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24-bit/192kHz music downloads and why they make no sense (2012)
- teach 3mo ago(2012)
- lokar 3mo agoI wonder how many people think that 24 bit audio encodes 50% “more”
- recursive 3mo agoIt is 50% more headroom above the noise floor in logarithmic decibels.
- z_open 3mo agoAs they say, most people listen to their music with equipment. Audiophiles listen to their equipment with music.
- nntwozz 3mo agoThis is perfect, thank you this goes straight into my long-term memory bank. 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).
- fecal_henge 3mo agoYou Edison shill.
- _kb 3mo agoThose wax cylinders are a modern hack. The curved surface distorts the real artistic intent. The only way to appreciate the true beauty of sound is a the purity of soot etchings on a phonautogram.
- nntwozz 3mo agoI will add this to my repertoire :) I had a good laugh listening to the sample at https://en.wikipedia.org/wiki/Phonautograph https://en.wikipedia.org/wiki/Phonautograph The sound is very pure indeed.
- mingus88 3mo agoThat’s true, but I consider myself a collector. Think of how a comic book collector operates. 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.
- pimeys 3mo agoI might be something from the middle. Yes, I did spend a hefty 5000 euros to my headphone setup. And yes it sounds absolutely magical and every day I'm happy listening to music with it. But I also have a large multi-terabyte music collection, I follow new music, go to concerts, go to parties, talk about music with my friends in signal group chats. It's a hobby, and when you get a bit older and start having some savings, if you love music treating yourself with a better system is not that crazy.
- UmYeahNo 3mo agoWhen I got old enough to finally afford those toys I discovered I couldn't hear above 16khz anymore.
- pimeys 3mo agoIt is not only that. It's the spacing, how the bass sounds, separation of instruments. There's so many interesting headphones in the midrange to try out. Compare the Hifiman HE1000se to Heddphone 2 GT, or to Focal Clear MG and you'll understand. Also with HEDD you get a handcrafted device made in Berlin. And if you go with nicer cables, they are very beautifully done and feel great. There is no difference in sound of course. Some people like jewelry, I can get similar enjoyment from beautiful audio equipment and cables.
- az226 3mo agoWhat’s the quality of this trove? As in bitrate or similar.
- pimeys 3mo agoDepends. I'm more into finding certain masters. And some of the albums are DSD tape transfers. DSD if that was the original recording format, if it was mixed and PCM was needed, DXD flac. And so many CDs of course.
- eimrine 3mo agoWhy not to listen to the equipment from time to time? And why not to learn how class D sounds?
- dijit 3mo agohuh... So I guess the programmer equivalent is distributing .pdb's (or, symbols)
- Blackthorn 3mo agoPretty 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.
- gizajob 3mo agoYou’d be adding 150khz and 8bits of nothing.
- viccis 3mo agoIf 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 3mo ago24/192 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 3mo agoYou 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 3mo agoYeah, 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.
- Blackthorn 3mo agoIt's tough to tell without specific names, but I imagine a lot of particularly old* VSTs were written to use naive sawtooths rather than perfect band-limited ones, which would have terrible aliasing at 44.1 khz. Oversampling those would help a lot! * Some people are still making this mistake, despite information on the (many) ways to do it the right way being widely and freely available!
- trashcluster 3mo ago24 bits is now ubiquitous and 32 bit is becoming the norm in recording studios.
- lysace 3mo agoThat use case is literally addressed in the first sentence.
- evo 3mo ago32-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.
- metalman 3mo agosheeesh , 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 3mo agoIf it has been transmitted via hollow-core fibres it will obviously sound hollow.
- Tsarp 3mo agoThis 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 3mo agoTo 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.
- ubercow13 3mo agoWould 192khz audio result in less sizzle and distortion? Or more audible band IMD from the sound >22khz
- meowface 3mo agoCorrect. I've paid for Tidal for a decade because I just like the peace of mind that it's closer to the original recording. I'm sure it's mostly placebo, but I like it.
- yellowapple 3mo agoIt's also sort of an inverted “Van Halen demanding a bowl of M&Ms with the brown ones removed” thing for me, too. The vast majority of my Tidal listening happens over Bluetooth, so that 24bit/192kHz FLAC stream is just gonna get downsampled to 16bit/48kHz anyway because that's all any Bluetooth speaker or headset is capable of doing — but the fact that it's an option in the first place signals that other things are being done right, too (namely: that Tidal's whole “we're the streaming service that pays artists the most per listen” premise actually has some semblance of merit rather than being complete marketing bullshit; while recording quality ain't the strongest signal possible for that, it's certainly a good sign when musicians/publishers are willing to send over the highest-bitrate lossless recordings they've got and not just the same ol' compressed-to-shit MPEG audio you can yank off YouTube for free).
- dist-epoch 3mo agoThe whole audiophile industry is built on stuff which doesn't make any sense 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"
- bellowsgulch 3mo agoThe latter is probably true, but the former does actually happen, and it's easy to accidentally do--lossless or not.
- lmc 3mo ago> 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 memory" causes audible "jitter". 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.
- wat10000 3mo agoIn addition to that, while it is possible to hit a delay and run out of buffer because memory access is slow (the most obvious would be if the input got swapped to disk at an inopportune moment), but the audible effect is really obvious. This isn't some subtle "oh my music sounds ineffably worse" effect, it's "my computer is glitching and my music is unlistenable."
- justsomehnguy 3mo agoAm486DX/100 was enough to decode and listen an MP3 at 22KHz (and maybe mono?) and was more than enough to listen for 44/16/2 PCM. It's 31 y.o. today.
- cozzyd 3mo agoWhat a human centric view. I like my music to scare neighbor's pets.
- Aldipower 3mo ago[dead]
- PcChip 3mo agoI'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 3mo agoProper 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.
- rasz 3mo ago"proper" resampling was expensive in 1997 when Intel was introducing fixed sampling AC'97, but was below noise floor of CPU load meter in 2007 when Microsoft released Vista killing hardware mixing.
- PcChip 3mo agoI'm also one of those audiophile crazies that obsesses over which metals to use in cabling, power filtering, swapping opamps, and builds their own DACs, amps, and speakers
- PaulDavisThe1st 3mo agoIf you're not on a US-based IP, you should check out https://src.infinitewave.ca/ https://src.infinitewave.ca/ It is an incredible resource to see the quality of the resampling algorithms used by the actual production software likely used in any digital audio workflow. You will see that while the best are indeed almost 100% transparent, many are not.
- nok22kon 3mo agoI remember using Adobe Audition for resampling audio, this site shows I had good intuition your software is among the best, but not pitch black best :)
- speak_on 3mo agoAt 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 3mo agoAre 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.
- clawlor 3mo agoMax representable frequency is half the sampling rate (nyquist-shannon theorem), which is still a bit above normal but IIRC the extra headroom has something to do with eliminating aliasing
- Blackthorn 3mo agoIndeed. And what is the max frequency that a human can hear?
- Rotundo 3mo agoDepends on age of the listener, on average, 30 to 50 year olds hear a maximum frequency of 14 to 16 kHz.
- Blackthorn 3mo agoRight. Which are quite below 1/2 of 44.1k!
- OkayPhysicist 3mo agoSure, but those are averages. I'm 30-ish, and my hearing doesn't cut out until somewhere in the 21kHz range. When I was younger, it was even higher. One of my roommates in college had one of those anti-rodent high-frequency noise generators, we almost came to blows over it.
- haunter 3mo agoThe more the bits the better the music, easy as one two three Don't forget to buy the new low oxygen platinum plated HDMI cables for the better experience! /s
- 0l 3mo agoObligatory mention of https://xiph.org/video/ https://xiph.org/video/ which clears up a lot of misconceptions.
- jerf 3mo agoIf you can't hear the squeals of the plants [1] in the studio's reception area, are you really getting the full experience of a piece of music? [1]: https://www.cnn.com/2023/03/30/world/plants-make-sounds-scn https://www.cnn.com/2023/03/30/world/plants-make-sounds-scn
- Blackthorn 3mo agoOh great. And here I thought that fantasy literature where forest elves could hear the screams of the plants they stepped on when they walked was just that -- fantasy.
- SketchySeaBeast 3mo agoTriffid music.
- rahimnathwani 3mo agoThe article says "I've run across a few articles and blog posts that declare the virtues of 24 bit or 96/192kHz by comparing a CD to an audio DVD (or SACD) of the 'same' recording. This comparison is invalid; the masters are usually different." 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
- zamadatix 3mo agoWhile 100% true, I'd phrase B) as: "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.
- black_knight 3mo agoI usually A/B test the different versions before choosing my canonical one. I will listen to the same sections in each version, flipping back and forth to hear the differences. It is incredible how much finding the right master improves the experience of listening to a track. Often times that means I end up with a hi-res version, but not always.
- Arodex 3mo agoI completely accept that human audition has limits that are easy to determine by playing a pure sound. But is it the same with music, where multiple frequencies are played and interfere with each other? Aren't some harmonics or effects created by these "inaudible" frequencies? 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.
- nok22kon 3mo agowhen beams of ultrasounds interract they can produce audible frequencies this has practical applications
- WarmWash 3mo agoFoobar2000 has an extension that allows you to blindly test whether you can tell the difference between two tracks.[1] The prime use is to compare different encodings of the same song from the same lossless master. 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 https://www.foobar2000.org/components/view/foo_abx
- pimeys 3mo agoBut try out to stream that mp3 from your home server in lower bitrate to save data, e.g. as opus. And now you suddenly hear the lossy encoding. We store files in the highest quality because it gives us the option to encode the music without audible loss of quality.
- vitamark 3mo agoI personally encode flacs as 192/256k opus from the start and that's fairly enough for most data save purposes, so no reencode for streaming is needed that's a bitrate of 1GB per 9-12h, and for cases when it's too much I just have cached music on my device (I'm lucky to have mostly empty storage on my 256gb phone)
- me551ah 3mo agoNobody downloads music these days and everybody just streams. Audio at 24 bit still takes a small fraction of the bandwidth that 1080p video takes, so I don’t understand the hate for it. 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
- mingus88 3mo agoCounterpoint: bandcamp is doing well. Vinyl sales are doing well. 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.
- zamadatix 3mo agoI don't think the hate is about people who know it doesn't actually sound different if the audio file is 16 bit or 24 bit or necessarily about receiving a few more bytes than they need, it's about the pushes by these types of streaming services/offerings or people insisting that it's supposed to be any better for listening when it's not. 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.
- pimeys 3mo agoWait, what? I do download everything I listen. And Roon is quite popular in the music communities. How else you can make sure you have that correct mastering of your favorite album?
- glimshe 3mo agoJust get one of those "hi fi" valve amplifiers from Amazon you see under $100. The valve already distorts the sound, so you don't need to bother paying more for low distortion anywhere else in the audio chain. Saved you thousands of dollars, done!
- PaulDavisThe1st 3mo agoDistortion is why people love the sound of vinyl. And its all good! It's perfectly fine to say "I prefer the sound when the whole mix (or just that guitar) ends up being subject to interesting and possibly harmonically relevant distortion at low levels". Just don't say "The version with the distortion is more accurate than the one without", because that's a lie.
- hackingonempty 3mo ago@xiphmont also made an amazing video response to the many responses he received to this article. Using analog equipment he busts a bunch of myths and demonstrates what really happens with digital audio. https://video.xiph.org/vid2.shtml https://video.xiph.org/vid2.shtml or on YT if you can't play it https://www.youtube.com/watch?v=cIQ9IXSUzuM https://www.youtube.com/watch?v=cIQ9IXSUzuM
- niccl 3mo agoThank you for posting this. I thought I knew a bit about what was going on with audio sampling and reproduction, but I learned a surprising amount from this well presented introduction
- justin66 3mo agoA classic.
- pringk02 3mo agoOne of the best educational videos on any topic ever imo. I can't help but watch it again each time. So incredibly well paced and accessible
- hobonation 3mo agoCounter: An ultra high bit rate solves the problem and you can stop worrying if it's the weakest link. 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.
- hackingonempty 3mo agoThe point of this article and video is there is no problem with 16-bit 44-kHZ PCM. It thoroughly covers the audible range and is there is absolutely no need for more when distributing music for humans to listen to. 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.
- RijilV 3mo agoI hate to be the one to break it to you, but high end skis make tradeoffs which are harmful to beginner or intermediate level skiers... also there's sorta no thing as "best ski". what you'd want for high speed bombing double blacks is going to be different from off piste or moguls or snow park fun.... double also, skis wear out. Depending on who you want to believe it's as low as 20-30 days. Which, granted the average skier is at something like 5 days a year. but if that's you... triple also? 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.
- hobonation 3mo agoYou're acting like I don't have everything from Hellbents all the way to Fisher GS skis all the way into Vole Scaled Skis for Telemarking. Just don't buy shit, and you don't get to blame the equipment. And none of them are broken after 20 days unless it's low tide or I fuck up on a cliff band. I ski a minimum of 100 days a year, and the only thing you can notice after 60 days is some slight decambering on softer skis. I will say that boots tend to get soft around 100 days. But usually dealing with that's also a skill issue; get good balance, and you don't have to have the boot hold your sorry form in place. It's how people basically skied for a thousand years with leather boots: They were good.
- stego-tech 3mo agoI cannot hear the difference between 16/44.1 (and by extension, 16/48) and High-Res Content generally, be they HDCD, SACD, or just straight-up Masters from Qobuz. This is on multiple sets of equipment, ranging from El Cheapo earbuds all the way to HD800 cans and full-fledged tower speakers being bi-amped. 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.
- Cider9986 3mo agoI can't hear the difference between 128 kbps opus and FLAC.
- stego-tech 3mo agoAnd that's fine! I've got a flatmate who loves 320kpbs MP3s on studio monitors, I've got musician friends who swear by CD-audio and Sennheiser HD200s, and others who love how vinyl uniquely degrades over time on big speakers. The takeaway from these sorts of posts, at least in my opinion, should be two-fold: * Understand the physical limits of human senses and perceptions to help inoculate yourself against outright scams and grifts * Liberate you from the "tech grind" and allow you to enjoy what you like, how you like it.
- waffletower 3mo agoFor typical listening (though humans can perceive bone-conducted vibrations up to 100 kHz or even 120 kHz) 16-bit-fixed/44.1kHz is a high-fidelity transport format. As a DSP researcher, I prefer 32-bit-float/44.1kHz as a transport format. I often upsample to 32-bit-float/188.2kHz or even 32-bit-float/192kHz for signal processing applications such as high-fidelity reverberation via direct and FFT convolution. While the author advocates for the transport to ear use case, I would argue that 24-bit/192kHz provides greater fidelity and resolution for sound processing. I found the pedantic arrogance of the author to be annoying. But yes, the sampling theory is an important consideration -- but so is the quality of the actual digital filters used in the DAC->ADC pipeline. They are much more forgiving and less lossy at 192kHz.
- deleted 3mo ago[deleted]
- HelloUsername 3mo ago(2012) https://news.ycombinator.com/item?id=3668310 https://news.ycombinator.com/item?id=3668310 316 comments (2014) https://news.ycombinator.com/item?id=8689231 https://news.ycombinator.com/item?id=8689231 424 comments (2015) https://news.ycombinator.com/item?id=10520639 https://news.ycombinator.com/item?id=10520639 228 comments (2017) https://news.ycombinator.com/item?id=15127633 https://news.ycombinator.com/item?id=15127633 428 comments (2019) https://news.ycombinator.com/item?id=19318898 https://news.ycombinator.com/item?id=19318898 314 comments
- ChrisArchitect 3mo ago(2012) Some previous discussions: 2023 https://news.ycombinator.com/item?id=34698427 https://news.ycombinator.com/item?id=34698427 2022 https://news.ycombinator.com/item?id=30138561 https://news.ycombinator.com/item?id=30138561 2019 https://news.ycombinator.com/item?id=19318898 https://news.ycombinator.com/item?id=19318898 2017 https://news.ycombinator.com/item?id=15127633 https://news.ycombinator.com/item?id=15127633 2015 https://news.ycombinator.com/item?id=10520639 https://news.ycombinator.com/item?id=10520639 2014 https://news.ycombinator.com/item?id=8689231 https://news.ycombinator.com/item?id=8689231 2012 https://news.ycombinator.com/item?id=3668310 https://news.ycombinator.com/item?id=3668310
- rz2k 3mo agoMy good enough amplifier and DAC combo claims up to 24bit/192kHz, I use a cheap optical interface from my computer that claims up to 32bit/192kHz, and the streaming service I use serves most albums at 24bit/44.1kHz. 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.
- PcChip 3mo agoyou might want to see if your DAC re-clocks incoming optical, if not then it's relying on the cheap clock generator from your computer
- rz2k 3mo agoSome people have claimed to hear an improvement with an external clock on a Wiim Ultra, but I do not think it is possible to re-clock the WiiM Amp Ultra with an outboard clock. When I play from the computer, I'm not sure whether it is using the clock on my Mac, the clock on the optical interface, or the WiiM's clock. However, I do not notice any difference in fidelity when I use the Qobuz software player on my Mac or use Qobuz Connect to allow the player to directly stream from the source, so either it isn't a difference that I can hear, or the WiiM's internal clock is used for both sources.
- rasz 3mo agoThat cheap clock source is good enough for 16 GHz PCIE signaling, but Im sure you will be able to hear the jitter!
- porridgeraisin 3mo agoAfter that "audio player that lays out the file in contiguous memory to avoid cpu jitter" example from above, this one is my new favourite audiophile trope. "cheap clock on the computer" I giggled.
- LarsAlereon 3mo agoThe main benefit for me is that digital watermarking becomes completely inaudible with high-res audio, but I can sometimes clearly hear it in standard resolution.
- dlcarrier 3mo agoThere is a good reason to distribute it though, and compressed it doesn't really change the file size. 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.
- sholladay 3mo agoMusic producer here. High resolution audio is useful for editing and anywhere there might be downstream processing or format conversion that may or may not be high quality, let alone lossless. The article covers that pretty well. 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.
- nok22kon 3mo agothermal noise allows about 18-22 bits of real precision at audio level voltages, so it's plausible that 16 bit is somewhat limiting
- PaulDavisThe1st 3mo ago16 bit may limit it on the input side, but the question is more about human hearing's sensitivity on the "output" side ...
- guenthert 3mo agoWhat? You still operate your ADCs without active cooling? I'm writing in jest, but long time ago, -hp- used actively cooled FETs (not a very popular approach today as that caused problems with condensation and we have better FETs now).
- PaulDavisThe1st 3mo ago> I can tell the difference between my renders that are 16 bit vs 24 bit. established using double blind testing, I assume?
- manoDev 3mo agoThey make sense for so called audiophiles who don’t understand Nyqist frequency theory. It’s like photographers who are confused about the difference between raw and bitmap (jpeg), videographers confused about the difference between linear raw vs log vs gamma encoded, etc. Just because a data format with higher bit depth/sampling frequency/whatever exists for editing purposes, doesn’t mean it’s “better” or makes sense as a consumption format for a finished work.
- casion 3mo agoThey make sense for sound designers and derivative artists (e.g. sampling, which is a real artform). Forms of manipulation bring inaudible content into the audible range. Of course that doesn't mean audiophiles aren't being audiofooled by it, but there is legitimate usage.
- eimrine 3mo agoNyquist describes harmonic frequency while music can contain non-harmonic oscillations, also Nyquist theory tells about the maximum possible frequency, but this same frequency can be placed in such a way with respect for quantization points that it becomes a silence. I like your point about editing, because trying to mix too much tracks in 44100 or 48000 makes spontaneus click sounds which are not supposed to be here.
- geraldmcboing 3mo agoThe OP is a bit off with their description of why pro audio engineers work in higher bit rates and sample rates. We use 24bit to preserve low level sounds eg reverb, breaths etc and use 32bit float when recording as the headroom is so massive clipping is not an issue (other than of course still neeing to avoid overloading microphones with max SPL - cleanly recorded distorted sound is still a fail). Unclipping 32bit float feels like voodoo - I did a test, recording fireworks & unclipping the 32bit float recordings. I use microphones that can 'hear' up to 100kHz (Sanken CUX100K) and for film sound design playing 192kHz audio at half and quarter speed the results are very significant, and reveal there IS 'content' above human hearing. Irrelevant for general listening but very important for sound design.
- PaulDavisThe1st 3mo agoHave you ever actually checked the number of actual bits your ADC can use? Most 24 bit converters struggle to get to 18 bits. Nobody uses 32 bit float for recording (to do so is just to capture at least 10 bits of noise, most of that being brownian); its strictly a format for mixing and processing. You don't get any more resolution from 32 bit floating point than you do from 24 bit integer formats, but the result of "clipping" is less dramatic, hence the appeal of the format. While there is some evidence that non-auditory human sensory perception may be sensitive to ultrasonic acoustic waves, it's pretty weak right now, and somewhat in the "woo" zone. It may turn out to be significant, or it may not. I wouldn't base an audio production workflow that requires 4x the cpu power and 4x the disk space on such tentative claims, but you're welcome to.
- nullc 3mo ago> Nobody uses 32 bit float for recording Yes they do, almost all high end field recorders used for film work are 32-bits now and have been for much of the last decade, often with some fancy preamp integration so that there is no expertise required for gain staging the recording. (I believe the implementations use a second matched 24bit ADC with 48 dB less gain in front of it). The result obviously doesn't have a noise floor which is lower (as the noise of a room temperature _resistor_ gets in the way of that even at the 24-bit level) but they have more dynamic range so that your recording isn't ruined by hard clipping some unexpected loud sound. It's a big improvement for practical usage, and also likely does improve SNR somewhat because you can run higher gains without as much fear that you'll ruin the recording. The reason it would pay off is that the SNR loss you get from splitting the signal is easily smaller than the SNR loss you would get from gain reduction to avoid clipping. (maybe... capsule self noise is also limiting... at these levels, and usually people aren't using microphones designed for the lowest possible self noise unless they're doing something special)
- hgoel 3mo agoI still insist on the higher bitrate stuff. I don't expect to notice the difference, I just think that music where the artists have bothered to prepare those files is probably recorded with more care than otherwise. I'm not generally listening to big artists where this can just be expected, and while I don't have any evidence to support my belief, I choose to continue believing it. I'm not interested in finetuning everything in my life for efficiency.
- codedokode 3mo ago192 kHz vs 48 kHz can make a difference if you slow down the audio. If you pitch shift down 2 octaves, the ultrasonic range 20-80 kHz turns into 5-20 kHz and there will be large difference between 192 kHz and 48 kHz sources. However, I do not know if it would sound good because the mixing engineer cannot hear those frequencies and mix them properly, or the microphone might not catch it or some of the material could be recorded with lower quality. Also, sadly consumers are getting used to low quality audio nowadays - they often listen to lossly compressed audio on social media (sometimes decompressed and re-compressed several times) which is then re-compressed to send to bluetooth headphones, or played back on an awful smartphone speakers. Streaming services also use compressed audio.
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- ycui7 3mo ago24-bit was created because microphone want to record large dynamic range without gain switching circuit. 96kHz was created to better reproduce 20kHz high frequency, so the digital noise shaping filter does not need to be super sharp right at the Nyquist frequency. Both were introduced for a sound technical reason. beyond that, most are marketing non-sense to cheat consumers.
- ryankrage77 3mo agoI decided to test for myself, downloaded Lacinato ABX and tested a 32-bit 352.8Khz flac I had lying around, to the same file downsampled to 16-bit 44.1KHz. I couldn't tell any difference. Then I tried 192k mp3... still no difference. Couldn't reliably differentiate 128 or 64kbps mp3 either. I had to go down to 32k before I could be certain which was which, and even then I still had to listen carefully. Think I need to get my ears checked. I know I can't hear much above 15-16KHz but I didn't think it was this bad.
- DiabloD3 3mo agoYeaaaaaaaaaahhhh.... you might be a tiny bit deaf. OTOH, we know nothing of your audio equipment nor how its setup.
- ryankrage77 3mo agoHiFiMan Sundara headphones, focusrite scarlett 2i2 interface. Although, it turns out the Scarlett is set to 48KHz anyway, and I can't seem to change it easily under linux. Not that it seems to matter for my ears, lol. EDIT: Did some more ABX testing with a CD-quality track that I'm much more familiar with ('Introduction' from the Mirrors Edge soundtrack, which has been my go-to for comparing audio gear for the last decade). I could sometimes distinguish 128k mp3 this time, though interestingly, I got it consistently wrong rather than right. For some reason the compressed version seems to be my preference. Dropping to 96k mp3, I got it right 100% of the time - though only because there was a very noticeable difference in the stereo positioning of the first sound, rather than a difference in the quality of the sound. I think if it were mono I would still be unable to tell.
- DiabloD3 3mo agoSundaras aren't great, but are sufficient for this. Scarlett series is also good enough. At 48khz, with a tone generator (not rando Youtube videos or anything), you should be able to clearly hear up to around 16khz (as in, can tell pitch of tone), and be able to tell a tone is being played at all up through 18khz, and hopefully up to 20khz.
- kstenerud 3mo agoWhat's really interesting to do with all of these people arguing over audio formats (as always happens on HN) is to point a frontier model at this thread. In a nutshell: nullc, rahimnathwani, zamadatix and vor_ know their shit, and geraldmcboing and PaulDavis are technicially correct but talking past each other. speak_on and TheOtherHobbes are confidently wrong. And also: 44.1 kHz captures the entire human audible spectrum with room to spare, and 16-bit already goes beyond anything useful for listening. The higher resolution / sample size format is useful for production or archival purposes only. The two main reasons why you hear a difference between the two formats: (1) it's likely a different master, (2) tiny gain differences in the signal (salesmen use this trick, but it's also easy to do it by mistake).
- UltraSane 3mo agoThe only real advantage it has is being able to re-encode it into any lossy format you want. With modern storage capacity the size isn't much of an issue.
- BoingBoomTschak 3mo agoNo mention of Toole? Obligatory considering the subject, even if it's about acoustics and not digital audio: https://www.routledge.com/Sound-Reproduction-The-Acoustics-and-Psychoacoustics-of-Loudspeakers-Rooms-and-Headphones/Toole-Olive-Welti/p/book/9781032761930 https://www.routledge.com/Sound-Reproduction-The-Acoustics-a... And a talk by the author covering some of the material for those who prefer that: https://www.youtube.com/watch?v=zrpUDuUtxPM https://www.youtube.com/watch?v=zrpUDuUtxPM
- saidnooneever 3mo agothe make sense for DJs and producers. preferably WAV full quality. in DSP it matters a lot, so if mixing digitally or producting remixes etc. its useful to have more and larger samples to work with