFWRedditGifs

No, your journey has just begun.

chiru.no

chiru.no/dl/

>.mp3

Just requested a .dff

Listen to chiru.no:8081/highres.flac to stream 352KHz audio

starts acting buggy when I try to change volume mid-stream

>352KHz audio
why?

out of nowhere a DSD BRO appears

*fistbump*

goddamn i am eating up my storage with this habbit

It's the future.

my dirty little secret is that i downsample everything down to 24/96 PCM so i can feed my room eq and digital crossovers but its still pretty damn gut

i got native DSD on my PMP so i aint to fussed

No son, you've got a placebo addiction. There is no cure other than not being a weak minded bitch, which is clearly impossible for you.

>tru autist

Don't talk to me about that money pit.

My Grado Stylus just broke....and I just bought some new records. Im fucking pissed. Hard to beak a DAC.

>have literally had bad lead out groves in records snap needles

Shits fucked I'll stick to owning the album for the artwork and downloading a 24bit rip from soulseek

Here's my audio formats of choice:
.acb
.adx
.awb
.bcstm
.bfstm
.brsar
.brstm
.hca
.miniusf
.rwav
.sns
.wem

>768000 Hz
>32 bit
>autistic
An actual autist would have the knowledge to know this actually decreases quality.
What you're looking at is just yet another retarded "enthusiast"

...

Directly stimulates your prostate

There's always this

>collect Audio with highest possible sampling rate
>buy $5 headphones
>pretend you're audiophile
Sup Forums

All this shit just to listen to those 2 Death Grips songs

>not sampling your music at 13.5MHz

But from what source was it sampled from? Theres no point of high sampling rates if the master is of lesser quality to begin with.

>linnrecords.com/linn-what-is-a-studio-master.aspx
Studio Master files are encoded at 24-bit or higher, and currently up to 192kHz. This is so close to analogue quality that it is virtually impossible for the human ear to perceive any difference. Therefore we feel this is the best format in which to be offering our music. This is the level that most music is recorded at these days, and that is the resolution that we offer it to you, so it doesn't get any better!

>Not using Opus 1.2@64kbps for everything
Your ear can't tell the difference.

$extensions = array('3gp', 'aa3', 'aac', 'ac3', 'aif', 'aiff', 'amd', 'ape', 'asf', 'aud', 'avi', 'd00', 'dff', 'divx', 'dsf', 'flac', 'flv', 'gbs', 'hes', 'hsc', 'it', 'kss', 'laa', 'm2ts', 'm2v', 'm4a', 'm4v', 'mad', 'mid', 'mkv', 'mod', 'mov', 'mp2', 'mp3', 'mp4', 'mpc', 'mpeg', 'mpg', 'mtm', 'nsf', 'ogg', 'oma', 'opus', 'rad', 'raw', 's3m', 'sa2', 'sid', 'smk', 'sol', 'spc', 'stm', 'str', 'swf', 'tak', 'ts', 'tta', 'umx', 'vgm', 'vgz', 'vob', 'voc', 'wav', 'webm', 'wma', 'wmv', 'wv', 'xa', 'xm');

> virtually impossible
What a fucking cop out.

They sure ain't getting my money.

>24bitx192khz=~33MB/min
Point being that virtually all modern studio masters are 24bit and 192khz, so if you exceed this you are in essence sampling nothing extra but using way more space to achieve the same results.

I grabbed a dsf song that was about 85MB/min so apparently this is 52MB/min of wasted space.

Who gives a shit? It's all placebo anyways.
Double blind test is like a curse word around here.

not using MP3 320 or AAC so that devices can actually fucking play the music. flac fags fell for the meme so hard

Autism. Just use regular perfect flac's.

I care if going to the trouble to download or rip music/video only to double my storage requirements and increase cost to in essence store nothing extra.

>not building an ADC out of MESFETs and sampling at 44.1GHz

Every single one of you is a massive pleb.

the irony is that when that will get downsampled by your audio driver you will lose quality over the same file at 44.1k

anything higher than 24/48 is a waste of space in production

Not true, when applying digital processing before exporting you're going to want 32bit/192KHz.

I was taking the piss.
There is little point sampling your music above 40-60kHz for listening purposes.

funny how iphones still to this day max out @ 24/48.... its almost as if they knew something

Anything over 16/44.1 is a waste for listening. 24/44.1 for multitrack recording, but always bounce down a final mix to 16bits. 48 is fine but it's completely indistinguishable. The human ear tops out at around 21Khz, and 44.1 provides you with up to 22,050Hz.

Source: work at a music studio, we laugh at idiots like you and opie.

>I've never touched an audio tool in my life

what fucking smack are you talking bitch nigger ?

this is a photo from the actual transcode from a Denon 302 to RME ADI-2Pro

real signal path - real cutting edge gear

hahahahahaHAHAHAHAAHHAHA
Oh god, you actually bought this shit? Hahahahahaa

>he can't taste the clock jitter of his cpu in his audio playback
massive pleb

No megadouche - I torrented the resulting super awesome .wv file for free

Obligatory.

dat drop

Even knowing what that shit is makes you fucking weird imo.

anything over 16/44 is a meme

>doped GaAs
>FET
>not sampling at 1.411 THz with InP HBT
what a massive pleb

i could say the same about you tripfags

close for general listening (not production) 24/48 is the sweet spot when you actually have to store a half decent hi res library on a pmp

...

>what fucking smack are you talking bitch nigger ?
en.wikipedia.org/wiki/Nyquist–Shannon_sampling_theorem
If you understood how digital music works, you wouldn't be yet another retard who things bigger numbers = better
Sampling that far above nyquist literally makes the quality worse

>Sampling that far above nyquist literally makes the quality worse
not him but a complete stranger here, I think you should rather learn to signal processing better. it does not make it worse if the quantisation noise can be kept the same. in fact, it provides better suppression of the harmonic frequencies and therefore makes sure the signal does not contain any components higher than 20 kHz.

>provides better suppression of the harmonic frequencies and therefore makes sure the signal does not contain any components higher than 20 kHz.
Wrong. The reason 44khz is used is exactly because anything above that is superfluous at best and degrading in playback at worst.

If you need a dumbed down explanation with pretty pictures to help you understand, you can look here
xiph.org/video/vid2.shtml

>dumbed down explanation
perhaps that is exactly what you need in the topic of what is aliasing.

>Wrong
I'm sorry that you can't change the physics and the fact that you'll get the alias of your entire signal spectrum occupying between 22.05-infinity kHz with a decaying coefficient over frequency.

>aliasing
Simply, objectively wrong.
This is exactly why you need to watch the video. It deals exactly with the wrong perceptions you have.

Jesus christ, i need to get into the audiophile industry ASAP so that I can cash in on retards like this guy.
Pic related

nice reading comprehension you've got there

>namecalling
>exemplifies with a digital data transmitting component where medium quality does not matter as long as forward error correction can overcome bit error rate
and you infer all this out of my true statement that a 48 kHz DAC provides a more pure signal with less aliasing than a 44.1 kHz DAC? projecting on Sup Forums has gone up to a new level.

>The reason 44khz is used is exactly because anything above that is superfluous at best and degrading in playback at worst.

(You) are a fool. 44.1 was picked because it had something to do with the video frames of the helical scan adat like tape format they were storing the digital information to - it was a hardware decision - not an aural one

Telarc / Soundstream were doing the very first digital hifi recordings way before Sony at 50khz but were using linear tape with many parallel tracks and werent locked to anything

also research has proven ultrasonics do have have harmonics that modulate the audible human hearing range

Someone obviously already cashed in on this guy but maybe I could still get 1200$ out of this guy by just making some bullshit 5$ "anti-aliasing" box that sits between the source and his headphones

nice counter arguments with indisputable proofs