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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.
I also spent a lot of time ripping my old CDs to FLAC and trying different MP3 and AAC encoder settings to get playback that felt transparent enough to me. I could never tolerate Sirius/XM radio streaming due to the horrid compression I heard with every futile attempt. I still seem to have more sensitive hearing than most people around me, but in my 50s I know it isn't what it once was.
I never had huge budgets, but did strive for hi-fi in my limited ways. I used things like toslink and HDMI to send raw PCM data from Linux to my Yamaha A/V receiver's DACs + amplifier to drive somewhat nice Polk tower speakers. But then COVID-19 happened, and this stuff was packed up to move house.
Nowadays, music playback is streaming with mundane "subwoofer + satellite" PC speakers or MP3 playback with a mini-SD card permanently parked in my car's infotainment system.
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 releases meeting the A) criteria, not just 44.1 kHz @ 16-bit ones.
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).
If I have an option to get a 16bit version of a recording or a high-res version, I choose the highest quality version very time
Same with a physical copy. A limited edition, better quality vinyl LP is more attractive if you are going through the trouble of curating a collection.
I’ve been curating a music library of digital files since before the iPod was released and I will always go for the highest quality version out of principle. I can always downsample it to any thing that makes sense.
However, the article claims that the final distribution doesn’t need to have a bit depth of more than 16. That does not match my experience. I can tell the difference between my renders that are 16 bit vs 24 bit. I cannot tell the difference between 44.1 kHz and higher sample rates, and that’s consistent with the math (Nyquist-Shannon), but bit depth is a different matter. Would be fun to participate in a double-blind test that includes my own tracks and others.
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.
https://www.carwow.co.uk/blog/carwow-quarter-mile-400-metre-...
https://en.wikipedia.org/wiki/List_of_N%C3%BCrburgring_Nords...
I’ve played with the nice toys, and they are nice, but for 100x the price, they barely deliver 1.5x the experience.
[1]: https://www.cnn.com/2023/03/30/world/plants-make-sounds-scn
There's multiple YouTube channels that I listen to as podcasts, that are professionally created and the creators presume that exported audio works like studio audio, so what you end up with is really quiet audio that can't be turned up without pre-processing.
If we distributed audio the same way we work with it in a studio, we could forgo a lot of problems.
Also, the human ear does have enough dynamic range to make 24 bits worthwhile, though that much dynamic range is rarely used in recordings, and that high of a bit depth provides no benefits within a small dynamic range. A 192 kHz sample rate, on the other hand, is always useless.
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.
[1]https://www.foobar2000.org/components/view/foo_abx
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.
As for how this relates to audio compression, in particular in the context of 2012. you are making a tradeoff of storage size and decompression cost. Maybe that doesn't matter to you, but maybe it either did in 2012 or still does.
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.
It would have cost the same for the entire stack to be 16bit/44.1kHz at every step, but with excessive resolution I can control the volume anywhere. The bits right before the analog conversion at the end are essentially the same whether I turn down the volume in the software player, the operating system, or the DAC/amplifier.
(2014) https://news.ycombinator.com/item?id=8689231 424 comments
(2015) https://news.ycombinator.com/item?id=10520639 228 comments
(2017) https://news.ycombinator.com/item?id=15127633 428 comments
(2019) https://news.ycombinator.com/item?id=19318898 314 comments
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
If I like an artist that I find on streaming, I buy an LP and get a lossless download for free. I still have a music library and I will never rent my favorite music.
Artists prefer to connect directly with their fans and BC is probably the best platform for people who care to pay and support acts directly. They have high res downloads and I import them.
Also the playback rate and the file rate are different topics. The former can get into scenarios more like the audio processing section of the article e.g. I had this one shitty headset for work which required me to set the volume to 1-2 (out of 100) on the computer and I could actually blind test tell when it was in 16 bit or 24 bit mode because it was cutting and boosting it so much it effectively lost precision in 16 bit mode.
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.
https://video.xiph.org/vid2.shtml
or on YT if you can't play it https://www.youtube.com/watch?v=cIQ9IXSUzuM
Now in terms of realistic audio encoding, 16 bit at 44.1 kHz is designed to be a faithful representation as far as human hearing is concerned. Can someone with a trained ear potentially tell the difference between that and 24 bit at 192 kHz? In a studio environment it's possible. Most audiophile claims are dubious and a blind A/B test catches them out on most of it but the Nyquist-Shannon sampling theorem does not directly apply to quantized samples, it's about exact samples and with quantization, sampling rate is intertwined somewhat with the quantization depth.
So I guess the programmer equivalent is distributing .pdb's (or, symbols)
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"
[1] https://www.audioasylum.com/messages/pcaudio/119979/
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.
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.
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.
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.
Some previous discussions:
2023 https://news.ycombinator.com/item?id=34698427
2022 https://news.ycombinator.com/item?id=30138561
2019 https://news.ycombinator.com/item?id=19318898
2017 https://news.ycombinator.com/item?id=15127633
2015 https://news.ycombinator.com/item?id=10520639
2014 https://news.ycombinator.com/item?id=8689231
2012 https://news.ycombinator.com/item?id=3668310
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 oxygen platinum plated HDMI cables for the better experience!
/s