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24/192 Music Downloads Are Very Silly Indeed (2012)

• sp332 9y ago
  (2012) but still an excellent article. Edit: And the linked "Show & Tell" video is a great way to get some "intuition" about the sampling theorem. https://video.xiph.org/vid2.shtml https://video.xiph.org/vid2.shtml
  • tomphoolery 9y ago
    Still relevant, as well.
  • zymhan 9y ago
    Yeah, this post title should be dated, I definitely read this exact post years ago. I mean it references Steve Jobs in the present ffs.
  • sctb 9y ago
    Thanks! We've updated the headline.
• tomc1985 9y ago
  While the extra quality is lost on listeners, from experience I've found that super-HQ source material can change your results (for the better) when fed through distortion/compression/warming effects processors
  • mrob 9y ago
    Is that because of the HQ audio, or because the processing software introduces less artifacts at those sample rates/bit depths? I.e. could you get the same results by up-sampling CD quality audio?
    • trevyn 9y ago
      I imagine it's because lossy codecs are tuned for human perceptual limitations, and when you process audio, it can "pull" areas of the sound that are otherwise hidden from your perception into perceptual ranges, analogous to how fiddling with the brightness/contrast of a highly compressed JPEG image can accentuate the artifacts.
      • anyfoo 9y ago
        We are not really talking about lossy codecs here, though, are we?
        • trevyn 9y ago
          Ah, my mistake. But perhaps you could consider 24/192 -> 16/48 a lossy codec in this context; the same argument does apply, for the same reasons.
          • anyfoo 9y ago
            That way it does make sense, editing might require higher fidelity than we can perceive in the final project to avoid artifacts.
        • kuschku 9y ago
          Actually, the article is not talking about it, but the topic very much is lossy codecs. Specifically, if 24/192 AAC is worth it compared to 16/44.1 AAC (and the answer to that is yes, although the answer to 24/192 WAV is no)
      • microcolonel 9y ago
        But what about lossless 16/44.1KHz audio? It's not a "compressed" version of 24/192Khz. Chances are that your 192KHz music will be low pass filtered anyway, 192KHz is a trick that sound chipset vendors added so that integrators wouldn't have to make high quality analog filters for their ADCs. There's likely nothing above 22.05KHz except shaped noise, production artifacts, and very, very quiet ultrasound noises which have nothing to do with the music.
    • tomc1985 9y ago
      I think it's because processes that affect things at the individual sample level (which distortion/compression often do) have a lot more detail to work with. The author mentions this a bit in the article
    • TazeTSchnitzel 9y ago
      More bits means: • Less quantisation noise, so your noise floor is a bit lower and therefore, when you're mixing stuff together, you accumulate less noise • More numerical room to play with, you can scale or shift the volume up and down more without clipping and with less loss of accuracy With 16-bit CD audio, you can't just convert to 24-bit and suddenly have less noise. You might get more room, though. As for higher sampling rates (more kHz, if you will), I think Monty mentioned some benefit regarding less sharp aliasing filters (can have a larger transition band from 20–48kHz, say, rather than from 20—22.1kHz), but it's not something I understand the benefit of well.
      • dspig 9y ago
        filters with a sharper cutoff will 'ring' more, smearing sharp changes in the signal out over time. It's a pretty subtle effect when it's all happening above 20 kHz though
    • squeaky-clean 9y ago
      For distortion and compression, the only possible difference I can think of is aliasing, and yeah you could just oversample your input, then downsample again on the way out. This is really really common in most professional plugins, often with an adjustable amount to balance quality and CPU usage. I also have some gear that aliases at a volume you can't hear, but when you plug it into an analog distortion pedal, the aliasing in the high frequencies becomes apparent. This would be avoided if it had a higher sample rate so the aliasing was far out of the audible range. For other sorts of effects like spectral processing, pitch shifting, the extra detail above 22khz really does make a difference, especially if pitching down.
    • davidgerard 9y ago
      Because modern production can involve hundreds or even thousands of digital effects. So 24-bit, or even 32-bit float just because you can, is standard. As Monty demonstrates, it's a fraudulent waste to try to sell the result as a product to the end listener.
  • acchow 9y ago
    > Its playback fidelity is slightly inferior to 16/44.1 or 16/48, and it takes up 6 times the space. The article is highly technical. Does anyone have a way to describe this phenomenon intuitively?
    • mikewhy 9y ago
      "Diminishing returns"?
    • hamandcheese 9y ago
      24/192 can capture audio well beyond what we can hear. These sounds can, though, lead to distortion that is within our hearing range, depending on the equipment used. Therefore, rather than just being useless extra data, it can be actively harmful to the listening experience.
      • tomc1985 9y ago
        It's not that much different to think of than 480p vs 4K/2160p -- dramatically higher resolution, increasing the width and length of the 'grid' that sound is recorded into -- except that in the case of audio fidelity, quality after a certain point causes problems with playback systems, resulting in audible artifacts and distortion.
        • pzone 9y ago
          That analogy works if you modify it: - 480p vs 2160p is measuring the resolution of your cell phone propped up on a pillow at the other end of your living room - experimental evidence shows that your eyesight is not good enough to pick up on the increased resolution, you've maxed out your sensory perception - Your phone stutters trying to stream at 4k so the playback might actually be worse.
          • tomc1985 9y ago
            All good points! Indeed all that extra data makes the decode stage that much more difficult. Same happens with audio too, but high-res audio isn't nearly as bulky as video or nearly as resource-intensive to decode
  • sp332 9y ago
    Sure, when you're editing music it can be helpful to have super high sample rates and lots of bits to work with. But for listening, it's just a waste of space.
    • kobeya 9y ago
      He was talking about listening.
      • sp332 9y ago
        tomc1985 said "fed through distortion/compression/warming effects processors" which is editing, not listening.
        • tomc1985 9y ago
          Yeah, editing
          • ioquatix 9y ago
            Lots of pre-amps can do this in real time, and some installation audio systems do it as part of pre-processing (e.g. compression, limiter, etc).
  • jotux 9y ago
    This is specifically addressed in the link under the section titled "When does 24 bit matter?"
  • InclinedPlane 9y ago
    Well, yeah. For properly mixed audio, you don't need anything beyond 44-khz/16-bit. If you need to do extensive processing, or you're working with raw audio and want the extra margin to be able to recover a sound when it may be too quiet (or too loud) or otherwise compromised then more samples and greater bit-depth can be highly useful.
  • jchanimal 9y ago
    I've heard (and it passes the smell test) that even if your ears can't resonate at frequencies higher than human hearing, the space you are in can, and your ears are sensitive to the differences in arrival time that are smaller than the pitches we can't hear. So at least in high sample rate music the imaging could potentially be cleaner. Assuming super great everything else.
    • stephen_g 9y ago
      What you're sort of describing - having more of the higher frequencies resonate together - can actually cause intermodulation distortion (which itself can be in the audible range), which can make it sound worse... I haven't had time to check the linked article but if it's the one I'm thinking of I'm pretty sure it goes into this. In the end though, double blind testing (again, I think mentioned in the article) shows that there's a threshold (a bit below 48kHz, probably why 44.1kHz was chosen for CD audio) after which people cant distinguish between higher sample rates at any more accuracy than randomly guessing.
      • jchanimal 9y ago
        I think it's the intermodalation distortion or as I call it "the room sound" that your non-conscious mind perceives. The best example of this I've heard is when listening to identical masters on SACD/DSD vs PCM audio. Notably different sound when reproduced via loudspeakers.
    • __david__ 9y ago
      That's one of those things that sounds true without putting thought into it but is ultimately false. A 44.1kHz sampled sounds can encode phases with offsets smaller than 22µs (1/44.1k). How much smaller? I'm not exactly sure, but I believe it's dependent on your sample size. My guess is 1/2^16 (assuming sample resolution of 16 bits), but I might be off by a factor of 2 (or indeed 100% wrong) there. Caveat emptor. If I'm right then that's 346 pico-seconds, which I'm betting is smaller than your brain can differentiate.
      • deleted 9y ago
  • dangayle 9y ago
    It's also great if you you're doing a bunch of time/pitch mangling.
  • kazinator 9y ago
    In the amplitude domain, you definitely need more bits for that type of processing, and many other types. Digital distortion basically simulates high gain with soft clipping, which basically takes a narrow slice of the amplitude domain and magnifies it. The extra resolution has to be there for that not to turn into ugly sounding shit with artifacts.
• mmahemoff 9y ago
  Mirror: http://evolver.fm/2012/10/04/guest-opinion-why-24192-music-downloads-make-no-sense/ http://evolver.fm/2012/10/04/guest-opinion-why-24192-music-d...
• 0xCMP 9y ago
  I've always been able to tell when something on Spotify I've heard before is lower quality now. I can instinctively tell in my car when that's the case. It's mostly due to how loud certain parts of the song are or how it sounds at higher volumes. I guess, given what he says, that wouldn't be the case if made sure they were the same dB.
  • anyfoo 9y ago
    It is well known that music media for "popular" consumption are often intentionally mastered with much less dynamic. This is called "compression", but has, on the surface, nothing to do with compression of bits* . It's the dynamic that's compressed, making quieter and louder sounds be closer together. For markets with people that are more likely to care about sound quality, though, a much larger dynamic range is preserved. This is why the same album often sounds better on vinyl than on digital media[1]. It has nothing to do with the media, it's the superior mastering that was consciously chosen. The Wikipedia article on the Loudness War[2] offers a good explanation. * Well, technically, music with compressed dynamic has less entropy, so can be encoded at a lower bitrate without loss. [1] See this database for example: http://dr.loudness-war.info http://dr.loudness-war.info [2] https://en.wikipedia.org/wiki/Loudness_war https://en.wikipedia.org/wiki/Loudness_war
    • baddox 9y ago
      It's true that much popular music is heavily compressed, but you normally wouldn't encounter two differently-compressed masters of the same song. Two exceptions would be radio (some stations used to apply their own compression, and perhaps they still do) and some high-quality releases aimed at audiophiles that are produced from a less-compressed master. I wouldn't think that streaming services like Spotify would re-compress songs.
      • anyfoo 9y ago
        I was a bit imprecise, but the "high-quality releases aimed at audiophiles" is essentially what I was referring to. The record stores are full of remastered vinyl editions.
• squarefoot 9y ago
  FTA: "192kHz digital music files offer no benefits. They're not quite neutral either; practical fidelity is slightly worse. The ultrasonics are a liability during playback." I would expect the samples to be interpolated before the final DA conversion takes place, so no extra ultrasonics should be involved. And there would be filters for them anyway, we still have to use them for 44.1, 48, 96 etc.
  • sp332 9y ago
    But if you're filtering out ultrasonics, then you've guaranteed that the extra samples will be wasted. There won't be any difference in the reconstructed signal at all.
    • zkSNARK 9y ago
      As an engineer, those extra samples are never wasted, and as a listener, I could care less about the "wasted space." Drive space is meaningless at this point. When I buy a piece of audio, give me the highest quality available and I will make my own decisions regarding the frequency I want to listen to it at. The reasons behind not giving the user the highest quality possible all sound like a misguided excuse to prevent piracy.
      • sp332 9y ago
        Ok... but the point stands that the signal will be the same if you low-pass filter it no matter how many samples you take. You might as well distribute the file at 44.1 kHz and then interpolate to something higher yourself, if you want to play with it later.
      • enneff 9y ago
        > as a listener, I could care less about the "wasted space." In general, bandwidth is more of a concern than disk space. Most media people consume is never stored on the local device.
      • IgorPartola 9y ago
        So 192/1024? 1024/4M? 1G/1T? Like is there a practical limit for you when you say "highest quality possible"?
• mrob 9y ago
  Super-high sample rate could make a difference if you're sampling it and pitching it down heavily to make a bass-line. But in practice, I think most such audio has very little information above 20kHz. Ordinary microphones are designed for audible frequencies, and ultra-sonics are easily lost in production. What music actually contains interesting ultra-sonic information for slowed-down listening?
  • TazeTSchnitzel 9y ago
    If anything, wouldn't the high sample rate be a liability when trying to shift pitch? If it captures too much outside the range of human hearing, then that ultrasound might suddenly become audible when you change pitch, which could sound weird.
    • mrob 9y ago
      Exactly, and the weirdness might be interesting.
      • deleted 9y ago
        [deleted]
      • TazeTSchnitzel 9y ago
        Ah, right. And if ultrasound is actually much like regular sound, it may sound perfectly normal.
• jahnu 9y ago
  I've said it before, this video demo is one of the very best I've ever seen. https://xiph.org/video/vid2.shtml https://xiph.org/video/vid2.shtml So well prepared, so well presented, so little that could be removed without ruining it. I aspire to do such good demos but always fall so short.
  • esaym 9y ago
    Wow, that was awesome!
  • kaishiro 9y ago
    Thank you, this really is a great watch. Particularly the zero point stuff re: stairstepping - I had no idea this was merely a function of display.
  • osrec 9y ago
    09:30 to 10:30 explains the effects of quantisation extremely well! Overall, that is one of the clearest, no frills, to the point videos I've ever seen.
  • combatentropy 9y ago
    > this video demo is one of the very best I've ever seen I have said the same thing. I was a film major, a video producer, and a tech writer (now a programmer), and I am in awe.
  • IgorPartola 9y ago
    That is fantastic. The presenter manages to pull of "charismatic engineer" really well.
    • obrajesse 9y ago
      Well, he is a very charismatic engineer...
  • AceJohnny2 9y ago
    Monty Montgomery, the author, hangs around here as id=xiphmont :)
• pmalynin 9y ago
  My go to comparison track for sound quality for a new device, headphones, DAC, amp, codec etc, is "Speak To Me" from Dark Side Of The Moon. I don't hear a difference between various rips that are 16/128 or 24/192, but I have noticed a difference listening to the Blu-ray version of it (which is many-many gigabytes in size). It is a definitely interesting experience, but the way I can describe it is as an absolute absence of noise. Every single version but this one exhibits noise at the start (the heartbeat sound) as the sound goes from very quiet to loud. But, to be fair, it could just be different masters.
  • terinjokes 9y ago
    Which would be in line with the article: a SACD release pressed onto a CD-R sounded better than the CD release, because the SACD release had a better master.
  • nightcracker 9y ago
    You should try downsampling the Blu-ray version yourself and comparing then.
  • ycombinete 9y ago
    Hah. I use the Dark Side of the Moon album for comparisons as well. I find that you need to use something that you are very well acquainted with. Oddly enough I also use Electric Heart, by Sia, as it has such a bizarrely flooded mid-range :)
  • anentropic 9y ago
    almost certainly different masters :)
• Veratyr 9y ago
  Perhaps this is a silly question but the explanations mostly make perfect sense to me but in the case of sample rate, what happens when two waves with very different frequencies overlap? Say I've got an 18kHz wave and a 9kHz wave and the 9kHz wave is ever so slightly out of phase. Then imagine there are 10 different waves under 20kHz all interfering with each other in different wages. Is it still possible to reproduce everything accurately? And on bit-depth and dynamic range: Given that much audio doesn't use the full range available to it, wouldn't higher bit depth increase the fidelity in the range the audio does fill? The article talks about the bit depth and range only in terms of the maximum volume but what about fidelity? What's the minimum difference in volume the human ear can hear?
  • TazeTSchnitzel 9y ago
    IIRC, I think the article's point about maximum volume was about how you'd need to turn the volume up to deafening levels before the small differences would be noticeable.
    • Veratyr 9y ago
      It doesn't seem to say anything like that. Looking at it again, I see this: > It is also worth mentioning that increasing the bit depth of the audio representation from 16 to 24 bits does not increase the perceptible resolution or 'fineness' of the audio. It only increases the dynamic range, the range between the softest possible and the loudest possible sound, by lowering the noise floor. However, a 16-bit noise floor is already below what we can hear. But I don't understand, why is this? Is the step in volume between two values fixed regardless of the bit depth? If so, why?
      • lgunsch 9y ago
        The video linked in the top-most comment will beautifully answer your questions. It's also on YouTube. You can think of it as in the sampling rate provides a mathimatical solution to the waveform - it's simply 100% accurate all the time. The bit depth adds extra "buckets" to divide the volumes into . If you have 2 bits, you then have volume levels 0, 1, 2, 3.
• flatfilefan 9y ago
  when I listen mp3s via 24/192 M-audio spdif to denon 1705 there is a huge differenc to just optical from the motherboard. why is that then?
  • anyfoo 9y ago
    There can be dozens of completely different reasons why that might be the case, without requiring inaudible frequencies to be involved.
• kuschku 9y ago
  This article makes the same mistake that is done frequently with video. So, let’s look at a similar issue with video. Your display is likely only 720p, or 1080p, but a 4K video on youtube will still look a lot better, although technically it should have no visible difference. But the reality is, we don’t get uncompressed video, or uncompressed audio. We have a choice between audio or video compressed with lossy codecs, at 16bit/44kHz or 16bit/96kHz, or 4:2:0 video at 1080p or 4:2:0 video at 4K. And just like you need 4K 4:2:0 mp4 video to get even close to the quality of uncompressed 1080p 4:4:4 video, you also need far higher sampling rate and depth of highly compressed audio to get a quality similar to 16bit/44.1kHz PCM. That’s the real reason why 24bit/192kHz AAC downloads were even discussed.
  • TazeTSchnitzel 9y ago
    > Your display is likely only 720p, or 1080p, but a 4K video on youtube will still look a lot better, although technically it should have no visible difference. Not at all, bitrate differences are audible. > And just like you need 4K 4:2:0 mp4 video to get even close to the quality of uncompressed 1080p 4:4:4 video, you also need far higher sampling rate and depth of highly compressed audio to get a quality similar to 16bit/44.1kHz PCM. …how so?
    • kuschku 9y ago
      > Not at all, bitrate differences are audible. Even with the same bitrate, you’ll need 4K video to get quality comparable to uncompressed 1080p. For example, mp4 defines that the brightness channel Y should be stored with full resolution, and the color channels Pb Pr should be stored at half resolution. So in a 4K mp4 video at lossless bitrate, the actual colors will only be stored at 1080p. This functionality is called Chroma Subsampling. Audio Codecs do something very similar, causing these exact issues.
      • astrange 9y ago
        It doesn't matter what resolution the color is stored at, as long as the high frequencies are the same as the Y channel. You won't have lost anything. Of course, cross-channel intra prediction would work better[1] but 4:2:0 is pretty good quality considering you can throw out 3/4 the pixels. [1] https://people.xiph.org/~unlord/spie_cfl.pdf https://people.xiph.org/~unlord/spie_cfl.pdf
• flavio81 9y ago
  Good points in the article, but it has some flaws. The problem whenever somebody writes about digital audio, is that it is very tempting to hold on to sampling theory (Nyquist limit, etc) and totally discard the problems of implementing an actual Analog-Digital and Digital-Analog chain that works perfectly at 44100Hz sample rate. I agree with the assesment that 16 bit depth is good enough; even 14 bit is good enough and was used with good results in the past (!). However, the problem is with the sampling rate. > All signals with content entirely below the Nyquist frequency (half the sampling rate) are captured perfectly and completely by sampling; Here lies the problem. This is what theory says, however, when using 44KHz sample rate, this means that to capture the audio you need to low-pass at 22KHz. And this is not your gentle (6, 12 or 24db) low-pass filter; no, this needs to be HARD filtering; nothing should pass beyond 22KHz. And this must be on the analog domain, because your signal is analog. To implement such a filter, you need a brickwall analog filter and this is not only expensive, but it also makes mess with the audio, either 'ringing' effects and/or ripple on the frequency response and/or strong phase shifts. So on Analog-to-digital in 2017, converters should be operating at a higher rate (say, 192KHz), because this makes analog filtering of the signal much easier and without side effects. Now, for Digital-to-Analog, if your sample rate is 44KHz, you have two alternatives: a) Analog brickwall filtering, with the problems noted above or b) filtering on the digital domain + using oversampling the article mentions: >So the math is ideal, but what of real world complications? The most notorious is the band-limiting requirement. Signals with content over the Nyquist frequency must be lowpassed before sampling to avoid aliasing distortion; this analog lowpass is the infamous antialiasing filter. Antialiasing can't be ideal in practice, but modern techniques bring it very close. ...and with that we come to oversampling." So they are mentioning alternative (b). The problem is that oversampling does not solve all problems. Oversampling implies that the filtering is done on the digital domain and there are several choices of filtering you could use, for example FIR (Finite Impulse Response), IIR (infinite impulse response), etc. And each one of these choices have side effects... In short, the problem is that with 44KHz sampling rate, your filter cutoff (22KHz) is too short to your desired bandwidth (20Hz-20KHz). Using a sample rate of 192KHz gives the DAC designer much more leeway for a better conversion. And CONVERSION is the key to good digital sound. >What actually works to improve the quality of the digital audio to which we're listening? It is interesting that the author mentions things such as "buying better headphones" (agree), but he never mentions "Getting a better Digital to Analog converter", which is highly important !! On the other hand, he backs up his claim that "44KHz is enough" with an interesting AES test i was already aware of in the past: >Empirical evidence from listening tests backs up the assertion that 44.1kHz/16 bit provides highest-possible fidelity playback. There are numerous controlled tests confirming this, but I'll plug a recent paper, Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback, done by local folks here at the Boston Audio Society. This is a very interesting paper, and I did have the copy, however the test equipment should be checked. There are systems and better systems. The AES paper cited above had the particularity that the ADC and DAC used were provided by exactly the same machine (a Sony PCM converter), with the same strategy: no oversampling, brickwall analog filters. I can bet (99% sure) that the brickwall filters were identical on the ADC and the DAC on that machine; Murata-brand filters in a package. The devil, as they say, is in the details.
• RachelF 9y ago
  To be replaced soon by "4k video is very silly indeed" If you look at the angular resolution of the eye, unless you are sitting very close to the screen, you can't resolve 4k video.
  • combatentropy 9y ago
    Yes, I've measured my eyes' resolution to be about 60 pixels per degree. I made a black-and-white grid and set it as my desktop picture, with a one-to-one magnification. Then I backed away until I could not discern the pixels. There's even a moire effect, like with cameras, as I reached my threshold. Then I used trigonometry to calculate my angle of view. Most living-room TVs are probably placed at a 15- to 25-degree angle of view. 1920x1080 is enough for up to a 32-degree horizontal angle of view, which is 7 feet away from a TV with a 55" diagonal, for example. I will say, however, that movie theaters look a little better with 4K, since 30 degrees is supposed to be the back row, and the front row might be 60 or so.
  • jaquers 9y ago
    Can't tell if this is sarcasm or not, but I think that's sort of the point. You want a high enough pixel density so you can't perceive pixels at normal viewing distance, and I can definitely see pixels on my 42in 1080p screen from across the room.
    • sundvor 9y ago
      I'm going to go for "sarcasm" there. I use a 65" 4k TV with my HTPC in the living room, but often do desktop-ish things on it. (Logitech's wireless keyboards are great). The difference in resolution between 1080 and 2160 is huge. 1080 is just fuzzy.
      • quickthrower2 9y ago
        65 freaking inches! Of course you are gonna need 4k
  • stephen_g 9y ago
• rconti 9y ago
  Off topic, but I've noticed in the gym, sometimes the music playing from the instructor's iPhone through one of those cheap Ion Block Rocker bluetooth speakers (big boxy one, looks like a guitar amp) has very very noticeable pitch sag in various parts of various songs. It's via bluetooth, and the source is Pandora. I even googled Pandora and pitch change, and found some forums with folks discussing bluetooth causing this to happen, which has never been my experience. It occurred to me that slowing the playback (eg, causing the pitch to sag) could be a fantastic way of dealing with a connection that's too slow. It would be far better than stuttering, even though many of us would find the pitch change really annoying. The instructor and my wife simply can't hear it, or say "I thought that was just part of the song". Anyway, has anyone ever heard of this happening? Do certain products do it? Were the Pandora forums wrong and this is actually a Pandora problem? Of all of the bluetooth problems I've had in my life, I've never experienced this before, so I tend to lean towards it being a data rate issue on the cell connection and Pandora slowing the music down.
  • jaquers 9y ago
    At my greatest aspiration, I'm no more than a casual music listener - however I am very conscious of pitch. On more than one occasion, I've noticed that a song I'm familiar with seems off pitch, as in lower or higher key than I think of it in my head - and I seem to notice that particularly around shitty speakers or bluetooth or some combination of the two. Glad to know I'm not crazy :) To your point, I would lean towards bluetooth as I've never associated the phenomena with Pandora. My friends and I mostly use Spotify.
    • rconti 9y ago
      I use spotify as well. And what I describe is a sag for 5 or 10 or 15 seconds, not just a song that's playing too slowly. it's really weird.
    • colejohnson66 9y ago
      Or people who pirate their music by ripping YouTube videos that are just the songs pitch shifted to avoid automated copyright strikes
• yzhou 9y ago
  the devil is in the DAC and ADC. You just can't turn 16bit/24bit data directly to/from analog without much loss. The 1/65536 accuracy voltage divider simply don't exist. So, you have to up-sample the songs to high rates with less bits, like 1bit to 6bits, then do the conversion, and get the best SNR you can. In this sense, there's simply a lot of advantage of using 24/196 since the above conversion can result in less loss and higher SNR
  • arrrvalue 9y ago
    why was this down-voted too? it's the truth!
• tzs 9y ago
  What about for non-humans? Consider a dog that lives with a musician who plays, for example, trumpet. The musician plays the trumpet at home to practice, and also records his practices to review. A trumpet produces significant acoustic energy out to about 100 KHz [1]. When the musician plays live the dog hears a rich musical instrument. When the musician plays back his recordings, half the frequency range that the dog could hear in the live trumpet will be gone. I'd imagine that this makes the recorded trumpet a lot less aesthetically pleasing to the poor dog. [1] https://www.cco.caltech.edu/~boyk/spectra/spectra.htm https://www.cco.caltech.edu/~boyk/spectra/spectra.htm
  • QAPereo 9y ago
    In my experience as a long time trumpet playing dog owner, the dog is generally thrilled when there is less richness in the noise, and generally less trumpet playing period. I say this with experience owning many dogs and having played the trumpet in a range from really badly and inexpertly, to pretty good... they don't care. Trumpets in person are just way too loud and a bit too much in general for most dogs.
    • 72deluxe 9y ago
      And perhaps too loud for humans? I only say this because most people are happy listening to music on CDs but when in the presence of a live band (eg an orchestra) it is suddenly obvious how incredibly loud it is. My brother is a drummer and I find it incredibly loud; I am a bass player and I don't play loud although he sometimes complains that there's "too much bass". Perhaps we just go deaf in our relative audio spectrum.
  • Florin_Andrei 9y ago
    How do we know that the trumpet still sounds harmonically pleasing above 20 kHz? It may sound nice to the human ear, and a horrible mess to anyone with ultrasonic hearing capabilities. Have we asked the dogs?
    • kazinator 9y ago
• dbills333 9y ago
  If you sample the waveforms at 192khz you are still sampling lower sound frequencies at a higher rate which does produce a better quality sampling of the waveform. All digital audio looses information from the original wave that is between the sampling times. You will not have a pure sine wave in digital format with this nethod of sampling. Stop confusing frequency range and sample rate. Its such a fundamental face palm. Nobody is saying listening to sound at a frequency above 22k is improving anything.
• plg 9y ago
  This is fabulous
• dep_b 9y ago
  Without even reading the article I already know that it's extremely hard that hearing the difference between 16/44.1 and 24/192 indeed is really hard if not impossible. I only use higher bitrates when creating music so when I mix or record I don't need to make the levels go almost into red to get a hot enough signal, I can do all kinds of changes like slowing down or up, distortion, EQ, compression without losing any perceived detail in the end. If I record on 16 / 44.1 then start manipulating the sound I start losing detail immediately. But in the end more than 20 / 88.1 won't do much for you. 16 / 44.1 might be a bit less than ideal for music with a lot of dynamics but it's absolutely fine for most purposes.
  • anyfoo 9y ago
    No, not "really hard if not impossible", it is impossible. The article itself cites that no one has ever been observed to be able to do so.
    • dep_b 9y ago
      More than 44 is useful for tempo matching while mixing
      • anyfoo 9y ago
        Yes, the article also mentions how 24/192 can be useful for production and editing.
    • ehsankia 9y ago
      Yeah, when you leave the possibility open, everyone wants to believe they are that special one with amazing ears that can tell the difference where no one else can, but it's always placebo.
• duderific 9y ago
  Kind of a side topic, but does anyone know what's up with Sirius XM satellite radio? Whenever I listen to that in my wife's car, I find the sound quality to be obnoxiously bad, to the point that I choose to listen to plain old FM if I'm driving her car. The higher frequencies (hi hats for example) are mushy and sound like they are warbling, and the sound just generally has a lack of depth. What is the explanation for this, or am I just imagining it?
  • kenwjr 9y ago
    No, I hear it too and would like to know how it works. I think some stations get more fidelity than others, but it's almost always compressed as you described.
  • dangayle 9y ago
    Probably compressing the living daylights out of it. By squashing it down, they flatten the amplitude in the music and things with a lot of transients (by definition, drums) suffer. It results in a lot of fatigue, because the higher the compression the more it turns into white noise, essentially. Sirius XM might be one of the last to standardize on a reasonable LUFS setting of 12 to 16.
  • TD-Linux 9y ago
    I notice it too, and it's because the music is extremely low bitrate - usually 48kbps, but sometimes as low as 32. It's HE-AACv2 - not as good as Opus, but close. Its characteristic artifacts include destroying transients, like your hi-hats.
  • com2kid 9y ago
    Highly compressed AAC. Answers on Quora say about 48kb/s of bandwidth, so equivalent to about a 96kb/s MP3, which sounds about right. Bandwidth allocation is per channel, talk only channels are even worse and some words are barely comprehensible.
• kevin_thibedeau 9y ago
  Makes me long for the day when consumer audio products were proudly labeled as having a 1-bit DAC. Nowadays you couldn't sell audio hardware that made such truthful statements about its internals. Gotta have the bits.
  • byproxy 9y ago
    Hey, they still do. And it's 'audiophile.' DSD is 1-bit, but with a very high sampling rate (2.8224 MHz). Though, as far as I know there isn't actually any quality boost vs PCM.
    • kevin_thibedeau 9y ago
      That's my point. The golden ears wouldn't deign to buy a "cheesy" 1-bit DAC. It has to be recast as something else to fool them.
• Const-me 9y ago
  If the only thing you do with your music is listen, then yes, 24/192 delivers questionable value compared to the more popular 16/44.1 or 16/48 formats. However, all musicians I know use these high-rez formats internally. The reason for that, when you apply audio effects, especially complex VST ones, these discretization artifacts noticeably decrease the result quality. Maybe, the musicians who distribute their music in 24/192 format expect their music to be mixed and otherwise processed.
  • zaroth 9y ago
    TFA specifically addresses this and agrees 100% with you on it. Develop in 24/192, but ship in 16/48. Good point about supporting better downstream remixing!
    • daurnimator 9y ago
      I feel like there is some parallel to open source and GPL type licenses: you can never know which of your customers may wish to remix or further work on your materials, so you should ship the "source" material.
  • kazinator 9y ago
    I do not believe it. 24 bits is definitely needed for processing (better yet, use floating-point). Not 192 kHz; no friggin' way. Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works.
    • WalterSear 9y ago
      Unless you are altering the time or pitch, which, these days, you are more than you are not. But no, musicians aren't releasing things at ultra-resolutions because they expect others to reuse their work. The ones that are, are providing multitracks.
    • mbell 9y ago
      > Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works. That isn't entirely true. e.g. It's common for an audio DSP to use fixed point 24bit coefficients for an FIR filter. If you're trying to implement a filter at low frequency then there can be significant error due to coefficient precision, that error is reduced by increasing the sampling rate.
• arrrvalue 9y ago
  and yet, we are nowhere near being able to electronically reproduce a live acoustic music performance. have you ever walked by a bunch of musical sound coming out of a room and thought to yourself, "wow, those live musicians sound great" only to discover it was just a stereo playing? nope. as engineers we will never solve this problem as long as the "44.1kHz is good enough" dogma is perpetuated. here's a question. why are frequency and bit depth the only two variables under discussion here? how does the human ear locate a sound in space? suppose I place a series of 20kHz tone generators along a wall (and that I can still hear 20kHz :) and trigger them at different times, and record the session in stereo at 44.1kHz with a standard X-Y mic setup. will I be able to reconstruct the performance?
  • flyinghamster 9y ago
    In my own experience, the vast majority of consumer audio gear doesn't even take full advantage of 16/44.1. I've made live recordings of my classical chorus, and if I want it to be listenable on typical consumer gear, I have to apply a significant amount of dynamic range compression - otherwise, it will be either too quiet, or it won't handle the peaks when played at concert volume. That being said, I'm using quite a bit less compression than the loudness-war-type mastering that is all too typical with pop music.
    • chungy 9y ago
      That's the funny thing about those tools -- applied properly, it can be a great asset. Applied carelessly, you end up with clipping and brickwalls of audio.
  • romwell 9y ago
    >as engineers we will never solve this problem [reproduce a live acoustic music performance] as long as the "44.1kHz is good enough" dogma is perpetuated It's the opposite. We are never going to solve this problem if we are going to focus on things that have nothing to do with the problem. Compare and contrast: >as engineers we will never solve this problem as long as the "copper wires are good enough" dogma is perpetuated Also, please read the article. The author specifically lists advances in audio tech they think are worthwhile to consider, such as surround sound. This actually addresses the problem you mentioned (reproducing the live performance) and the question you asked, i.e. >here's a question. why are frequency and bit depth the only two variables under discussion here? They are not, at least not in the article. Here it's because that's what's in the title, and not everyone gets to the end of the article. Some comments do talk about the importance of having a good DAC for a good sound.
• lz400 9y ago
  There's a very straightforward practical reason why you might want to keep downloading 24/192. The people who put those together are on average "audiophile" snobs so the rips tend to be perfect, often from high quality sources like SACDs, hdtracks, etc. The source of the recording is usually the best master known for that record (special remastered editions, etc.). If you download mp3/spotify chances are the copy you download is from a worse source. Sure, you don't need 24/192, a well ripped 320kbps mp3 would be the same in practice, but looking for 24/192 in the internet is an easy way of getting better quality music on average in practice.
  • derefr 9y ago
    Similarly, seeing 24/192 on a rip from an analogue source (e.g. vinyl) usually implies that a high-quality player+amp+ADC was used to digitize the album.
  • kazinator 9y ago
    So, in other words: bite the bandwidth bullet and get the 192 material, but for Pete's sake, do yourself a favor and downsample to 44.1.
    • lz400 9y ago
      If you have the patience, sure. In my desktop I keep everything on the big file source and listen to it that way. I also have dedicated DACs/AMPs that can decode all the way up to DSD. I know it probably doesn't make a difference but whatever, storage is cheap. For my phone I listen to spotify.
  • mbell 9y ago
    > The source of the recording is usually the best master known for that record (special remastered editions, etc.). Usually what is happening here is you're getting a master that hasn't been compressed to death (see loudness wars). Vinyl is a shit source for 'quality', but most records aren't compressed to death so they can still sound better on a good set of speakers due to the dynamic range.
  • gruez 9y ago
• insulanus 9y ago
  Excellent writing. Information conveyed concisely, but targeted at non-audiophiles.
• gwbas1c 9y ago
  "Under ideal laboratory conditions, humans can hear sound as low as 12 Hz[8] and as high as 28 kHz..." From https://en.m.wikipedia.org/wiki/Hearing_range#Humans https://en.m.wikipedia.org/wiki/Hearing_range#Humans That explains why some people can tell the difference between 44.1 khz and the higher-resolution sampling rates. It also means that the ideal audiophile sampling rate is somewhere between 58khz and 60khz, not 192khz.
  • TheRealDunkirk 9y ago
    You just need these speakers (http://magico.net/product/ultimate.php http://magico.net/product/ultimate.php) to reproduce those frequencies! The problem, of course, is whether even "prosumer" recordings keep frequencies higher than 20 or 22kHz, even if they were recorded on equipment that sampled at 192? Seems like most would LPF the rest away in the process of mastering for the masses.
• sqldba 9y ago
  It really needs a TL;DR because it's all buried in so much fluff I didn't understand why. (I understand the rest of you all really enjoy that and wouldn't call it fluff but I'm not an audio engineer).
• Mediterraneo10 9y ago
  Not entirely silly. Yes, the purported benefits of this high-fidelity audio are imaginary or even include undesirable traits. However, when most "remasters" of pop music today involve the dynamics being boosted to "loudness wars" standards for a target audience of people listening to the music through earbuds in the street, the 24/192 downloads or SACD releases are often the only way to hear the album with real dynamic range. I’m not sure how big that market is these days, though. I recently decided to move from a block in the city to a house in the surrounding countryside purely to get a quieter listening environment and let me really enjoy the high-dynamic-range recordings I have – I listen to a lot of classical music, especially the avant-garde with its even greater range, e.g. the Ligeti cello concerto starting from pppppp. Yet even among my friends who are really obsessed with such music, seeking a better listening environment seemed an extreme measure to take. So, people who a generation ago would have invested in higher-end equipment (not audiophile snake oil, just better speakers) and who would have sought silence are now giving in to listening to music on their phones or cheap computer speakers. It’s a big shift.
  • hashmal 9y ago
    How does 24/192 help with dynamic range?
    • akvadrako 9y ago
      That's the 24-bit part, which provides more play room then 16-bit. Normally 16 is plenty, by consider an example track which is heavily compressed so only the top 1% of range is used: 16-bits = 2^16 * 0.01 = 655 discrete levels 24-bits = 2^24 * 0.01 = 168000 discrete levels
      • Mediterraneo10 9y ago
        No, that has nothing to do with it. See my reply above.
        • akvadrako 9y ago
• donatj 9y ago
  Contrary to what he says, I can most certainly see some IR remotes. Not super bright but enough to notice. Same goes for the IR lights in my Kinect.
  • rblatz 9y ago
    Have you had cataract surgery? I know that can have an impact on visible spectrum depending on the replacement lens they put in.
    • donatj 9y ago
      Nope, I'm afraid not. I never noticed until maybe 3 years ago, so I think it's a newer development. I'm 31. I also got glasses this year for the first time in my life, which the need for came on rather suddenly... So maybe it's linked with some kind of degeneration? I really can't say. I honestly didn't think to mention it to the doctor, but now I'm thinking maybe I should have.
      • rblatz 9y ago
        I'm not an ophthalmologist, so if you are concerned about your eye health talk to one. That being said I was referencing that the human eye (retina) is capable of seeing a wider spectrum than we typically see. The natural lens filters out part of the spectrum (UV) that the synthetic replacement lenses do not filter out.
  • rikkus 9y ago
    I expect this is because some actually produce light in the visible spectrum too.
  • xiphmont 9y ago
    "The original version of this article stated that IR LEDs operate from 300-325THz (about 920-980nm), wavelengths that are invisible. Quite a few readers wrote to say that they could in fact just barely see the LEDs in some (or all) of their remotes. Several were kind enough to let me know which remotes these were, and I was able to test several on a spectrometer. Lo and behold, these remotes were using higher-frequency LEDs operating from 350-380THz (800-850nm), just overlapping the extreme edge of the visible range. "
• tigerBL00D 9y ago
  If you are downloading to simply consume, then sure. But as others have noted this article assumes that mastering is always the last step in the editing process. Anyone who ever remixed, sampled or DJ'ed will disagree. If we keep that in mind, then the setup of the problem changes significantly and most arguments made here do not apply to the download itself.
  • krisdol 9y ago
    Right. I used to DJ throughout college and tried out a variety of formats. At high amplification, artifacts become apparent. VBR0 and 256kbit MP3s did not hold up, at all. Even smaller speakers and headphones didn't hide some of the compression artifacts for me. 320kbit CBR MP3s were... I'd say, generally OK if you did not plan on skewing their tempo much outside of a very small range (give or take 5% speed). Really bad artifacting becomes audible quickly beyond that range. Maybe it was placebo, but I also found differences between FLAC and 320kbit MP3 discernible when working on big, more powerful speakers. But, with FLAC, it didn't matter if the sound was played at 10% tempo or extreme amplitude, audio was always crisp, clear, and free of compression artifacts (obviously). On my laptop speakers or earbuds, no way would I be able to tell the difference between any of these today.
  • hashmal 9y ago
    > Anyone who ever remixed, sampled or DJ'ed will disagree. Could you expand on this?
  • stephen_g 9y ago
    He specifically addresses audio engineering (which can be interpolated to DJing and remixing). As he says, what you need in that case is 24bit audio to have more dynamic range to play with. >48kHz is still useless to human hearing. This particular article doesn't touch on compression as well, which is probably the biggest thing for remixing or DJing. You really want uncompressed or at least high-bitrate AAC.
• mailslot 9y ago
  I don't know. It could be very useful once cellphone makers find out how to blast audio directly into nerve impulses. New and innovative audio filters. Humans with custom DNA. Etc.
• josteink 9y ago
  I've done signal and sound-processing courses at the university. I know the Nyquist-Shannon theorem. I know all about samples, and digital sound not been square staircases. I know and understand how incorrect down-sampling from high frequencies can cause distortion in the form of sub-harmonics in the audioable range. I know about audible dynamic range and how many decibels of extra range 8-bits are going to give you. I know all this, but I still have to admit: if there's a hi-res recording (24-bit, >48kHz) available for download/purchase, I'll always go for that instead of the "regular" 16-bit 44.1/48kHz download. I guess escaping your former audiophile self is very, very hard. Anyone else want to admit their guilty, stupid pleasures? :)
  • TheSpiceIsLife 9y ago
    More is better right? Deriving pleasure from listening to music has a large subjective component. So if I've paid more for a track and / or I got the most amount of bits I could I'll probably enjoy it more. Also makes for great conversation topics.
  • xiphmont 9y ago
    I collect vintage headphones, and especially relish the truly absurd ones I can't believe anyone thought were a saleable item. I'm up to about 300 different models.
    • TheRealDunkirk 9y ago
      Would love to see a shot of the collection on /r/headphones.
      • specto 9y ago
        Yeah seriously, that's really cool.... and on that note what headphones does he actually use?!
  • tomc1985 9y ago
    raises hand
• Kiro 9y ago
  Am I the only one who have no idea what 16/44.1 or 16/48 mean? I initially thought 192 here was referring to 192 vs 320 etc but this is apparently about something completely different?
  • draugadrotten 9y ago
    16/44.1 - 16 bits at 44.1 kHz frequency. 16/48 is 16 bits at 48 kHz frequency. Both refers to the sampled but uncompressed, "raw" audio. 192 and 320 usually refers to the bitrate for mp3 compressed audio, which indeed is something different. The mp3 compression removes more and more details from the original audio to fit "inside" this bitrate speed window, hence higher is better because less original content is removed. Only great ears can hear what was removed from a 320 stream.
  • dspig 9y ago
    bit depth / sample rate common sample rates are 44.1 and 48 kHz and multiples: 96 kHz ... 192 kHz
• kevinsaptell 9y ago
  Exactly in 2017 it looks like silly but i t has been popular in sometime.
• abhirag 9y ago
  This seems the best place to plug one of my favorite musicians, 24192(https://m.soundcloud.com/x24192 https://m.soundcloud.com/x24192)
• chaboud 9y ago
  So, having come from an image-processing background to put in 15 years of experience at audio processing companies, I've seen the xiph arguments, or something like them, several times. Heck, I worked for Sony (where that precious 22KHz cliff was made) for close to ten years... in professional media... many of those years with Sony professional audio people. And the "44.1KHz is enough" or "48KHz" is enough" people are, sadly, kind of dumb. How do I know? Because I was dumb, too. Being a coding/math/audio/video badass, after a few years of industry experience, I rattled off some mouthy kid quip, saying "well, I don't know why we do 24/96, since nobody can hear above 20K anyway..." And a very talented, very knowledgable, and generally reserved engineer suddenly perked up one eyebrow and said, incredulously "because temporal and frequency response are inherently linked..." That look was in 2001, and I still remember that feeling of dread sinking in, realizing that I had no idea what he was talking about, no concept of why that would matter. I knew about Nyquist and could write a quick FFT, but he'd spent four years getting a degree in pure audio engineering at the most selective program in the country. That look, which I completely remember today, was like a deeply disappointed parent after a kid has just been bailed out of jail, from one of the nicest engineers I know. It was late, I was brash, and I pressed on, asking him what he meant. "What's a transient look like, spectrally?" he asked. He waited for my blinks to make audible sounds (due to the apparent hollowness of my head), then he asked "and how many channels of audio do we listen to?" He watched me stand there, like a doofus, for what seemed to me like several minutes (probably 5-10 seconds), then he went back to coding. It didn't hit me until weeks later, and I didn't really internalize until years later, that he was hinting at inter-ear phasing the other facilities of our auditory systems besides frequency response. Years later, I read up on Georg von Békésy's incredible work (including positional acuity), and I worked as a tech lead at Digidesign on the first-generation Venue live sound system which operated at 48KHz but with incredibly low latency (processing steps were 1, 2, or 16 samples) due to the requirements of, for instance, vocalists using in-ear monitors. Along the way, I ran across Microsoft engineers who thought that ~10-20ms inter-channel timing consistency would be okay in Windows Vista (it wasn't), conducted blind tests between 96KHz and 44.1KHz audio (for people who were shocked to immediately notice differences), came across plenty of hot-shot kids who said exactly the same kind of stuff I'd said, and saw postings from xiph making a mix of valid and grossly sophistic arguments ranging from "here's how waveform reconstruction from regularized samples work" (good) to "audio equipment can't even capture signals beyond this range" (dumb). At times, I thought about setting up refutation articles, then I realized, like many, that I had actual work to do. Von Békésy's work points to positional inter-ear phase fidelity of roughly 10µs. What's the sampling interval at 44.1KHz? >22µs? Good luck rebuilding that stereo field at 44.1... The trick is that there is a really serious diminishing return on audio sampling rate. 4KHz to 8KHz is enormous... 8KHz to 16KHz is transformative... 16KHz to 32KHz is eye-opening... 32KHz to 48KHz is appreciable... 48KHz to 96KHz is.... pretty minor, especially in the age of crappy $30 earbuds, streaming services delivering heavily compressed audio that will be crammed into a BT headset that may or may not be connected with additional compression, and all of the convenience that those changes bring. You may detect it in some audio if you're really listening, if you know what to listen for, and it may present advantages in system design (converters, processing, etc). From a data-rate perspective, the low-hanging fruit has already been picked. But people who smugly say that there is "no difference", that audiophiles are buying "snake oil", are letting their ignorance show, and that's including that kid that I was, 16 years ago. I've since moved out of pure pro media to consumer devices, where precision takes a back seat to the big picture a lot of the time. When discussing an audio fidelity multi-channel problem with a possible vendor last year, I expressed my concern about the inter-channel timing assurance slipping from 1µs to 50µs in product generations. "Depending on the sampling rate, that's several samples of audio", I said. A very senior engineer on our side (Director equivalent) quickly admonished me, saying "it's microseconds, not milliseconds", to which I said "I know... Which is why it's several samples, not several thousand..." From the look on his face, I'm 100% sure that he didn't understand me at the time, but I hope he put it together eventually. In the end, the industry has moved in the opposite direction of 24/192 for a long time. If we can get back to normalization of CD quality audio, I'll be happy.
• Applejinx 9y ago
  I work in this industry, and I have produced what is almost certainly the most high-performance dither to date, which works through noise-shaped Benford Realness calculations: http://www.airwindows.com/not-just-another-dither/ http://www.airwindows.com/not-just-another-dither/ I mention this to say that I can absolutely make 16 bit 'acceptable' or 'listenable', even for audiophiles. I do that for a living. And yet… Monty is wrong. To cover the range of human listeners, the required specs even through use of very insensitive double blind testing (which is geared to substantially indicate the PRESENCE of a difference between examples if that's present, and does NOT similarly indicate/prove the absence of a difference with a comparable degree of confidence: that is a horrible logical fallacy with realworld consequences) are more like 19-21 bit resolution at 55-75K sampling. Beyond this, there's pretty much no problem (unless you are doing further processing: I've established that quantization exists even in floating point, which a surprising number of audio DSP people seem not to understand. There's a tradeoff between the resolution used in typical audio sample values, and the ability of the exponent to cover values way outside what's required) That said, it is absurd and annoying to strive so tirelessly to limit the format of audio data to EXACTLY the limits of human hearing and not a inch beyond. What the hell? I would happily double it just for comfort and assurance that nobody would ever possibly have an issue, no matter who they were. Suddenly audio data is so expensive that we can't allow formats to use bytes freely? That's the absurdity I speak of. Our computers process things in 32-bit chunks (or indeed 64!). If you take great pains to snip away spare bits to where your audio data words are exactly 19 bits or something, the data will only be padded so it can be processed using general purpose computing. It is ludicrous to struggle heroically to limit audio workers and listeners to some word length below 32 bit for their own good, or to save space in a world where video is becoming capable of 1080p uncompressed raw capture. Moore's law left audio behind years ago, never to be troubled by audio's bandwidth requirements again. Sample rate's another issue as only very nearby or artificial sounds (or some percussion instruments, notably cymbals) contain large amounts of supersonic energy in the first place. However, sharp cutoffs are for synthesizers, not audio. Brickwall filters are garbage, technically awful, and expanding sample rate allows for completely different filter designs. Neil Young's ill-fated Pono took this route. I've got one and it sounds fantastic (and is also a fine tool for getting digital audio into the analog domain in the studio: drive anything with a Pono and it's pretty much like using a live feed). I've driven powerful amplifiers running horn-loaded speakers, capable of astonishing dynamic range. Total lack of grain or any digital sonic signature, at any playback level. My choice for sample rate at the extreme would be 96K, not 192K. Why? Because it's substantially beyond my own needs and it's established. I'm not dissing 192K, but I wouldn't go to war for it: as an output format, I would rather leave the super high sample rate stuff to DSD (which is qualitatively different from PCM audio in that the error in DSD is frequency-sensitive: more noise in the highs, progressively less as frequency drops). Even with DSD, which is known to produce excessive supersonic noise even while sounding great, the scaremongering about IM distortion is foolish and wrong. If you have a playback system which is suffering from supersonic noise modulating the audio and harming it, I have three words you should be studying before trying to legislate against other people's use of high sample rates. "Capacitor", and "Ferrite Choke". Or, you could simply use an interconnect cable which has enough internal capacitance to tame your signal. If you have a playback system that's capable of being ruined just by 192K digital audio, your playback system is broken and it's wrong to blame that on the format. That would be very silly indeed. I hope this has been expressed civilly: I am very angry with this attitude as expressed by Monty.
• cabbagehead 9y ago
  "Gentlemen, meet your ears" Shame about the sexism
• silimike 9y ago
  There's something satisfying about having really fat audio files on the hard drive.
• daveheq 9y ago
  Funny that the author says we haven't found any such people in the past 100 years of testing with truly exceptional hearing, such as a greatly extended hearing range, so they probably don't exist, yet I was tested at around 13 as having 33KHz upper range, which allowed me to hear a beetle walking over leaves and grass in a basement window sill at about 20 feet away... Unfortunately I lost a lot of hearing through concerts, headphones, and traffic, but I can still tell the difference between an MP3 and CD-quality music much of the time.
  • YSFEJ4SWJUVU6 9y ago
    Your describing of “MP3” as a level of quality doesn't really give credence to your claim.
  • Applejinx 9y ago
    That's not even an additional octave over 20K. It amazes me when people behave as if there's a magical hard limit that's really, really precise and applies to all humans. I think you probably understand that people get locked into arguing for victory, and if you tell them you were tested with thus and so range (as a young person, which is plausible) they will simply call you a liar. My own experience is this: when I was a kid, I got an ear wax problem, and had it removed (hasn't recurred). It was a horrible painful nightmare with nasty tweezers and water squirters, and I was just a little kid… but afterwards, sound (especially very high frequency sound) was a revelation. Later, when I was a little older, the advent of digital audio (at first, in record albums) was a nightmare to me, because I couldn't understand how or why that stuff sounded SO BAD. And of course my early experience of CDs was pretty nightmarish: pod people music, with all emotion and humanity weirdly excised. That's what got me into audio: wanting to understand this, and then later, fix it. I did actually succeed: I can produce and mix and process digital audio that young me would not be horrified by. But especially if I had to meet that higher bar, I won't be able to do it at less than say 22 bit/80K, well engineered. If I get to use all my current tricks I could do it at 20 bit/80K: I can cheat word length easier than I can compensate for a nasty brickwall filter. 24/96K is widely prevalent and enough, given good converters. I'm not convinced 192K is at all necessary, but the more people crusade against it, the more contrarian I get ;) I've got a Beauty Pill album mastered in 24/192K and it sounds freaking phenomenal. Mind you, I have professional equipment designed to handle that.
• sideshowb 9y ago
  Minor niggle: we can see ultraviolet just a little because it makes various bits of our eyes glow. Perceptually this shows up to us as a little bit of fuzziness near a UV source. But no, we can't see it with anything like the resolution we see visible light.
• lightedman 9y ago
  I want as much resolution as possible, because when you do like I do and introduce pitch-shifting after the fact (re-tuning a Floyd Rose for every other song is not really conducive to a jam session) 24/96 just doesn't cut it at all, and you get bad artifacting further than a half step either way.
• tinix 9y ago
  I challenge y'all to take two copies of the same professionally mastered track, 16 vs 24 bit... and phase invert one, and put them on top of each other. What you can then hear is basically like an auditory diff. Do you not hear anything? Yeah... all that sound is what amounts to lost fidelity when you down-sample. You can't just argue on bit-rates alone... It's about having head room in the mix and room for more fidelity. Sure, if you mix shit badly, you can't hear the difference, but that's missing the point entirely. It's not nearly as bad as the time someone tried to tell me that Opus was an adequate audio file format for music, but still... frustrating. I know I don't have super-human hearing, and I can easily hear the difference. Just because everyone can't hear the difference, or can't tell that they can hear the difference, doesn't negate the fidelity loss of lower bitrates. Further, given a large enough speaker stack, it's not about what you can hear any more, it's about what you can feel. And never-mind the benefits of a 24bit DAW pipeline... Hello, low latency?