This really is driving a muscle/super car, or drinking expensive wine. At the end none of specs or tests matter. It is a form of art. If it makes the listener feel better (even if its just psychological) then its probably worth it.
munchler 4 minutes ago [-]
To expand on this a bit, I appreciate some audio overkill because, if I do hear sizzle or distortion, it eliminates one possible reason and helps me figure out what’s actually happening.
It’s like having gigabit internet to my house: I don’t actually need it, but when a website is slow, I know the problem isn’t in my internet connection.
cozzyd 8 minutes ago [-]
What a human centric view. I like my music to scare neighbor's pets.
PcChip 6 minutes ago [-]
I'm curious if the audio was being sent bit-perfect to the DAC for all of these tests (ALSA direct), or if it was being run through the audio mixer and being resampled
I can always tell if my 44.1 songs are being resampled to 48 because they're being run through the OS mixer
dist-epoch 2 minutes ago [-]
Proper audio resampling should not be identifiable. Of course, the OS mixer probably doesn't do proper (CPU expensive) resampling.
But a quality audio player should account for this and do it's own.
speak_on 5 minutes ago [-]
At a minimum, anything above 16/44.1 requires far more than just files: monitors, a treated room, listening position, DAC, etc... but most importantly - a trained ear. That last one is the most uncomfortable truth.
Blackthorn 1 minutes ago [-]
Are you, per chance, a dog posting on the internet? Since 44.1khz sample rate is already past the range of the human ear, regardless of training.
So I guess the programmer equivalent is distributing .pdb's (or, symbols)
Blackthorn 7 minutes ago [-]
Pretty good analogy. Thing is though, the person who receives the 16-bit, 44.1khz music file can always upsample it to 192khz and not lose anything in the process (heck, lots of audio stuff oversamples internally to this level or beyond, for extra aliasing headroom!). I'm not sure about expansion from 16bit to 24bit though, downward expansion isn't necessarily perfect.
viccis 14 minutes ago [-]
If you try to use empiricism when it comes to certain groups audiophiles, you are going to be sorely reminded that it's basically the equivalent of healing crystals for a different type of person. 24/192 is useful for mixing/mastering, but completely unnecessary for the end product to distribute for listening.
evo 8 minutes ago [-]
24/192 is also great for digital synthesizers--if you're generating a waveform like a sawtooth that has theoretically instantaneous transitions, they can eat as much frequency as you can give them. Running at 44khz loses noticeable high-end content.
Most modern digital synths have already caught onto this and run internally at much higher sampling rates even if their output gets downsampled, but sometimes you run across a vintage plugin that runs at the host audio rate and working in a higher sampling rate is audible.
Blackthorn 5 minutes ago [-]
You can generate perfect band-limited sawtooth waves at 44.1khz, there are multiple techniques for doing this and most production digital synthesizers use them.
Oversampling gives you headroom for aliases for the rest of the synth that is more vulnerable to it.
evo 2 minutes ago [-]
Yeah, I was oversimplifying a blit, the raw waveforms are usually okay, but I distinctly remember old-school VSTs where you couldn't achieve a nice saw lead at 44.1.
Aldipower 6 minutes ago [-]
Even with mixing/mastering 96khz is enough for persisting to files. But as another commenter said, 192 is useful, if you bend and stretch samples!
lokar 13 minutes ago [-]
I wonder how many people think that 24 bit audio encodes 50% “more”
recursive 10 minutes ago [-]
It is 50% more headroom above the noise floor in logarithmic decibels.
dist-epoch 9 minutes ago [-]
The whole audiophile industry is built on stuff which doesn't make any sense
My favourite: "audiophile-grade" audio players which allocate a single contignuous buffer of RAM into which they load/decode the whole .WAV/.FLAC file, because supposedly the CPU "jumping" between "fragmented audio" causes audible "jitter".
Of course, they don't know that what looks like contiguous memory to user-code is probably discontinuous in kernel/physical RAM.
Didn't check in many years, I wonder if they created kernel level players to account for that, to have "true contiguous memory"
metalman 12 minutes ago [-]
sheeesh , measly 24-bit/192kHz
of course it makes no sense, unless it is downloaded through low oxyegen wire, which somehow and unfathomably, must have been omited or forgotten.
b3orn 8 minutes ago [-]
If it has been transmitted via hollow-core fibres it will obviously sound hollow.
trashcluster 13 minutes ago [-]
24 bits is now ubiquitous and 32 bit is becoming the norm in recording studios.
evo 5 minutes ago [-]
32-bit float has become popular in filmmaking/field recording equipment lately because, with a microphone preamp that supports it, you can capture the entire dynamic range of the microphone--there's no accidental clipping if you drive the gain stage too hard.
It's a bit redundant for a skilled technician, they're already used to setting the gain staging, inbound compression, and feathering the mics to avoid this in 24-bit, but if you're handing a boom mic to a novice and have a scene where e.g. someone's whispering and another person's screaming, it can be nice to not have to worry about it.
lysace 9 minutes ago [-]
That use case is literally addressed in the first sentence.
Aldipower 8 minutes ago [-]
[dead]
haunter 5 minutes ago [-]
The more the bits the better the music, easy as one two three
Don't forget to buy the new low oyxgen platinum plated HDMI cables for the better experience!
/s
Rendered at 16:58:41 GMT+0000 (Coordinated Universal Time) with Vercel.
It’s like having gigabit internet to my house: I don’t actually need it, but when a website is slow, I know the problem isn’t in my internet connection.
I can always tell if my 44.1 songs are being resampled to 48 because they're being run through the OS mixer
But a quality audio player should account for this and do it's own.
So I guess the programmer equivalent is distributing .pdb's (or, symbols)
Most modern digital synths have already caught onto this and run internally at much higher sampling rates even if their output gets downsampled, but sometimes you run across a vintage plugin that runs at the host audio rate and working in a higher sampling rate is audible.
Oversampling gives you headroom for aliases for the rest of the synth that is more vulnerable to it.
My favourite: "audiophile-grade" audio players which allocate a single contignuous buffer of RAM into which they load/decode the whole .WAV/.FLAC file, because supposedly the CPU "jumping" between "fragmented audio" causes audible "jitter".
Of course, they don't know that what looks like contiguous memory to user-code is probably discontinuous in kernel/physical RAM.
Didn't check in many years, I wonder if they created kernel level players to account for that, to have "true contiguous memory"
It's a bit redundant for a skilled technician, they're already used to setting the gain staging, inbound compression, and feathering the mics to avoid this in 24-bit, but if you're handing a boom mic to a novice and have a scene where e.g. someone's whispering and another person's screaming, it can be nice to not have to worry about it.
Don't forget to buy the new low oyxgen platinum plated HDMI cables for the better experience!
/s