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.
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.
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.
Hydrasynth aliases like a mad thing. My flagship synth ended up being Summit, and its oscillators are digital but run at a crazy high sample rate. Did likewise with some Chord Organ modules: that Teensy board it was built on could do chord audio at 300k and over a megahertz if you were just generating one wave as simply as possible. The freedom from aliasing really helped the sound, for all that it's a 12 bit analog output. A squarewave is a 1 bit signal…
> 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.
So if your synthesizers do not use proper band-limited oscillators then 192KHz is _FAR_ too slow. You'd want to be running at hundreds of KHz, perhaps a few MHz.
In reality synth software that doesn't sound like crap uses band limited oscillators and should work okay at 48KHz too. That said, even if the oscillators are band limited it may be the case the varrious modulations aren't band limited properly, as getting those wrong won't sound instantly wrong (in particular because you have to modulate to make it wrong, and the underlying change of the modulation may make it harder to tell its wrong).
Though also in those cases if you're not counting on every step being properly band limited then 192KHz may be an improvement but you're still probably getting some meaningful aliasing. I think given how fast computers have become relative to digital audio there is probably a good case to just make any "modular synth" run at 32-bit 480KHz or even 4.8MHz through every stage that could process the audio.
Maybe 192KHz really is enough to suppress the aliasing artifacts but I think to be convinced of that I'd want to see a system that supported both and validate that the difference between a downsampled 48KHz output from the two modes was below -90dB or something.
Or otherwise you can just declare that the aliasing is part of the sound and then there are no right choices... 24khz sampling, 48k, 192k ... who cares, use what you like best. :)
> I think given how fast computers have become relative to digital audio there is probably a good case to just make any "modular synth" run at 32-bit 480KHz or even 4.8MHz through every stage that could process the audio.
1. It should run at FP64 if you want to preserve filter resonances, etc.
2. At 10x/100x fixed-rate oversampling, even a modern "fast" CPU will have very few cycles per (higher-rate) sample to run the DSP for 1 "module" of the software modular. Forget about interconnected modules, multiple tracks, or polyphony. For this kind of "analog"-style processing, it's better to run adaptive-rate algorithms (think SPICE) instead of wasting compute on unnecessary extra audio samples.
so an 8-core zen4 should be able to sustain more than 300 gflops of 64-bit multiply-adds. At 480khz that's 625k operations per sample. I'll grant the 100x oversampling was probably too ambitious. :P
For adaptive rate I think the issue there is you have a hard-realtime constraint for this usage (even if you wouldn't mind rendering offline, you kinda have to hear it realtime to tweak it-- after all you might tweak it in a way that brings out an artifact you like and then be disappointed by the render). Also in the case of a whole modular system having all sorts of different parts needing to be part of the adaptation loop seems pretty hard to me.
My thinking was just in general that 192k is really not enough to prevent aliasy algorithms from messing up. If you are alias safe you can probably run at 48k and be fine. If you're not, you really want to go much higher.
I'm travelling without my Zen 4 machine, or I could test it. ;) Oh well, Compiler Explorer is enough to look at these microbenchmarks on your own.
These simulations are single core to avoid core-to-core latency. Number of cores isn't relevant unless you want to run independent voices/channels and sum them at the end.
So you start with a very optimistic ~90 GFLOPs of 64-bit FMA on Zen 4. Unfortunately, not all operations are clean multiply-adds. Realistically, you'll need trigonometric functions and LUTs, which are quite slower. Btw, the tradeoff between when to compute vs LUT is very fragile and can change due to a ton of factors (notably integrator algorithm).
Then the data you are operating on won't fit cleanly in AVX-512 registers, requiring spills to L1 cache. Ok, still fast on a modern core.
Of course, the peak theoretical number assumes clean vectorization with double-pumped AVX-512... which also won't happen in practice. Classical DSP will fare better (https://www.youtube.com/watch?v=Ssq0a-YdamM) but SPICE integrators are inherently branchy and divergent. Especially for adaptive integrators, you'll waste a lot of operations trying to "lock in" at the exact time point where the waveform turns a corner. Apple Silicon is better at this messy, branchy code.
So yeah, it's possible-but-hard to hit hard-realtime under these conditions.
32-bits are great for recording too because they do an incredible job of capturing the dynamic range without having to be precise on the preamp settings. It removes an entire job from the recording workflow.
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.
We had a really nice crystal decoration that I happened to put on top of one of my TV speakers and, wouldn't you know it, it had this resonant frequency somewhere around specific human speech frequencies that drove us absolutely bonkers until I figured out the cause and moved it.
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.
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.
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.
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.
It'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).
I tried Tidal nearly a decade ago, and the audible fluttering effect caused by their audio watermarking totally ruined certain types of music, like choral recordings, strings and such. It was obviously apparent on $20 ear buds driven by any device, far beyond the more stereotypical audiophile gripes.
I opened a support ticket but they never responded. After that it was difficult to take their lossless claims seriously when the labels were providing such garbage source material. Their whole value prop was totally hollowed out.
I don't know whether the labels still impose such horrible practices, but I largely gave up on streaming services after that experience and now focus on keeping good digital archives of my physical library.
I'm pretty strongly in the camp of trust the science and measurements for audio stuff. Thus I suspect its mostly just better sounding masters, but I was shocked at how much I noticed the sound quality of Tidal compared to Spotify when I switched.
I'd distinguish between differences that anyone can detect but some may not care about, and differences that may not be objectively detectable at all. Muscle cars, at least, are different in a way that anyone can see. Push that pedal to the floor and it feels different from a Honda Civic or whatever. Whether that difference is actually interesting or good is, of course, a matter of taste. Whereas audiophile nonsense is often indistinguishable even to the connoisseur and depends entirely on some form of self-deception. Still could be worth it, depending on what one considers worthy.
That’s actually a really good comparison, especially because - yes I can hear the difference between an excruciatingly lossless digitization of a piece of music that I’m intimately familiar with, played back on expertly configured hardware… but the difference is so little, that most of the time, I’m find just listening to it at medium high quality streaming on a pair of <$50 headphones.
I’ve played with the nice toys, and they are nice, but for 100x the price, they barely deliver 1.5x the experience.
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"
> 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.
In 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."
I remember playing 44khz 16-bit stereo MP3s encoded at 128 kbit/sec on a 133 Mhz 486.
It gobbled like 90% of the CPU and I had to make sure I gave it a pretty large buffer so it didn't stutter when an app claimed CPU for more than a second, but it worked.
Depends on the vendor and a cache amount. But yes, 22KHz was a thing for a simple Notepad activities at best and at 11KHz you didn't care much 'cuz it wasn't that different from the usual.
I have an external audio card, if I put it on a laptop I can hear the modem-like sounds. I wonder why it is so sensitive, should not DAC produce strong signal that cannot be easily affected by radio waves?
Also my headphones are extremely sensitive. I can touch the ring and sleeve of a jack with a finger, and touch a metal bed frame with a tip and I hear quiet clicks as I move the tip along the metal. Sometimes I do not even need to touch the jack with a finger. It doesn't work with small objects like a knife though.
audiophiles (https://forums.stevehoffman.tv/threads/turntables-with-pace....) also claim that turntables can be rated on "timing, rhythm, and pace" in which supposedly the timing of the music can be affected by the turntable's mass and other properties.
How this would occur without also producing grossly audible pitch distortion never seems to be discussed.
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
"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.
I'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
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.
Yeah, we use Secret Rabbit Code for ours, though we have access to the sox code now and that is "perfect". We might change to that as the default sometime this year.
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.
Max 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
I don’t have great hearing, so I’m not sure I can really weigh in here (thanks punk concerts in my teens). I remember similar arguments around screens and 60Hz vs ‘the human eye’. I think a lot of people, myself included, can easily perceive the difference between 60Hz and something higher- given the right conditions. I would not be so quick to disregard claims of more sensitive hearing.
The human threshold-of-hearing curve intersects the threshold-of-pain curve at about 20 kHz.
Above that frequency (or thereabouts) the sound has to be so loud that it will literally instantly damage your hearing before you can hear it.
This has been replicated across many studies for more than 100 years.
Flicker threshold is completely different. You can’t damage your vision by increasing the FPS, and it has always been commercially desirable to use a lower frequency because that is cheaper.
Would you agree that a trained engineer could identify artifacts produced by an imperect conversion process? If you lean "yes", then that's your answer: AD/DA is not a Rust function perfectly implementing Nyquist theorem, it's a collection of imperfect physical components many of which introduce artifacts into the audio path. This thread is not about the theory of human hearing, the electronic components are literally imperfect.
They're no more imperfect than the pickups on an electric guitar, the assembly inside the microphone, the circuit in the compressor and everything else in the analog signal chain that exists long before AD happens.
But the central point is that there's no reason to pick on the digital elements in any particular way. Recorded music in 2026 is a pretty good recreation of the original acoustic pressure waves when it is intended to be, but (a) not perfect, even in the pure analog domain and (b) it is frequently not intended to be.
The central point is that AD conversion can and will introduce artifacts. DA process wil intrduce more artifacts. The "imperfect" is a huge range and AD/DA converters play a role in that. We are not talking about "golden cables" bs here, conversion does introduce measurable artifacts in the audio path. The more tracks you record the more artifacts you have. Can everyone hear them? Definitely no. Can they be heard - yes, I can hear the difference between an old Digidesign interface and Grace Design interface.
No, the central point is that the analog signal handling before AD introduces vastly more "artifacts" than the AD or DA does.
In addition, nobody cares about "measurable" artifacts (or rather, they should not). What matters are "audible" artifacts. We have measuring equipment that is vastly more sensitive than human ears (e.g. your recording equipment that can pick up signals far above 22kHz). What's measurable is not particularly interesting - what's audible is.
Artifacts do not sum linearly, because they do not originate from correlated sources (unless you're doing something rather unusual).
Glad you can hear the difference between two converters, but I trust you've tested it in a double blind setting?
(I responded on this topic in this thread already) Look up articles on practical limitations of AD/DA converters and why a seemingly counter-intuitive claim that the difference between 44.1 kHz and above is noticeable is actually a completely accepted practical reality (aliasing, lowpass filters, etc)
You need at least twice the frequency range for sample rate in order to represent the original signal. That's slightly misleading though, that's from the Nyquist-Shannon sampling theory and it's a mathematical fact but that is true for exact numerical samples, once you add in quantization that muddies the water a bit. Taken at the extreme, it's straightforward to see why a 1 bit quantization per sample at 44.1 kHz would not capture a perfect representation of some analog signal even if there's only a 1 kHz frequency component to the signal. If we instead decide to sample at 10 MHz but still one bit quantization, now that 1 kHz frequency component can be much more accurately represented even though we're still using the worst quantization possible. Don't think of quantization like a square wave or a step pattern, think of it as "the signal is closer to here than any other discrete value".
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.
As I responded below, you are confusing math with physical reality. A true 44.1 kHz converter can't realistically capture frequencies above 18-20 kHz due to the limitations of filters used in the process. A perfect lowpass brick-wall filter just does not exist - they all introduce artifacts, which a trained ear can identify. You don't need to be a dog to hear the difference, just someone who does not assume that Nyquist theorem can be magically applied in the real world (and, ideally, someone who utilizes high quality converters with oversampling).
To draw a design parallel: pixel-perfect design isn't something we are born with, noticing tiny details is a developed skill.
And yes, you are on point: oversampling is used extensively, but this just points at the exact issue: Nyquist theorem gave us a math algorithm, we still need to account for the electronic component imperfections.
Oh, dear, that AES 2014 paper from Meridian (which was trying to push its controversial proprietary MQA audiophile system the same year) was widely criticized on audio forums when it came out, ranging from the rectangular dithering method to the use of a hard metal tweeter that could cause IM.
I don't. Do you? I am not a researcher. Saying that, do you have a double-blind study handy on MP3 256 vs 320 actual audible differences? If not, can you yourself hear the difference? If you can - it might be an illusion.
That extra 4.1 khz sample rate is for headroom for a low pass filter (and not necessarily a brick wall one). Leftovers or any such artifacts are below the noise floor, which is also an important part of the physical reality.
Would be happy to see an actual, real study to prove that humans can notice, but to my knowledge none exist that confirm they can. Not even any on teenagers or younger (the only group that can even hear close up 20khz).
The most impactful for noticing the difference? Again, I would argue it's the trained ear. If you have plenty of mixing experience then all these details add up, and a treated room becomes the most critical - agree with that.
Other than the top engineers in the industry. This is a discussion that always ends up in the "double-blind study" vs actual real engineers working in the industry.
If you want to hear the difference between an audio file recorded at 44.1 and 88.2kHZ, then you need slow the audio playback down. Otherwise, a trained ear cannot physically hear the difference.
44.1 is "enough" only in theory. This assumes a physically impossible steep filter. Realistically, frequencies around 20 kHz will create audible artifacts (aliasing). So yes, a trained ear can tell the diffrenece between 44.1 and even 48 kHz. Like many other commenters in this thread, you are mixing up math theory with physical limitations of AD/DA converters. Oversampling is a common way to address this limitation, but strictly speaking 44.1 kHz is not as obviously "enough" as it seems.
Do you have citations for this claim? The "golden ears" argument is often employed by audiophiles, but even the cheapest converters oversample by up to several hundred times as well as employ antialiasing filters.
> Realistically, frequencies around 20 kHz will create audible artifacts (aliasing)
The energy of the signal components above the Nyquist is generally very low, and very few double blind tests have given any indication that humans can detect the resulting aliasing (even though many people claim to be able to do, almost always in non-double-blind environments).
Genereally very low for a single track? What about 200 tracks? Badly written synthesis, or badly recorded live instruments, or bounced and re-bounced dozens of times... we are not talking about the quality-defining aspect here. You can produce an excellent mix on KRKs connected directly to a MacBook.
This space is not driven by a single precise formula. 48/96 kHz helps some engineers to produce better sounding mixes. Can everyone hear the extended range of Adam tweeters? Probably not. But some can, and they benefit from that. Even if there is no double-blind study to prove this in absolute terms.
If you recorded 200 tracks of the same instrument, so that the partials above Nyquist were all broadly the same, then sure, summing the tracks would include summing 200 copies of the aliasing results too.
But very little music is like that, and the energy profile above Nyquist will differ dramatically. Consequently, you're not summing a set of identical aliasing results, and in general, the results will still be undetectable to almost everyone.
Jacob Collier routinely works with 300+ tracks in Logic. He doesn't worry about this sort of thing, and neither do the Grammy voters who love what he does.
Oh 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.
Those 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.
That’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.
I 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.
It is not only that. It's the spacing, how the bass sounds. There's so many interesting headphones in the midrange to try out. Compare the Hifiman HE1000se to Heddphone 2 GT 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. We people like jewelry, I can get similar enjoyment from beautiful audio equipment and cables.
Depends. 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.
The 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
While 100% true, I'd phrase B as "for a particular song, the mastering can make a difference regardless if the digital sampling was done at perceptibly perfect levels" just to be clear that the statement applies to files with higher bitrates as well.
I 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.
I 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.
Foobar2000 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.
I 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)
Counterpoint: 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.
I 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.
Wait, 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?
Just 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!
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.
The 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.
I 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.
You'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.
@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.
Thank 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
I 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.
And 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.
> Understand the physical limits of human senses and perceptions to help inoculate yourself against outright scams and grifts
Also understand that while there is an upper limit, we are all different within that. I can hear the difference between 128Kbps and FLAC, at least for some content, but not 256Kbps, maybe not 192. For some content (spoken word etc.), 64Kbps, sometimes less, is perfectly acceptable (to me). There was a time I could hear the difference between some encoders, but that was decades ago and anything in active use is pretty damn good (and my ears are not what they used to be) unless you really crank the bitrate down or tweak other options daftly.
Some time ago, yes. For 128 and there-about & below, not 192+ hence I'm less certain about that (but I'm pretty sure I wouldn't be able to tell the difference).
Not recently, so it is possible that improvements in encoding methods, and changes in my ears, could mean that I'd get a different result now.
I’ve not tried encoding my own MP3s in at least a decade, but when I was doing so, 128 kbps was instantly distinguishable to me on anything with cymbals, especially hi-hat: it loses that shimmery sound. At 192 kbps I could tell if I really, really tried, but it was so minute I didn’t really care. I was never able to reliably tell the difference between 256 and 320 kbps rips.
The thing I didn't understand with higher quality music files is that it's not like the entire song is different and better when you go from 64 to 128 kbps opus, it's just these super minor details that get changed. It was enlightening doing an abx test, but I still use flacs because it's nice not worrying about the quality mattering.
> I can't hear the difference between 128 kbps opus and FLAC.
A reasonable definition of transparency for high bitrate compressed audio is "Can the worst files be distinguished by a listener trained in what artifacts sound like". Maybe also add in having to use a high discrimination listening setup, including not running excessively loud (increases masking).
If that's not the test you're doing, it's unsurprising. At moderately high bitrates no one can reliably distinguish them on arbitrary samples: most inputs are easy.
If you test on known-difficult "killer samples" you'll probably easily distinguish them, even without first being shown what to look for, and certainly after.
During the development of Opus I created many 'trained listeners' and selected many killer samples, and I don't recall* ever encountering a tin ear that couldn't be taught to ABX any high rate samples, though some people are obviously much better at it.
I'm not sure I'd recommend it though: learning to identify artifacts has a frequent side effect of making low rate audio like the HE-aac used in SirusXM absolutely intolerable. I'm bothered by it even when I hear cars driving by using it. :)
[*] My memory for such things sucks, so I could be wrong-- but my point that it's not expected remains.
Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart.
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.
Those things have huge drivers and are probably too big for a lot of rooms.
Unless you (and your neighbors) absolutely want to have that thumping sound and you go out of your way to kill unwanted bass reflection you're probably better off with the HS5 or something similar.
> Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart.
As referenced in the article, a common explanation for those audible differences is that the high-resolution version of the album is sourced from a different master.
This is easy to disprove by downsampling from a 24/192 source to 16/44.1 Even if the downsampling is (close to) ideal there are obvious differences.
In fact if you can't hear the difference between 24/192 and 16/44.1 you shouldn't be working in audio. (Doesn't apply to consumers. Does apply to musicians and engineers.)
It's like being colour blind.
And if you don't understand the math behind quantisation, you shouldn't be posting pseudo-scientific videos where you use an oscilloscope and a cheap spectrum analyser - both tools with very limited resolution - to "prove" your point.
16 bit isn't enough for hard, objective reasons. One is that the noise spectrum of quantisation is not simple. Most people assume it's something close to plain white noise, but it really isn't. It's actually a very complex spectrum with some prominent peaks at specific subdivisions of the sample rate. Those frequency peaks are significantly above audibility. 24-bit quantisation shrinks them below audibility.
The other is that most people can hear dither/noise-shaping at 16-bits. That adds a single bit of noise which should - if you're being very literal - be far below the threshold of audibility. But it clearly isn't.
These two facts are related.
The more complex reason is that listening is an active perceptual process. The brain does a huge amount of processing to separate sources and place them in a perceptual field which includes information about perceived object type, distance, and ambience cues. Some of those cues are very quiet, and we don't hear them linearly.
So using sine waves as some kind of perceptual reference for audibility is nonsensical. We hear much more complex signals in an active way, and if there's information missing in the quiet parts - which there is with limited quantisation - then the signal simply isn't accurate.
I agree with most of your points, but saying you shouldn't work in audio if you can't tell the difference between 192khz and 44.1khz is a bit elitist imo. And saying you're color blind if you can't tell the difference is like saying you're blind if you don't have 20/20 vision and shouldn't draw. You can always use meters to check for aliasing artifacts.
It's not like all of your samples and virtual instruments are 192khz or even 96k. Many are 48khz or even 44.1k.
I think there are many cases where people never need to go above 44.1khz unless you maybe have saturation on the master bus. I agree that good dithering is important though and think that there hasn't been enough research on that so far.
> 16 bit isn't enough for hard, objective reasons. One is that the noise spectrum of quantisation is not simple. Most people assume it's something close to plain white noise, but it really isn't. It's actually a very complex spectrum with some prominent peaks at specific subdivisions of the sample rate.
What you are describing is the result of blunt truncation. If you use the most basic (“uniform” or “rectangular” a.k.a. “RPDF”) dither, the spectrum is in fact flat, as demonstrated by the video you are likely alluding to and calling “pseudoscientific” (https://youtu.be/cIQ9IXSUzuM?t=12m50s). If you sum two uniform dithers together, you get what pretty much everyone uses (“triangular” or “TPDF” dither) which, in addition to decorrelating the mean quantisation error from the signal, also decorrelates the standard deviation, eliminating noise modulation and leaving a correlation only in still higher-order moments like skewness and kurtosis.
You can even try it for yourself with SoX. Find a 24-bit track, quantise it with dither to 16-bit, calculate the difference between both tracks, blow up the difference and take its spectrogram and it will be completely flat. Or listen to the difference (mind the volume) and see if you can make out anything meaningful.
And then remember that this difference would normally sit at roughly -93 dB FS, so to hear it in a typical room, you would have to be listening at deafening levels. You claim that it “clearly isn’t” below the threshold of audibility but it’s not clear how you arrived at that conclusion. You then claim that the audibility of that noise floor is somehow related to what you said before about the effects of undithered quantisation, even though those effects stop being relevant the moment you apply any sort of dither.
> We hear much more complex signals in an active way, and if there's information missing in the quiet parts - which there is with limited quantisation - then the signal simply isn't accurate.
It’s not missing. You can do a similar test where you “bury” your source material in the 16-bit dither noise floor, blow it up again, and you’ll be able to detect it under the noise.
$ sox source.flac -b 16 quiet.flac gain -100
$ sox quiet.flac loud-again.flac norm -1
$ open loud-again.flac
> As referenced in the article, a common explanation for those audible differences is that the high-resolution version of the album is sourced from a different master.
In this case, it was my brother's own 24/192 recording, down-mixed by him to CD format with the intent that it be transparent. I believe he said his software was supposed to be dithering, but this was ~25 years ago and I can't really confirm the details anymore.
Even more likely, high frequency ringing in the higher res file, caused by the converters, has the same effect analog distortion via tubes does creating the perception of clarity where there is none.
No one can hear the difference between properly mastered high res files. I will happily put money on it.
This is an extremely hard comparison to do well. I'll give a few examples as to why:
Small differences in gain are ABX able much more readily than differences in noise at the 16 vs 24 bit level. So if the signal chain gives even a small difference in gain between the samples that's what you'll track. A reasonable conversion path to 16 bits for mastering will also apply dithering and some kind of brickwall limiting (you have to limit after the dither or as part of the dither as dither can change levels!), and this can result in gain changes. The DAC may behave differently or have outright bugs for some configurations too.
This is particularly true wrt reconstruction filters for sample rate differences. And if you were comparing 44.1k and 192k then the physical DAC itself was likely running at a different rate and its _analog_ filters are probably better optimized for one vs the other (this is less true for 48k vs 192k, as the hardware likely runs at the same rate for both). So one answer to this comparison can be "on this particular hardware this rate is better than that rate"-- but that's a implementation property not a property of format choice.
You might think, "okay I'll use a mathematically perfect down and up conversion process and run the DAC in the exact same configuration for all cases". But even then you run into issues like after reconstruction the _inter sample_ peak levels will be higher than the levels of the samples, so you have to handle that and in a way that doesn't produce a gain difference between the two configurations. (probably by running your perfect process and finding the gain level that results in no limiting, then making the gain of the original match).
And then for the high rate vs non-high rate you have to deal with the fact that most amplifiers are not particularly linear (compared to well constructed software at least!) and that any real speaker is very far from linear. This means that the presence or absence of ultrasonics will change the audio in the 0-20khz band.. Before you think "well that could be a reason that high rate is better" observe that if there was some consistently good effect from the ultrasonics you could just bake it into the low rate sample.
> but in my 50s I know
Yeah if you're in your 50's you're absolutely not hearing differences way up above 20khz (especially if you're male), I bet you can't even hear CRT flybacks from 100 yards anymore. :P
I don't mean to discount your experience: I don't really doubt that it was real. But answering the general question of the necessity of low vs high rate probably takes a team of experts, armed with test gear and the designs of the HW/SW in question, to vet the test configuration. Testing a _particular_ configuration without the ability to distinguish its implementation quirks from format-fundamentals is much easier and that's what most attempts to test this question are actually testing.
> Small differences in gain are ABX able much more readily than differences in noise at the 16 vs 24 bit level.
This was common knowledge at least as far back as the mid 80s, when every hifi shop and salesguy knew to ensure the bit of gear with the highest profit margin got played an almost imperceptible bit louder than the gear the customer came in to buy during back to back testing.
It's also a reason why double-blind testing is important. If someone doing the setup is expecting one piece of kit to sound better, if it doesn't they'll check the configuration more, and difference in gain can come from many sources. So errors that result in higher gain in favor of the "better" candidate go uncorrected, while ones that favor the worse tends to be fixed.
Point being: it doesn't even require an unscrupulous sales person to get similar results to an unscrupulous sales person! :P
This was supposed to be running the DACs to match the source configuration, not resampling into some common format. I think that is an unavoidable part of the whole end-to-end ABX test concept.
Maybe it would be interesting to up-sample back into 24/192 and play both in that mode. But then people would argue about what type of up-sample to use.
I was in my mid 20s for this test. I understand my high-band hearing was better back then.
Speaking about up-sampling, Im curious to know your opinion on the benefits of it. Im sending CD resolution audio as well as web streams from soundcloud.com to cambridge audio azur 840C and its not clear if its the up-sampling that makes the difference or their per channel wolfson dac arch. The iPod Video with their dac sounded amazing with just normal AAC files compared to the iPods before or after it.
Any high end audio dac is internally running at a much higher sample rate-- that's what it takes to get their delivered performance with the silicon that's available. Its up-sampling process was designed by the designers of the DAC with intimate knoweldge of its analog properties.
Second guessing it by upsampling in front of it seems dubious to me. It might help in some cases where the DAC designers were thinking of different objectives or just didn't do a great job. It might also help with some other issues, like if the dac is timed off the input clock and the input clock sucks and the upsampler retimes the signal.
Of course the upsampler designers could also get it wrong, be aliasing the hell out of the results, and happen to like the sound of the corrupted audio. :P
The effects are all objectively measurable however-- with expensive equipment at least. I think I'd want to set test results with a particular hardware combination before sticking an upsampler in it. OTOH, if there already was one there because it's just some built in feature of some kit I wanted to use otherwise, I wouldn't worry much about it. Particularly if that kit has been reviewed by people with proper test gear and they didn't decide that it was broken.
Ah, 20kHz and CRT flybacks.. when I was a child I could of course hear that (in Europe that would be 15625 Hz), when I studied electronics and TV repair we could all hear that, and because we had the equipment we "tested" what we could hear using a function generator. The limit for conscious hearing for me was somewhere around 17kHz. Or not 18kHz for sure.
But I think I lost the ability to hear the flyback not long after I passed twenty. The world turned silent as far as that's concerned (before, you could hear it anywhere and everywhere, in shops, homes, some workplaces..)
The "20kHz" thing is kind of a myth for most people, at least that's what it looked to me after all the testing we did at school. I think it can influence what you hear, somehow, but in any case it's for very young people.
> Most people have no idea how much their high frequency hearing degrades as they age because it plays approximately no role in your life, but it's real, dramatic, and as far as I know happens to everyone.
I agree completely. I recall some discussions a long time ago on RMMGA (Usenet: rec.music.makers.guitar.acoustic) where some distinguished and experienced, but middle-aged guitarists got practically angry when a young guy described the sound of a certain type of newly-introduced strings "harsh" and "like fingernails on a blackboard" when used on a particular guitar.
The difference was, of course, that what the young guy could hear is something which stopped existing at least when you had passed 30.. I was at an age where I too couldn't hear that kind of sound from strings, but it was still not that long ago and I remembered and had noticed the difference, i.e. that I could not hear what I could hear before. For example the huge difference between fresh strings and week-old strings (and that fact has, over the decades, saved me tons of money which I would otherwise have spent on replacing strings all the time..)
That would be how you'd go about telling, sure enough. You can't go by 'frequencies' or distortions or anything like that, these analog departures from convincing reality aren't how digital failings manifest.
You try to hear the brickwall by the muffled, enclosed quality and possibly by the weird pre-ring blurriness of the filter making things sound more vague than they have to be, and you hear the truncation not because it is audible 'distortion' as we know it, but because depth collapses and it sounds like it's coming from the speakers and not being a separate space behind/around the speakers. At no point will it be the most glaringly obvious thing but it'll never be 'distortions' as we imagine them, it's more a 'pod people' lack of personality thing.
Like a much subtler version of listening to AI music :)
I'm quite happy with 24/96 as suitable overkill for anything I might want to hear or do. Neil Young went hard on the proposition that 192 was necessary. Sold the Ponoplayer, I had one but it died on me, battery failed eventually. It really did sound awesome beyond just about any other listening device I've ever heard…
24 > 16 is not debatable. Sample rates are more complex because then higher the clock rate the more you get distortions from jitter and the design of the DAC/ADC. Most converters introduce different artefacts at different sample rates, especially at the prosumer end, so you're not comparing like for like.
The last couple of generations of converters have gotten a lot better, so 192kHz today is likely to sound cleaner and smoother than it did ten years ago, where there was a good chance the clock was quite jittery.
Personally I don't think it's worth the extra bandwidth for playback, but I can understand why some people might want it.
Generally all of these "debates" come down to people who think math > circuitry. All real designs are imperfect trade-offs. They all have issues, and arguing as if converters are perfect when they never are, and the imperfections can be benched objectively, is... not very scientific.
>Generally all of these "debates" come down to people who think math > circuitry. All real designs are imperfect trade-offs. They all have issues, and arguing as if converters are perfect when they never are, and the imperfections can be benched objectively, is... not very scientific.
There is one purely objective benchmark: a true blind test. You can believe if something is different or not, but if nobody's capably of hearing the difference, does it matter?
One possibility (pure speculation) is a bad antialiasing filter. The Nyquist frequency at 44.1ksps is 22.05kHz, which is only ~10% above the audible band. This means that you need a rather sharp filter both when downmixing and when playing to avoid potentially audible aliasing into the audible band or attenuation within the audible band.
If you look at a site like audiosciencereview.com and pull up measurements of a DAC or ADC, you can find graphs of the antialiasing filter response. Some are great and some are not.
One could think of 16/44.1 PCM as being a codec that is potentially perfect but requiring some degree of care to encode and decode correctly.
This. It may be a niche. But music that works with volume as part of the composition can be amazing.
But most music today has heavy compressors in the pipeline that kills dynamic range in favor of allowing you to hear almost even whispers, even in traffic or a city with ear pods.
But if you're from the first group, as you said it's more noticable the benefits of having better codecs and bit depth vs heavily compressed top billboard songs where even listening the master track from the studio, falls into diminishing returns.
> 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.
I used to think the same. But I realized that downsampling hi-res music to 16/44.1 isn't a transparent conversion. So now I prefer the one downsampled to 16/44.1 by an expert in production env. I almost always download 16/44.1 flac files because of this.
https://news.ycombinator.com/item?id=48774112 <-- Here is a test. I would have said the same and I played all samples quite often. It is extremely subtle, but even with a cheap headset it is possible to hear differences, where the high quality version is just a little bit clearer.
For 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.
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[ 3.0 ms ] story [ 110 ms ] threadSo 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.
So if your synthesizers do not use proper band-limited oscillators then 192KHz is _FAR_ too slow. You'd want to be running at hundreds of KHz, perhaps a few MHz.
In reality synth software that doesn't sound like crap uses band limited oscillators and should work okay at 48KHz too. That said, even if the oscillators are band limited it may be the case the varrious modulations aren't band limited properly, as getting those wrong won't sound instantly wrong (in particular because you have to modulate to make it wrong, and the underlying change of the modulation may make it harder to tell its wrong).
Though also in those cases if you're not counting on every step being properly band limited then 192KHz may be an improvement but you're still probably getting some meaningful aliasing. I think given how fast computers have become relative to digital audio there is probably a good case to just make any "modular synth" run at 32-bit 480KHz or even 4.8MHz through every stage that could process the audio.
Maybe 192KHz really is enough to suppress the aliasing artifacts but I think to be convinced of that I'd want to see a system that supported both and validate that the difference between a downsampled 48KHz output from the two modes was below -90dB or something.
Or otherwise you can just declare that the aliasing is part of the sound and then there are no right choices... 24khz sampling, 48k, 192k ... who cares, use what you like best. :)
1. It should run at FP64 if you want to preserve filter resonances, etc.
2. At 10x/100x fixed-rate oversampling, even a modern "fast" CPU will have very few cycles per (higher-rate) sample to run the DSP for 1 "module" of the software modular. Forget about interconnected modules, multiple tracks, or polyphony. For this kind of "analog"-style processing, it's better to run adaptive-rate algorithms (think SPICE) instead of wasting compute on unnecessary extra audio samples.
For adaptive rate I think the issue there is you have a hard-realtime constraint for this usage (even if you wouldn't mind rendering offline, you kinda have to hear it realtime to tweak it-- after all you might tweak it in a way that brings out an artifact you like and then be disappointed by the render). Also in the case of a whole modular system having all sorts of different parts needing to be part of the adaptation loop seems pretty hard to me.
My thinking was just in general that 192k is really not enough to prevent aliasy algorithms from messing up. If you are alias safe you can probably run at 48k and be fine. If you're not, you really want to go much higher.
These simulations are single core to avoid core-to-core latency. Number of cores isn't relevant unless you want to run independent voices/channels and sum them at the end.
So you start with a very optimistic ~90 GFLOPs of 64-bit FMA on Zen 4. Unfortunately, not all operations are clean multiply-adds. Realistically, you'll need trigonometric functions and LUTs, which are quite slower. Btw, the tradeoff between when to compute vs LUT is very fragile and can change due to a ton of factors (notably integrator algorithm).
Then the data you are operating on won't fit cleanly in AVX-512 registers, requiring spills to L1 cache. Ok, still fast on a modern core.
Of course, the peak theoretical number assumes clean vectorization with double-pumped AVX-512... which also won't happen in practice. Classical DSP will fare better (https://www.youtube.com/watch?v=Ssq0a-YdamM) but SPICE integrators are inherently branchy and divergent. Especially for adaptive integrators, you'll waste a lot of operations trying to "lock in" at the exact time point where the waveform turns a corner. Apple Silicon is better at this messy, branchy code.
So yeah, it's possible-but-hard to hit hard-realtime under these conditions.
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.
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.
I opened a support ticket but they never responded. After that it was difficult to take their lossless claims seriously when the labels were providing such garbage source material. Their whole value prop was totally hollowed out.
I don't know whether the labels still impose such horrible practices, but I largely gave up on streaming services after that experience and now focus on keeping good digital archives of my physical library.
I’ve played with the nice toys, and they are nice, but for 100x the price, they barely deliver 1.5x the experience.
https://www.carwow.co.uk/blog/carwow-quarter-mile-400-metre-...
https://en.wikipedia.org/wiki/List_of_N%C3%BCrburgring_Nords...
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"
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.
It gobbled like 90% of the CPU and I had to make sure I gave it a pretty large buffer so it didn't stutter when an app claimed CPU for more than a second, but it worked.
[1] https://www.audioasylum.com/messages/pcaudio/119979/
Also my headphones are extremely sensitive. I can touch the ring and sleeve of a jack with a finger, and touch a metal bed frame with a tip and I hear quiet clicks as I move the tip along the metal. Sometimes I do not even need to touch the jack with a finger. It doesn't work with small objects like a knife though.
And this is not the case of 'stronger than', it's 'strong enough to be caught up by anything resembling an antenna'.
You hear the interference because some analogue tract of your system:
Do transmit the radio waves everywhere
Does receive the inference from the all electric things around.
And now I'm skipping 'cuz I'm inna a bar and it's more interesting.
How this would occur without also producing grossly audible pitch distortion never seems to be discussed.
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.
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.
your software is among the best, but not pitch black best :)
There is also https://src.hydrogenaudio.org/ (with no IP based restrictions, AFAIK).
The human threshold-of-hearing curve intersects the threshold-of-pain curve at about 20 kHz.
Above that frequency (or thereabouts) the sound has to be so loud that it will literally instantly damage your hearing before you can hear it.
This has been replicated across many studies for more than 100 years.
Flicker threshold is completely different. You can’t damage your vision by increasing the FPS, and it has always been commercially desirable to use a lower frequency because that is cheaper.
In addition, nobody cares about "measurable" artifacts (or rather, they should not). What matters are "audible" artifacts. We have measuring equipment that is vastly more sensitive than human ears (e.g. your recording equipment that can pick up signals far above 22kHz). What's measurable is not particularly interesting - what's audible is.
Artifacts do not sum linearly, because they do not originate from correlated sources (unless you're doing something rather unusual).
Glad you can hear the difference between two converters, but I trust you've tested it in a double blind setting?
Who has the best ears? What can they detect?
I know from my 20-ish year mixing experience that I can hear the difference when mixing. Is it good evidence? No. So we can agree to disagree then.
I'm not disagreeing with you. I'm really curious about the limits of what people can hear, what can be taught and what is rare.
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.
A quick search returned this PDF with a nice diagram of what aliasing looks like: https://download.tek.com/document/76W_30631_0_HR_Letter.pdf
To draw a design parallel: pixel-perfect design isn't something we are born with, noticing tiny details is a developed skill.
And yes, you are on point: oversampling is used extensively, but this just points at the exact issue: Nyquist theorem gave us a math algorithm, we still need to account for the electronic component imperfections.
Do you have more convincing sources?
Would be happy to see an actual, real study to prove that humans can notice, but to my knowledge none exist that confirm they can. Not even any on teenagers or younger (the only group that can even hear close up 20khz).
The energy of the signal components above the Nyquist is generally very low, and very few double blind tests have given any indication that humans can detect the resulting aliasing (even though many people claim to be able to do, almost always in non-double-blind environments).
This space is not driven by a single precise formula. 48/96 kHz helps some engineers to produce better sounding mixes. Can everyone hear the extended range of Adam tweeters? Probably not. But some can, and they benefit from that. Even if there is no double-blind study to prove this in absolute terms.
But very little music is like that, and the energy profile above Nyquist will differ dramatically. Consequently, you're not summing a set of identical aliasing results, and in general, the results will still be undetectable to almost everyone.
Jacob Collier routinely works with 300+ tracks in Logic. He doesn't worry about this sort of thing, and neither do the Grammy voters who love what he does.
Don't forget to buy the new low oyxgen 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).
I had a good laugh listening to the sample at https://en.wikipedia.org/wiki/Phonautograph
The sound is very pure indeed.
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.
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.
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. We people like jewelry, I can get similar enjoyment from beautiful audio equipment and cables.
And so many CDs of course.
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
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.
this has practical applications
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
We store files in the highest quality because it gives us the option to encode the music without audible loss of quality.
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)
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.
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.
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.
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.
And none of them are broken after 20 days unless it's low tide or I fuck up on a cliff band.
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.
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.
Also understand that while there is an upper limit, we are all different within that. I can hear the difference between 128Kbps and FLAC, at least for some content, but not 256Kbps, maybe not 192. For some content (spoken word etc.), 64Kbps, sometimes less, is perfectly acceptable (to me). There was a time I could hear the difference between some encoders, but that was decades ago and anything in active use is pretty damn good (and my ears are not what they used to be) unless you really crank the bitrate down or tweak other options daftly.
You've established this with double bind testing, correct?
Not recently, so it is possible that improvements in encoding methods, and changes in my ears, could mean that I'd get a different result now.
A reasonable definition of transparency for high bitrate compressed audio is "Can the worst files be distinguished by a listener trained in what artifacts sound like". Maybe also add in having to use a high discrimination listening setup, including not running excessively loud (increases masking).
If that's not the test you're doing, it's unsurprising. At moderately high bitrates no one can reliably distinguish them on arbitrary samples: most inputs are easy.
If you test on known-difficult "killer samples" you'll probably easily distinguish them, even without first being shown what to look for, and certainly after.
During the development of Opus I created many 'trained listeners' and selected many killer samples, and I don't recall* ever encountering a tin ear that couldn't be taught to ABX any high rate samples, though some people are obviously much better at it.
I'm not sure I'd recommend it though: learning to identify artifacts has a frequent side effect of making low rate audio like the HE-aac used in SirusXM absolutely intolerable. I'm bothered by it even when I hear cars driving by using it. :)
[*] My memory for such things sucks, so I could be wrong-- but my point that it's not expected remains.
You're right it's just minor details.
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.
As referenced in the article, a common explanation for those audible differences is that the high-resolution version of the album is sourced from a different master.
In fact if you can't hear the difference between 24/192 and 16/44.1 you shouldn't be working in audio. (Doesn't apply to consumers. Does apply to musicians and engineers.)
It's like being colour blind.
And if you don't understand the math behind quantisation, you shouldn't be posting pseudo-scientific videos where you use an oscilloscope and a cheap spectrum analyser - both tools with very limited resolution - to "prove" your point.
16 bit isn't enough for hard, objective reasons. One is that the noise spectrum of quantisation is not simple. Most people assume it's something close to plain white noise, but it really isn't. It's actually a very complex spectrum with some prominent peaks at specific subdivisions of the sample rate. Those frequency peaks are significantly above audibility. 24-bit quantisation shrinks them below audibility.
The other is that most people can hear dither/noise-shaping at 16-bits. That adds a single bit of noise which should - if you're being very literal - be far below the threshold of audibility. But it clearly isn't.
These two facts are related.
The more complex reason is that listening is an active perceptual process. The brain does a huge amount of processing to separate sources and place them in a perceptual field which includes information about perceived object type, distance, and ambience cues. Some of those cues are very quiet, and we don't hear them linearly.
So using sine waves as some kind of perceptual reference for audibility is nonsensical. We hear much more complex signals in an active way, and if there's information missing in the quiet parts - which there is with limited quantisation - then the signal simply isn't accurate.
It's not like all of your samples and virtual instruments are 192khz or even 96k. Many are 48khz or even 44.1k.
I think there are many cases where people never need to go above 44.1khz unless you maybe have saturation on the master bus. I agree that good dithering is important though and think that there hasn't been enough research on that so far.
What you are describing is the result of blunt truncation. If you use the most basic (“uniform” or “rectangular” a.k.a. “RPDF”) dither, the spectrum is in fact flat, as demonstrated by the video you are likely alluding to and calling “pseudoscientific” (https://youtu.be/cIQ9IXSUzuM?t=12m50s). If you sum two uniform dithers together, you get what pretty much everyone uses (“triangular” or “TPDF” dither) which, in addition to decorrelating the mean quantisation error from the signal, also decorrelates the standard deviation, eliminating noise modulation and leaving a correlation only in still higher-order moments like skewness and kurtosis.
You can even try it for yourself with SoX. Find a 24-bit track, quantise it with dither to 16-bit, calculate the difference between both tracks, blow up the difference and take its spectrogram and it will be completely flat. Or listen to the difference (mind the volume) and see if you can make out anything meaningful.
And then remember that this difference would normally sit at roughly -93 dB FS, so to hear it in a typical room, you would have to be listening at deafening levels. You claim that it “clearly isn’t” below the threshold of audibility but it’s not clear how you arrived at that conclusion. You then claim that the audibility of that noise floor is somehow related to what you said before about the effects of undithered quantisation, even though those effects stop being relevant the moment you apply any sort of dither.> We hear much more complex signals in an active way, and if there's information missing in the quiet parts - which there is with limited quantisation - then the signal simply isn't accurate.
It’s not missing. You can do a similar test where you “bury” your source material in the 16-bit dither noise floor, blow it up again, and you’ll be able to detect it under the noise.
In this case, it was my brother's own 24/192 recording, down-mixed by him to CD format with the intent that it be transparent. I believe he said his software was supposed to be dithering, but this was ~25 years ago and I can't really confirm the details anymore.
No one can hear the difference between properly mastered high res files. I will happily put money on it.
Small differences in gain are ABX able much more readily than differences in noise at the 16 vs 24 bit level. So if the signal chain gives even a small difference in gain between the samples that's what you'll track. A reasonable conversion path to 16 bits for mastering will also apply dithering and some kind of brickwall limiting (you have to limit after the dither or as part of the dither as dither can change levels!), and this can result in gain changes. The DAC may behave differently or have outright bugs for some configurations too.
This is particularly true wrt reconstruction filters for sample rate differences. And if you were comparing 44.1k and 192k then the physical DAC itself was likely running at a different rate and its _analog_ filters are probably better optimized for one vs the other (this is less true for 48k vs 192k, as the hardware likely runs at the same rate for both). So one answer to this comparison can be "on this particular hardware this rate is better than that rate"-- but that's a implementation property not a property of format choice.
You might think, "okay I'll use a mathematically perfect down and up conversion process and run the DAC in the exact same configuration for all cases". But even then you run into issues like after reconstruction the _inter sample_ peak levels will be higher than the levels of the samples, so you have to handle that and in a way that doesn't produce a gain difference between the two configurations. (probably by running your perfect process and finding the gain level that results in no limiting, then making the gain of the original match).
And then for the high rate vs non-high rate you have to deal with the fact that most amplifiers are not particularly linear (compared to well constructed software at least!) and that any real speaker is very far from linear. This means that the presence or absence of ultrasonics will change the audio in the 0-20khz band.. Before you think "well that could be a reason that high rate is better" observe that if there was some consistently good effect from the ultrasonics you could just bake it into the low rate sample.
> but in my 50s I know
Yeah if you're in your 50's you're absolutely not hearing differences way up above 20khz (especially if you're male), I bet you can't even hear CRT flybacks from 100 yards anymore. :P
I don't mean to discount your experience: I don't really doubt that it was real. But answering the general question of the necessity of low vs high rate probably takes a team of experts, armed with test gear and the designs of the HW/SW in question, to vet the test configuration. Testing a _particular_ configuration without the ability to distinguish its implementation quirks from format-fundamentals is much easier and that's what most attempts to test this question are actually testing.
This was common knowledge at least as far back as the mid 80s, when every hifi shop and salesguy knew to ensure the bit of gear with the highest profit margin got played an almost imperceptible bit louder than the gear the customer came in to buy during back to back testing.
Point being: it doesn't even require an unscrupulous sales person to get similar results to an unscrupulous sales person! :P
This was supposed to be running the DACs to match the source configuration, not resampling into some common format. I think that is an unavoidable part of the whole end-to-end ABX test concept.
Maybe it would be interesting to up-sample back into 24/192 and play both in that mode. But then people would argue about what type of up-sample to use.
I was in my mid 20s for this test. I understand my high-band hearing was better back then.
Second guessing it by upsampling in front of it seems dubious to me. It might help in some cases where the DAC designers were thinking of different objectives or just didn't do a great job. It might also help with some other issues, like if the dac is timed off the input clock and the input clock sucks and the upsampler retimes the signal.
Of course the upsampler designers could also get it wrong, be aliasing the hell out of the results, and happen to like the sound of the corrupted audio. :P
The effects are all objectively measurable however-- with expensive equipment at least. I think I'd want to set test results with a particular hardware combination before sticking an upsampler in it. OTOH, if there already was one there because it's just some built in feature of some kit I wanted to use otherwise, I wouldn't worry much about it. Particularly if that kit has been reviewed by people with proper test gear and they didn't decide that it was broken.
But I think I lost the ability to hear the flyback not long after I passed twenty. The world turned silent as far as that's concerned (before, you could hear it anywhere and everywhere, in shops, homes, some workplaces..)
The "20kHz" thing is kind of a myth for most people, at least that's what it looked to me after all the testing we did at school. I think it can influence what you hear, somehow, but in any case it's for very young people.
> Most people have no idea how much their high frequency hearing degrades as they age because it plays approximately no role in your life, but it's real, dramatic, and as far as I know happens to everyone.
I agree completely. I recall some discussions a long time ago on RMMGA (Usenet: rec.music.makers.guitar.acoustic) where some distinguished and experienced, but middle-aged guitarists got practically angry when a young guy described the sound of a certain type of newly-introduced strings "harsh" and "like fingernails on a blackboard" when used on a particular guitar.
The difference was, of course, that what the young guy could hear is something which stopped existing at least when you had passed 30.. I was at an age where I too couldn't hear that kind of sound from strings, but it was still not that long ago and I remembered and had noticed the difference, i.e. that I could not hear what I could hear before. For example the huge difference between fresh strings and week-old strings (and that fact has, over the decades, saved me tons of money which I would otherwise have spent on replacing strings all the time..)
You try to hear the brickwall by the muffled, enclosed quality and possibly by the weird pre-ring blurriness of the filter making things sound more vague than they have to be, and you hear the truncation not because it is audible 'distortion' as we know it, but because depth collapses and it sounds like it's coming from the speakers and not being a separate space behind/around the speakers. At no point will it be the most glaringly obvious thing but it'll never be 'distortions' as we imagine them, it's more a 'pod people' lack of personality thing.
Like a much subtler version of listening to AI music :)
I'm quite happy with 24/96 as suitable overkill for anything I might want to hear or do. Neil Young went hard on the proposition that 192 was necessary. Sold the Ponoplayer, I had one but it died on me, battery failed eventually. It really did sound awesome beyond just about any other listening device I've ever heard…
The last couple of generations of converters have gotten a lot better, so 192kHz today is likely to sound cleaner and smoother than it did ten years ago, where there was a good chance the clock was quite jittery.
Personally I don't think it's worth the extra bandwidth for playback, but I can understand why some people might want it.
Generally all of these "debates" come down to people who think math > circuitry. All real designs are imperfect trade-offs. They all have issues, and arguing as if converters are perfect when they never are, and the imperfections can be benched objectively, is... not very scientific.
There is one purely objective benchmark: a true blind test. You can believe if something is different or not, but if nobody's capably of hearing the difference, does it matter?
If you look at a site like audiosciencereview.com and pull up measurements of a DAC or ADC, you can find graphs of the antialiasing filter response. Some are great and some are not.
One could think of 16/44.1 PCM as being a codec that is potentially perfect but requiring some degree of care to encode and decode correctly.
High-dynamic-range material benefits from lots of bits.
But most music today has heavy compressors in the pipeline that kills dynamic range in favor of allowing you to hear almost even whispers, even in traffic or a city with ear pods.
But if you're from the first group, as you said it's more noticable the benefits of having better codecs and bit depth vs heavily compressed top billboard songs where even listening the master track from the studio, falls into diminishing returns.
I used to think the same. But I realized that downsampling hi-res music to 16/44.1 isn't a transparent conversion. So now I prefer the one downsampled to 16/44.1 by an expert in production env. I almost always download 16/44.1 flac files because of this.
Higher rate sampling is just like storing integers to 3 decimal places, or archiving an upscaled DVD.
I recommend you actually read the article. I vaguely recall they did it in video form too.