25 ms·

There is no point to distributing music in 24-bit/192kHz format.

• yzhou 15y ago
  The hearing of ears is a time-domain thing, not a frequency domain thing. It's the frequency response of all the frequency components added together. people might not be able to respond well to a single high frequency tone, but might respond well to a combination of tones.
  • xiphmont 15y ago
    No. It's both. The basilar membrane is a loosely tuned resonator. The hair cells placed on it fire beginning on the positive zero crossing. So, to a first approximation, the ear is in fact a filterbank. There is a time domain component in that the cochlear nucleus contains nerve cells that watch multiple hair cells at a time and correlate the firing in several different ways. Some attempt to discriminate pitch, some convolve and correlate in-phase firing energy, some look for tones to end, etc. This information is then forwarded on to the brain. However, getting back to your point, no hair cells will fire if the basilar membrane doesn't move, and it's tuned to a frequency range.
• Andys 15y ago
  This is a really convincing article that makes me want to set up a double blind test for myself with my own equipment. In my own tests I believed that I couldn't tell the difference between 16/44 and 24/96 on high quality loudspeakers, but I could with high quality headphones. The studies cited all seem to use loud speakers in testing. Also worth noting, the article states that obtaining 24/96 source material sometimes means you get better mastered material, which still sounds better after down-sampling back to 16/44.
  • hackermom 15y ago
    You weren't just believing things. The difference between 44khz and 96khz sample rate is very noticable even with mediocre audio equipment. It's an overstatement to refer to the situation as a "hi-fi case". 16/24 bits however makes no difference at all except on the size of the material.
• coopersloan 15y ago
  Huh, I think people truly advocating 192 as a distribution format will be few and far in between, a really good and cheaper sampling system can be put together at 96. Still, a lot of things in this article perplex me. Human hearing is limited to 20k because frequencies higher than that are perceived as painful? Dont agree with that one. 24 bit doesn't offer any advantages to sound quality? Sheesh. And the crux of the argument is intermodulation distortion increases when you try to represent more frequencies? Isn't that an argument for a faster power amp?
  • brazzy 15y ago
    "Don't agree with that one" and "Sheesh" are pretty weak counters to detailed, objective arguments based on extensive research and decades of test data.
    • coopersloan 15y ago
      Principle frequencies well above 20k as well as their sympathetic harmonics are pretty easily audible by me, try it. 24 bit is also extremely easy to hear. Arguably more important during the recording phase when headroom is valuable. Its just as easy to qualify everything with "placebo effect", as it is to be dismissive
      • tompagenet2 15y ago
        There will always be people with slightly wider sensory perception than others, but are you honestly saying that you can easily hear sounds "well above" 20kHz? Have you tested this in a double blind test? I know that sounds like a lot of effort, but it would be interesting. You would be somewhat unusual if you could hear, say, 22kHz.
        • nullc 15y ago
          Not somewhat unusual. A one of a kind specimen. If you test this and hear something, it's almost certainly because the ultrasonic signal is being distorted by your amplifier and speakers and you're hearing distortion products that are ending up at frequencies you can hear.
• cmer 15y ago
  There's a lot of scientific-sounded content in this, but unfortunately most of it couldn't be further from the truth. I'm an ex-audio engineer and studied digital and analog audio engineering; this has been debated to death over the last 15 years. Digitally recording a triangle is the best example of why 48kHz is very limiting. The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 5kHz and of many very high-pitch harmonics. Most of these harmonics are above 20kHz. The harmonics are what makes it sound like a triangle, not the frequencies below 20kHz. This is why the triangle is one of the hardest instruments to digitally record. It always sounds like crap. In theory, it's true that the human hear can't hear above ~18kHz, but it can hear the influence of the very high pitch harmonics on a lower frequency. EDIT: here's more data backing what I said http://www.cco.caltech.edu/~boyk/spectra/spectra.htm http://www.cco.caltech.edu/~boyk/spectra/spectra.htm EDIT 2: typos, frequency mistake
  • tompagenet2 15y ago
    Hey cmer, thanks for posting I don't think I understand quite what you're saying and wondered if you could explain more. You and the article both say that humans can't hear above about 20kHz. If there are higher frequencies that create a harmonic at a lower frequency (e.g. a 33kHz harmonic that produces a sound at 16.5kHz) then surely that lower harmonic (16.5kHz in this case) will be recorded by the original recording equipment assuming it is recording at a frequency at least twice that of the highest audible frequency (let's say that this would be 48kHz, although there might be other DAC-related reasons to go higher). I'm possibly being very daft here!
    • cmer 15y ago
      Let's make things super simple. Let's say you record 4 sine waves at a 192kHz sampling rate: 15kHz, 30kHz, 45kHz and 60kHz. All 4 frequencies will be captured and the 15kHz frequency will sound different to your hear because its harmonics. If you take this recording and master it for a CD (44.1kHz), you'll effectively get up to ~20kHz (since they're a low pass filter starting at around 16-18kHz). This means that only our first frequency will be captured: 15kHz. It will be exactly the same as if you only recorded 15kHz alone. The harmonics don't modify the fundamental frequency, they just trick the human hear. But when they're gone, they have no effect whatsoever. Hope this helps! EDIT: the frequency numbers I used are actually somewhat of a bad example. Harmonics are never exactly double, triple the fundamental. Those would be mostly inaudible. But you get the idea.
• Derbasti 15y ago
  He raises a lot of valid points. However... 192 kHz is clearly overkill for listening. Not so for further editing of the data. Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible. 44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. A bit more headroom is well needed there. That bit about intermodulation distortion is complete bogus. He talks about problems when resampling high-fs audio data. However, you would never do that. You would digitally process 192kHz all the way. Only your loudspeakers or ears would introduce a high-pass filter, and a rather bening (flat) one at that. There is certainly no aliasing going on there unless you resample (wrongly). Intermodulation distortion is not the fault of the sample rate. I mayored in hearing technology. Calling 192/24 worse than 44.1/16 is total BS. How useful it is is a different debate.
  • deleted 15y ago
    [deleted]
  • ferongr 15y ago
    >Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible. This [1] (widely accepted in the scientific audio community) study's conclusions disagree with your assertion. >44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. This is not the 1980s, hardware has progressed beyond that point. Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems. Also see [1] >That bit about intermodulation distortion is complete bogus. He talks about problems when resampling high-fs audio data. I did not notice that in the article. It talks about IMD in the context of the analog chain and the transducers following the DAC, and it's possible that high frequencies can increase it. [1] http://www.aes.org/e-lib/browse.cfm?elib=14195 http://www.aes.org/e-lib/browse.cfm?elib=14195
    • Derbasti 15y ago
      > Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems. True, but they do so using (long, high-quality) high-cut filters. And these filters are pretty sharp, as they have to close within, say, 18-22.1 kHz. You can design them as linear-phase FIR filters with oversampling and all the good stuff, but physics dictates that sharp filters introduce distortion. A sharp filter like that is audible.
• sliverstorm 15y ago
  So, presuming we take this example: http://people.xiph.org/~xiphmont/demo/jaggy2.png http://people.xiph.org/~xiphmont/demo/jaggy2.png The key to reproducing the original signal from the digital signal is a low-pass filter that rejects everything above the sampling rate, correct? That is to say, what I am getting at is while the original signal can be reproduced, it requires properly tuned, and probably reasonably high performance, hardware to remove the higher frequency components of that square wave. Can you count on consumer grade hardware to do this well?
  • mindslight 15y ago
    Your question is somewhat amusing. A standard CD player uses 1-bit DAC (it's either on or off) at a yet-higher frequency to achieve better linearity. Filtering is quite easy in the analog world.
  • nullc 15y ago
    Yes, thats basically it. They do this _exceptionally_ well in fact. Typically the technique used inside DAC is to digitally upsample the signal (by duplicating samples, often to a few MHz— also allowing them to use a low bit-depth DAC) then it applies a very sharp "perfect" digital filter to cut it right to the proper passband (half the sampling rate). The analog output then contains only a tiny amount of ultrasonic aliasing which is so far out that it's easily rolled off by simple induction in the output. This isn't just theory. Here is a wav file I made at a 1kHz sampling rate, where every other sample is -.25/.25: http://people.xiph.org/~greg/1khz-sampled.wav http://people.xiph.org/~greg/1khz-sampled.wav (so a 500Hz tone, the highest you can represent with 1kHz sampling). Feeding that file to a boring resampler (I used SSRC, but anything should give roughly the same result— a least when not quite so ridiculously close to nyquist, most will attenuate near-nyquist data extensively) and get this: http://people.xiph.org/~greg/1khz-sampled-to-48khz.wav http://people.xiph.org/~greg/1khz-sampled-to-48khz.wav Here are the two signals plotted against each other: http://people.xiph.org/~greg/1khz-to-48khz.png http://people.xiph.org/~greg/1khz-to-48khz.png As you can see— the 500Hz sinewave is reconstructed perfectly. (Of course, a 500Hz square wave would not be (you'd get a sinewave out) but this is because a 500Hz square wave contains energy far beyond the nyquist of 1kHz sampling). Here is a spectrograph of the same signal http://people.xiph.org/~greg/1khz-to-48khz-spec.png http://people.xiph.org/~greg/1khz-to-48khz-spec.png showing that the tone is indeed pure (the faint background noise is the dither the resampler applies when requantizing its high precision intermediate format back to 16 bits).
• wickedchicken 15y ago
  For an article containing a lot of "well, if you knew signal processing..." there are two fairly major oversights: 1) Any well-designed system is going to have headroom. Period. Just because 48kHz can capture the frequencies the human hear theoretically, it's always good to have a little wiggle room. This comes into play even more with interactive situations: humans are particularly sensitive to jitter. Having an "overkill" sample rate lets you seamlessly sync things easier without anyone noticing. 2) 192kHz comes with an additional benefit besides higher frequencies: it also means more granular timing for the start and stop of transients. More accurate reverb would be the obvious example. I don't know if the human ear can discern the difference between 0.03ms and 0.005ms but it's something I don't see mentioned often.
  • Joeboy 15y ago
    > I don't know if the human ear can discern the difference between 0.03ms and 0.005ms but it's something I don't see mentioned often That's the time it takes sound to travel 8mm. Do you think you could tell if an instrument was positioned differently by 8mm?
    • barrkel 15y ago
      The ears distinguish directional audio in part from timing differences in what hits each ear. http://en.wikipedia.org/wiki/Sound_localization http://en.wikipedia.org/wiki/Sound_localization cites http://web.archive.org/web/20100410235208/http://www.cs.ucc.ie/~ianp/CS2511/HAP.html http://web.archive.org/web/20100410235208/http://www.cs.ucc.... that suggests the brain is sensitive to timing differences between ears as low as 10 microseconds, or 0.01ms.
      • xiphmont 15y ago
        It's not the timing differences, it's the phase differences. The ear is exceptionally sensitive to phase differences between the ears below 1kHz. This information is captured exactly (to well beyond the naive precision of the sampling clock) for any frequency below Nyquist.
    • 15y ago
• hackermom 15y ago
  There is no point with going over 16 bits, but there is definitely a point with going over 44.1khz, as it allows you to actually reproduce waveforms more accurately than 44.1khz. Try reproducing f.e. a sinewave accurately over 4-5khz with a sample rate of just 44.1khz - it cannot be done, and at this point we haven't even taken into account the issue of varying slew-rate characteristics of the thousands or so different DAC output stages in use in personal audio equipment. 44.1khz gives too much aliasing distortion, but 192khz is quite the overkill. Ideally, digital audio could sit on 16 bits of depth sampled at 96khz.
  • cmer 15y ago
    Couldn't agree more with you! 192kHz is overkill as a "final" format. 16 bits is very limiting for music with lots of dynamics (ie: classical). Very quiet sounds sound quite bad at 16 bits, but since most pop music has about 6-12db of dynamic range, it doesn't make much of a difference. I always thought the sweet spot would be 96-24. But the truth is, the market wants smaller and portable digital files, not higher quality music. Anything MP3 encoded will sound significantly worse than a CD anyways.
    • nullc 15y ago
      16 bits is not "very limiting" for anything, unless you think your ears themselves are very limiting. Many things are mastered poorly— recording engineers crushing the dynamics in order to get the loudest possible signals— mostly a problem for pop music, but nothing is immune. It's been observed that the various 'higher-definition' recordings have less brutal mastering— no doubt owing to the different audience they are marketed to. But this isn't a property of 24-bit vs 16-bit distribution.
  • nullc 15y ago
    No. This really is not the case. The article _specifically_ addresses this misconception. The signal reproduced from your 44.1kHz sampled digital input is not a stair-step like some broken waveform editor might display: On output it goes through a matched reconstruction filter (which may, in fact, be digital and involve an oversampled DAC or it could be analog though those are harder to build without compromise). After the reconstruction filter the output is _EXACT_, assuming the input only contained energy below the nyquist (well, and was sufficiently far away from the reconstruction lowpass). So even a 5khz sine wave is reproduced perfectly with 44.1kHz sampling.
• yzhou 15y ago
  A person can not hear a 22kHz tone doesn't mean he can not hear a sound that contains 22kHz components. For example, a square wave contains lots of high frequency harmonics, the more higher frequency harmonics it have, the "squarer" the square wave gets. An ideal square wave forms ideal "0" "1" states. A person's ear might not be able to hear a 22Khz sine wave tone, but he might be able to sense the steepness of "0" "1" state.
  • schiffern 15y ago
    Except that human ears don't directly perceive the waveform, only the frequency decomposition.
  • nullc 15y ago
    First, if you can "hear the steepness" you really can hear higher frequencies, but I assume you meant "maybe you can hear higher frequencies but not higher frequency _tones_"... People have suggested this. It's been tested in rigorous double blind tests— involving both real music signals as well as special test tones (Linked from the article). The tests were unable to show that people could hear the ultrasonics. Moreover, there isn't any physiological basis to expect people to be able to. You can't expect a stronger result than that. Common 48KHz audio already goes a bit beyond what adults are known to be able to hear, so you've already got some headroom for "but what if a few people hear better than anyone the researchers have been able to find!".
  • lwat 15y ago
    There's been countless A B X tests that showed conclusively that the vast majority of people cannot tell the difference between samples with and without the 22kHz+ frequencies.
• nileshtrivedi 15y ago
  What I would love to have is: independent instrument/vocals tracks along with a default recommended "mix". The default mix would be used for normal playback and independent tracks would be great for custom mix / karaoke etc. Is this too unrealistic to expect? Has something like this been tried before?
  • xiphmont 15y ago
    I'd love that too... damn, should have put it in the article... Closest I've found is to take the .mogg files out of the Guitar Hero games and use those to make new mixes. :-)
  • moondowner 15y ago
    Some songs when released as a single have Acapella and Instrumental versions of them as well. There are also compilations with only acapellas and compilations with only instrumental versions of the songs. And when you have them, just use something in the like of Ableton Live and that will be it. I think that that's what you mean right? It will be a great idea to have tracks released as several `layers` so that the user can choose which of them to play and which not, for example the bass/beats layer, layer with melodies, layer with the percussions, layer with the vocals of course, but that sounds like semi-studio production.
  • k-mcgrady 15y ago
    There are two reasons I don't think this will happen: 1. People would use the tracks to create custom remixes which they would then distribute. What happens when a remix becomes more popular than the original track? Artists generally have to pay other artists to remix their songs (usually via royalties). 2. Creativity. When an artist creates something they want you to hear it the way it was intended. Allowing you to remix it however you like takes away a lot of the creative control from the artist.
    • moondowner 15y ago
      Regarding remixing. Artists usually don't "pay" each other, but return the favor - if it's the right term to say. E.g. artist A remixes a song of artist B and artist B in turn does the same for artist A. Or if they are all on the same record label artist A does a remix for artist B and later B makes a collaboration with A. I've noticed this in electronica/edm music artists at least. And another important remark: some artists are flattered when someone asks them to make a remix for their song. (Imagine you're an artist and your idol asks you to make a remix of his song.)
• leouznw 15y ago
  I know a bit of sound engineering, waves and so.. I totally agree with the title and the first 60 lines of article, and I add my POV: 1. Most of the people doesn't care, 2. What apple did is just about marketing, 3. Most of the people who says that care is pretending, 4. Zeppelin still rock the shit in a poor quality mono mp3 recorded by a drunk guy in the audience of a concert in 73.
  • kokey 15y ago
    I do care, but I'm not the average user. Apple has always catered well for those in audio and video, up to professional levels. These are markets that retain Apple users, even when Steve Jobs was between Apples. It seems like Apple is only requesting masters to come in a higher resolution, not that consumers will generally end up with these. I think this is entirely fair since before you want to modify something (e.g. to remaster it for iTunes) you want to start off at a good quality high resolution. That said, if Apple also allows high quality recordings to be sold, it will be useful. For example of their acapellas, instrumental tracks or samples, it would be convenient for others who want to want to remix it, and iTunes would be a platform for this trade. Also for tracks DJs play. Most compression throw away a lot of the bass which people can't hear, but this is bass you can feel rumbling through your guts on a big sound system and is part of the experience. For the rest, they were happy with low rate AAC files on the early iPods, they are happy with the sound coming from their crappy little iPod dock, for them it won't make a difference as long as it's a chart music track from a memorable and impressionable time of their life.
• rbreve 15y ago
  Unless you are a dj or producer and would like to sample or time stretch the tracks. That's why Beaport offers a wav download option, that many djs/producers prefer.
• noonespecial 15y ago
  I was under the impression that two inaudible high frequency tones could interfere with each other to create an audible interference pattern. (I think known as a "beat frequency"). If this is the case, then all of the arguments in the world about the maximum audible single frequency are irrelevant. Imagine music composed entirely of these beat frequencies and performed with a pair of oscillators between 25kHz and 35kHz. Without higher resolution encoding, it would be audible IRL but the recording would be silence.
  • zb 15y ago
    If the beat frequency is audible, it will be on the recording. Obviously.
    • noonespecial 15y ago
      That would suppose that the recording device precisely matched the orientation of the listener, and the recording was not created digitally in (multi-track fashion for example). There would have to be air space in order for the interference pattern to set up in. So you'd be right if your mics were head spaced and in the venue. But you'd still have secondary data, with the original lost.
      • ef4 15y ago
        > But you'd still have secondary data, with the original lost. By that standard, the original is always lost unless you have a completely holographic recording. 192kHz doesn't help with that problem at all.
  • nullc 15y ago
    Alas, they don't— you can easily demonstrate this for yourself. Startup an audio editor and generate tones at 25k and 28k (make sure you can't hear them— otherwise you have severe distortion screwing up your test) then play both at once. You will not hear a 3kHz tone. The tone you get from an acoustic beat is not a real tone— it's a perceptual quark that requires you to be able to hear the tones in the first place.
• agentgt 15y ago
  I know this is slightly tangential but are hi-end DACs really worth it? I have always been amazed how much audiophile DACs cost ($300-1000). The reality is I listen to 320kbps music that was most likely recorded at 44100. DAC technology is not exactly new. So why the price? Another tangent: To me it seems audio engineering should fix the "woofer". That is it seems subwoofers have terrible distortion.
  • TylerE 15y ago
    A low-end dedicated DAC is likely to be a substantial upgrade over a built-in soundcard (I'm assuming we're talking PC sound here). A PC case is a pretty noisy place, electrically - I know one one work PC I had once you could actually here the mouse move, if you had heaphones on and cranked the volume with nothing playing - horizontal and vertical movement had different frequencies. The move from a low end ($150-300) DAC to one much more expensive will be considerably less drastic, and likely won't matter until you've dropped at least $5k in to the rest of your system. That said, you may already own a DAC without realising it...as long as you're taking the singal out _digitally_ (e.g. SP-DIF or digital coax) to an external receiver, you're already in a pretty decent place.
    • agentgt 15y ago
      Oh yes I agree an off board DAC is better. I own the Fiio E7 which I highly recommend for laptops and only costs $80.00. In fact I run a 50 foot USB cable from my laptop to my DAC and the improvement is much better than running a 50 foot 3.5mm TRS. But the high end ones that are 24 bit 192khz that cost $1k (Cambridge Soundworks DAC magic comes to mind) I have to seriously doubt I'm going to hear it. I really only hear the DAC difference (compared to my laptop and FIIO) when I use headphones.
• sjwright 15y ago
  I must say I get rather irritated when people spend time worrying about dubious 'tweaker' methods to improve their audio, when the most under-performing component of most people's sound equipment also has the lowest-hanging fruit: The room itself. When people ask me where they should spend money to improve the quality of their hi-fi or home theater system, in nearly every case my response will be something like "get a thicker rug" or "put something on this wall to absorb sound reflections, even if it's just a bookshelf." Beyond that, I'd tend to say something like "stop being so paranoid about what you think you can't hear, and enjoy the damn music."
  • xiphmont 15y ago
    Amen. Someone got the point of the article.
  • agentgt 15y ago
    I agree the damn room matters so much more. Also the distortion of most speakers is already the bottle neck in most people's systems.
  • Anechoic 15y ago
    Agree one-hundred percent about the room (although the prescription isn't always as simple as "get a thicker rug" etc). The other issue regarding high-frequency sound reproduction is that in most cases, the loudspeaker won't be outputting much beyond 22-25 kHz (assuming very good quality loudspeakers, cheap consumer grade units might struggle to hit a -6 dB point at 18 kHz) and even for the speakers that have usable output at that range, the directivity at those frequencies will be so narrow that your head will have to be locked in the perfect "sweet spot" to hear anything.
    • marcusf 15y ago
      I might sound/be stupid for asking, but what's the actual physical response from something at 22 kHz+? I have a hard time picking up a pure sine > 17 kHz. I doubt I'd get any aural response from anything at 22 kHz, so what's the deal?
• zzygan 15y ago
  This is a good article, however the guy who has been pushing this for years and years now, is a man called Dan Lavry. In fact he wrote a very good, rigorous explanation a few years back,in very readable and well written form. http://www.lavryengineering.com/documents/Sampling_Theory.pdf http://www.lavryengineering.com/documents/Sampling_Theory.pd...
• jensnockert 15y ago
  I just want floating point, then this silly loudness war would end (to some extent, since you can make the mix almost infinitely loud).
  • FrankBooth 15y ago
    It would make no difference at all.
  • jasomill 15y ago
    So you want your music clipped or limited by the DAC? Or do you envision amps and speakers capable of generating shockwaves powerful enough to level entire cities? Hotblack Desiato, is that you?
• blackhole 15y ago
  You always record stuff at 24-bit/192 kHz for many reasons usually involving minimizing analog artifacts and to give you a lot of information to work with. You use 32-bit float wavs to transport stuff around so you don't have to worry about normalizing levels and clipping. Lossless formats drastically improve the quality of transients by an enormous degree. But every single objection to this is either ignoring the points of the article, or talking about the benefits of recording at high fidelity, when this entire article is pointing out that once you have _finished a mix_, there is no reason to distribute things in 24-bit/192kHz. Most speakers can't even play about 20kHz anyway, which makes the entire point moot. I don't care if you have a bajillion kHz, the speakers can't play about 20 kHz, so your screwed.
  • mfarris 15y ago
    You're getting two entirely different things mixed up. 192 kHz is the sample rate. 192,000 slices per second. It does not refer to the audible sound spectrum. 20 kHz in speakers refers to the cycles per second of the audible waveform. Normal human hearing rage is 20 hz - 20 kHz. For most people, it's less than that. A speaker can certainly play back music sampled 192,000 times per second. Most of them can't play tones that are higher pitched than 20 kHz, which is fine because mostly only dogs can hear up there anyway.
    • blackhole 15y ago
      I am not getting these things mixed up, because the sample rate is related to the maximum frequency that can be stored, and lo and behold, look at all these people claiming that those higher frequencies matter. 44.1 kHz sample rate can only encode tones up to about 22 kHz, whereas 192 can encode frequencies of up to 81 kHz, and those people up there are arguing that these higher frequencies are exactly why 192 kHz is superior. Now, if you want to say that sampling a tone at 44100 times per second somehow won't sound as good than 192000 times per second, I'm not saying that isn't possible, but I don't really take that claim seriously at all. The fact is, simply distributing music in lossless format carries the vast majority of audible improvements. Arguing over whether or not its 24-bit or 16-bit or making a chunk of sound last 5.2 microseconds instead of 22.67 seems incredibly stupid to me, because you're better off simply improving the mix itself then fiddling over such microscopic differences. These things only become relevant if your mix and performance and recording equipment (or synths) are absurdly close to perfection. This becomes even LESS relevant in an age of indie-musicians.
• untangle 15y ago
  This article is one of the most lucid and accurate that I have read on this topic. However, one thing that's missing here (and in nearly all other similar pieces) is a full discussion of the prerequisites of the sampling theorem. For example, the signal must be bandwidth-limited (and no finite-time signal can be). But this is a minor concern, as there are many elements in the analog domain of the recording and playback chains that serve as low-pass filters - starting with the mics. So bandwidth-limiting is effectively achieved. For a similar reason, the discussion of the "harmful" effect of high-frequencies to playback electronics and loudspeakers to be a bit overdone IMO. Peruse the excellent lab results of modern audio gear on Stereophile's web site. You'll find that bandwidths exceeding 30kHz are rare. One last thing. When doing subjective "testing," keep in mind that what some folks are hearing may be limitations of their gear. For example, most DACs derive their clocks for higher sampling rates (88/96/176/192) by clock-multiplier circuits. IOW, 44kHz and 48kHz are the only ones clocked directly by a crystal. These multiplier circuits are often noisy, contributing to jitter. The audible effect of this jitter is hard to predict. Bob PS As an avid audiophile, I find the clash of subjectivists and objectivists on this normally-buttoned-down forum to be a bit of a trip.
• polshaw 15y ago
  I have to say that was probably the most comprehensive dealings with the issue of sample-rates I've ever come across. I'm not going to make the mistake others have of claiming falsehoods (all of which i've read so far have been debunked to my satisfaction by the HN users-- i'm impressed, guys). As pointed out, mastering has vastly greater effect on the audio quality (and is often pretty poor[1]), and is the reason vinyl records often can sound better than their digital counterpart, despite being an inferior technology[2]. The DAC used also has a massive effect on the sound once you get into decent quality equipment. Like the author, i'd also love to see some expansion of mixed-for-surround music. [1] a lot because of loudness wars, as pointed out in the post, but also just due to a lack of time/care/love(/demand?). [2] http://www.hydrogenaudio.org/forums/index.php?showtopic=61758 http://www.hydrogenaudio.org/forums/index.php?showtopic=6175... This thread explores the bit-depth of vinyl records, beginning with a claim of a maximum 11-bit resolution-- limited by the width of a PVC molecule the record is made from.
• hackinthebochs 15y ago
  One thing I don't see addressed is the experience of feeling frequencies that can't directly be heard. There was a study done with a particular piece of classical music, with and without a particular inaudible component to it. The presence of the inaudible component drastically changed the listeners perception of the music. They described it as more dark or creepy (perhaps not the actual words used, but it matches the sentiment). The point is that there may be value in reproducing frequencies that we can't "hear", as inaudible notes can alter the experience of the music. *not the study I was referring to but its along the same lines: http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=5291285 http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=5291...
  • hackinthebochs 15y ago
    Mmm loves me some inappropriate downvotes.
    • lwat 15y ago
      Probably because you couldn't come up with the actual study you mention.
  • notahaxor 15y ago
    The author completely ignores infrasonics and writes under the incorrect assumption that our only perception of wave pressure comes from our eardrums. I've never been able to enjoy listening to my favorite classical music on headphones or even smaller speakers, and it's largely because of the effect you describe. At this point I'm resigned to preserving my treasured (and cumbersome) vinyl collections. Maybe if Apple comes up with some snazzy marketing term (e.g. "Retina") for 24/192 or even 24/92, and starts distributing it on iTunes, things might start to change.
    • mdarens 15y ago
      You don't need a higher sample rate to capture or play back infrasonic pressure waves, but most recordings are mastered to remove DC offset and rumble <20Hz, as reproducing those components requires specialized equipment, such as a rotary subwoofer.
• Anechoic 15y ago
  For those of you who are interested in just how much of a golden ear you truly are: download Harmon's "How to Listen" software for Windows or Mac OS X http://harmanhowtolisten.blogspot.com/ http://harmanhowtolisten.blogspot.com/ (scroll down). Harmon requires its trained listeners to pass tests based on this software before participating in juries to evaluate Harmon products. It doesn't directly address the sample rate/bit depth issues discussed in the linked article, but it does address a lot of the issues brought up in the HN discussion, so you can have a chance to see how much those characteristics really matter. You may be surprised.
• jlft 15y ago
  In normal listening conditions and for most people the difference between 16/44 and 24/192 is inaudible. Given a 5 minute song, if I have the choice to download a 11MB file (320kpbs MP3) or a 330MB file (24/192) I would of course choose the 11MB file. The sound quality is perfectly acceptable and the file size much more convenient to manage (storage, backups, etc.). In terms of the convenience of managing the file size and sound quality I think 320kbps MP3 is the best compromise. Here's a file size comparision of a 5 minute stereo song: MP3 128kbps > 5 MB MP3 320kbps > 11 MB Uncompressed 16/44 > 50 MB Uncompressed 24/192 > 330 MB When talking about sound quality there is a much more relevant issue: the amplitude compression (distortion) abuse used by mastering engineers and producers that totally destroys the dynamic and life of the sound. That is a real issue. When buying a song there should be two versions to choose from: A) "Loud", dynamically destroyed / distorted version. B) Normal, dynamic, non-distorted version. Today only version A is available to buy.
  • rdtsc 15y ago
    But then for every 10 people like you there is 1 person who is willing to pay 20x as much so they can get a "higher fidelity" product. For a producer and manufacturer the rational approach would be to cater to that craziness and extract as much money from it as possible. In other words if you are selling HDMI cables, spend $2/cable to make it, then sell most for $5 and then re-brand some and sell for $500. If only takes 1 out of 100 people to buying that to make the same profit. You know these people are obsessed and irrational so you cater to that. And that's basically how we end up with ridiculously overpriced Monster cables and recordings distributed to customers @ 192kHz.
    • jlft 15y ago
      Agreed, that market exists. My point is, why discuss the subtle difference between 16/44 vs 24/192 when there are far more audible and damaging practices going on in the music industry. For example, aggressive compression and brick limiting which adds distortion to achieve maximum loudness ('loudness wars').
• mistercow 15y ago
  > Can you see the LED flash when you press a button? No? Not even the tiniest amount? I used to be able to see it when I was a kid (it looked very faintly red), but I just tried it and couldn't see it at all. That's actually a little bit disturbing.
  • xiphmont 15y ago
    I would guess that's because some earlier remotes used a higher frequency IR emitter that was in fact touching into the red. These days, the various IR communication protocols have been standardized and virtually all use 920nm, 940nm or 980nm emitters, all of which will be invisible. I mentioned the Apple IR remote specifically because it's a remote most people reading TFA will have, and it's known to be a 980nm emitter.
  • sjwright 15y ago
    Almost certainly because it was a different remote control when you were a kid. Some remote controls emit a lot closer to visible light than others.
• jwatte 15y ago
  The sampling theorem is for static signals and perfect filters. Turns out, music isn't static. Once you have transients in the program, you need higher bandwidth or you will end up with phasing effects (time domain aliasing.) This is plain from the math! Filters are also not perfect (but good oversampling filters are not the weakest link) Further, even perfectly dithered 16 bit data can't go 20 dB below the quantization floor, unless you give up on frequency response on the high end. Again, this is plain math. With a calibrated 105 dB low-distortion sound system, in a quiet room, I can hear imperfections from 16 bit, 44 kHz material, especially in soft flutes and triangle type percussion. Of course, D class amplifiers, and MP3 encoding, do worse things to the signal, so let's start there. But 20 bit, 96 kHz (or at least 64 kHz) are scientifically defensible, when analyzing the math and the physics involved. No snake oil needed!
• JangoSteve 15y ago
  Even without debating the science and signal processing arguments raised... In any test where a listener can tell two choices apart via any means apart from listening, the results will usually be what the listener expected in advance; this is called confirmation bias and it's similar to the placebo effect. It means people 'hear' differences because of subconscious cues and preferences that have nothing to do with the audio, like preferring a more expensive (or more attractive) amplifier over a cheaper option. The human brain is designed to notice patterns and differences, even where none exist. This tendency can't just be turned off when a person is asked to make objective decisions; it's completely subconscious. Nor can a bias be defeated by mere skepticism. Controlled experimentation shows that awareness of confirmation bias actually increases rather than decreases the effect! Doesn't that completely negate his conclusion, that there is no point to distributing 24/192 music? If people want to pay for 24/192, and even he just admitted that they will legitimately enjoy it more, how can you conclude there is no point? Life is short. I want to enjoy things. Whether or not my enjoyment can be quantified or scientifically defended, I really don't give a shit. But that's okay, if you don't want to sell me 24/192 music, Amazon will. Between this and DRM-free content, it's no wonder I buy all my music from Amazon these days.
  • cwp 15y ago
    Well, if we accept that argument, then just about any means of signalling "this sounds better!" will work. How about we choose something that doesn't waste bandwidth?
    • JangoSteve 15y ago
      That's true. It's kind of the Monster Cable model. BTW, I'm not saying that marketing and whatnot should deceive less technical consumers and trick them into spending more money than they should (which is basically what Moster Cable does). But when you explain to technical people why something like 24/192 isn't better (other people in this thread have pointed out, this isn't totally accurate in the first place), and they understand what you're saying but still prefer it, by all means, let them buy it.
  • jpdoctor 15y ago
• blahblahblah 15y ago
  I mostly agree with the article in the context of distribution of a final mix. However, the article ignores one glaringly obvious reason to distribute in 24/192 format: to allow the listener to be a participant in the creative process, enabling better results for amateur musician listeners who want to sample or remix the audio or for DJs to get better results when altering the tempo for beat matching one track with another, etc. Of course, if you're going to do that, you might as well distribute in a multi-track format instead to maximize flexibility for the end user (Want to sing karaoke? Just turn off the lead vocal track for playback).
  • nullc 15y ago
    Yea, and and if the bandwidth/storage is at all an issue 6x size bloat from 24/192 pays for 6 separated tracks. (Actually more, because multitrack is more losslessly compressible while 24/192 is less). If you're already providing multitrack then 24 bit audio would make sense... otherwise, meh.
• citizenspaced 15y ago
  I don't understand why anyone gets down on 24-bit consumer audio. Specifically because CD-quality 16/44 audio has midrange distortion present during complex passages that is completely eliminated and non-present in 24/96 sources. Listen to "Us and Them" off a 16/44 CD version of the Pink Floyd album Dark Side of the Moon. When it kicks into the chorus, it becomes totally distorted and everything in the midrange bleeds into each other. It's a mess. Then, try listening to the 24/96 Immersion box set copy or a vinyl-sourced 24/96 rip and you'll find it's gone. When the song gets complex and loud, everything remains totally clear, each instrument stands on it's own, it doesn't become an awful distorted jumble. You could argue that it's just the quality of the master that makes the difference; but if you take a copy of the original transcoded to 16/44 and compare it again with the 24/96 copy you can hear the same effect. Why would anyone argue against high-resolution audio anyway? Sure, most everyone will probably just continue downloading 16/44 MP3s, but at least give us the option to have 24bit FLACs of the stuff we really like. Please and thank you.
  • dedward 15y ago
    You could argue it's the quality of the master, and the mastering process, and you'd be right. That's a no-brainer. "but if you take a copy of the original transcoded to 16/44 and compare it again with the 24/96 copy you can hear the same effect." I could believe that, but do you mean to do the transcoding yourself? IN this case you become the engineer, and the tools you use and all that become vital as well. Having heard stunningly awesome CD's of DSOTM on a homebuild heathkit amp and some old speakers and not believing my ears when I saw what the setup was, I'm skeptical... can't help it.
• jaekwon 15y ago
  The article AFAIK states little about distortions introduced in remixes & samples. I would expect certain high frequency samples, when mixed together to overlap in time, would introduce moire artifacts (beats).
  • nullc 15y ago
    Not unless you pass them through a non-linear filter like a distortion effect. (yes, the ear-brain system is non-linear too— but apparently it filters out the ultrasonics before they do anything measurable in this regard)
• aiurtourist 15y ago
  Science be damned! Onwards with subjectivity! • 24-bit audio is magical. When I recorded myself playing guitar in 24-bit and played it back through my amp, it sounded like I was still playing. 16-bit sounded like a CD. • With MP3s, 192 kbps is a huge step up from 128 kbps. 192 doesn't exhibit any of the "swooshiness" heard in the upper range of 128 kbps MP3s for regular rock/pop/hiphop music.
  • inghoff 15y ago
    ahem 192 kHz. We're talking sampling frequency, not bitrate.
  • danbmil99 15y ago
    I believe you are confusing 192kbit compression with 192 khz sampling rate. Not at all the same thing.
    • aiurtourist 15y ago
      Whoops! Guess I'll read more carefully, then.
  • thesunny 15y ago
    The article states that greater than 16 bit has value in recording, just not in playback. If you took your 24 bit recording and translated the best 16 bits of it than output it through your 24 bit DAC then they are saying you won't here a difference. I say output through your 24 bit DAC so you aren't simply hearing a better DAC.
• neuroelectronic 15y ago
  The article states the point to using this format; keeping master-quality originals.
• ChuckMcM 15y ago
  TL;DR - long and detailed information about why if you got music in 24/192 format you couldn't tell the difference between it and 16/48 music. I chuckled because this is so true, and yet tell that to the people who buy oxygen free copper 'monster' cables for their speakers, being careful to align the arrows with the direction of the music from the amplifier to the speaker. People, even otherwise reasonable people, will swear up and down they can hear the difference.
• WalterBright 15y ago
  My hearing has declined over the years, to the point where audiophile gear is a complete waste of money. For example, I can no longer hear the difference between a cassette tape and an LP. I still listen to and enjoy music all day, but no longer worry at all about the sonic quality of it. My advice to you younger guys is to keep the windows rolled up while driving. I have no other explanation why my left ear is much worse than my right.
• naughtysriram 15y ago
  I think 192kHz is the sampling rate used by the A2D converter vice verca. It is not the actual frequency of the sound (data).
• tintin 15y ago
  I think this only applies to headphones. People also 'hear' sound with there body (skin). Maybe you could call it experiencing sound. And then there are resonating sounds that cannot be heard but help to create other sounds. But maybe this won't apply to a recording because your will record the result and not the tones that make the result. This is a great article but I'm still not convinced people cannot have a sensation of sound out of there hearing range.
  • tintin 15y ago
    Never mind. I read 192kbps instead of 192kHz. 24 bit might have some advantages but 192kHz not.
• diminish 15y ago
  Would someone explain should I use 44.1 or 48Khz?
• joccam 15y ago
  Sometimes less is more. The debate goes on. Why not just let the music play? And by that I mean high resolution music. All you need is one person who can hear high frequencies, and all the technical mumble-jumble becomes hogwash. People actually _believe_ the 20KHz argument that anything above is inaudible. That's hogwash. I know because I can hear (or sense) higher frequencies, and I do not have the absolute best ears I've ever "met." For example, last week I attended a A/V equipment event with very high-end equipment. It was packed --- over 600 people for one evening. 6 rooms of equipment. I'm sure all six served the same fare according to the 20-20KHz argument of this piece, yet they all sounded quite (or even extremely) different. The 20 KHz argument is a myth. For people who can't hear the difference, no problem. But please do refrain from ruining or hobbling music for the rest of us... who can hear a wider frequency range. Yes, some people are color blind. Does that mean the rest of us shouldn't use color? I hope not. Music is an important wholesome and potentially emotional part of human life. Please do not cap it with "false optimizations". 24-bit/192 KHz is not inferior to CD quality sound. If you don't believe me, try a Linn system sourced on a Klimax DS with some high bitrate Linn classical music (or the Beatles Masters USB release!). If you can't hear the difference compared to low bit-rate (including CD quality) material, I assure you someone can. The low bit-rate will sound flat, hollow, less lively, or/and more coarse. Any number of problems exhibit at inadequate bit levels. Vinyl is analogue quality (no discrete digital distortion). CD quality is a large step down from vinyl. A/V is just trying to get vinyl like quality from digital. We don't need nay-sayers impeding progress. If you can't hear the difference, please let someone who can hear make the informed decisions. Thanks.
  • _delirium 15y ago
    It's not a myth, but a fact established in laboratory studies. Your anecdotal claims to hear frequencies that scientific evidence suggests you cannot hear doesn't overturn science. I'd be convinced if you correctly identified which speakers were reproducing 21 kHz frequencies in a double-blind test, though.
    • joccam 15y ago
      Isn't science verified through (wait for it...) experimentation? So how does my hearing not invalidate your science? That's the problem with the theoretical science. When it's false, it's false. Come up with a new hypothesis; this one's false as it pertains to human hearing. There's information theory, and then there's auditory reality. Reality confounds the theory as applied to hearing. I don't know where the fault lies, and I don't really care. But it's really annoying and frustrating having people nix progress out of idealistic theory, "laboratory" studies, and ignorance. The experiments (my experiences and numerous others) don't lie. Double-blind is great, but I can already tell the differences between all six rooms of equipment from last week. One of the rooms was so extreme, I wanted to run out of the room due to discomfort (but I was polite and stayed all 30 minutes). In other words, double-blind was unnecessary. Someone whose ears I respect a great deal, loved that room. Even golden ears don't all hear the same. But I don't need double-blind to confirm trivial experience. The proof is already in the listening.
• deleted 15y ago
  [deleted]
• rbanffy 15y ago
  Minor nitpick > The FLAC file is also smaller than the WAV, and so a random corruption would be less likely because there's less data that could be affected. At the same time, if you flip a bit on a WAV file, you may hear a "pop" sound. On a FLAC file, the whole encoding block may be inaudible (or worse).
• tammer 15y ago
  I find mp3 and aac compression artifacts to be monstrously irritating. I have no idea how the majority of the world seemingly can ignore them. Further, I can hear a difference between 44.1kHz and 96kHz. Whether you can hear that difference is up to you. (The word-length is a red herring - there's no new information contained in a 24-bit recording vs 16.) IMO anything less than flac and you're missing something. Higher sampling frequencies do add to the sound, but in a way that is almost invisible to the untrained ear. Perhaps these should be distributed at a premium the way SACDs and similar "audiophile" formats were in the past?
• neilalbrock 15y ago
  A few years ago I became really interested in recording music. I had been writing a little with a friend, using whatever crap equipment we could afford, the results weren't amazing but we were having fun and staying focussed on the music itself. Then we starting recording other people. I became obsessed with gear, software and all the associated toys that go with any technical pursuit. I'm a programmer, so it's easy to understand how that happens but I totally lost sight of the music, spent way too much money and equipment that was nowhere near being required and generally lost the plot. I was tracking everything 24-bit/96kHz and bemoaning the loss of quality when I mixed down for CD. Anyway, the TL;DR version of what followed was that we recorded quite a bit, lost interest in making our own music and then the whole adventure came to an end. Now my gear is leaving via eBay and I'm finding my way back to just playing guitar and trying to write good music. 24-bit/192kHz - pointless. Give me a small venue and a guy with an acoustic guitar any day.
• bryanlarsen 15y ago
  One of the strongest things that makes this article credible is that in it we have the author of Ogg Vorbis recommending that we stop using Ogg Vorbis (and all other lossy compression formats).
• tcarnell 15y ago
  Has anyone had a look at their hi-fi amp recently? If probably probably doesn't handle much more than 80 kHz and your speakers probably dont respond to anything over 20 kHz. So yes, 192 kHz is pointless UNLESS you intend using it for studio quality editing/mixing - and I'm sure Steve Jobs would not have encouraged this!
• thewisedude 15y ago
  I am told that a similar argument can be made between TV's that display at 120 Hz as opposed to 240 Hz. i.e there is no discernible difference!
• jdc0589 15y ago
  There is no harm in releasing higher quality uncompressed or loss-less tracks. At the worst they will bring in some new customers, such as myself, that currently will not buy music online. Why would I pay $10 for an album as a highly compressed download when I can pay the same price for the CD and rip it to FLAC myself? I realize I am in the minority here, but as CDs phase out even more, there has to be some other way for consumers to obtain high quality versions of tracks. Footnote, you don't have to have a >$10,000 setup to benefit from higher quality tracks (compared to the downloads that sometimes have 'questionable' quality). I have two systems, a full range stereo (front left and right) setup for nearfield listening at my desk thats +/- 1DB from 50hz-20khz. The other is a stereo setup in my media room; 2 way quarter wave transmission line, +/-3DB 40hz-20khz. The point is, there are a lot of people with less than $1200 in audio gear that still want lossless tracks made available. Who cares if the human ear can't discern much of the extra information, we still want it.
• yu 15y ago
  From Footnote 1: [...] If we were to use the full dynamic range of 24bit and a listener had the equipment to reproduce it all, there is a fair chance, depending on age and general health, that the listener would die instantly. The most fit would probably just go into coma for a few weeks and wake up totally deaf.