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.
* Some people are still making this mistake, despite information on the (many) ways to do it the right way being widely and freely available!
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.
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.
My favourite: "audiophile-grade" audio players which allocate a single continuous 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 continuous 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 continuous memory"
Thanks for the laugh... this is absolutely bonkers. In case anyone is wondering, before sound hits our ears it has to go through a digital to analog conversion, which takes place on hardware independent of the CPU, operating with its own clock and buffers etc.
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.
Don't forget to buy the new low oxygen platinum plated HDMI cables for the better experience!
/s
[1]: https://www.cnn.com/2023/03/30/world/plants-make-sounds-scn
On a tangent, whenever someone mentions LP sounding warmer or whatever I like to point out that I prefer wax cylinders (a.k.a. phonograph cylinders).
It may be simultaneously true that:
A) Humans cannot tell the difference between 44.1kHz/16-bit audio and any higher resolution, and
B) For a particular song, the best commercially available 44.1kHz/16-bit version may not be the best commercially available version
"The quality of the particular mastering can still make a noticeable difference regardless of the ability for the digital sampling rates to perfectly represent it perceptually"
Just to be clear that the statement applies to any such releases, not just 44.1 kHz @ 16-bit ones.
To try to imagine something similar: the human eye is unable to see UV light, yet fluorescent paint has a visible quality of its own compared to "normal" pigments.
It kind of changed me a bit when I ran through 20 lossless tracks I had re-encoded to various mp3 bitrates and realized that even on a fancy system, it can be really hard if not impossible to discern even moderate lossy from lossless.
If you are an audiophile geek, really think about if you want to try this, the reality check might crack your foundations.
I use a DAC by focusrite which can do 24-bit, and if I want to listen to higher fidelity audio on my planer headphones then I should be able to. Why should I limit myself to 16-bit
You can the focus on other things.
Example: I Bought the best skis possible. Now I know I need to just focus on my skills and not blame the equipment.
The problem is the people spreading myths and disinformation out of ignorance or to promote their enterprise.
The weak links are producers/mastering-engineers, speakers/headphones and the room when using speakers.
https://video.xiph.org/vid2.shtml
or on YT if you can't play it https://www.youtube.com/watch?v=cIQ9IXSUzuM
That’s not why I go for High-Res stuff, though.
It’s all about archival, at least for me. With a 24/192 Master in FLAC or ALAC, I can downsample to whatever the destination form factor is. I can transcode to a 320kbps MP3, or a 16/48 WAV stream for a smart speaker, or a 24/96 stream for the theater. The point isn’t that I can hear the difference, it’s the fear that I might lose something irrecoverable by sticking with lower-quality files for bulk storage. Once data has been discarded, it cannot be retrieved, and that influences my preference for storage (and is also why my BD/UHD rips are into MKVs, no re-encoding).
Now that being said, I will absolutely hem and haw and ABX different releases to determine if I opt for the 16/44.1 CD rip of an album from the 80s or the new 202X remaster in 24/192 (spoiler: almost always the former), and I absolutely prefer anything with classic instruments (Jazz, Classical) in higher-quality formats because of a subjective perception of a wider, clearer sound stage, though this is almost certainly a psychological effect from performing in concert bands and orchestras rather than physical or objective in nature.
Like I tell newcommers: if it sounds better enough to you to warrant the purchase price, then that’s all that really matters. Enjoy the hobby.
There's some ways to do band-limited distortion but...they aren't nearly as widespread, easy, or universal as band-limited oscillators.
Ring modulation is funny though because you'd ideally want the sidebands to modulate down by default rather than filter them out, that's why you're using it.
192 for mixing and mastering can be useful especially if you're doing a lot of effects, especially anything that pitch shifts. But I've seen low quality phone-microphone recordings make it to the master; if you capture lightning in a bottle, it hardly matters what the settings were, what the microphone was, or anything else.
I’ve played with the nice toys, and they are nice, but for 100x the price, they barely deliver 1.5x the experience.
teach•53m ago