Slashdot Mirror


Why Distributing Music As 24-bit/192kHz Downloads Is Pointless

An anonymous reader writes "A recent post at Xiph.org provides a long and incredibly detailed explanation of why 24-bit/192kHz music downloads — touted as being of 'uncompromised studio quality' — don't make any sense. The post walks us through some of the basics of ear anatomy, sampling rates, and listening tests, finally concluding that lossless formats and a decent pair of headphones will do a lot more for your audio enjoyment than 24/192 recordings. 'Why push back against 24/192? Because it's a solution to a problem that doesn't exist, a business model based on willful ignorance and scamming people. The more that pseudoscience goes unchecked in the world at large, the harder it is for truth to overcome truthiness... even if this is a small and relatively insignificant example.'"

126 of 841 comments (clear)

  1. Can we stop using the word "truthiness," please? by Anonymous Coward · · Score: 5, Interesting

    I know, Stephen Colbert is Reddit's hero and they're starting to infiltrate this site as well, but seriously. Call them lies. That's what they are, that's what they -deserve- to be called. Are people really that passive-aggressive and afraid of expressing themselves that they won't call someone who lies a liar any more?

  2. Re:The article writer is a deaf idiot by Microlith · · Score: 2

    Does it matter when the dynamic range is shot to hell?

  3. yeah, just use monster cables. by Anonymous Coward · · Score: 5, Funny

    lossless formats and a decent pair of headphones and a set of really expensive MONSTER CABLES will do a lot more for your audio enjoyment than 24/192 recordings.
      There, ftfy.

    1. Re:yeah, just use monster cables. by mudshark · · Score: 2

      Warm. Sparkling. Punchy. Silken. Pristine. Thumping. Brilliant. Dynamic. Crystalline.

      And the best modifier of them all: Audiophile-quality.

      How's that?

      --
      In other news, astrophysicists have announced that they now know what all that dark matter is: it's stupidity.
  4. Pro recording by koan · · Score: 2, Insightful

    I record my performances at 96 kHz sample rate, I have to say that the music sounds much better at 96 kHz than 48 kHz I think (feel?) because the higher sample rate gives audio effects like reverb a lush, deeper sound.
    The more sample units per second give the effects more to work with, in addition, even though you can't hear above and below certain frequencies recording those inaudible frequencies has an effect on the final product.

    You may be able to find some scientific proof of this but for me it's an ear thing, higher sample rates sound better.

    --
    "If any question why we died, Tell them because our fathers lied."
    1. Re:Pro recording by Anonymous Coward · · Score: 2, Informative

      I recently remixed a classic recording for sony records. The files where rolled off of tape at 24bit/96k. 48k I can understand but 96k is pointless. WAAAAAAY beyond the range of human hearing. In the old days, things like cymbals and brass could really stick out because the encoders and decoders where just not where they are today.

      Anyone that tells you they can hear the difference between 48k and 96k is dreaming. Its the quality of the recording that counts more than anything these days.

    2. Re:Pro recording by smi.james.th · · Score: 3, Insightful

      44.1kHz will be able to capture the basic information of the signal, as the human ear can hear to 20kHz in some cases, and Nyquist's theorem says that to recover the information you need to sample at least double the highest frequency. Oversampling (i.e. 192kHz) allows much more room to develop a good anti-aliasing filter. It may be that the reverb is phase-shifted somewhat with standard AA-filters, but ones designed for the higher sampling rate can have more linear phase. Also, higher sampling rates allow for better reconstruction of the actual wave form, if you're interested in music rather than just information. So yes, sampling a telephone call at 192kHz would be stupid, but if you're an audiophile, doing it for music is quite reasonable.

      --
      One thing I know, and that is that I am ignorant...
    3. Re:Pro recording by king+neckbeard · · Score: 4, Informative

      That doesn't make sense. 48k and 96K are sampling rates, so the problem wouldn't be in encoding and decoding. If there was a quality problem, it would be analog to digital converters those transferring to digital formats are using and the digital to analog converers a sound system has. You seem to be conflating sampling rate and bitrate. There have been dramatic improvements for the same bitrates in the last 20 years.

      --
      This is my signature. There are many like it, but this one is mine.
    4. Re:Pro recording by Bassman59 · · Score: 5, Informative

      I recently remixed a classic recording for sony records. The files where rolled off of tape at 24bit/96k. 48k I can understand but 96k is pointless. WAAAAAAY beyond the range of human hearing. In the old days, things like cymbals and brass could really stick out because the encoders and decoders where just not where they are today.

      Anyone that tells you they can hear the difference between 48k and 96k is dreaming. Its the quality of the recording that counts more than anything these days.

      The difference is that the antialiasing filters are much simpler and have a gentler roll-off when sampling at 96kHz. The high-order filters necessary to ensure adequate attenuation at Nyquist and above when sampling at the lower rates have this tendency to ring.

    5. Re:Pro recording by Graff · · Score: 2, Informative

      Oversampling (i.e. 192kHz) allows much more room to develop a good anti-aliasing filter.

      *whoosh*

      As the whole point of the article goes right over your head! You do not need any anti-aliasing. If you sample at 40 kHz with a decent equipment and a good 20 kHz low-pass filter then you can completely and faithfully recover a signal of less than 20 kHz by applying the Whittaker-Shannon interpolation formula.

      Now we generally sample at 44.1 kHz in order to have some oversampling to take care of non-ideal filters and such. This is 10% oversampling and it's far more than you need with modern equipment and algorithms. By doing all this properly you will get the exact waveform back. There will be no aliasing to anti-alias.

    6. Re:Pro recording by Iniamyen · · Score: 2

      You'll get disagreements about what phase shift/jitter is actually audible. It might be a good idea to frame an argument about this with data, rather than the "musicians with highly sensitive hearing *will* notice it" argument that the article purposefully and explicitly avoided.

    7. Re:Pro recording by GrahamCox · · Score: 3, Informative

      You're right but only in theory. You must have a low-pass filter to prevent aliasing - ANY signal beyond that will be aliased (and sound appalling). Thus the filter needs to have a brick-wall characteristic which is impossible. So by sampling at a much higher rate, the filter can be a lot more practical. The 10% "extra" you get with 44.1kHz sampling is insufficient space to implement a decent filter - that sampling rate is something of a historical accident anyway.

    8. Re:Pro recording by justforgetme · · Score: 2

      My point exactly. Storage is not a factor so when you have the choice in a studio setting why not keep the best resolution you can?
      Now for playback you obviously wouldn't want to log 2GB files for a 3min hiphop song, in that case you can just downsample to
      flac/48khz or even 256kbps mp3. Hell, most consumers are happy with the crappy 112/128kbps rips they get extracted from youtube videos.

      --
      -- no sig today
    9. Re:Pro recording by MikeBabcock · · Score: 4, Insightful

      When I listen to music, its not for the data -- its for the feeling. You should try listening to music for the feeling too ;-)

      My opinion.

      --
      - Michael T. Babcock (Yes, I blog)
    10. Re:Pro recording by Pieroxy · · Score: 2

      Why not 1000KHz? 1GHz? I mean, the wave form will be much, much much nicer. On a screen. Because your ears will hear the exact same thing.

      All that you describe is theory of signal. Of course, the wave form will be much better reconstructed at 192KHz. Of course. The real question is: can anyone make out the difference *with their ears*?

      The answer is no.

    11. Re:Pro recording by Pieroxy · · Score: 2

      Yes, for many purposes, its good enough. But its obviously not perfect.

      Of course. But why would you need perfection? Are your ears "perfect"?

    12. Re:Pro recording by thegarbz · · Score: 5, Insightful

      My favourite audiophile rebuttal quote:

      "If your hifi costs more than your music collection you have missed the point." - Unknown Source

    13. Re:Pro recording by petermgreen · · Score: 3, Informative

      Recording a signal with high fidelty is NOT a matter of just taking samples at defined intervals. If you do that you will get aliasing (higher frequencies getting converted to lower frequencies by the sampling process). So before you sample you need an "anti-aliasing filter" to remove signal components above the nyquist point.

      However filter design is a compromise, a filter with a steep response in the frequency domain will have a long impulse response in the time domain. A filter that doesn't cause phase distortion will cause pre-echo when fed with an impulse signal. Further making high order analog filters reliable and well behaved is difficult.

      Similarlly at output many digital to analog conversion methods will produce unwanted copies of the signal beyond the nyquist point, again a filter (known as a reconstrution filter) is needed to remove these.

      96KHz gives you a much bigger "gaurd band" between the audio signal and the nyquist frequency so your anti-aliasing and reconstruction filters can be much less aggressive.

      Using oversampling (running your recording/playback devices at higher than the sample rate you are storing the music at) and doing most of the filtereing digitally can remove the issues with high order analog filters being unstable but it can't change the fundamental issue that a filter with a sharp response in the frequency domain will have a long impulse response in the time domain or that a filter with no phase distortion in the frequency domain will have pre-echo in the time domain.

      --
      note: i'm known as plugwash most places but i screwd up registering that here somehow in the past and now can't register
    14. Re:Pro recording by scary_jeff · · Score: 5, Informative

      I also spent 4 years studying an EE degree, and although it was not especially focused on signal processing, I now work for a large pro audio company.

      Some of the issues pointed to in this and other posts regarding oversampling and AA filters are not really relevant to the subject at hand, given the technology currently in use. A statement like 'oversampling at 192 kHz' shows a lack of knowledge regarding the kinds of audio converters that have been in use for a good while now. A Delta Sigma ADC running with an Fs of 48 kHz might often be oversampling at 3.072 MHz or 6.144 MHz. Anti aliasing filters that many people have mentioned are implemented digitally inside the converter (no need for external analog filters, which may well exhibit many of the problems mentioned), and actually have extremely good pass band ripple.

      Look at datasheets for converters from manufacturers such as TI (burr brown), cirrus [page 36 here has detailed plots of 48, 96, and 192 kHz pass pand characterisitcs for the device, highlighting the fact that increasing the sampling rate does not improve pass band ripple for this device (also note the scale is 0.02 dB/div)], AKM, Wolfson micro You will find pass band pass responses that are flat to within less than +/- 0.05 dB over the audible range, and stop band attenuation in excess of 100 dB, whether sampling at 48 kHz or 192 kHz. If you can find anything in actual converter datasheets that points to better converter performance from selecting a higher sampling rate, I would be interested to see it.

      All in all, the basics of sampling theory don't really help people to understant the real world issues in designing a moden high end audio device. And in the end, surely the proof of the pudding is in the blind tests, that never seem to show that anybody can tell any difference when moving to higher rates? Even if there were a few people who could hear this difference in some perfect listening envirmonment, would it really make sense for everyone else to go out and buy 192 kHz equipment?

    15. Re:Pro recording by adolf · · Score: 4, Insightful

      The problem with low-pass filtering was resolved eons ago with a concept called "oversampling."

      Only the earliest and ruddiest of CD players (and a lot of computer sound cards) had a brick-wall filter at ~22.5 KHz. The rest of them resampled the input by 4x or 8x, or converted the original signal to PWM, and then applied the anti-aliasing filter at a frequency several octaves above the range of human hearing.

      This hypothetically pushed the nastiness inherent of a steep filter to a realm well outside such that humans could hear, and at least far beyond the limited confines of a CD.

      Welcome to 1985, where your stated concerns are both accurate and already solved.

    16. Re:Pro recording by Dogtanian · · Score: 2

      [The 44.1 kHz] sampling rate is something of a historical accident anyway

      Yes, it dates back to the days when the only method of *recording* CD-sized amounts of data (i.e. hundreds of megabytes) was to use a video cassette recorder (with a suitable interface) as a backing store, which dictated some of the technical aspects.

      --
      "Slashdot - News and Chat Sites Deviant". (Click "homepage" link above for details).
  5. Re:The article writer is a deaf idiot by Aboroth · · Score: 5, Funny

    I find your well-reasoned and respectfully written response to be full of helpful counterpoints and useful references. I wish to subscribe to your newsletter.

  6. Re:The article writer is a deaf idiot by bmo · · Score: 4, Insightful

    >There is a huge problem with file sizes

    Not any more, pumpkin.

    We hit the terabyte size in drives a couple of years ago. There's no reason to be buying this format vs "archive quality" cd-audio or other lossless.

    Buy/rip lossless. Transcode to lossy as needed. Anything else and you're being ripped off.

    I listen to real music with real instruments. The "swish" you get in high-frequency percussion with lossy algorithms is annoying as fuck.

    --
    BMO

  7. Re:I can tell the difference by Aboroth · · Score: 5, Informative

    You are missing the point of the article. 192KHz is not 192kbps.

  8. Re:The article writer is a deaf idiot by DeathFromSomewhere · · Score: 4, Insightful

    Double blind test results or I will continue to believe that you are suffering from Illusory superiority.

    --
    -1 overrated isn't the same thing as "I disagree".
  9. Pfft. by bmo · · Score: 5, Funny

    I have a PhD in Digital Music Conservation from the University of Florida. I have to stress that the phenomenon known as "digital dust" is the real problem regarding conservation of music, and any other type of digital file. Digital files are stored in digital filing cabinets called "directories" which are prone to "digital dust" - slight bit alterations that happen now or then. Now, admittedly, in its ideal, pristine condition, a piece of musical work encoded in FLAC format contains more information than the same piece encoded in MP3, however, as the FLAC file is bigger, it accumulates, in fact, MORE digital dust than the MP3 file. Now you might say that the density of dust is the same. That would be a naive view. Since MP3 files are smaller, they can be much more easily stacked together and held in "drawers" called archive files (Zip, Rar, Lha, etc.) ; in such a configuration, their surface-to-volume ratio is minimized. Thus, they accumulate LESS digital dust and thus decay at a much slower rate than FLACs. All this is well-known in academia, alas the ignorant hordes just think that because it's bigger, it must be better.

    So over the past months there's been some discussion about the merits of lossy compression and the rotational velocidensity issue. I'm an audiophile myself and posses a vast collection of uncompressed audio files, but I do want to assure the casual low-bitrate users that their music library is quite safe.

    Being an audio engineer for over 21 years, I'm going to let you in on a little secret. While rotational velocidensity is indeed responsible for some deterioration of an unanchored file, there's a simple way of preventing this. Better still, there have been some reported cases of damaged files repairing themselves, although marginally so (about 1.7 percent for the .ogg format).

    The procedure is, although effective, rather unorthodox. Rotational velocidensity, as known only affects compressed files, i.e. files who's anchoring has been damaged during compression procedures. Simply mounting your hard disk upside down enables centripetal forces to cancel out the rotational ruptures in the disk. As I said, unorthodox, and mainstream manufactures will not approve as it hurts sales (less rotational velocidensity damage means a slighter chance of disk failure.)

    I'd still go with uncompressed .wav myself, but there's nothing wrong with compressed formats like flac or mp3 when you treat your hardware right

    --
    BMO

    1. Re:Pfft. by smpoole7 · · Score: 3, Insightful

      Doood ... just, dood. You originally posted this, word for word, elsewhere (http://www.investorvillage.com/smbd.asp?mb=1911&mid=10609989&pt=msg). Either you are a bug-eyed alien, a prankster, or a combination of the two.

      For those who aren't in on the secret, you can look up "rotational velocidensity" -- on the Urban Dictionary. It is the supposed loss of bits in a file over a time, which is absolutely ludicrous. Digital is digital. It's ones and zeroes. Files stored digitally don't degrade, unless you're talking about media degradation (ex., CDs and DVDs can possibly suffer from loss of data over time).

      Dood also talks about files "repairing themselves," which is somewhere south of ridiculous.

      But enough of this. I fell for it and actually answered it.

      ("Digital dust." Heh.)

      --
      Cogito, igitur comedam pizza.
    2. Re:Pfft. by Frosty+Piss · · Score: 5, Funny

      No one never told you about backups and hashes?

      I think the parent knows all about hashish.

      --
      If you want news from today, you have to come back tomorrow.
    3. Re:Pfft. by bmo · · Score: 2

      >My understanding is that it's better to use CDs with gold reflective layers, rather than silver, as silver is prone to tarnishing. Is that correct?

      Taking this question seriously because there is an actual serious answer to this.

      No. A bit is a bit is a bit. Gold reflects infrared better, but not enough better that it makes a difference in the end.

      The biggest risk to CDs is voids in the lacquer on the top. Any scratches or holes in the lacquer top, the aluminum layer underneath oxidizes and vanishes. Hold up an old CD to the light and look at the constellation of holes.

      Screen printed CDs last much longer than "silver top" CDs that just have the lacquer layer.

      Honest to gawd literal bit rot, that is.

      --
      BMO

    4. Re:Pfft. by tkrotchko · · Score: 2

      "For those who aren't in on the secret"

      I think you were the only one.

      --
      You were mistaken. Which is odd, since memory shouldn't be a problem for you
  10. Re:The article writer is a deaf idiot by tapspace · · Score: 2

    > I listen to real music with real instruments. The "swish" you get in high-frequency percussion with lossy algorithms is annoying as fuck

    Seconded. Many things sound fine (not great, but OK) in medium to low bitrate MP3 or OGG or AAC or whatever. Some things sound terrible, and when they do, it sucks to listen to.

  11. The bit depth does matter by gnu-sucks · · Score: 4, Insightful

    As a former audio engineer with some ranking success, I can tell you that it's true -- delivering high-sample rate audio as an end format is really pointless. It hardly makes sense in a studio, and definitely is illogical for the distribution of a final mix.

    However, there is an increase in quality using 24 bit. Most people just assume increasing the bit depth is the same as increasing the sample rate, but this is incorrect and short-sided. With higher bit depths, you can get your analog components operating a little further away from the noise floor. This also makes dithering much less noticeable (the noise you hear when you crank the volume up as a song fades out). Why? There are more "levels" for each sample to be recorded into. It's like going from 16 to 24 bit color. You would notice this.

    For the 192 KHz fans out there, there is direct and proven mathematical reasoning for why 44 KHz audio is plenty. That, and your equipment probably can't produce it. Your converters probably suck at this frequency, and your ears definitely can't vibrate that quickly. More samples doesn't "smooth out" the waveform.

    1. Re:The bit depth does matter by jmv · · Score: 3, Insightful

      I would say that theoretically, 44 kHz is enough, but in practice the filtering is a bit of a PITA. WIth 48 kHz, you can use shorter filters and it's much easier to convert to-from other widely used sampling rates (e.g. 8 kHz and 16 kHz for telephony/VoIP). Otherwise, I fully agree that 192 kHz is totally stupid.

    2. Re:The bit depth does matter by PhrostyMcByte · · Score: 2

      It's not the music that matters so much as the mixer/DAC.

      High bit-depth playback matters greatly for any system that controls volume at the mixer stage rather than the amp stage. This is very important for PCs, where it is common to keep the amp (speakers) at a fixed volume and control the actual listening volume from the operating system's mixer.

      If you keep the volume all the way up on your mixer, controlling your listening only at the amp stage, then a 16-bit pipeline is plenty. Highly integrated hardware such as an MP3 player can easily get away with this.

      Otherwise, a 24-bit mixer and DAC can be very useful as it allows you to turn the volume down really far on the mixer while still retaining all the detail of your 16-bit music for the amp to boost back to listenable levels. It's still not perfect (such a low line level will inevitably be noisier) but it's still much better than a 16-bit pipeline.

    3. Re:The bit depth does matter by TheGratefulNet · · Score: 3, Insightful

      as a current audio engineer (doing dac's and spdif circuits), let me inform you that 88.2, 96, 176.4 and 192 are well alive and working well and showing some really impressive test measurements.

      I can't hear any better than cd redbook (even then its better than my aging hearing) but I sure can see it on the test gear I use to design my own gear with.

      its cheap, too. wolfson dac chips are $10 or so, give or take. that's a current high-end pick and it tests very very well. so well that most analog buffers can't keep up and many power supplies are not low noise enough.

      I do agree tha 192k is overkill for final delivery. I also shoot photos and I downmix to 8bit jpg but I insist on getting 24bit raw images, doing all my processing at 24bit color, then finally going down to 8 again for jpg saving. audio is exactly like that, too.

      but in photo, you are either slim (8bit jpg) or really a pig and taking up far too much room. in audio, there are many grades. you can be 88.2 (relative of 44.1 cd) or 96k (never intending to go down to cd silver disc format). you can record at a multiple of 96k (first multiple is 192k) and then downconvert to 96k for user distribution.

      dacs at 24/96 are a GOOD break point for performance (chip and circuit) and cost. files are not that big at 2496 either, really. 192 is nuts for end users but 2496 is quite good.

      --

      --
      "It is now safe to switch off your computer."
    4. Re:The bit depth does matter by toejam13 · · Score: 2

      I agree with you. I've done blind tests between 48kHz and 96kHz and I cannot hear a difference. I used to hear a difference between 44.5kHz and 48kHz when I was younger, but it is getting harder as I age. Personally, I cannot see why 192kHz samples would be released outside of the studio.

      I can hear the difference between 16-bit and 20-bit, but not so much between 20-bit and 24-bit. At that point, the noise floor for the media has gone below that of other components, so you really can't tell.

    5. Re:The bit depth does matter by bill_mcgonigle · · Score: 2, Interesting

      and your ears definitely can't vibrate that quickly

      Your ear drums top out at 20KHz, but some of the small bones in your ear will vibrate up into the 60s' and that passes on auditory information. This can help provide clues for positioning, at least.

      --
      My God, it's Full of Source!
      OUTSIDE_IP=$(dig +short my.ip @outsideip.net)
    6. Re:The bit depth does matter by jrumney · · Score: 2

      High bit-depth playback matters greatly for any system that controls volume at the mixer stage rather than the amp stage.

      Which is why sound cards that do mixing/volume control in the digital domain have an upscaling step before their final mixing stage. So the source material still doesn't need to have more bits.

    7. Re:The bit depth does matter by niftydude · · Score: 2

      As a former audio engineer with some ranking success, I can tell you that it's true -- delivering high-sample rate audio as an end format is really pointless. It hardly makes sense in a studio, and definitely is illogical for the distribution of a final mix.

      Maybe this is true for people who just want to listen - but what about non-studio music nerds that want to play around and sample and remix tracks? Amateur musicians would like as high-sample rate audio as possibly, so that any down-mixing artefacts don't accumulate.

      The only argument for not distributing the full sample rate audio in the current environment of high bandwidth and high disk space is if you believe that music creation should start and end in studios. I can't express how much I disagree with that sentiment.

      --
      You can never know everything, and part of what you do know will always be wrong. Perhaps even the most important part.
    8. Re:The bit depth does matter by hechacker1 · · Score: 2

      And pretty much everything you said is true in some sense. Given not so superb equipment for mixing, recording, and playback, simply having the slight room for aliasing filters and frequency information can improve the final product that gets output at 16/44.

      But as the article says, if you do it right the first time, there's really nothing to be had going for more than 16 bits, and 44kHz, it should encapsulate the entire range of human hearing in any normal situation.

      I'm really glad the article was posted. It cleared up some of my misconceptions.

      And now I know the final product at 16/44 is just fine if done right.

    9. Re:The bit depth does matter by tlhIngan · · Score: 3, Insightful

      I agree with you. I've done blind tests between 48kHz and 96kHz and I cannot hear a difference. I used to hear a difference between 44.5kHz and 48kHz when I was younger, but it is getting harder as I age. Personally, I cannot see why 192kHz samples would be released outside of the studio.

      I can hear the difference between 16-bit and 20-bit, but not so much between 20-bit and 24-bit. At that point, the noise floor for the media has gone below that of other components, so you really can't tell.

      First, most studio masters are 48kHz. Finding 96kHz or even 192kHz mastered audio is HARD. The range and selection of media capable of those sample rates is extremely low. Maybe under 100 Blu-Rays have 96kHz audio tracks, and far fewer have 192kHz tracks. And 96kHz has been around since the DVD days, and we still get audio mixed at 48kHz.

      They do, however use 24bit sampling.

      As for why go 96kHz or 192kHz, it's quite minor. For this, we need to explore sampling theory.

      First, you have an analog signal. Then you MUST pass it through a low-pass filter (called an anti-aliasing filter) that bandlimits the input signal so it doesn't exceed the Nyquist limit (which will cause aliasing in the sampled waveform).

      The trouble spot is the analog filter. If we assume that human hearing stops precisely at 20kHz, at 44.1kHz, we have to have a filter that basically has a stop band from 20kHz to 22.05kHz. It takes a lot of work to do this and the filters tend to be pretty big if you want to achieve filters that have flat passbands and low phase-distortion.

      At 48kHz, you have a stop band of 20kHz through 24kHz, which makes for a much easier design. At 96kHz, you have a LOT of stop band. Enough so that you can perhaps set the passband higher (you have to block frequencies above 48kHz, so you can start your stop band somewhere between 24-25kHz which should cover the majority of people's hearing. And you'll have a whopping 24kHz or so for the stop band, making for a very clean filter with gentle rolloff (which generally gives you better passband performance - flatter response on the pass band, and very low phase distortion).

      At 192kHz, that's really getting excessive - even if you set the pass band at a ridiculous 48kHz to cover every possible human and dog, there's a pile of bandwidth available for the stop band.

      96kHz audio may sound better if you're young (or a dog), but a good chunk of the older population has hearing that rolls off starting around 16kHz or so.

      Hence why the vast majority of works are sampled at 48kHz - it really is good enough and those that can hear ultrasonic will lose the ability in a few years.

    10. Re:The bit depth does matter by BlackPignouf · · Score: 3, Informative

      I insist on getting 24bit raw images

      Maybe you should insist more, because they're no such thing as a 24bit/channel camera.
      AFAIK, the highest bit depth you can get is 16bit/channel on high end medium-format sensors.

    11. Re:The bit depth does matter by serviscope_minor · · Score: 3, Insightful

      However I've always doubted it as, it's can be defeated with a pen and paper.

      Basically, what you're saying is that you have no background in maths, but you can disprove a very well known and thoroughly proven theorem by sketching lines on a peice of paper.

      You can also draw a triangle with bent edges to disprove Pythagoras too, if you like.

      You can also disprove Fermat's Last theorem by showing 1782^12 + 1841^12 = 1922^12 on many common calculators, too.

      It is well known that the Nyquist frequency (and that frequency only) cannot have the phase or amplitude reconstructed correctly. *every* *single* frequency below that can, no matter what you think your bar graphs look like.

      22kHZ will not be reconstructed correctly. 21.9999kHz will be, and that's still above the threshold of hearing.

      Since you've gone to the effort of drawing them, now draw them after they've run through an analog filter. That's a little bit harder...

      The sampling errors you refer to add noise over the entire frequency spectrum. This is well known and the article even addresses it very obliquely (noise floor).

      --
      SJW n. One who posts facts.
  12. Re:The article writer is a deaf idiot by Sparohok · · Score: 5, Insightful

    When you can tell the difference between 44.1/16 and 192/24 in a double blind trial, come back and we'll talk.

    Subjective opinions about audio quality, particularly those accompanied by words like "deaf" or "idiot", are worse than useless. Subjective listening is deeply suggestible and unreliable. Claimed differences among any acceptably well designed audio electronics virtually always disappears under rigorous and controlled testing.

    To give just one example, listeners reliably prefer the louder source in subjective testing, even if the difference is not consciously perceptible. If a 192/24 D/A is just 0.1db louder than a 44.1/16 source, listeners will tend to describe it in all sorts of subjective terms... "edgier," "richer," "more forward," "cleaner impact," "deeper soundstage" etc when in fact it is simply a little louder.

  13. Re:Audiophiles by Lanteran · · Score: 2

    If you buy your music over the 'net, flac isn't an option, and CD stores are dying. One of the many reasons piracy is still so popular among audiophiles.

    --
    "People don't want to learn linux" hasn't been a valid excuse since '03.
  14. Re:44KHz by belg4mit · · Score: 3, Informative
    --
    Were that I say, pancakes?
  15. the poster at xiph never heard of Monster Cable by circletimessquare · · Score: 4, Funny

    "Because it's a solution to a problem that doesn't exist, a business model based on willful ignorance and scamming people."

    which happens to be a business model that works, unfortunately

    --
    intellectual property law is philosophically incoherent. it is your moral duty to ignore it or sabotage it
  16. Re:What if... by enoz · · Score: 4, Funny

    Your cat is not "listening", it is simply tolerating that annoying racket that you call "music" in exchange for food, body heat, clean kitty litter, etc.

  17. Re:The article writer is a deaf idiot by Anonymous Coward · · Score: 3, Insightful

    If you're sure you can hear a difference, why don't you ABX and prove it (or give strong evidence for it)? It's easy to hear a difference if you think you're supposed to, or if you paid a lot of money for speakers, etc. But its a lot harder to hear differences if you're doing a double blind test.

    It's certainly OK to allow your emotions to take over if it makes you feel better to know you're listening to 24/192, but that's different than there actually being a perceived difference. You feeling better listening to 24/192 is an opinion, but whether you can actually perceive a difference is fact; lots of people confuse the two, so don't feel too bad.

  18. Re:44KHz by PPH · · Score: 2

    Either Harry Nyquist or Claude Shannon probably could have. But they are both dead now. So we will have to take Monster Cable marketing department's word for it now.

    --
    Have gnu, will travel.
  19. Re:The article writer is a deaf idiot by DeathFromSomewhere · · Score: 4, Insightful

    Did you listen to it double blinded? No? Then I don't care what your confirmation bias tells you that you heard. The difference is beyond your ability to hear, but not beyond your ability to deceive yourself into believing what you want to believe.

    --
    -1 overrated isn't the same thing as "I disagree".
  20. Re:The article writer is a deaf idiot by Sparohok · · Score: 5, Informative

    A group of sixty audio professionals and audiophiles did a series of controlled double blind trials published in the Journal of the Audio Engineering Society. They found no perceptible degradation caused by a 16-bit/44.1kHz A/D/A.

    http://www.aes.org/e-lib/browse.cfm?elib=14195

  21. Re:44KHz by GumphMaster · · Score: 3, Informative

    The Nyquist-Shannon Sampling Theorem basically shows that if an analogue signal contains no frequency higher than B Hz then sampling at any rate greater than 2B Hz is adequate to reproduce the signal without aliasing. In the case of audio recording intended for the human ear, the highest audible frequency is about 20kHz and the minimum sampling rate to cover that should be 40kHz. This is (partly) where the 44100 HZ sampling rate of CD audio comes from. In practice sampling is usually performed faster than required by the theorem (though not four times faster). The theorem is not sufficient in itself to guarantee perfect reproduction and is limited by the ability of real systems to match the mathematical ideals during sampling and reproduction. Reproduction is, however, typically very close.

    The 192kHz sampling that is the subject of this thread is capable of capturing frequencies well beyond the capability of a human ear to hear, or any typical speaker system to reproduce.

    --
    Patent litigation: A doctrine of Mutually Assured Destruction... in which everyone seems willing to push the button
  22. Re:The article writer is a deaf idiot by smi.james.th · · Score: 4, Interesting

    No loss from the original sampling, i.e. they didn't loose any information in the compression. Most music is sampled at (correct me if I'm wrong someone?) 44kHz, I forget how many bits, I think 16. The thing being touted is sampling it at 192kHz with 24bit resolution, which is much higher on both counts, and therefore, in theory, should produce better quality reproduction of the sound based on oversampling and reduction of the signal to quantization noise rate. The point the TFA makes is that human ears can't hear the difference, although I think that some audiophiles may beg to differ.

    FWIW, I have quite bad ears, a recording needs to be quite bad before I notice it. I'm an electronic engineer though, so I know all the theory...

    --
    One thing I know, and that is that I am ignorant...
  23. Re:Can we stop using the word "truthiness," please by Anonymous Coward · · Score: 5, Funny

    You mean like, honkies, spics, niggers, dune coons, prairie niggers, kykes, faggots, chinks, canucks, wops, guineas, krauts, and polocks? I think that's everybody anyway, my apologies if I left out any group, I try to be an equal opportunity offender, challenging people to be adults and get over their group identitied. Criticism welcome. Cowardly disapproval spurned.

  24. Re:Can we stop using the word "truthiness," please by MobileTatsu-NJG · · Score: 5, Funny

    I know, Stephen Colbert is Reddit's hero and they're starting to infiltrate this site as well, but seriously. Call them lies. That's what they are, that's what they -deserve- to be called. Are people really that passive-aggressive and afraid of expressing themselves that they won't call someone who lies a liar any more?

    Okay, everybody, listen up: Anonymous Coward is having a rough day so let's all be extra nice to him!

    --

    "I like to lick butts!" by MobileTatsu-NJG (#32700246) (Score:5, Informative)

  25. Re:The article writer is a deaf idiot by smpoole7 · · Score: 2

    > If you can't hear the difference ...

    I certainly can. I'm glad to hear others say that, too. I thought it was just me.

    We have an analogous problem in broadcasting -- everyone wants to use compressed formats to save space and upload/download time. Files are thrown all over the Web now. (I haven't seen a reel tape in years, though I think we still have an old reel-to-reel somewhere just in case. Political season coming up, after all.)

    The problem is REALLY bad when you repeatedly encode. For example, our digital automation systems wants to compress files. Our studio to transmitter links (STLs) want to compress to save bandwidth. HD Radio compresses the SNOT out of the audio. Honestly ... some of the crap that I hear on the radio now is so bad I don't know how anyone can listen to it. It swishes, it glitches, it swarms, it sounds brittle, it's awful.

    I made a rule in our facilities a few years ago that if it wasn't at least 256 Kilobits, we wouldn't air it. This annoyed some people -- one guy had to dump and entire music library that he'd spent a week putting into the system -- but it was awful.

    Maybe there's no point in 192/24 for kids listening to pirated music on $20 MP3 players, but I refuse to believe that most people can't hear the difference. Heck, I'm getting old and I'm half deaf nowadays, and I can immediately hear the difference. There's just no comparison.

    --
    Cogito, igitur comedam pizza.
  26. Re:Audiophiles by Sarten-X · · Score: 4, Insightful

    For the rest of us on /. haven't we had all of our music in FLAC for a decade now? I don't even listen to music much and mine is.

    My music is mostly stored in whatever the default is for YouTube videos that I've saved locally. I'm apparently even less of a music fan than you are.

    Fun fact: I'm also an audio technician. Yes, I can hear the occasional damaged sound, but I'm not enough of an asshole to care.

    --
    You do not have a moral or legal right to do absolutely anything you want.
  27. Re:44KHz by smi.james.th · · Score: 2

    There may be no theoretical benefit, but since there's no such thing as an ideal sampler or filter or quantiser, it has many practical benefits.

    --
    One thing I know, and that is that I am ignorant...
  28. Re:Can we stop using the word "truthiness," please by xiphmont · · Score: 5, Informative

    Truthiness refers to a specific kind of lie-- a lie that sounds true, and that a large segment of people really want to be true. The kind of thing that's close enough to true for AM radio talk show hosts.

    And now... I'll get off your damned lawn. Don't forget to take your teeth out before falling asleep.

  29. Re:The article writer is a deaf idiot by icebike · · Score: 4, Funny

    Not if you don't know any better. ;-)

    Seriously, its been so long since I've seen a live band I don't know what a drum is supposed to sound like.
    At my age my ears are not so hot.

    --
    Sig Battery depleted. Reverting to safe mode.
  30. Re:The article writer is a deaf idiot by kyrio · · Score: 2

    The point is that the 44KHz 16bit track has already been compressed from the original recording. However you rip that track, lossless or lossy, it doesn't matter; you're still not getting the original track.

    Knowing this, it doesn't mean that the tracks some sites are selling as 192KHz 24bit are from the original sources, or will even sound better, either. The original track could have been recorded with bad equipment or settings. In other cases, when doing comparisons on CD tracks vs high resolution tracks from sites like HDTracks, you can sometimes find that the HDTracks track is just the CD track with increased reported resolution/file size - possibly due to the inability to acquire the original material, though it could also be as simple as pure greed and laziness. Not that all of the albums on those sites are fakes, but a few of them have been found to be ripoffs.

    There's also the fact that it's extremely unlikely anyone can tell the difference between an encode at 96KHz vs 192KHz. If they are both properly encoded from the same source, it's unlikely there will be any audible difference between them.

  31. Re:The article writer is a deaf idiot by xiphmont · · Score: 4, Insightful

    Indeed. One of the overlooked but highly important issues with sampling rates is that although you can represent up to Nyquist in a periodically sampled signal, that is the limit for infinite length recordings. For finite-length recordings, it isn't all or nothing, represented perfectly or not at all -- instead the uncertainty (read: representation error) increases as you approach Nyquist.

    Too bad Shannon and Nyquist are dead. It seems they've completely misunderstood the math. How embarrassing they passed on before you could correct their mistake. Now they'll never know.

  32. Re:The article writer is a deaf idiot by Jafafa+Hots · · Score: 5, Funny

    I used to think like you. Spent thousands on audio equipment.

    Now that I'm deaf in one ear I listen to MP3s through $24 headphones.

    Being deaf saves a lot of money.

    --
    This space available.
  33. Re:Can we stop using the word "truthiness," please by SternisheFan · · Score: 3, Funny

    If George Carlin were still alive, he would mod you up right now.

  34. Re:The article writer is a deaf idiot by DeathFromSomewhere · · Score: 4, Insightful

    Double blind test or gtfo. The peer reviewed research says you can't hear it. Talk is cheap, show us some data.

    --
    -1 overrated isn't the same thing as "I disagree".
  35. Re:The article writer is a deaf idiot by pla · · Score: 4, Interesting

    There is a huge problem with file sizes (so both hard drive space and download bandwidth) with lossless files, so no, it's not entirely without problems.

    I own (legally, even) somewhere on the order of 2500 CDs.

    I have ripped all of them to FLAC (lossless).

    Total size, under 600GB. I could easily fit my entire collection on a single HDD five years ago. Today, they don't even count as the biggest single directory on my home file server (hell, not even third place - Though in fairness, I do collect historically-significant Linux distro ISOs).

    FWIW, even ripped raw rather than compressed as FLAC, they would still fit on a single 2TB drive. Audio really doesn't present all that much of a problem these days.

  36. Re:The article writer is a deaf idiot by russotto · · Score: 2

    The thing being touted is sampling it at 192kHz with 24bit resolution, which is much higher on both counts, and therefore, in theory, should produce better quality reproduction of the sound based on oversampling and reduction of the signal to quantization noise rate.

    Sure, but with the loudness war, they're not really using the 16 bits they have, so what's the point?

  37. It really doesn't matter by wbr1 · · Score: 3, Insightful

    Educating people is fine, but the elitists will always say swear that x is better than y, even if it is provably otherwise. Just like some people will swear they saw Elvis working as a hooker at the Rt. 97 truck stop blowing Jesus.

    --
    Silence is a state of mime.
    1. Re:It really doesn't matter by Toonol · · Score: 3, Insightful

      This topic is a good barometer for the general quality of the Slashdot readership, which (rumor has it) has been declining. If we ever reach the point where over half the comments are 'audiophiles' defending these impossible-to-hear improvements, we'll know that Slashdot has reached the tipping point, and it will be time for any remaining rational people to leave.

  38. Re:Can we stop using the word "truthiness," please by Anonymous Coward · · Score: 3, Insightful

    I think Truthiness covers half truths too. A half truth is that 24-bit/192kHz audio is higher quality than 24-bit/96KHz audio.

    The whole truth is that only your house cat would be annoyed at 96KHz, or an audiophile dog.

  39. Re:44KHz by ChrisMaple · · Score: 4, Interesting

    Your ear samples at about 20 kHz.

    That's a profound misrepresentation of how hearing works.

    Here's an oversimplified and inaccurate explanation. The ear's mechanism relies on different frequencies providing the highest level of excitation at different places. Your trained nervous system recognizes each different place as a different tone.

    For most people, there is no place where sounds above 20 kHz will irritate a nerve ending enough to send an impulse to your brain. Thus, no sound higher than 20 kHz is audible, and 20 kHz corresponds to a 40 kHz sampling rate. (One sample at the low point on the wave, the next sample at the next high point, etc.

    --
    Contribute to civilization: ari.aynrand.org/donate
  40. Re:The article writer is a deaf idiot by Gr8Apes · · Score: 2, Insightful

    The point is that the 44KHz 16bit track has already been compressed from the original recording. However you rip that track, lossless or lossy, it doesn't matter; you're still not getting the original track.

    Actually - it wasn't compressed - it was the limits of the recording equipment at the time. 192KHz/24 bit wasn't common in the 80s.

    Knowing this, it doesn't mean that the tracks some sites are selling as 192KHz 24bit are from the original sources, or will even sound better, either. The original track could have been recorded with bad equipment or settings. In other cases, when doing comparisons on CD tracks vs high resolution tracks from sites like HDTracks, you can sometimes find that the HDTracks track is just the CD track with increased reported resolution/file size - possibly due to the inability to acquire the original material, though it could also be as simple as pure greed and laziness. Not that all of the albums on those sites are fakes, but a few of them have been found to be ripoffs.

    Unless the track are genuine 192KHz/24 bit tracks, that is true. CD tracks can sound as good or better than 192KHz/24 bit tracks, it all depends upon settings. CD tracks can also sound worse than 92KHz encoded MP3s, again, it depends upon settings.

    There's also the fact that it's extremely unlikely anyone can tell the difference between an encode at 96KHz vs 192KHz. If they are both properly encoded from the same source, it's unlikely there will be any audible difference between them.

    This, however, is patently false. Given appropriate equipment and a person with a reasonable ear (mine aren't even that great and they suffice) and you can definitely tell the difference between 92KHz and 192Khz, and even straight CD tracks they were encoded from. It does require that the original source have enough depth that something is lost, however. Simple electronica, or other music that samples heavily from trivial sources will not provide enough depth to tell.

    At this point my entire collection is lossless (CD quality at a minimum), and yes, it even makes a difference in my car, which has a halfway decent audio system. The other vehicle needs new speakers and an amplifier, the former sound blown and the latter was never clean to begin with, enough so that I pretty much haven't listened to music in it in years, just haven't gotten around to replacing it as it was only short trips anyways.

    --
    The cesspool just got a check and balance.
  41. Re:The article writer is a deaf idiot by DeathFromSomewhere · · Score: 3, Informative

    I don't care how highly you think of yourself, until you show me some data you are a worthless troll.

    --
    -1 overrated isn't the same thing as "I disagree".
  42. Re:The article writer is a deaf idiot by Toonol · · Score: 4, Insightful

    They won't believe you. They believe their ears must be superior to those pseudo-audiophiles. Your post should have ended all discussion, but *sigh* it won't.

  43. Re:The article writer is a deaf idiot by smi.james.th · · Score: 4, Insightful

    Fair point. The people who go on about 24/192 probably don't really listen to the kind of music which is affected by the loudness war. Most audiophiles I know are heavily into jazz or classical music, the recordings of those usually try to be quite faithful to the original.

    --
    One thing I know, and that is that I am ignorant...
  44. Re:The article writer is a deaf idiot by DeathFromSomewhere · · Score: 4, Insightful

    1. Find post asking for results of a properly conducted double blind test.
    2. Ramble on about your various stereo equipment for a couple paragraphs, show a complete ignorance of confirmation bias.
    3. Completely fail to provide the requested evidence, wasting every ones time.
    4. ???
    5. Profit!

    --
    -1 overrated isn't the same thing as "I disagree".
  45. Not quite accurate on the human senses by izomiac · · Score: 2

    Of course this is ludicrous.

    No one can see X-rays (or infrared, or ultraviolet, or microwaves). It doesn't matter how much a person believes he can. Retinas simply don't have the sensory hardware.

    I wouldn't be so sure... $10 IR filter goggles. The human senses do have limits, but they're rather soft and fuzzy. First, there's genetic variation in the exact sensitivity range (e.g. some people can perceive further into the "infrared" spectrum than others, it's a common high school & college lab experiment). Plus, pedantically, everyone can detect IR up to 3,000 nm at least, cooking would be highly impractical otherwise, and Beethoven felt for vibrations so he could continue composing/performing despite his deafness (IOW, our senses overlap, very important for concert goers that like to feel the bass).

    Second, and more importantly, the raw signals are integrated by the brain in a semi-predictable pattern (obviously it's a self-teaching neural network, so people process things differently, although there are common trends). An insect has a compound eye with dozens or hundreds of photoreceptor units. Individually, they're not terribly sensitive, but when integrated provide a much clearer picture. It's akin to how photographers can merge multiple overlapping images to create gigapixel-level quality.

    Given harmonics, pinna distortion and such, it wouldn't surprise me if hair cells do not impose an absolute limit on hearing, as the article states. OTOH, I doubt that 192 kHz offers any real sound improvement, but I don't think you can argue that with just biology, as there are few, if any, definites in that subject.

  46. Re:The article writer is a deaf idiot by sribe · · Score: 2

    A snare brush rustles at 192/24 instead of sounding like rustling paper.

    While that's true, it would definitely also be true at 64/24, and likely at 64/20 I think. While 44/16 is a marginal format that with good D/A conversion can merely deliver what most equipment is able to reproduce, 192/24 is *way* beyond what anyone can hear.

  47. Re:The article writer is a deaf idiot by xero314 · · Score: 4, Funny

    I'm not deaf, but I've never spent more than $10 on headphones.

    You'll be in for one heck of a shock the day you hear what music actually sounds like.

  48. Re:The article writer is a deaf idiot by Gr8Apes · · Score: 2

    There are lots of double blind tests. Most that mean anything are between CD quality and above. No difference found after a year plus of testing. If you want to hear some differences in what's left out when items are compressed A refutation of the validity of double-blind audio tests The main point would be that a well mastered CD is better than a poorly mastered 192kHz/24 bit recording, and the same goes for a poorly mastered CD vs a 192 encoded well mastered piece. However, when the original quality material is of like quality, many can tell the differences until they get to CD quality. After that, a smaller segment can tell. What's been destroying music is the large group of folks who've never heard anything that wasn't put through a pipe filled with a wet sponge first. If that's all you've been exposed to, even the clear trill of a bird might sound unpleasantly harsh in its clarity.

    --
    The cesspool just got a check and balance.
  49. Re:The article writer is a deaf idiot by maccodemonkey · · Score: 2

    Sure, but encoding at lossless (which is what I do for albums that are important to me, rest are 192 kbps iTunes purchases) is entirely different than just wasting space. Lossless has a tangible benefit, whereas as the article points out, outside production, stuff like 24 bit audio does not.

    It's the equivalent of encoding beyond lossless, just adding extra bits on top of a lossless encode that you'll never hear ever.

  50. Logic seems flawed by KingTank · · Score: 2

    Difficult to explain, but it reminds me of how some people say that there's no point having a frame rate higher than 30 fps. No, your eyes can't actually see the screen flickering above that frame rate, but that doesn't mean it looks perfectly fluid. The author is assuming the point of diminishing returns is actually a point of no returns, which may be far from the truth.

  51. Re:The article writer is a deaf idiot by Larryish · · Score: 3, Funny

    $24 earphones?! You lucky devil.

    When I was a wee lad, we had to listen to music through paper cones pressed to our ears. And they weren't real paper, mind you, but a great bloody lot of wasps nests glued together with our own spit.

    Youngsters just have no idea.

  52. Re:The article writer is a deaf idiot by mug+funky · · Score: 3, Interesting

    the trick is getting noise from the real world to sit quietly below the 7 dB loudness that a 16 bit noise floor gives us with an ideal listening environment (ie 83 dB SPL when presented with pink noise at -20dBFS in digital land).

    i really hope EBU R-128 gains more momentum. it's been adopted in the broadcast industry very fast, but that's preaching to the choir. i don't think it'll ever make headway in the music industry unless apple rename it "iLevel" and insist on it - rejecting any music submitted to their store that doesn't meet the spec that they totally invented.

  53. 16 bits isn't enough dyanamic range, sort of. by Animats · · Score: 2, Insightful

    If it weren't for the fact that all popular music has its dynamic range compressed to provide maximum loudness for the entire song, dynamic range would be be a problem.

    The problem is that, on soft passages, where the high 8 or 10 bits are zero, you're listening to 8 or 6 bit audio. That quantization can be heard. This is a problem for classical recordings made without any dynamic range compression. Of which there are very few.

    This is an issue only if you listen to classical music in a very quiet environment. It doesn't matter for car audio. It doesn't matter for Apple's trendy crap earbuds. So almost nobody cares.

    1. Re:16 bits isn't enough dyanamic range, sort of. by serviscope_minor · · Score: 2

      I love how arguing with maths gets modded insigntful.

      The quantization adds the same noise across the entire spectrum, regardless of the amplitude of the signal.

      If the loudest sound is on the threshold of pain, then the noise floor is below the threshold of hearing.

      Look at the PSD figure. With a sine wave at -105dB, the noise floow is still 20dB
      down in power.

      If you can hear quantization artefacts, then you're either suffering from confirmation bias or the piece hasn't been mastered to use the full range correctly.

      --
      SJW n. One who posts facts.
    2. Re:16 bits isn't enough dyanamic range, sort of. by raxx7 · · Score: 2

      It can be heard, but you need to turn your volume way up.
      Try this:
      sox -V -t sl -r 44100 -b 16 /dev/zero silence.wav trim 0 1:00
      This first command will produce a WAV consisting on a stream of 16 bit null samples.

      sox -V -t sl -r 44100 -b 24 /dev/zero -b 16 dithered_silence.wav trim 0 1:00
      This second command will convert a stream of 24 bit null samples to 16 bit, adding dithering noise in the process.
      This will be a good representative of the noise floor "intrinsic" to a 16 bit format.

      Play the dithered_silence.wav file and turn up the volume knob until you can hear the noise.
      Then play the silence.wav just to check that the noise was coming from the file, not from your playback system.
      And then ask yourself: do you really listen to music that loud?

      Also, a warning: many classical music recording have way higher noise floors, due to ambient noise.

  54. Re:"Truthiness" is a dumb word by retchdog · · Score: 5, Interesting

    no it isn't. verisimilitude is, roughly, the quality of being believably realistic. truthiness is like "verisimilitudinous lying," i.e. the apparent realism is misleading, often toward the exact opposite of the truth.

    --
    "They were pure niggers." – Noam Chomsky
  55. Re:44KHz by tftp · · Score: 5, Informative

    There may be no theoretical benefit, but since there's no such thing as an ideal sampler or filter or quantiser, it has many practical benefits.

    Here is a quick example. You sample at 44 kHz. The first Nyquist zone is from 0 to 22 kHz, the second one is from 22 to 44 kHz (with flipped spectrum.)

    Now, say that some [mechanical] harmonic from some instrument has frequency of 33 kHz. We don't hear those with our ears (parts of the ear are too massive to vibrate fast enough) so no harm done. The orchestra is playing as usual.

    But now record this orchestra with an imperfect antialiasing filter (there are reasons why a perfect one wouldn't do you much good anyway.) The 33 kHz harmonic falls into the 2nd Nyquist zone. It will be played back as if it was (22 kHz - 11 kHz = 11 kHz.) Can you hear 11 kHz? Most people hear it just fine. Think about it for a moment. There was no 11 kHz signal in the original spectrum; there was 33 kHz, an inaudible one. The artifact showed up because a [lossy] mathematical operation was performed on the data that describes the signal. The resulting distortion produced an audible tone where none was present originally.

    However if you encode at, say, 128 kHz sampling rate, things change. First, the antialiasing filter - even if it is of the same architecture - will have its cutoff way below the Fs/2. This means that signals of the second Nyquist zone will be attenuated by many tens of dB - essentially they can be completely eliminated because nobody cares what you do to ripple and phase above 30 or 40 kHz. Second, for the alias to show up it has to be in LF radio band now, starting at 128 kHz. Microphones aren't even mechanically capable of picking up those frequencies. And finally, if that 33 kHz harmonic passes through the filter (with the same mediocre attenuation as in the first example) ... it will be played back as 33 kHz, and it won't go anywhere. The amplifier will filter it, and the speakers will attenuate it greatly. In other words, a serious distortion that was present when you are sampling at 44 kHz disappears when you are sampling at a much higher rate.

  56. Why not lossy-compress 24bit/192kHz? by Dr.+Spork · · Score: 2

    I think I can find a compromise that should work for everyone: Why not just run the needlessly good 24 bit 192 hHz music file though a lossy compressor that does psychoacoustics well - something like AAC or maybe even OGG? Everyone agrees that the vast majority of the data in 24/192 can be thrown away with zero perceptible loss. Fine, let's do it. But let's do the bit discarding in some principled way, guided by a reasonable psychoacoustic model. Isn't that a lot better than indiscriminately downsampling to 16/44.1? By anyone's lights, a 16/44.1 FLAC at 1100 kbps will not sound better than a 24/192 OGG at 1100 kbps - or even 700 kbps, for that matter. The nice thing about this plan is that we have good models for the human threshhold of detection. Scientists claim that 16/44.1 is so good that any improvements on it will not be detected. Maybe, but what if they're wrong? Why not start with the data rich source and apply our acoustic models to throw out only the data that we know is FAR FAR FAR BEYOND our threshhold of detection? It would still be most of it, but at least we'd know we're throwing out the RIGHT data.

  57. Re:The article writer is a deaf idiot by BobNET · · Score: 3, Funny

    Yeah, what would a guy named xiphmont know about signal processing?!

  58. Re:The article writer is a deaf idiot by DMUTPeregrine · · Score: 5, Informative

    My last hearing test has shown that I can hear up to 21khz. I play Tin Whistle, Great Highland Bagpipe, Ceilidh Pipe, and Guitar. I have heard the rattle of a live sax. I have heard a delicate triangle ringing out over a live orchestra. I have heard live trumpet. I've spent quite a bit of time training my ears to hear those sounds.

    I have consistently failed to find a difference between the following in ABX tests I have run:
    192/24 and 44/16 .wav
    96/24 and 44/16 .wav
    44/16 .wav and FLAC, encoded with the FLAC reference encoder
    My reference tracks have been Pink Floyd's "Time", Sirenia's "Meridian", Bach's "Herz und Mund und Tat und Leben" part 7 conducted by Nikolaus Harnoncourt.
    The reference system was a PC with an Asus Xonar Essence sound card, a Rogue audio Perseus pre-amp, a pair of Rogue M-180 monoblock power amps, and Vandersteen Signature 2ce speakers. (My father's sound system and my PC).

    Of course, msobkow will claim that since I like Highland Bagpipes my hearing is inferior, and I can't hear the differences because he's better than me.

    That said, I do like having music in 192/24. Why? Because I can play with it. I can edit it, there's more headroom. If I feel that "Another Brick in the Wall" just needs a tin whistle part, well, I'll have an easier time editing it in without distortion. But for listening? Nope.

    --
    Not a sentence!
  59. Re:The article writer is a deaf idiot by mug+funky · · Score: 2

    we're talking about sample rates (kHz). you seem to be talking about bit rates (kbps).

  60. Re:The article writer is a deaf idiot by mug+funky · · Score: 4, Interesting

    training doesn't make one's senses better. it trains the observer's brain to relay the appropriate signals, rather than ignoring them.

    i can spot a boom mic in shot almost subliminally. i can spot jitter of all kinds, motion-compensation artifacts, compression artefacts, spots on film (white and black), and can even tell if a cameraman was running out of film, and when the roll was likely to end by looking at the subtle increase in spottiness. other people can't spot these things.

    that said, my eyes are pretty poor. my ears are pretty poor, but i can spot when a (perceptibly) lossy source has been used in a master well before i whip out the spectral view. other people can't.

    that said, decent mp3 (lame preset standard, or even medium) flies by undetected. ditto the equivalent transparent settings in all audio encoders. ditto a decent h.264 compared to the film scans it came off, when viewed with the same chroma sampling (otherwise it'd be cheating to compare 4:4:4 with 4:2:0).

    my wife can tell you every ingredient that goes into a tiny sample of food. i need twice as large a sample to correctly identify only half as many ingredients. my senses are trained (though not as well), but not as sensitive. good thing considering i work in media production, not food.

    my point - you're fooling yourself if you think you have better senses than an average joe - you've just trained you brain to pick different things. they probably enjoy the movie more than you...

  61. Re:The article writer is a deaf idiot by formfeed · · Score: 4, Funny

    On warm summer nights I enjoy sitting on my front porch, with a dry gin made from hand-picked juniper berries, some artisan cheese and bread made out of flour that has been milled before sunrise. And if I am in the mood for it, I also enjoy 192kHz music with my bat friends. For us discerning people this is just a standard of living.

  62. Re:The article writer is a deaf idiot by Prune · · Score: 2

    They only determined there's no immediately detectable conscious difference. Now consider this research: http://jn.physiology.org/content/83/6/3548.full So frequencies we don't consciously notice affect brain activity. Thus your reference is not as conclusive as you imply; still need studies to eliminate the possibility that inaudible frequencies do not impact the brain's perception of audible frequencies in a subtle manner over long listening. I've been suggesting we need long-term listening blind tests with psychological assays for about a decade, but haven't found volunteers that want to go through the trouble.

    --
    "Politicians and diapers must be changed often, and for the same reason."
  63. Re:The article writer is a deaf idiot by mug+funky · · Score: 2

    44.1 was chosen to fit reasonably well in an NTSC video signal... there's some antique A/D converters out there that output composite and intended to use VHS tapes as media.

    48 would have been better, and this was rectified with DVD, but the music industry lags behind...

  64. Re:The article writer is a deaf idiot by Prune · · Score: 2

    Blind tests show that we perceive ultrasound: http://jn.physiology.org/content/83/6/3548.full So I suggest you GTFO. Albeit the effect is not conscious, no one has ruled out that it cannot subtly affect the perception of audible sound over long periods of time to the point where a conscious preference may develop in long term listening, without subjects of a study being able to describe the specific difference. In fact, this is more than plausible, given the reference I posted and others like it.

    --
    "Politicians and diapers must be changed often, and for the same reason."
  65. Re:The article writer is a deaf idiot by MikeBabcock · · Score: 4, Insightful

    Not wanting to go deaf, I use high quality devices with low THD percentages so I can listen at lower volume with maximum impact. Most people don't realize that high volumes are much less necessary as noise is removed and SNR goes up. With a very low noise level, you can play music at relatively low volumes that sounds incredibly good, whereas the high THD injection from a pair of crappy headphones or terrible stereo will cause you to turn up the volume repeatedly to counteract the noise.

    --
    - Michael T. Babcock (Yes, I blog)
  66. Re:The article writer is a deaf idiot by Forever+Wondering · · Score: 2

    The ability of the wealthy to afford large hard drives does not mean file sizes aren't an issue for other less fortunate people. My hard drive is 75 GB and most of that is taken with important stuff, as is my external drive, so there's not much room for music and compression matters quite a lot.

    I think it's time for you to reacquaint yourself with current disk drive pricing. About six months ago, I got some 2TB drives at about $200 each. The 1TB models were half that and the 500GB even less. And, it you can't retrofit internal SATA drives, they have equivalent [self-powered] USB ones. So, I'm guessing $75 would allow you to upgrade your present system.

    --
    Like a good neighbor, fsck is there ...
  67. Re:The article writer is a deaf idiot by macslut · · Score: 4, Funny

    I could maybe save you an additional 50%. I have a friend who is also deaf in one ear. You could go halfsies and spend only $12 on a headphone. Which one of your ears works?

  68. Re:The article writer is a deaf idiot by MikeBabcock · · Score: 2

    No offense, but what was the THD rating on the equipment you used for listening? It really does make a difference. If you listened with a sound card in a PC, you probably lost most of the difference to EM noise.

    --
    - Michael T. Babcock (Yes, I blog)
  69. Re:Twenty years ago, all mp3 encoders were really by arose · · Score: 2

    The mighty wiki disagrees: "The reported completion date of the MPEG-1 standard, varies greatly: a largely complete draft standard was produced in September 1990, and from that point on, only minor changes were introduced.[2] The draft standard was publicly available for purchase.[14]"

    --
    Analogies don't equal equalities, they are merely somewhat analogous.
  70. Its also called a factoid by tkrotchko · · Score: 5, Informative

    Many people think a "factoid" is a small fact. Actually a factoid is something that sounds true, but is actually false.

    --
    You were mistaken. Which is odd, since memory shouldn't be a problem for you
  71. Re:The article writer is a deaf idiot by H0p313ss · · Score: 2

    Though in fairness, I do collect historically-significant Linux distro ISOs.

    Wow, I'm really impressed by that. Do you have the Linux disto that Jefferson wrote the constitution on or the one Hitler used to build the V2 rockets?

    Oh come on, everyone knows that Jefferson ran BSD and Hitler insisted on OS/2.

    --
    XML is a known as a key material required to create SMD: Software of Mass Destruction
  72. Re:Can we stop using the word "truthiness," please by TomHeal · · Score: 2

    You missed the "soulless" AKA Gingers.

  73. Re:The article writer is a deaf idiot by Pieroxy · · Score: 2

    Audiophiles are some of the most amazing people I've ever seen. I've seen some buy $5000 power cords. Yes, that's five thousand dollars.

    These guys should be left alone. Just shield any cable with gold and sell them for a couple of thousand bucks, making a 98% margin. That's what they want!

  74. Re:The article writer is a deaf idiot by Pieroxy · · Score: 2

    No professionally conducted double blind test has found any difference above 16/44. None. Even including people that claimed they could tell the difference before the test weren't able to differentiate anything above 16/44. The only ones that claim that are people that have never taken a properly conducted AB double blind testing.

    Don't you find it intriguing? It's a bit like telepathy. Some claim they are able to do it. But it has never been proven and boy, have there been a number of tests on this subject! This doesn't prevent some mono zygotic twins to claim they could feel their sibling's accident from 1000km away.

    You sound just like them.

  75. Sampling rate by shadowmas · · Score: 3, Insightful

    96KHz isn't the audio frequency. It doesn't mean that the audio contains a 90Khz tone. It's the sampling rate. The higher the sampling rate smoother the signal.

    Human perception wise a audio signal recorded at 96KHz sampling rate might well be indistinguishable from one sampled at 192Khz, but so is the file size between these files for practical purposes. I don't deceive my self thinking that I'm hearing better sound from a 192Khz file, specially considering that I'm using a basic pair of headphones on a my basic phone to listen to them. But my thinking is that future technologies might let you do interesting things with the extra bit of data which is useless to us right now. So given the choice I opt to get the higher sampled versions. Kind of like with digital pictures which are too noisy or blurred, but which might be cleaned up with future algorithms to give us a slightly more useful picture.

  76. Re:The article writer is a deaf idiot by kyrio · · Score: 2

    Your comment points out a huge issue with some sites that release high resolution audio, especially if it's older music.

    For the last decade, people have been upmixing regular stereo CDs to 5.1, and doing it extremely well. There have been many cases where a few years later the studio releases its own 5.1 version, using the original material (supposedly) and it comes out sounding worse than a stereo upmix that some guy made in his basement. You can search Demonoid for classic examples of this happening, or just to get your hands on some of the upmixes, if you're interested (you'll have to be able to play DTS files).

    Back to the point, I wouldn't be surprised if a large amount of the "classic" albums that are released with higher resolutions are just upmixes, which account for situations like HDTracks' Rolling Stones collection being released in multiples of 44. At the very least, it looks like the source material wasn't recorded in the highest quality possible, or maybe at the time the highest possible just wasn't where we are now.

  77. Re:Can we stop using the word "truthiness," please by geekgirlandrea · · Score: 3, Informative

    There was already a perfectly good word for that.

  78. Re:I want my 15 minutes back by gl4ss · · Score: 2

    Then do some double blind tests that show that you can actually hear the difference.

    the guy wants 480khz.

    he'd show the difference with a oscilloscope. would be blatantly obvious there. couldn't hear it of course, but you could show a difference when pumping silence!

    --
    world was created 5 seconds before this post as it is.
  79. No smooth by DrYak · · Score: 5, Informative

    The higher the sampling rate smoother the signal.

    Well... no. There's enough information in a low sampled curve. As TFA explains it, the output isn't "jagged" when played back in analog.

    Human perception wise a audio signal recorded at 96KHz sampling rate might well be indistinguishable from one sampled at 192Khz

    as explained in the article:
    - Yup the human ear won't hear anything aboe 20kHz sounds, because it doesn't have any receptors for that.
    But there are some real-world problems that come into the mix. No audio installation is perfect. You always get distortions.
    - Thus, a 192kHz sampled file could contain frequencies up to 96kHz. These are sound which can't be heard in theory. In practice if you throw 96kHz frequencies to a sub-optimal speaker, the speaker can barf a lot of distortions, including distortion below the the 20kHz. So not only are you trying to output a sound that can be heard, but you force the speaker to produce bad noise *which* is audible.

    But my thinking is that future technologies might let you do interesting things with the extra bit of data which is useless to us right now.

    Hard to do anything with those bits at all. We simply lack the anatomic feature to do anything with them. Unless you do something like transpose everything at lower frequencie (slow down everything 2x = move everything 1 octave lower). At which point you aren't really outputing the original sound anymore. You're simply using the data to produce new sounds that weren't here to begin with.
    The only practical use-case for this would be zoologist studying animals whose sound are beyond the human hear range. In that case "moving everything a couple of octave down" would help the scientist have an approximation with which he can work (to find rythms or other variation that are inaudible in the original frequency range). But that has nothing to do with hearing music made by human, for humans, with instruments designed for human hearing ranges.

    Kind of like with digital pictures which are too noisy or blurred, but which might be cleaned up with future algorithms to give us a slightly more useful picture.

    The situation with pictures is slightly different. What you're speaking about is spacial frequency. I.e.: resolution.
    And human eyes can percieve way much more than some blurry low-res pictures. And in addition to that, there's this thing called zooming which makes perfectly sense to record picture at higher resolution. Because looking at details is simply looking at the same picture at another scale.

    The "visual equivalent" to 192kHz sounds would be recording colours outside the human range. Like recording also infra-reds, microwaves, ultraviolets, and X-Rays.
    Things that can't never been seen, because human lack the corresponding apparatus. The only way to get someting out of this extra data would be to transpose it into the visible domain. Thus use pseudo-colours to display levels of low infrared (heat), etc.
    Just like the "zoologist" use-case above, there are a lot of scientific use-case where that could actually make sense (as an exemple, think about all the data collected by astronomers).
    But in no way is it useful to record X-Rays to enjoy a painting by some known artist. The painting was done by a human painter, for human public, using colours chosen for their effect on an un-aided human visual system, disposed on a canvas in a way which is pleasing to the eyes.
    (Well, okay. I know that some scientist use infra-red or X-ray image of paintings to analyse how they were done, what are the layers underneath or if there's even another picture over which the current one was painted. But these are scientist analysing the paint, so we're agin on the "scientific analysis" use-case).

    24/192 makes sense as an intermediate format to avoid rounding errors, aliasing during filtering, etc.
    There could be also some scientific value to keeping

    --
    "Sufficiently advanced satire is indistinguishable from reality." - [Tips: 1DrYakQDKCQ6y52z6QbnkxHXAocMZJE61o ]
    1. Re:No smooth by L1mewater · · Score: 2

      Where are you buying 192kHz audio? I think maybe you're confusing 192kHz with 192kbps.

    2. Re:No smooth by Anonymous Coward · · Score: 2, Informative

      No, but it is *aliased*. The waveform between two samples is a simple interpolation. It is probably pretty close to the original sound, but there will always be some error too.

      You need to re-read Nyquist. The reason for the 2x minimum limit is to avoid aliasing.

      You don't need the "waveform between two samples" because you're reconstructing the sine wave at the highest frequency those samples represent. Any other waveform will contain harmonics above the limit, and should be filtered out before sampling.

    3. Re:No smooth by Dogtanian · · Score: 2

      No, but it is *aliased*. The waveform between two samples is a simple interpolation.

      Bzzt. No, you're wrong- that's absolutely *not* how Nyquist assumes the wave is going to be reconstructed, it's just a "refined" (but equally wrong) variant of the same widespread "join-the-dots-reconstruction" misconception that the article already explained was wrong.

      As far as I'm aware (correct me if I'm incorrect here), one could in theory get the original signal by taking the "join the dots" version *and* then filtering out all the frequencies above the original range. (That second step being very significant).... or even by applying the same filter to an "impulse train" of value spikes representing the sample values of the original wave.

      --
      "Slashdot - News and Chat Sites Deviant". (Click "homepage" link above for details).
    4. Re:No smooth by Dogtanian · · Score: 3, Informative

      You may not be able to hear the higher frequencies, but when they're sampled with a too low sample rate, you'll be converting waveforms you can hear.

      Nyquist assumes that the signal to be sampled does not contain any frequencies higher than half the sampling rate. Any that exist thus *are* expected to be filtered out beforehand, otherwise aliasing will occur.

      Try it for yourself on paper.

      The "samples" do *not* represent the final "reconstructed" wave (are you suggesting the same "join the dots reconstruction" misconception that most people have about Nyquist?). My understanding of Nyquist (probably incomplete and far from perfect, but still miles better than most people's fundamental misunderstanding) is that this sample output has to be filtered so that all the harmonics above half the sampling rate are removed. Since Nyquist only says you get perfect reconstruction for frequencies up to that limit, there's no contradiction there.

      A "perfect" square wave (which can never actually be created in the real world) has harmonics of infinite frequency, and even a "real-world" as-near-square-as-makes-no-difference-wave will contain very high harmonics. If one was to do a Fourier transform on a square wave, filter out all the frequencies above the human range of hearing, then convert it back to the familiar (spatial domain) wave form, it wouldn't be square any more.

      Therefore, you can't sample a square wave using standard techniques anyway.

      --
      "Slashdot - News and Chat Sites Deviant". (Click "homepage" link above for details).
    5. Re:No smooth by Twinbee · · Score: 4, Interesting

      I often 'zoom' into music (i.e. play it slower) for sheer fun. I often like to hear the details of a tune.

      --
      Why OpalCalc is the best Windows calc
  80. Re:"Truthiness" is a dumb word by Winchy · · Score: 2

    Don't we already have the word "specious" for things that look true but are not? So might "intentionally specious" be a better definition?

  81. Re:Can we stop using the word "truthiness," please by Wraithlyn · · Score: 4, Funny

    Only if your definition of "perfectly good" is "so convoluted that nobody EVER uses it". ;)

    Let's be honest here, verisimilitude exhibits a superlative and ostentatious preponderance of syllables.

    --
    "Mind, as manifested by the capacity to make choices, is to some extent present in every electron." -Freeman Dyson
  82. Re:Why does Photoshop have 16bit colour? by mvdwege · · Score: 4, Informative

    BS. If the overtones of a flute high C and a piccolo high C are both under 22Khz, then sampling at twice that will catch all the overtones, and replaying the sample at the same rate will perfectly reproduce them.

    And if the overtones are over 22Khz, but their lower-order harmonics aren't, the sampling will pick up the harmonics and reproduce them perfectly, even without the existence of the original overtone.

    There is no subjectivity in that. An oscilliscope will show you that the overtones and/or their harmonics are all there.

    The only step that decides whether or not the overtones have any influence is the quality of the low-pass filter. At 44Khz that can be a bit iffy, so using 48Khz to get a little more headroom is nice, but in practice you won't be able to hear a difference with anything above that.

    --
    "I know I will be modded down for this": where's the option '-1, Asking for it'?
  83. Re:44KHz by adolf · · Score: 2

    Comprehension, FTW.

    Saying 20KHz is the upper limit of human hearing, is the same as saying that a human may run no faster than 27MPH. You're arguing against something which, obviously, is completely arbitrary.

    20KHz is a rule of thumb, not a hard-and-fast limit. I'm glad to hear that you can hear up to 24KHz (and yes, it is an annoying sound), but you simply serve to counter-balance all of the other folks in the world who can't hear a lick past 5KHz (yes, really -- there''s lots of 'em).

    (This being Slashdot, I refer you to the MTBF of a hard drive.)

    I myself annoyed the hearing-testers at school when I was a kid, because they'd push the "Go" button on the automated tester and I'd keep giving them a thumbs-up for every progressively-higher tone...even though I could hear them telling me the test was finished and I could see that they'd stopped writing. I have no idea how high my hearing used to go. When I finished my own partially-documented tests, I could still hear the tones from the other testing stations from other kids who followed instructions better.

    I used to hear 38KHz peizo remote controls, plain as day, though quiet. I was only 7 or 8 at the time.

    Can I hear that now? No. Not a chance. I've got a hole around 4KHz, another around 8KHz, and it trails off to nothing lot long after that. The tinnitus takes care of much of the rest, if things are quiet (and if things are loud, it just gets worse in the very long-term).

    Too many concerts, too much time listening to angry music, and too much time playing FOH engineer, along with a few hundred thousand miles driving cars seems to me to be an adequate explanation for the loss in my case.

    I still listen better than most folks, in that I can interpret what I'm hearing to mean a specific mechanical or electrical issue after decades of careful self-training, but I can't always hear everything that they can.

    To this end, I'm in favor of higher sampling rates for recorded music. Why? Because even though I can't hear it anymore, I remember what 38KHz sounds like and if there's any musical information there, some kid will hear it and --hopefully-- enjoy it.

  84. Re:Can we stop using the word "truthiness," please by Kamiza+Ikioi · · Score: 4, Funny

    You willfully leave out nerds, geeks, dorks, and spazzes? Obvious /. bias! ;)

    --
    I8-D
  85. Re:Can we stop using the word "truthiness," please by Anonymous Coward · · Score: 5, Informative

    At that sample rate a 15kHz tone has only three samples. With only three samples there's no way to accurately draw the waveform. With three samples there's no way to discern between a sine wave, a square wave, or a sawtooth wave.

    I wish you guys would get this right. There is absolutely no way you can tell the difference between a 15kHz sine wave, square wave, or sawtooth wave (apart from amplitude, perhaps).

    Sawtooth waves have even and odd harmonics, and square waves only have odd ones. This means that the first harmonic of a 15kHz sawtooth wave would be at 30kHz, and the square's 3rd harmonic would be at 45kHz. As you pointed out, even if you could hear them, you'd have to have damn good speakers to reproduce.

    Three samples is enough to reproduce the 15kHz fundamental per Nyquist.

  86. Re:Can we stop using the word "truthiness," please by dgatwood · · Score: 2

    You're actually wrong. Human ears are relatively good at hearing phase relationships and volume relationships between sounds, as these are key components in determining a sound's direction. Thus, even though you cannot hear the fact that it has turned into a sawtooth wave, you can at least potentially hear that the peak is at the wrong point in time, and you can almost certainly hear that the amplitude is reduced inconsistently from wave to wave.

    This paper is also wrong in its claim that 20 kHz is "generous". It isn't. I've done listening tests and have successfully heard high-pitched whines up to... it was either 22 or 23 kHz (which was where I stopped trying, not where I stopped being able to hear), and I'm not even all that young. Admittedly, this is at relatively high amplitude, but the notion that most people can't hear 20 kHz is just plain wrong, and if you start out with that fundamentally wrong premise, you pretty much have to question all the other assumptions, too.

    They also make the fundamentally incorrect claim that everything below the nyquist limit is sampled perfectly. This is also provably and trivially false. The Nyquist theorem says no such thing. It merely says that signals above that limit will result in "folding", causing aliased frequencies below the limit, which means that any frequency below the Nyquist limit can be captured without aliasing. However, music is not a single frequency in isolation; it is a bunch of frequencies interacting in complex ways. The Nyquist theorem says nothing about the phase of a signal near the Nyquist limit being consistent relative to other signals at lower frequencies, and in fact, it is not. Nor does the Nyquist theorem state that the frequency will be captured in a way that maintains consistent amplitude as you approach the limit; indeed, it isn't.

    Read the Wikipedia article about the Kell factor in display technology, and you'll understand why this is a problem. Notice that with display technology, there is no anti-aliasing filtering involved (because the signal is a known signal that is entirely below the Nyquist limit), so this roughly maps onto what would happen if you could magically create a perfect anti-aliasing filter on the input side. You don't become nearly artifact-free until the frequency you are sampling is about 2/3rds of the Nyquist limit. This is an indisputable fact.

    Admittedly, these artifacts are less objectionable in audio because of the anti-aliasing filtering that occurs (both on input and output), but no filter can magically "fix" that inconsistent amplitude. It represents actual information loss—the signal is equally likely to be a constant 15 kHz tone with constant amplitude as it is to be a signal that varies on either side of 15 kHz with a variable amplitude—and once that precise phase and amplitude information is lost, it is impossible to definitively reconstruct it.

    In other words, this article is just plain wrong, almost top to bottom.

    Besides, the real question is not whether 44.1 kHz is "good enough". It provably isn't, if you care about faithful reproduction over the entire human hearing range. The question is whether the information in the top octave of human hearing is in any way useful or important, to which the answer is "probably not". That's not the same thing as saying that 44.1 kHz or even 48 kHz sampling rate faithfully reproduces the entire range of human hearing, though, but rather it is merely saying that most people don't care about its deficiencies. A 48 kHz sampling rate is "close enough" up to about 16 kHz, which is a broad enough frequency range to be "good enough" for all practical purposes.

    --

    Check out my sci-fi/humor trilogy at PatriotsBooks.