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Why Distributing Music As 24-bit/192kHz Downloads Is Pointless 841

Posted by Soulskill
from the but-all-those-bits-sound-so-good dept.
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.'"
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Why Distributing Music As 24-bit/192kHz Downloads Is Pointless

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  • by Anonymous Coward on Monday March 05, 2012 @11:11PM (#39256967)

    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?

    • by MobileTatsu-NJG (946591) on Monday March 05, 2012 @11:47PM (#39257219)

      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!

    • by xiphmont (80732) * on Monday March 05, 2012 @11:53PM (#39257261) Homepage

      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.

      • Re: (Score:3, Insightful)

        by Anonymous Coward

        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.

      • by tkrotchko (124118) on Tuesday March 06, 2012 @04:19AM (#39258913) Homepage

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

  • by Anonymous Coward on Monday March 05, 2012 @11:17PM (#39256995)

    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.

  • Pfft. (Score:5, Funny)

    by bmo (77928) on Monday March 05, 2012 @11:25PM (#39257051)

    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

    • Re:Pfft. (Score:4, Insightful)

      by smpoole7 (1467717) on Monday March 05, 2012 @11:37PM (#39257159) Homepage

      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.)

  • by gnu-sucks (561404) on Monday March 05, 2012 @11:25PM (#39257057) Journal
    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.
    • by jmv (93421) on Tuesday March 06, 2012 @12:07AM (#39257363) Homepage

      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.

      • by toejam13 (958243)

        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.

        • by tlhIngan (30335) <(ten.frow) (ta) (todhsals)> on Tuesday March 06, 2012 @01:58AM (#39258161)

          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.

    • 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

      • by jrumney (197329)

        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.

    • by TheGratefulNet (143330) on Tuesday March 06, 2012 @12:30AM (#39257499)

      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.

    • Re: (Score:3, Interesting)

      by bill_mcgonigle (4333) *

      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.

    • 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

  • "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

  • by wbr1 (2538558) on Tuesday March 06, 2012 @12:16AM (#39257419)
    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.
    • by Toonol (1057698) on Tuesday March 06, 2012 @12:40AM (#39257601)
      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.
  • by Animats (122034) on Tuesday March 06, 2012 @01:44AM (#39258061) Homepage

    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.

  • by Dr. Spork (142693) on Tuesday March 06, 2012 @01:50AM (#39258095)
    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.

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