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Can You Really Hear the Difference Between Lossless, Lossy Audio?

CWmike writes "Lossless audio formats that retain the sound quality of original recordings while also offering some compression for data storage are being championed by musicians like Neil Young and Dave Grohl, who say compressed formats like the MP3s being sold on iTunes rob listeners of the artist's intent. By Young's estimation, CDs can only offer about 15% of the data that was in a master sound track, and when you compress that CD into a lossy MP3 or AAC file format, you lose even more of the depth and quality of a recording. Audiophiles, who have long remained loyal to vinyl albums, are also adopting the lossless formats, some of the most popular of which are FLAC and AIFF, and in some cases can build up terabyte-sized album collections as the formats are still about five times the size of compressed audio files. Even so, digital music sites like HDtracks claim about three hundred thousand people visit each month to purchase hi-def music. And for music purists, some of whom are convinced there's a significant difference in sound quality, listening to lossy file formats in place of lossless is like settling for a Volkswagen instead of a Ferrari."

147 of 749 comments (clear)

  1. Depends on the bitrate by Anonymous Coward · · Score: 5, Informative

    Usually if the bitrate is above 256kb/s, i dont notice any difference.
    Ofcourse it still effects some songs (especially the percussion parts).

    1. Re:Depends on the bitrate by jlfose · · Score: 5, Interesting

      It could be dependent on the gear that playback occurs on and the quality of the listener's ears. In watching Stan Lee's new show about "superhumans" it becomes clear that some people have, by training or genetics, better reflexes then the bulk of humanity. On my home gear I can't tell the difference above 160Kbs, but I'm more then willing to believe that some people can, either because they have much better gear to listen to, and/or they have superior hearing.

    2. Re:Depends on the bitrate by Anonymous Coward · · Score: 4, Insightful

      I'd say it depends on what you're listening to.

      Most people, including most slashdot armchair pundits, who listen to Lady Gaga or some similar shit will never notice the difference. However, if you listen to things like Tchaikovsky's "1812 Overture", you will notice just how crappy lossy codecs really are. Especially towards the end.

    3. Re:Depends on the bitrate by Anonymous Coward · · Score: 2, Insightful

      I'd say it depends on the person. Much like there are people who can see more colours, I have no doubt that there also exists people who can perceive subtle differences in sound far better than normal people. In fact, I assume there's probably a term for it, but I don't feel like looking it up.

      So while a lot of audiophiles (or perhaps most) are just saying they can hear differences between lossless and virtually lossless... I assume to look "cool", or whatever the appeal is of self-identifying as an audiophile, there's probably a handful that actually CAN. Not to say those $10000 audio wires aren't a complete scam, but it would be foolish to say that there aren't people who have no problem telling the difference between 256kb/s and lossless.

      I may be wrong of course, but for a while people didn't think tetrachromacy existed either. And like tetrachromacy, synesthesia, or hyperthymesia, I imagine there's a number of people who possess these traits, but simply aren't aware that they do, assuming that everyone does and that it's normal. Although for the last, I imagine that would be a lot easier to determine.

    4. Re:Depends on the bitrate by Anonymous Coward · · Score: 3, Insightful

      That's a very apt description. Genetic factor, age, absence of damage, training to understand the difference/subtleties of overtones, and of course the equipment to playback sounds truly. I found the wired article about Peter Lyngdorf and Steinway building speakers good enough to detect the difference between an American and German manufactured pianos a fascinating read. http://www.wired.com/reviews/2012/10/steinway-lyngdorf-model-ls-concert/

    5. Re:Depends on the bitrate by cayenne8 · · Score: 4, Informative
      Well, the reproduction environment and the equipment makes a lot of difference too.

      I mean, if you're only listening to ear buds (even $$$ ones are limited in bass response, etc), or in a car (one of the worst listening environments conceived)....then sure it won't make a difference, and portability makes a lot of sense too.

      However, in a nice listening environment, with good equipment...it is worth the effort IMHO.

      For instance, I have a pair of Klipschorns ...paired with a couple of the much older models of the Decware SET amps , running mono to each channel..plus an older 15" 800W Klipsch sub, etc......

      Even with my older ears, I can hear differences in recordings and formats. Not as well as I used to be able to, but I figure, WHY would I want anything less than the best I can get for the given time/situation? When listening at home, I rip my music to flac, and have it play on my living room stereo.

      And hey....kinda fun to watch the Flintstones in concert volume on tv too from time to time, or hell, once hooked the MAME machine to it....Robotron 2084 is fun with the room shaking around you.

      God, my neighbors used to hate me when I live in a place where I had to share walls...

      --
      Light travels faster than sound. This is why some people appear bright until you hear them speak.........
    6. Re:Depends on the bitrate by Joce640k · · Score: 5, Informative

      You can actually practice listening to music, it's something you learn.

      Sometimes the difference between two sets of speakers can be as little as one clarinet in the middle of an orchestral piece. On one set it sounds good, on another it doesn't (or it's hardly there at all).

      It's not something you can pick out just by putting on a rap CD for ten seconds and turning the bass up to maximum in a store (which is how most "HiFi" systems are chosen these days and why the manufacturers produce so much garbage).

      --
      No sig today...
    7. Re:Depends on the bitrate by Bengie · · Score: 4, Interesting

      For me, MP3 knocks out a lot of highs no matter the bitrate. Listening to most Jazz really brings out the flaws of MP3.

    8. Re:Depends on the bitrate by Joce640k · · Score: 3, Insightful

      It's not mutually exclusive. Some of us manage to listen to more than one type of music..._including_ classical.

      --
      No sig today...
    9. Re:Depends on the bitrate by Bengie · · Score: 3, Interesting

      No amount of equalizer tuning will fix a bad lossy compression. When I listen to any music with real horn, string, or cymbals, MP3 literally hurts my ears. It will give me an ear ache and a headache after only a minute or two of listening. Other better compression algorithms like ogg will not do this, even at higher volumes. Pop music does not have this issue for me.

    10. Re:Depends on the bitrate by The+Mighty+Buzzard · · Score: 4, Funny

      Hey, it still counts if you're listening to it on an episode of Tom & Jerry.

      --
      Violence is like duct tape. If it doesn't solve the problem, you didn't use enough.
    11. Re:Depends on the bitrate by Panaflex · · Score: 4, Informative

      10 years ago, MP3 encoders couldn't encode decent cymbals and saxophones below 384kbps... it was just a stream of high pitched garbage.

      These days they're both really good encoders. I still prefer AAC over MP3 just because the high freq nuances are better captured, but at AAC@256 and MP3@320, the differences are practically imperceptible to my ears.

      The only time I'd look at lossless music is for Orchestral pieces. Compressed pieces still sound flattened and don't have the wideness because there's a lot more overtones, harmonics and variety of tones in live recordings. Microphones, recordings and engineering have adjusted in the past 5 years to compensate - so recent pieces are not too bad however.

      Like anything, it's best to just try a few different methods and see what sounds best to you.

      --
      I said no... but I missed and it came out yes.
    12. Re:Depends on the bitrate by asliarun · · Score: 5, Informative

      In my humble opinion, this old hoary debate will always remain a debate for several reasons. As you right mentioned, the reproduction environment in most cases is woeful at best. Most speakers are not even full-range to begin with, their cabinets resonate, their drivers cannot often keep up in complex multi-layered music, their passive crossovers do a half-assed job in distributing the sound to the various drivers, and so on. Then, the amps are weak so they start bottoming out and start clipping when the speaker impedance and phase dips sharply in certain frequency bands. Then the electronics, especially the capacitors and power supply cannot keep up. Then the cables are not fat enough or are not shielded enough so they load up the power amp even more. Then the pre-amp adds its own coloration to the already feeble signal coming from the source. Then the DAC does its own thing and further colors or degrades the source signal even more. Then the source adds its own share of noise and jitter to the audio signal that screws up not just the signal quality (bad enough) but even the timing of the music.

      On top of it, the room comes into play. The room adds its own coloration and effect that is often a far bigger factor that the audio system itself - boosting certain frequencies while muddying and deadening others, and even adding echoes, reflections, etc.

      Then there is the human being at the end of the chain. I personally can't even listen above 16KHz, and I have average ears. I suspect many people are like me too, at either end of our audible spectrum. On top of it, we humans hear music very differently - while our audio range may be fairly similar (20hz to 20khz by popular definition), our sensitivity to *variations* in tone and timing varies drastically - many often have off the charts sensitivity to even slightly off-key music (I do) or slightly off-beat music (I do not at all).

      All in all, a decent headphone setup is far far more revealing than a decent audio setup. At a thousand dollars, you can probably assemble a decent headphone, but an audio system will sound atrocious, unless you are willing to spend a whole lot more effort and research in second hand discrete gear OR are willing to do serious DIY.

      Anyway - I also wanted to say one thing - the thing that gets neglected the most in all this is actually the quality of the source recording - or what people call "mastering".

      Most people who say something like "SACDs sound far better than redbook CD" or "vinyl sounds far better than CD" are most likely saying this because a whole lot more care went into recording the SACD or vinyl compared to the cheaper mass market CD or mp3.

      If I look back at all the albums I have purchased or listened to (in whatever format), the one thing that stands out to me personally is that I have found less than 10% of them to be "recorded with care". And I'm not even being picky! Across the board, I can say that recording quality sucks when it comes to rock (which is what I listen to most often) - and I mean all kinds of rock.

      If Neil Young's initiative (and even his Pono device) and Dave Grohl's initiatives are successful in improving the audio quality of music in general, I strongly suspect it will be because recording quality will be done with greater care, not because they decided to use a fancier digital format or use higher number of bits and samples to store their music. While everything becomes a factor by the time the music reaches your ears (heck, by the time it is processed by your brain, you even have to factor in psychoacoustics and gear bias and the "burn-in" syndrome) - the recording quality in general needs to improve (except for the jazz and classical pieces that audiophiles love to love, and are hence recorded with care), and this improvement will arguably make the biggest difference in audio quality.

    13. Re:Depends on the bitrate by c++0xFF · · Score: 2

      The problem is the question. "Can you hear the difference?" can never be answered, exactly for the reasons you suggest. "Can you hear the difference under normal listening conditions?" is a much better question.

      I can already answer that question, too! The answer is: "Maybe, but only if you really try." Throw in a bit of poor acoustics and other real-world situations, and the 5% difference is lost.

    14. Re:Depends on the bitrate by Anonymous Coward · · Score: 4, Insightful

      I'd say it depends on what you're listening to.

      The people who care about the difference aren't even listening to the music. Totally different goals.

      Normal people use their stereo to listen to music.
      Audiophiles use music to listen to their stereo.

    15. Re:Depends on the bitrate by gTsiros · · Score: 3, Insightful

      ignore the DAC the amp the source and everything... ...except the speaker drivers themselves. even the best in the world are wildly non-linear.

      and then there's the air between your ears and the speakers

      another non-linearity

      Best source? .0001% THD. best amp? .0001% THD. Speakers? 1% THD haha good luck.

      --
      Looking for people to chat about multicopters, coding, music. skype: gtsiros
    16. Re:Depends on the bitrate by Cillian · · Score: 2

      We don't need to argue about whether they exist or not, or say they might or probably do exist. Get the people who claim to have magic ears, apply double blind testing, and now we know.

      --
      -- All your booze are belong to us.
    17. Re:Depends on the bitrate by jenningsthecat · · Score: 2

      You can actually practice listening to music, it's something you learn.

      Yes, I've noticed that as I've gotten older. Until I was in my late 40's I never cared much about bass, (the instrument, not the frequency range), in most songs. I heard it, but I felt the song would have been pretty much the same without it. Now, I delight in a good bass line - there's a lot going on there that I simply never heard before. I'm also much, much better at picking out the words a vocalist is singing - lyrics are more meaningful now, because I can hear more of what's being said.

      I've always loved listening to music; had lots of records and CD's, built and modded my own equipment, did listening comparisons between the same CD pressed in different countries, etc. But in mid-life, the depth of my appreciation for music has grown considerably, and I hear so much more detail in it than I used to.

      --
      'The Economy' is a giant Ponzi scheme whose most pitiable suckers are the youngest among us and the yet-unborn.
  2. A lengthy, thorough, and well-explained discussion by EmagGeek · · Score: 5, Funny

    There is a long discussion among very qualified individuals on this subject. You can read it here

  3. Depends on the source by Stentapp · · Score: 5, Insightful

    I am quite sure I prefer a lossy compressed version of a 24 bit, 96 kHz track than a lossless compressed version of a 16 bit, 44.1 kHz track.

    1. Re:Depends on the source by Hatta · · Score: 5, Insightful

      44.1hkz 16bit audio is completely transparent to the human ear. No one has ever been able to detect when a 16bit DAC ADC pair has been placed in a 24/96 audio path.

      Your preference for 24/96 audio as a listener is entirely due to the placebo effect. There are good reasons to master audio in high res, but for listening 16 bit 44.1khz audio is as good as anything.

      --
      Give me Classic Slashdot or give me death!
    2. Re:Depends on the source by fa2k · · Score: 4, Interesting

      Depends on how good the sound engineers are. A lot can be gained by higher resolution and sample rate in the mastering stage, but by using a good low pass filter and dithering (and dithering is not really necessary, http://developers.slashdot.org/story/13/02/27/1547244/xiph-episode-2-digital-show-tell ) basically all audible information is captured in 44.1kHz / 16. Your speakers probably don't go much above 20 kHz anyway, so anything beyond 44.1kHz will only cause distortion (aliasing), see post by MetalliQaZ "Debunked" below.

    3. Re:Depends on the source by fatphil · · Score: 4, Insightful

      > I am quite sure ...

      In other words, you've never done an ABX test and are just spouting ill-informed supposition. The ABX is the gold standard, get back to us once you can distinguish those sources that way with a 95% confidence level.

      --
      Also FatPhil on SoylentNews, id 863
    4. Re:Depends on the source by Rougement · · Score: 2, Insightful

      Mine are flat up to 50kHz. The problem with 44.1kHz is that the highest frequency possible is 22.5kHz, dangerously close to the upper range of human hearing of around 20kHz. Add to that possible DAC high pass filtering artifacts, etc and there's a good argument for moving to a sample rate of 48kHz or higher.

    5. Re:Depends on the source by QRDeNameland · · Score: 5, Informative

      Your preference for 24/96 audio as a listener is entirely due to the placebo effect.

      Well, in all fairness, listeners may actually hear perceptible differences between 24/96 and 16/44.1 audio sources due to different mastering, but of course that says nothing about whether they can actually tell the difference between the two bitrates when everything else is equal.

      This article is a pretty good explanation of why 16/44.1 is as good as anyone needs for playback.

      --
      Momentarily, the need for the construction of new light will no longer exist.
    6. Re:Depends on the source by femtobyte · · Score: 5, Informative

      You sure can hear the difference if you stick a 44.1kHz DAQ in a 96kHz signal chain before filtering out ultrasonic high frequency components (if there are enough to make a difference). The advantage of 96kHz recording isn't that it can capture any more human-audible frequencies than 44kHz can, but that you have a lot more leeway to prevent aliasing of signals above the Nyquist limit down into the audible range (a 25kHz tone sampled at 44kHz results in a spurious, highly audible (25-44/2)=3kHz aliasing signal).

      It's pretty much impossible to build analog frequency filters with a sharp cutoff (e.g. everything below 20kHz and below gets through, everything above 22kHz is -60dB attenuated), so recording at 44.1kHz sampling requires either being absolutely certain the original sound source has minimal high-frequency harmonics, or heavy analog filtering that cuts well into the audible high frequency range. With 96kHz sampling, it's much easier to build an analog filter that gradually rolls off high frequencies between 20kHz and 40kHz (...producing a >40kHz sound is tricky in the first place), preventing aliasing without the filter cutting into the audible range. Once digitized, it's trivial to make a *digital* filter with a perfect frequency cutoff to downsample the 96kHz to aliasing-free 44.1kHz.

    7. Re:Depends on the source by hairyfeet · · Score: 5, Interesting

      You are 100% correct, I have sat in a $100k studio with $5k reference monitors and heard my tracks played back at both 192k and at 44.1k and honestly? Couldn't tell the difference, i really couldn't. And while my midrange hearing may not be the greatest I'm picky as hell when it comes to low end and that is usually the first thing that goes when you compress but standard 44.1k? Couldn't tell the difference which if there was gonna be a difference i would have heard it on that system, it was top notch. I'm sure many here can bring citations showing double blind tests which i have no doubt show its all placebo, because if I can't hear it in a nice studio with the actual live instrument right beside it i doubt seriously anybody is gonna hear a difference with home gear, even high end home gear.

      --
      ACs don't waste your time replying, your posts are never seen by me.
    8. Re:Depends on the source by chipschap · · Score: 5, Informative

      44.1hkz 16bit audio is completely transparent to the human ear. No one has ever been able to detect when a 16bit DAC ADC pair has been placed in a 24/96 audio path.

      Your preference for 24/96 audio as a listener is entirely due to the placebo effect. There are good reasons to master audio in high res, but for listening 16 bit 44.1khz audio is as good as anything.

      As a former audio professional (specialized in location recording of choirs and orchestras) I must agree. But even my aging ears can hear the difference between 44.1 (or 48)kHz 16 bit uncompressed and a typical MP3. Side note: 24-bit has a few audible advantages for music with extremely wide dynamic range (from ppp to fff, say) where 16 bit will struggle a little at the very soft end.

    9. Re:Depends on the source by dgatwood · · Score: 5, Informative

      Speaking as someone who frequently does recording, your comment suggests that no one has done that test with classical music in a properly controlled listening environment using quality gear while giving the test subject the ability to control the volume arbitrarily. When you crank up the volume, the noise floor difference in soft passages alone should make the difference between 16-bit and 24-bit signal paths a dead giveaway, even for someone with moderate to severe hearing loss. It isn't even subtle. Of course, if the person doesn't turn it down for the loud passages, he/she will likely suffer hearing damage, but perhaps that's why he/she has moderate to severe hearing loss in the first place. :-D

      The 44.1 vs. 96 kHz difference is more subtle, requiring someone with top-notch hearing (very rare), headphones that can accurately reproduce frequencies above 20 kHz, and 96 kHz DAC hardware that does not have a bandpass filter starting at 16 kHz. If you fail to verify even one of those requirements, you would expect no one to be able to hear the difference, because there won't be any difference.

      --

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

    10. Re:Depends on the source by Dahamma · · Score: 4, Informative

      No, not at all like 640K.

    11. Re:Depends on the source by QRDeNameland · · Score: 5, Insightful

      kinda like 640K?

      Unless you want to argue that human hearing is improving similarly to Moore's law, then no.

      --
      Momentarily, the need for the construction of new light will no longer exist.
    12. Re:Depends on the source by hedwards · · Score: 5, Informative

      The point of the equipment is that you have quality in reserve as you go through the process of mastering the tracks. The more quality you have in reserve the more you're able to do before you start having to deal with artifacts and other nastiness. As with all such things, you have to think about the order in which you do things and the order in which you throw out data to get the best results.

      The point of buying lossless music isn't so much that it's better for listening to, it's that you can compress it however you like later on without having to worry as much about the sound quality you get. Since you have more data to work with, you can get a better quality at a lower bitrate than if you were starting with an already compressed track.

    13. Re:Depends on the source by hairyfeet · · Score: 5, Interesting

      Well to be really REALLY fair I have noticed it also matter if the original music was recorded in analog or digital, as I've taken some tracks we've cut in a classic studio with the analog 8-track and its really fricking hard to get those to sound really..."right" for want of a better term as its really hard to describe, when it is compared to digital.

      The closest I can get to describing it is this and sorry if you aren't a musician but they'll know of which I speak...you know how you have that great old tube amp for the guitar and it has that nice warm fat feel to it? Notice how the same amp when modeled digitally doesn't doesn't quite have the warmth? Its kinda...artificial sounding? That was the trouble we had, the tapes sounded nice and warm but trying to get that to switch over to digital was fucking hard, frankly it was easier to just cut the tracks again in a digital studio than it was to get the analog tapes to really convert well.

      Sorry if I'm not describing it correctly but music is one of those things where my terminology often fails me, its so hard to describe feelings and emotions and music for me is an emotional expression so I end up having to try to describe how I felt as I listened or played and my vocabulary fails me, the analog was a little fuzzy but it was warm and lived in feeling while trying to convert that to digital something was lost in translation, no other way I know how to say it. the same tracks recorded natively in digital sounded great, analog sounded great, but putting the two together was just something we never could get to work.

      --
      ACs don't waste your time replying, your posts are never seen by me.
    14. Re:Depends on the source by Dahamma · · Score: 3, Insightful

      You don't have to do a personal ABX test when there are many others who have done them and confirmed his statement. In fact, it's a much more powerful statement citing many others than just yourself. One is a statistic and the other is an anecdote.

      And for a MUCH more exhaustive and scientific discussion than any post on this article will ever make (anther post in this thread already linked it, but you must have missed it, and it's a great article): http://people.xiph.org/~xiphmont/demo/neil-young.html

    15. Re:Depends on the source by Jane+Q.+Public · · Score: 3, Interesting

      "There are good reasons to master audio in high res, but for listening 16 bit 44.1khz audio is as good as anything."

      The reasons for having "extra" fidelity in master recordings is the same reason for having high-resolution photos in "raw" format: there is lots more wiggle room for editing while still maintaining good enough fidelity that the end user can't tell the difference.

      For example: take a large (say 16M pixel) 8 x 10 photo, and reduce it to 4 x 5 at 600 dpi. Then take the same photo, edit it (for example, change some colors, remove a cloud from the sky, etc.) and reduce that to the same size and resolution. Even though the resulting photos are higher resolution (at arm's length) than the eye can perceive, they look different.

    16. Re:Depends on the source by juniorkindergarten · · Score: 5, Informative

      I will tell you now that the average person cannot hear to 20khz. Young children can. Anybody who has listened to loud music for any length of time have blown away the top couple of khz of their audio range.
      If you have ever gone to a rock concert and been near the front or gone to most dance clubs and you will have sustained hearing damage. If you have ever left one of these venues with ringing ears, or been around loud machinery and noticed the same, then you have sustained hearing loss. Your hearing will recover mostly after the trauma and that will be indicated by the subsiding of the ringing of your ears.
      If you want to find out how your good/bad hearing is, spend the money and see an audiologist. You will be surprised on to find out what your hearing is really like.

      --
      "Every security scheme that is based on secrets eventually fails." - Steve Jobs
    17. Re:Depends on the source by femtobyte · · Score: 2

      Right, which shows why 96kHz digital sampling *is* critical, even if you immediately digitally downsample on-chip before passing it along to the next device in the processing chain.

    18. Re:Depends on the source by Anonymous Coward · · Score: 4, Informative

      According to Wikipedia the audible range for human hearing is around 130dB. 16 bits can in best case offer a dynamic range of 96 dB, whereas 24 bits offer 144 dB.

      So it should be pretty obvious that you can't fit the entire audible range into 16 bits. This might not be relevant to modern day music. But if you want to record what the ear is actually capable of hearing (not including sound levels above the pain threshold) you will need those 24 bits.

    19. Re:Depends on the source by ozydingo · · Score: 2, Informative

      Two nits to pick:
      1) You can get arbitrarily close but you can't get "perfect" frequency cutoff.
      2) A 25 kHz tone sampled at 44 kHz gives you a 19 kHz tone. Remember the [-pi:0] (or [pi:2*pi]) frequency range comes first.A 41 kHz tone would get you a 3 kHz tone after sampling.

      Otherwise all true, which is why most recording devices do exactly that, sample at a high rate and digitally filter before downsampling to 44.1. But none of that has much to do with whether or not, once you've gotten past the aliasing problem as you say, you can tell the difference between a 44.1 kHz playback and a 96 kHz playback.

    20. Re:Depends on the source by femtobyte · · Score: 5, Informative

      1) Digitally, yes you can. Take the DFT of the data; zero out all components above your frequency cutoff; reconstruct the signal as the sum of below-cutoff frequencies. Voila, a perfect sharp cutoff. The only subtlety is that you can only choose an exact cutoff corresponding to some integral number of cycles in your sampling window, so you can't cutoff at exactly sqrt(e*pi)kHz --- but you do have plenty of wave numbers from which to select a perfect cutoff (increasing with the size of your DFT window).

      2) Untrue: a 44kHz *sampling rate* has a 44/2=22kHz Nyquist cutoff. Frequencies f>22kHz Nyquist limit "wrap around" to f-22kHz difference frequencies.

      But yes, I agree, on the playback side there's no audible difference between a (sufficiently well made) 44.1kHz and 96kHz DAC.

    21. Re:Depends on the source by jonsmirl · · Score: 3, Informative

      When the music gets soft in 16b you have a lot of zeros in front of the number. So you effectively only have a three or four bit signal being fed into the DAC. This is fixed point math, not floating. With 24b you can put all of those zeros in the front and still have eight or more bits to feed into the DAC. This is even more beneficial when the amp implements power supply volume control. PSVC raises the effective noise floor the DAC has to deal with.

    22. Re:Depends on the source by Entropius · · Score: 4, Funny

      Yes! Someday, instead of having real dog whistles, we'll just play back mp3's of dog whistles for our dogs, and those will only work if recorded in 24bit/96kHz!

      Also Monster Cable.

    23. Re:Depends on the source by Entropius · · Score: 5, Interesting

      OT, as a choral performer:

      Classical music has a stupid wide dynamic range, more than any other genre I know of, and (in particular) soprano sections have a nasty talent for pegging meters that were supposed to be set with plenty of headroom.

    24. Re:Depends on the source by ozydingo · · Score: 2

      (This is fun; I know we agree on all substantive points but I'm still going to take you on here :-)>

      1) This is a misconception. The DTFT only represents samples of the DFT, and you can only work with the DTFT with any real machine with finite computing resources. If you zero out the DTFT samples, you are *not* zeroing out all DFT samples in between them.
      Example MATLAB code:
      x = [1 0 0 0 0 0 0 0];
      X = fft(x);
      Y = X; Y(5) = 0; %zero out the highest frequency component
      y = ifft(Y);
      stem(abs(Y,8)) %Look at the pretty DTFT with zero amplitude at the pi frequency component!
      stem(abs(fft(y,256))) %Plot a finer sampling of the DFT. What happened to your perfect cutoff??

      2) True, despite the 22 kHz cutoff. f>22 kHz wraps around to the negative frequency region first. That is, w>pi wraps around to the [-pi:0] region before getting back into the [0:pi] region; remember, we have 2*pi periodicity in the DFT, and 0:pi here represents 0:22 kHz. 22:44 kHz is pi:2*pi, which by periodicity is the same as -pi:0. An aliased, rising tone falls continuously from fs/2 to 0 before rising again.

      Your move, good sir

    25. Re:Depends on the source by arth1 · · Score: 3, Informative

      But yes, I agree, on the playback side there's no audible difference between a (sufficiently well made) 44.1kHz and 96kHz DAC.

      No, but what makes a big difference is when you have a 48 kHz sound card that resamples everything to 48 kHz for an internal DSP stage that cannot be bypassed, and then back again. Yes, Soundblaster Audigy, I'm looking at you.
      44.1 -> 48 kHz gives a lot more audible artifacts precisely because they're so close. Think of it as audible moire.

      Also, for newer computer audio cards, if you have a choice, use 88.2 kHz for the internal rate instead of 96 kHz. The reason is that most high quality sound is in 44.1 which converts perfectly to 88.2. For 48 kHz, it's less of a problem in the first place, and likely also worse quality sound to start with.
      Of course, unless the rest of the audio path is good, it doesn't matter much, but if you like to listen to FLACs with high end headphones, it sure won't hurt to use 88.2 instead of 96 kHz.

    26. Re:Depends on the source by nabsltd · · Score: 5, Insightful

      The closest I can get to describing it is this and sorry if you aren't a musician but they'll know of which I speak...you know how you have that great old tube amp for the guitar and it has that nice warm fat feel to it? Notice how the same amp when modeled digitally doesn't doesn't quite have the warmth?

      The reason for this is that it's hard to capture distortion accurately.

      That "warm sound" is a result of the inacurracies of the tube amp. You may like it better (and that's just fine), but it is does not accurately reproduce the original signal. For me, it's really no different than the current "loudness war" where re-mastered releases are much louder. Many of today's listeners like that sound beter, but it isn't accurate.

    27. Re:Depends on the source by Omestes · · Score: 2

      The point of the equipment is that you have quality in reserve as you go through the process of mastering the tracks.

      This is how I've seen it as well. Its like the difference between a RAW file, and high quality jpeg. The jpeg is good enough for normal use, but you pretty much kill all the information you need for further editing during the compression process. The RAW is your master, but is pretty pointless for for anything else due to its size.

      --
      A patriot must always be ready to defend his country against his government. -edward abbey
    28. Re:Depends on the source by femtobyte · · Score: 3, Informative

      In a finite window, *any* signal can be represented as a sum of elements with frequencies corresponding to n=0 (DC offset), 1, 2, 3, ...., infinity integral cycles in the window. A signal corresponding to a non-integral number of cycles, e.g. 100.5, is indistinguishable over the window from some (infinite) combination of integral cycle waves. If you measured in a window twice as long, the 100.5-cycle signal would now be a unique, identifiable 201-cycle component. So, in an important sense, in a finite window the "intermediate" frequencies "don't exist" --- they can't do anything different from the (infinite series) of integral frequencies. Thus, you can create a cutoff that is as "perfect" as is meaningful in a finite window.

    29. Re:Depends on the source by Goaway · · Score: 2

      Try listening to them and then tell us the difference.

      Just because you can tell the difference between a 2 kHz sine and sawtooth wave does not mean you can do the same at 20 kHz.

    30. Re:Depends on the source by crgrace · · Score: 2

      It's pretty much impossible to build analog frequency filters with a sharp cutoff (e.g. everything below 20kHz and below gets through, everything above 22kHz is -60dB attenuated), so recording at 44.1kHz sampling requires either being absolutely certain the original sound source has minimal high-frequency harmonics, or heavy analog filtering that cuts well into the audible high frequency range. With 96kHz sampling, it's much easier to build an analog filter that gradually rolls off high frequencies between 20kHz and 40kHz (...producing a >40kHz sound is tricky in the first place), preventing aliasing without the filter cutting into the audible range. Once digitized, it's trivial to make a *digital* filter with a perfect frequency cutoff to downsample the 96kHz to aliasing-free 44.1kHz.

      But the fast majority of analog-to-digital converters used for audio use delta-sigma modulation. They are already sampling far above 96 kHz (delta-sigma modulation is a combination of oversampling and quantization noise shaping).

      Your argument is specious. If audio converters used Nyquist-rate ADCs I would agree with you, but they don't. The absolute vast majority of audio ADCs are of delta-sigma type so they are already doing your "trivial digital filter with a perfect frequency cutoff to downsample". It's an inherent part of the modulation.

    31. Re:Depends on the source by Overzeetop · · Score: 3, Informative

      Actually, you've proven the GP's point. You can't tell the difference if you are listening to the program. Turning a program up in the "soft sections" is exactly what you should never, ever do when listening to a program. You may as well put on the IR headset with compression that came with your TV so you can watch late night TV without disturbing your wife.

      Mastering is an entirely different ball of wax and, yes, you want all the headroom you can get. It's no different than photographers using RAW formats instead of JPGs (even lossless JPGs) out of the camera. You want all the bits you can get. But after your done mastering, dropping to 16bits isn't going to affect the outcome. That's the whole point of mastering - if we didn't want to be that soft, we would have engineered it to be louder.

      --
      Is it just my observation, or are there way too many stupid people in the world?
    32. Re:Depends on the source by ChrisMaple · · Score: 2

      Modern (last 20 years) audio ADCs are of the delta-sigma type that effectively sample at a multiple (8 or higher) of the output sample rate. Filtering is applied in the digital domain using FIR filters with very sharp corners and no phase distortion. An analog filter of similar quality is quite simply impossible.

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    33. Re:Depends on the source by dgatwood · · Score: 2

      Turning a program up in the "soft sections" is exactly what you should never, ever do when listening to a program.

      That's not necessarily the case. Consider a classical piece with three movements. The second movement is soft and slow. The entire CD is almost certainly mastered so that the relative volume from one track to the next is preserved—that is, the soft movement will be significantly quieter than the other two movements.

      If you are listening only to the soft movement, however, it is perfectly reasonable to crank up the volume so that you can hear more of the details—details that could easily be buried in the digital noise when listening to a 16-bit recording.

      --

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

    34. Re:Depends on the source by femtobyte · · Score: 4, Interesting

      The trick you're playing on yourself here is:

      x = [1 0 0 0 0 0 0 0]; % x is only defined on 8 samples over the interval. There are an infinite number of continuous signals that could be sampled this way.

      Following your procedure through to y:
      octave:5] y = ifft(Y);
      octave:6] y
      y =
            0.87500 0.12500 -0.12500 0.12500 -0.12500 0.12500 -0.12500 0.12500

      so y is also defined at 8 sample points; as for x, there are an infinite number of curves that could fit these. One of these curves is the sum of frequencies indicated by Y. But what does fft(y,256); mean? From the Matlab documentation,

      "Y = fft(X,n) returns the n-point DFT. fft(X) is equivalent to fft(X, n) where n is the size of X in the first nonsingleton dimension. If the length of X is less than n, X is padded with trailing zeros to length n."

      So, now you have y defined in a larger window (y = 0.87500 0.12500 -0.12500 0.12500 -0.12500 0.12500 -0.12500 0.12500 0 0 0 0 0 .... 0). See my response above to another poster's question: when you enlarge the sampling window, you "create" a lot of possible "intermediate" frequencies that "don't exist" (i.e. are indistinguishable from sums of integral frequencies in the shorter window). By padding y with zeros to a larger window, you're looking at a *different* signal from the un-padded y alone; consequently, you need the "extra frequencies" that you ascribe to the "non-sharp-cutoff" to correctly describe the different "y+0,0,0,0,...,0" signal (which is distinct from y). But that doesn't mean the cutoff isn't perfect as defined on the original signal x->y. In fact, if you periodically *repeat* y (y->y,y,y...,y instead of y->y,0,0,0...) you'll see the "sharp cutoff" still applies since the periodic signal is still the sum of the original frequencies in y.

    35. Re:Depends on the source by djdanlib · · Score: 5, Informative

      As a live sound engineer dealing with vocalists who do that regularly (sing at normal program levels and then BELT A PHRASE OUT)... let me say... ARGH.

      I put a steep compressor on someone who's prone to doing that, and let me tell you, it makes my life much easier. I can't fix the clipping, but I can make sure they don't cause the audience to cover their ears.

    36. Re:Depends on the source by muridae · · Score: 2

      In my 30s, my hearing is still fine above 20kHz. Those 'mosquito' type ring tones that float around the internet are still easily heard by my ears. Except for the ones right around 23kHz, those drop out before coming back at 24 and then fading down from there.

      So, maybe I've preserved my hearing; I doubt it since I've been to concerts with no ear plugs, listened to death metal through ear buds, and hung around server rooms and heavy machinery. Maybe I just started with good high range hearing, I dunno. I did have an audiologist test my hearing years ago, they were surprised by the high range. So while your 'average person' may not be able to hear it, I can hear a ton of rattling in badly compressed 16/44.1 audio. The cymbals and snare drums are horribly distorted compared to lossless. Flutes and some woodwinds suffer the same problem, hell even the harmonics of a soprano can get into the range of 'wow, that's a terrible mix/badly compressed'.

    37. Re:Depends on the source by micromoog · · Score: 3, Informative

      That's correct, there is no audible difference to a human between a 22kHz sine wave and a 22kHz any-other-shape periodic wave. Not to mention, no adult human can hear 22kHz anyway. I hear 16kHz. My 9-year-old can hear 19kHz. Get a frequency generator app and test yourself -- it's fascinating.

    38. Re:Depends on the source by Gr8Apes · · Score: 2

      I can honestly say at this point, I couldn't tell the difference between either of those either at 20kHz and a flat wave, because I can't hear 20kHz, and neither can more than 80% of the human race.

      --
      The cesspool just got a check and balance.
  4. One word: YES. by Anonymous Coward · · Score: 5, Insightful

    Caveat: You have to have decent headphones (not Apple earbud BS), and/or good speakers, but that's about it. The difference is negligible once you hit ~320Kbps MP3, in my opinion, but anything under 256Kbps, regardless of lossy format, you can *clearly* hear cymbal hits turning to an underwater splooshy mess.

    1. Re:One word: YES. by arth1 · · Score: 3, Informative

      Caveat: You have to have decent headphones (not Apple earbud BS), and/or good speakers, but that's about it. The difference is negligible once you hit ~320Kbps MP3, in my opinion, but anything under 256Kbps, regardless of lossy format, you can *clearly* hear cymbal hits turning to an underwater splooshy mess.

      Highhats are even worse than cymbals. Even at 256 kbps, highhats tend to sound like they're being hit with a bag of broken glass, and is the easiest way to identify lossy compression I can think of. Except, perhaps, some of Mike Oldfield's earlier works.

  5. I can hear a slight difference by jgtg32a · · Score: 5, Insightful

    I can't tell which one is better though.

  6. I grew up listening to music on the radio by BenSchuarmer · · Score: 3, Insightful

    ... and scratchy/poppy vinyl records. MP3s on my cheap ear buds are good enough most of the time.

    1. Re:I grew up listening to music on the radio by dugjohnson · · Score: 2

      I grew up with the same thing (AM radio, no less) and I've lost most of my highs in both ears and a lot of everything in my right ear at this point, so mono works fine for me...in fact, listening to some OLD recordings from the sixties and seventies when they really thought that separating the voices into different tracks was cool makes listening on headphones nearly impossible...I get left track only. Although a great take on the backup singers sometimes, depending on the mix. Frankly, if you stand behind me with a drum and a bass, I'm pretty much set for rest of my life.

      --
      My brain is overly lubricated
    2. Re:I grew up listening to music on the radio by Zemran · · Score: 4, Funny

      I listen in the truck with a blown exhaust and whilst getting high on the fumes, lossy or lossless? I have trouble noticing if the car radio is even turned on.

      --
      I love stacking my barbecues in the shed at the end of summer - you can't beat a bit of grill on grill action.
  7. No by Hatta · · Score: 5, Insightful

    No you can't. Not with any reasonably modern encoder and bitrates above 256. Anyone who tells you otherwise is experiencing the placbo effect. BTW, you can't tell the difference between 16bit/44.1khz audio and 24/96 audio either. And vinyl might sound "better" than digital to you, but digital is objectively more accurate.

    Audiophilia is saturated with woo. This is the same market that brought us $500 ethernet cables.

    --
    Give me Classic Slashdot or give me death!
  8. I usually can, but I rarely care. by Clueless+Moron · · Score: 5, Insightful

    I'm listening to a performance, not some audio benchmark. If a bit of loss bothers you, it must be some pretty damned uninspiring music you're listening to.

    And if you're listening on some random mp3 player with bud headphones while walking around doing stuff, compression loss is the least of your worries.

  9. In traffic, a VW will get me someplace by wiredog · · Score: 4, Insightful

    as fast as a Ferrari.

    Since I do most of my listening in a car, and am almost 48, I can't hear the difference between an mp3 and a vinyl album, or a cd, most of the time. Well, except for the lack of skipping. Ever try to listen to an LP in a moving car? But I digress. Sure, people who are younger and $pend lot$ of dollar$ on the Finest Audiophile equipment areound can tell. Me in my Chevy? Not so much.

  10. 44.1khz ought to be enough for anyone... by scorp1us · · Score: 5, Informative

    We recently discovered that human hearing beats the linear response assumptions used in lossy codecs. So yes, their criticisms are scientifically founded.

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    1. Re:44.1khz ought to be enough for anyone... by Hatta · · Score: 5, Insightful

      Unless you have people that can ABX the difference, no their criticisms are not scientifically founded. An actual blind test beats any theoretical reasoning any day.

      --
      Give me Classic Slashdot or give me death!
    2. Re:44.1khz ought to be enough for anyone... by Trepidity · · Score: 2

      In particular, nobody claims that lossy codecs use a perfectly accurate model of human hearing; they don't need to. The goal is to have a psychoacoustic model that captures enough of the general mechanics of hearing, to enable a bunch of constants to be tuned empirically. If the model doesn't come anywhere near to capturing anything important, that would be a problem, because you'd never be able to tune the constants. But once it captures the general outlines, much of the real work on lossy encoders over the past 10-15 years is on tuning a billion constants with listening tests. The goal is empirical transparency (people cannot distinguish the compressed version), not a scientifically valid model of human hearing. Pointing out that there are all sorts of slightly wrong things about the internal model isn't really important if you can't show that they produce audible differences in the end result.

    3. Re:44.1khz ought to be enough for anyone... by ImprovOmega · · Score: 4, Insightful
      Subject:

      44.1khz ought to be enough for anyone...

      Body:

      human hearing beats the linear response assumptions used in lossy codecs. So yes, their criticisms are scientifically founded.

      These have nothing to do with each other.

    4. Re:44.1khz ought to be enough for anyone... by scorp1us · · Score: 2

      Didja read the article? Some people can tell the difference down to one oscillation per second. That's not theoretical.

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  11. Debunked by MetalliQaZ · · Score: 4, Informative

    The concept of improving consumer listening experience using studio quality recording has been thoroughly debunked, right here on Slashdot...
    Why Distributing Music As 24-bit/192kHz Downloads Is Pointless

    --
    "Here Lies Philip J. Fry, named for his uncle, to carry on his spirit"
  12. It doesn't matter by Anonymous Coward · · Score: 5, Insightful

    The reason people use lossless compression for audio (i.e. FLAC or SHN) is not because they can tell the difference. Maybe you think you can, maybe you think you can't, but that's irrelevant anyway. The reason people choose lossless is that lossless is the only suitable solution for archiving. If you want to preserve your CD audio exactly as it appears on the CD, the only possible solution is lossless compression. If you choose lossy, you aren't making an archive or the original, but rather an approximation of the original.

    That's all there is to it.

    1. Re:It doesn't matter by Tamran · · Score: 2

      EXACTLY!

    2. Re:It doesn't matter by xorsyst · · Score: 3, Interesting

      Oh, for mod points.

      While I can't (mostly) tell the difference between the original CD and a ~140Kbs VBR MP3, I _can_ tell the difference between a 140Kbs VBR MB3 made from the CD source, and a 140Kbs VBR MP3 made from a 256Kbs VBR MP3.

      Lossless isn't for listening to, it's for archiving. And make sure you get the cuesheet, pregaps, etc. right when you're archiving too :)

      --
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    3. Re:It doesn't matter by tuffy · · Score: 5, Insightful

      And you never have to re-rip physical discs. 128kb/s CBR MP3 used to be the standard. Then 192 VBR. Then AAC. And so on and so forth. So by keeping a lossless archive, one will always be able to transcode to the latest-and-greatest lossy codec without a lot of hassle.

      --

      Ita erat quando hic adveni.

    4. Re:It doesn't matter by EvanED · · Score: 2

      That's my reasoning too. I've tried to ABX things, and at least with my equipment (I have decent headphones but not everything on the audio path is good) and I've sometimes been unable to discern a 128 kbps OGG from lossless. I'm under no illusion that it provides audible benefits over a

      And yet, even though I already had by CD collection ripped to higher-bitrate MP3s, I re-ripped to FLAC a few years ago. Why?

      (1) I wanted to re-rip anyway, as some CDs had ripping artifacts (clicks and pops) and I wanted to do a secure rip. (I bought dbpoweramp's ripper for this, and it worked quite well. No relation to the company except a pretty satisfied customer.)

      (2) Archiving. Suppose in 10 years some other compression format comes along and supplants MP3. What then? Well, I'd either have to transcode MP3->FutureFormat and suffer bigger losses (perhaps the codecs interact particularly poorly and even though 256 kbps sounds good with both, but the transcode is bad) or do another re-rip anyway. And in that time span, my CDs would be more likely to be damaged, destroyed, etc.

      (3) I want to add good metadata. I'm unhappy with the metadata that comes with most digital music purchases, for example; while in some sense it's more accurate than what I want, it behaves far worse with most players. For instance, I bought a collection of symphonies, and the Album Title of each is "Karajan Symphony Edition", the Artist is always "Herbert von Karajan" or something like that, and the track titles are always like "I. Moderato" -- but the actual composer of the symphony is nowhere present. That's a particularly bad case, but I would much rather have the Album Title be something like "Symphony No. 6", the Artist be "Bruckner, Anton", and then the actual album title and performers be in other metadata fields. So I am very slowly replacing all of the metadata in the tracks I've ripped. This interacts with the previous entry, because if I wanted to re-rip my collection in a different format I'd have to either re-do this work or automatically find correspondences between tracks. (Probably not too hard, but you never know.)

      (4) Storage is cheap. My CD collection takes up less than 100 GB in FLAC. That's way too much for my portable player, but in another few years it probably won't be. In the mean time, doing transcoding from lossless to MP3 or OGG is easy, and doesn't suffer from any additional problems of problem (2), and metadata is preserved well. If all you have is a laptop and no external drive that could be a problem even there, but I'm a space hog anyway so I barely notice that 100 GB.

      (5) There are no real drawbacks -- it's just a very small amount of extra storage cost (but not much; the collection of photos I've taken is three times the size of my music) and a token effort and a bit of CPU time to start an additional transcoding step to OGG for the portable player.

      (6) Given (5), why not? Who knows, maybe I'll have biotic ears implanted in 30 years that will make even lossless CD quality sound like crap. :-)

    5. Re:It doesn't matter by tuffy · · Score: 3, Informative

      Yeah an archive that may never be playable. The point of archiving is preservation, but a lot of good that FLAC archive would do someone who found it in 1000 years while sifting through the remnants of Earth - they will have a lot easier time finding a device that still exists that plays MP3 than they would FLAC or what have you.

      FLAC is about an order of magnitude simpler than MP3. I once implemented a decoder in about an hour over lunch just because I could. And because many lossless codecs feature error detection, they're much more likely to survive as a long-term archive than something like MP3 which doesn't even have a container or any reliable way to verify that the file's contents are correct.

      --

      Ita erat quando hic adveni.

  13. Any good studies? by Experiment+626 · · Score: 4, Interesting

    Anyone know of any good double-blind studies comparing people's ability to tell FLAC from 320kbps MP3? Googling just turns up people debating in forums whether you would be able to tell the difference rather than any serious academic research.

    1. Re:Any good studies? by FrankSchwab · · Score: 4, Interesting

      I don't know if it's a good study, but I did exactly this test. Ten or fifteen years ago.

      I took four musical selections (from the latest Rolling Stones album at the time, a solo piano performance, a classical orchestra, a female vocal), and encoded them at 128, 192, and 256 Kbps with the Fraunhofer codec of the day (remember that?). I re-expanded them to 44.1 KHz CD tracks, and put them on a burned audio CD (remember those?). Each selection on the CD had five versions - the first was always the original bit-for-bit copy from the source CD, then followed (in random order) the 128, 192, 256, and the original again.

      I made ten copies, and handed them out to the audiophiles in the office to play on their home stereos, and gave them a test sheet - I asked them to identify for each selection which version was 128, 192, 256, or the original. Nobody came close to having a "golden ear" that could reliably tell the 128Kbps versions from the others, much less the higher bitrates. Overall, there was a slight ability to detect the 128 kbps versions - it got selected as the lowest quality one more times than random chance would suggest, but even it was still well below 50% (I don't remember the exact numbers any more).

      And this was with ancient MP3 encoders.

      Frankly, if you think you've got the golden ear, first of all I pity you - I'm sorry that you have to put up with all the crap you're going to hear. Second of all, I really recommend running the same test - prepare the tracks, have a friend randomly order them (but keep track), and then see if you can identify them. Don't simply say "Of course I can" - Actually do it and prove it.

      And, if I can be an old man with a bit of advice for a minute: if you can't tell the difference, don't go out of your way to train yourself to tell the difference. It'll just be an annoyance to you for the rest of your life. Kinda like the person who taught me about the reel-change indicators on film at the movie theatres - I see it, and my whole body tenses up waiting for the change. I wish I had never known about it. I really appreciate the change to digital projection so I don't have to deal with those anymore. /frank

      --
      And the worms ate into his brain.
    2. Re:Any good studies? by Tyler+Durden · · Score: 2

      Kinda like the person who taught me about the reel-change indicators on film at the movie theatres - I see it, and my whole body tenses up waiting for the change. I wish I had never known about it.

      You're welcome.

      --
      Happy people make bad consumers.
  14. Oblig by jxander · · Score: 3, Interesting
    --
    This signature is false.
  15. mp3 vs wav by JonathanP.Bennett · · Score: 2

    Yes, I can hear the difference. When working in a small sound recording studio, I trained my ears to pick up on fine details. There was one day in particular I remember listening to a track, and wondering what the strange noise in the background of it was. I realized that I was hearing the audio artifacts from the mp3 compression. Not sure how Mr. Young figures that a CD is only 15% of the master, though. A CD is pure uncompressed audio. If you recorded and mixed in 44.1k audio, then your cd is an exact copy of your master.

  16. Difference is not in the listening. by Anonymous Coward · · Score: 4, Insightful

    The difference is the ability to transcode to different bitrates and formats without losing anything from the original source.

  17. the answer is obvious, isn't it? by v1 · · Score: 3, Insightful

    No you can't. Not with any reasonably modern encoder and bitrates above 256.

    And there's the rub of course. That general of a question can't be answered yes/no. It depends on a variety of factors, most notably the content, the codec, the bitrate, and the playback.

    I don't even know why this article submission got accepted. It's like asking "can you win a race against a Toyoda?" where do you even start with that....?

    --
    I work for the Department of Redundancy Department.
    1. Re:the answer is obvious, isn't it? by rudy_wayne · · Score: 5, Funny

      It's like asking "can you win a race against a Toyoda?" where do you even start with that....?

      Since Akio Toyoda is 30 years older than me, I'm pretty sure I could beat him in a race.

  18. Re:Audiophiles might. by msauve · · Score: 5, Funny

    Mine goes to fiveier.

    --
    "National Security is the chief cause of national insecurity." - Celine's First Law
  19. Sure, you can tell. by jtownatpunk.net · · Score: 3, Insightful

    If you've got decent equipment and a quiet environment. With cheapo earbuds, I don't notice the difference. With my good headphones, the difference is obvious. When I'm driving down the highway, I can't tell. In my living room, I can tell.

    With storage so cheap and bandwidth so plentiful, there's really no reason not to use lossless audio. My $40 Clip+ with a $25 miscrosd card can hold 40 gigs of content and can play FLAC. There's no reason to use a lossy format.

  20. Re:No by Spy+Handler · · Score: 4, Insightful

    Doesn't matter, the audiophile market is not rational (kind of like the wine market). After a certain quality threshold, say 256kbps mp3 or $100 bottle of wine, nobody can tell the difference in a blind test. Yet suckers keep paying money for $500 speaker cables and $1000 bottles of wine. Just stoking ego at that point.

  21. By my estimation... by fahrbot-bot · · Score: 3, Funny

    By Young's estimation, CDs can only offer about 15% of the data that was in a master sound track...

    ... Neil Young is neither a Mathematician or Audio Engineer.

    [ -- insert appropriate Neil Young lyric for satirical effect here -- ]

    --
    It must have been something you assimilated. . . .
  22. Nope, normally. by BLToday · · Score: 3, Insightful

    Nope. Not if the quality is high enough, I can't tell the difference 99% of the times. There are some musical instruments (harpsichord) and singers (Tori Amos) where compression is very obvious. The lossy version becomes almost unlistenable once you've heard the lossless version.

    On "normal" speakers I can rarely tell the difference, but on reference monitors the difference is noticeable on many tracks. Not terrible distracting but still noticeable.

    1. Re:Nope, normally. by Ichijo · · Score: 3, Insightful

      When you listen to music on electrostatic speakers, you can hear things you couldn't hear before. It makes normal speakers sound muffled as if you're listening through a pillow. So the speakers can mean the difference between hearing the mp3 compression and not hearing it.

      --
      Any sufficiently unpopular but cohesive argument is indistinguishable from trolling.
  23. Re:No by rudy_wayne · · Score: 5, Insightful

    In medical tests, people are given a placebo and yet claim to feel better or feel the same effects as people who are given the real medication. These must be the same people who rail against mp3s.

    Just because Neil young and Dave Grohl are famous musicians, it doesn't mean that they actually know what they are talking about. 40 years of exposure to loud music has probably damaged their hearing enough that they really don't know what they are hearing.

    Saying that A sounds better than B is completely subjective and affected by many things. Not just how the music was encoded, but the quality of the DAC used for playback and the quality of the speakers/headphones used.

  24. Trends by mpol · · Score: 2

    There have been more posts on Slashdot in the last 14 years on Slashdot about this topic. What I recall of them, is that people have been tested with blind and double-blind tests. And about ten years ago you could hear a difference between lossless audio and low-bitrate mp3's. The latter has less high and low, and mostly a certain "Hiss" sound through it. The preference was with the lossless audio then.
    What struck me in later tests, was that people seemed to favour mp3's above lossless audio. I reckon it has to do with getting used to the Hiss-sound in mp3's, and therefore having it as a preference. A big factor in music taste is how much you are used to hearing similar music and sounds, and the hiss-sound does make a usual sound.

    To be fair, I do think that mp3's in a high bitrate like 320 kbit are almost as good as lossless audio. Even though I prefer the lossless audio, just to be sure.

    --

    Well, don't worry about that. We can get you back before you leave. (Dr. Who)
  25. Re:Better question by noh8rz10 · · Score: 5, Funny

    it doesn't matter how lossy or lossless the file is if you're listening with shitty white earbuds.

  26. Will hi-def be mastered properly? by steveha · · Score: 4, Insightful

    I would pay more for audio tracks that are mastered properly.

    Far too much of the music released these days is mastered to sound "loud". A sound-level compressor removes the dynamic range, and then the music is gained up about as high as possible, or sometimes higher than that (gained so high there is hard-clipping).

    In the best case, the dynamic range is gone and the music loses some of the drama and impact it should have had. In the worst case, the sine waves are hard-clipped into square waves, which sounds terrible. Hard-clipping adds unpleasant harmonics and distortion and you definitely can hear this.

    I promise you that a properly mastered track at 16-bit/44.1 kHz will sound dramatically better than a poorly mastered one at 24-bit/96 kHz. Mastering trumps format.

    So if they are going to the trouble to make 24-bit/96 kHz tracks, I'm hoping that they will let the mastering engineers do their jobs properly! If they do, I would pay the extra money and bandwidth to buy the music in the higher-quality format.

    The music industry is convinced that most of their customers are idiots, unconcerned about sound quality, who can be distracted by shiny things or loud noises; so they try to make every album as loud as possible. But maybe, just maybe, they will be willing to try something different with the high-quality downloads.

    http://en.wikipedia.org/wiki/Loudness_war

    --
    lf(1): it's like ls(1) but sorts filenames by extension, tersely
    1. Re:Will hi-def be mastered properly? by steveha · · Score: 2

      Considering the lyrical content of what tops the charts, I think the music industry's assumption that their customers are idiots is quite correct.

      It's hard to disagree with your point. But the problem is that the music industry is remastering old music to be loud, as well as mastering the new music to be loud. I am now deeply suspicious when I see "newly remastered!" on a CD label. Once upon a time that was a promise of improved quality; these days it might mean a "loud" master that is actually worse than the original. And they are doing this, not just for death metal bands but for everything. For example, a Billy Joel pop album is not improved by being overcompressed, but:

      http://www.youtube.com/watch?v=3TlQo9k827c

      --
      lf(1): it's like ls(1) but sorts filenames by extension, tersely
    2. Re:Will hi-def be mastered properly? by TAG13 · · Score: 2

      I was going to say something similar to this. "Limiting" music is a much harsher treatment of music than encoding it as an mp3. For those unfamiliar with the audio jargon, limiting refers to squashing the dynamics of the sound. The loudest peaks of the song are brought down, and then the whole song is brought up in volume. The net result is that the quiet parts become louder. The current trend in music, especially pop music, is to severely limit the tracks. The loudness of a sound comes from the average volume of the sound, not the peak volume. So, limiting the track makes the song consistently as loud as possible, and thus the perceived loudness of the song is unnaturally loud. Loud things get people's attention, and thus you have the "loudness wars."

      Besides the distortion that tends to happen in the limiting process, you lose the dynamics of the song. In some songs I think that's fine and can be exciting, but severe limiting is used far too often. Take Dave Grohl's opinions on sound fidelity with a pinch of salt. Perhaps it's not his decision, but the music he puts out has severe limiting. He just made a documentary about a famous music studio, "Sound City." The audio in the film is nice and dynamic, but the soundtrack for the film (which includes original music that was performed in the film) has been limited hard. There's a video that discusses this case that might be interesting to some: http://youtu.be/O3aCNalLojQ

      When audio engineers are mixing and mastering songs for "hi-fi" formats like vinyl and SACD, they are much more delicate with their limiting, if they limit it. I think the hi-fi formats themselves don't offer much value to the listener*, but for songs are treated much better for those hi-fi releases. In the video I posted above, the guy compares the vinyl release of a Foo Fighters album to the CD version. He shows the waveform, so you can visually see the affects of limiting on audio (about 2:45 into the video).

      As for lossy vs. lossless: We've gotten really good with our lossy formats. Sure, it's getting rid of information, but it is carefully chosen information that humans ears don't easily pick up. I rip all of my CDs as 320kbps mp3, and I don't hear any difference. Even at 128kbps, only people really focusing in on sound quality will notice a difference. People listening to stuff in their car, listening to cheap ear buds, or just playing it as background music don't care.

      (*I think hi-fi formats can be great for archiving history. The human ear doesn't pick up on extra information of higher sample rate or analog playback, but there is still extra information there. For people dramatically manipulating the sound, it's often good to have the extra information. Maybe historians will have reason to comb through some of our recordings in the future and analyze the minutiae. I'm sure they'll appreciate the extra information.)

      For what it's worth, I'm study audio engineering in college and will be graduating soon. I've definitely got a lot to learn, but I think my studies of audio give a little weight to my opinions. Limiting and lossy audio is always being discussed in my circles. Hope I offered something useful.

  27. Re:No by noh8rz10 · · Score: 2

    dude, my approach is, so what? somebody worked hard to get a little pot of money, and wants to use the money on something that makes him happy. audiophile stuff makes him feel happy. it wouldn't make me feel happy for the price, but who am i to tell him otherwise? Life got a lot easier once i let people be their own people.

  28. Re:Better question by MarkGriz · · Score: 5, Funny

    Or not using Monster Cable

    --
    Beauty is in the eye of the beerholder.
  29. Re:Better question by Joce640k · · Score: 5, Informative

    This is the real point: People are so used to listening to music with no dynamic range, on ear buds, in crappy acoustic environments that they wouldn't know where to start listening for a difference.

    --
    No sig today...
  30. One question... by Junta · · Score: 2

    I know in imaging that having better than the human eye can see is important in intermediate products as visual manipulation on low fidelity content could produce visible artifacts. Is it the case for audio as well? If someone is going to resample audio for a remix, is there risk of the decreased fidelity ultimately manifesting in the final product?

    --
    XML is like violence. If it doesn't solve the problem, use more.
    1. Re:One question... by ozydingo · · Score: 2

      Yes, for both bit depth and sampling frequency. Here are two possible reasons why:

      1. Bit depth. Remix wants to amplify a sound in the original mix. At 16 bit depth, you have 2^16 possible values to cover everything from silent to max loudness. If you take a soft sound that uses only some of those values and amplify it, the result suffers from possibly noticeable quantization artifacts. This is like magnifying a small picture to produce a pixelated one.
      2. Sample frequency. Remix wants to frequency-shift / pitch-shift a sound in the original mix. Your sampling rate determine the max frequency you can encode, so any audio in a 44.a kHz file has a max frequency range of 22.05 kHz. Say you shift something down by an octave (factor or 1/2); the shifted sound will be cut off at 11.025 kHz.

      How much these effects are noticeable in typical mixes is up to the listener...

  31. Re:No by fatphil · · Score: 3, Insightful

    And if you put them up for a test, and told them which source was which in advance, I'm sure they'd be able to tell you the flaws in the one you said was the mp3 (or whatever). Even if you deliberately swapped the cables over.

    --
    Also FatPhil on SoylentNews, id 863
  32. sometimes, but lossy audio isnt the worst problem by AxemRed · · Score: 2

    I don't think that lossy audio compression is inherently hurting recorded music. Lossy is fine as long as good encoders and sufficient bitrates are used. At a certain point, no one can tell which is which (lossy or lossless) in a blind test.

    I mostly listen to MP3 encoded rock music. The loss of quality is very noticeable to me at 128kbps. The loss of quality is much harder to discern at 192, especially if a quality encoder is used. I use LAME -V 2 when I rip CDs and usually end up with average bitrates from ~190-215, and I can't tell the difference between those MP3s and the original CD.

    IMO there are bigger problems facing recorded music anyway. See: http://en.wikipedia.org/wiki/Loudness_war

  33. Re:A lengthy, thorough, and well-explained discuss by fredrated · · Score: 4, Funny

    You jerk! I clicked on that link!

  34. Re:No by osu-neko · · Score: 3, Funny

    Doesn't matter, the audiophile market is not rational (kind of like the wine market).

    Show me a rational market, and I'll have to inquire as to the nature and evolutionary history of the species of aliens participating in it.

    --
    "Convictions are more dangerous enemies of truth than lies."
  35. AIFF?, Flac!, Lossless in General. & Randomnes by neoshroom · · Score: 3, Interesting

    I've been into compressed lossless audio from the start. First, AIFF is definitely not one of the most popular lossless audio formats for distributing music because the popular formats are compressed lossless audio and AIFF is uncompressed. The top formats are FLAC, APE and ALAC. FLAC is the most popular because it is open-source and versatile. APE was highly popular in the late 90's and early 00's and still is with some because it has better compression than any of the other formats. However, as time went on hard drive space became more plentiful and mobile devices started popping up. APE achieves its superior compression via calculations that are more intensive than FLAC uses and thus more taxing on mobile devices. It is also less cross-platform-compatible. ALAC is Apple's Lossless Audio Codec and is a latecomer onto the scene. It has good iTunes support and slightly better compression than FLAC, but that's about it.

    Also, it is definitely possible to tell lossless audio from lossy audio, even at higher bitrates. Around 2002 I had a friend who completely mocked my lossless ways, even though I'm not one of those gold-cable audiophile people -- just a normal guy who likes his music. I just had a decent pair of Klipsh speakers with a subwoofer. My friend was so certain that this was all in my head and I was so certain that it was not that we devised a simple test. He would show me two identical-looking files in iTunes, just showing the titles. One was a high-bitrate AAC and the other a FLAC file. I could click on them to play them as much as I wanted. I was then to decide which was lossless and which was lossy. We did this with 10 files. It was basically double-blind as he didn't know which was which either until he took the computer back to check my answer. He set up 10 files this way. All in all the test took just 5 or 10 minutes.

    I got 9 of 10 right. It is hard to describe sounds, but the lossless music is "deeper," especially bass, guitar vibrations and high notes. This makes it obvious for many songs.

    However, I expect not everyone has hearing like this. I suspect this because one day I heard this annoying buzzing sound and asked my girlfriend about it. She couldn't hear anything. So, I searched all over for what was causing it. It turned out it was a television that was on, but that was on a non-channel so it was completely black on the screen. However, the CRT television emitted a sound from being on in a silent room that I found annoying and my girlfriend couldn't even hear. My sister could also hear it when I tested her later. I also sometimes find the sounds fluorescent lights make annoying too.

    Anyway, lossless is great and, yes, you can hear the difference if you have hearing which can hear the difference. It's sort of tautological, but it's the truth.

    --
    Big apple, new Yorik, undig it, something's unrotting in Edenmark.
  36. My Torture Test by JBMcB · · Score: 2

    The opening of Royal Oil by the Mighty Mighty Bosstones. It starts out with a quiet snare roll that gets progressively louder, joined by a simple bass line. I've yet to hear a lossy codec at any bitrate that doesn't turn it into watery gibberish.

    Disk space is cheap. Rip to FLAC or ALAC. For portables, 256kbps AAC seems to do the least amount of damage.

    --
    My Other Computer Is A Data General Nova III.
  37. Re:No by mhesd · · Score: 2

    digital is objectively more accurate.

    but music isn't

  38. Re:Better question by Tharkkun · · Score: 4, Insightful

    This is the real point: People are so used to listening to music with no dynamic range, on ear buds, in crappy acoustic environments that they wouldn't know where to start listening for a difference.

    Nor can they afford any better so while they are listening to a lesser quality, they couldn't begin to purchase equipment to give them what these artists say they are missing.

  39. People can't tell the difference above 128kbps by sc0pie · · Score: 2

    Coding Horror did a great experiment with their readers where they provided several samples of the same song at different bitrates and then had everyone vote on which they thought sounded best. The result? People could only tell the difference between 128kbps and everything else, and even that was not overwhelming. In fact, 160kbps beat CD!

  40. Re:Better question by coldfarnorth · · Score: 4, Informative

    Good point. Sadly, my $3k hearing aids don't seem to help either.

    Bitrate doesn't matter much if your ears are the lossy part.

    --
    Lets start refering to The War Against Terror by it's initials. . .
  41. Dodged the bullet by tannhaus · · Score: 2

    Thank God my hearing isn't worth a crap and I don't have yet another thing to geek over.

    As long as Frank Sinatra doesn't sound like Donald Duck, I'm cool with it.

  42. Re:No by zzsmirkzz · · Score: 2

    In medical tests, people are given a placebo and yet claim to feel better or feel the same effects as people who are given the real medication.

    People don't claim to feel better, they do feel better. There is no incentive for them to lie, in fact, there is a disincentive for them to do so. The reason behind the cause of the "placebo" effect is in the mind of the patient. The patient believes they should be getting better and then they do. Power of thought, belief and, if defined correctly, faith. Really, it is the power of consciousness which no one fully understands.

    This can be applied to apparent differences in audio formats. The observer believes that one source should sound better and then it does. Since qualifying better/worse is entirely subjective, objectivity has no place in the argument.

  43. Re:Better question by mwvdlee · · Score: 5, Funny

    Look, you want your 0's and 1's to look like stupid Comic Sans 0's and 1's or like high quality, stylish Zapfino 0's and 1's?

    --
    Slashdot social media options: AIM, ICQ, Yahoo, Jabber and Mobile Text. Why no MySpace?
  44. I knew this article was gonna be BS by SD-Arcadia · · Score: 2, Interesting

    "By Young's estimation, CDs can only offer about 15% of the data that was in a master sound track"
    And nothing of value was lost in the remaining 85% of the *data* that is inaudible to the human ear.

    "Young, in fact, created his own digital-to-analog conversion (DAC) service called Pono. Young has tweeted that the Pono cloud-based music service, along with Pono portable digital-to-analog players, will be available by summer."
    There's your cash-in scheme lurking behind all the BS.

    "Young's service would increase the quality, or sampling rate, of the music from 44,100 times per second in a CD (44.1KHz) to 192,000 times per second (192KHz), and will boost the bit depth from 16-bit to 24-bit."
    I would like to repeatedly hit you over the head with http://people.xiph.org/~xiphmont/demo/neil-young.html

    "The sample rate of a digital file refers to the number of "snapshots" of audio that are offered up every second. Think of it like a high-definition movie, where the more frames per second you have, the higher the quality."
    NO, do not think of it like that unless you're a charlatan. Refer to rebuttal on xiph.org.

    "Millions of people in the world are audiophiles."
    No doubt, Millions of people in the world are fools and they have money that could be yours.

    "It's just common sense that the higher the resolution -- the more data that's in an audio file -- the better the sound quality, Chesky said."
    Too bad this thing called SCIENCE has been trumping "common sense" for millenia now.

    "The site also recommends high-resolution player software such as JRiver, Pure Music, or Decibel Audio Player. The software, which basically turns your desktop or laptop into a music server or a digital-to-analog converter,"
    HILLARIOUS. I won't even begin to..

    "The most popular music server among audiophiles, according to Bliss, is an Apple Mac Mini."
    This is beautiful. I am not surprised in the least to see this audiophile-appleophile overlap.

    --
    https://dalgamotor.wordpress.com/ - Elektronik beyinlere ozgurluk asisi (Turkish)
    1. Re:I knew this article was gonna be BS by pauleir · · Score: 2

      The rebuttal you link to on xiph.org ignores research that illustrates that humans can in fact perceive frequencies far beyond the classical limit of ~20 kHz. Higher frequencies present essential localization cues. Higher sample rates, like 192 kHz., allow for the reproduction of higher frequencies (assuming playback equipment that can actually reproduce the higher frequencies) leading to recordings which are far more realistic than what is possible with the 44.1 kHz sampling rate.

      The difference between 24-bit and 16-bit amplitude resolution is like night and day. As someone that has recorded much contemporary and classical concert music, I can certainly attest to the huge difference between the two bit rates. If you listen to music with a wide dynamic range, then the comparison between the two bit rates is highly noticeable. Quiet sounds can be masked by quantization noise. You want the highest bit rate possible.

  45. Re:No by Waccoon · · Score: 2

    For chiptunes, I can hear a difference between 256 and 320, but just barely.

    The biggest factor is how the high frequencies are filtered out before the audio is compressed, because the filtering appears to be the same regardless of the final bitrate. Even ultra-high bitrate audio will sound awful if the stock frequency cutoff is used, and I have to fiddle with the settings in LAME to make my songs sound good, even at 320.

  46. Re:AIFF?, Flac!, Lossless in General. & Random by evilviper · · Score: 4, Interesting

    we devised a simple test. He would show me two identical-looking files in iTunes, just showing the titles. One was a high-bitrate AAC and the other a FLAC file. [...] I got 9 of 10 right.

    AAC (like MP3) is a frequency-domain codec, and can therefore never provide transparent audio. It has nothing to do with "deeper". but instead is an inability to represent transients... non-tonal components like percussive sounds and other noise.

    If you had performed the test with Musepack/MPC or even MPEG-1 Layer II at high bitrates, you would have failed the test.

    http://en.wikipedia.org/wiki/MPEG-1#Quality

    --
    Slashdot gets worse every day... Pipedot: News for nerds, without the corporate slant
  47. Re:Better question by Joce640k · · Score: 2

    Nor can they afford any better so while they are listening to a lesser quality, they couldn't begin to purchase equipment to give them what these artists say they are missing.

    Plus none of them have a special quiet room where they go to to sit down and do nothing but listen to music.

    They want music/noise constantly and as a backdrop to whatever else they're doing at the time.

    --
    No sig today...
  48. Depends on your ear... by jd.schmidt · · Score: 2

    There was an experiment I heard about on Radio Lab where several pieces of colored paper were given to someone to tell apart, supposedly this would identify someone with 4 color receptors rather than 3 (a small percentage of woman) and was largely analogous to color blindness tests (normally men). In theory people with 3 color receptors would be unable to tell the hues apart.

    What they found is that some people with lots of experience working with color could tell the color samples apart fairly easily, while most people literally could not. A lot had to do with training and life experience apparently. So yes, some people really see more colors than you because they are trained to as incredible as that sounds.

    Sound could be the same way. Plus, depends on your stereo system I guess.

  49. Re:Better question by Minwee · · Score: 2, Funny

    That's a myth. Monster cables are no better than cheaper products from other vendors.

    If you can hear a difference, then it's probably because you have your ethernet cable connected backwards.

  50. Re:A lengthy, thorough, and well-explained discuss by Minwee · · Score: 4, Funny

    You need to go deeper.

  51. Re:Better question by rocketjam · · Score: 2

    Shitty black earbuds okay then?

  52. Why you might care... by Overzeetop · · Score: 2

    No matter how much space is on my current player, I never have enough space to hold my whole collection (well, except on my 160GB iPod classic...but I digress). That means either juggling what is and isn't on my device, or compression, or both. And, its entirely possible that I might choose a player that doesn't work well with the format I've chosen (cough*mp3Pro*cough).

    Having a lossless version of everything means never having to worry about re-compression. My perception trails off between 200-230kbps. I can deal with 192 pretty easily, and 128 isn't the end of the world if I'm in my car or am on a cheap pair of earbuds. Heck, on my SwimP3, 64kbps is overkill. But a 200kbps that then gets re-coded to 128 can really end up with some weird sounding shit. So all my old CDs were ripped to FLAC. When I switched from Creative players to iStuff, I just recoded all of my library from FLAC to ALAC. No loss, no worries, no re-ripping. Most of what I buy today gets ripped straight to ALAC, but if I ever ditch apple, I can just recode it back over to FLAC.

    It matters that you get a lossless format because then you can convert it to any format that works for you. And if you change formats in the future, just re-code and never worry.

    --
    Is it just my observation, or are there way too many stupid people in the world?
  53. Re:Better question by t4ng* · · Score: 5, Informative

    I think the real point is that there are known limits to human hearing and many audiophiles fantasize about their hearing being superhuman. It just ain't so. Dynamic range compression is one thing, but perceptual compression, sample rate, and bit depth are a different matter. No audiophile has ever heard the difference between FLAC and 320Kbps mp3 audio in an ABX test at a statistical rate that is better than guessing.

    Any time this argument starts, I refer people to this well written article that lays out the limits of human hearing compared to the specifications of recording formats...

  54. Re:Better question by Lucas123 · · Score: 2

    Support your local electronics outlet. I buy all my audio and computer cables from You-Do-It Electronics in Needham, Mass. Not only do they have the least expensive, largest variety of cables, they also have actual experts on hand to help -- and no, they don't sell Monster.

  55. There are many factors by thetoadwarrior · · Score: 2

    I do think once you go belo 256 bit rate you start hearing issues or at least you do with some music. But then people also listen to these songs on shitty PC speakers, cheap headphons or worse yet their mobile's speaker. Lossless vs lossy doesn't matter as much when playing the music through poor speakers.

  56. Everybody here is missing a crucial difference by baka_toroi · · Score: 2

    Hearing the difference now isn’t the reason to encode to FLAC. FLAC uses lossless compression, while MP3 is ‘lossy’. What this means is that for each year the MP3 sits on your hard drive, it will lose roughly 12kbps, assuming you have SATA – it’s about 15kbps on IDE, but only 7kbps on SCSI, due to rotational velocidensity. You don’t want to know how much worse it is on CD-ROM or other optical media.

    I started collecting MP3s in about 2001, and if I try to play any of the tracks I downloaded back then, even the stuff I grabbed at 320kbps, they just sound like crap. The bass is terrible, the midrangewell don’t get me started. Some of those albums have degraded down to 32 or even 16kbps. FLAC rips from the same period still sound great, even if they weren’t stored correctly, in a cool, dry place. Seriously, stick to FLAC, you may not be able to hear the difference now, but in a year or two, you’ll be glad you did.

  57. Re:AIFF?, Flac!, Lossless in General. & Random by 0xABADCODA · · Score: 2

    ALAC is Apple's Lossless Audio Codec and is a latecomer onto the scene. It has good iTunes support and slightly better compression than FLAC, but that's about it.

    Apple's ALAC lossless codec is only a dozen C/C++ files (C for the actual codec, C++ for the file format). It's easy to understand, port, and include in other software. To build it you type 'make'. So from a source code perspective ALAC is much better... FLAC has many dozens of source files, assembly, uses automake etc so it's annoying to work with the actual source.

    Not that any of that matter to users, but to programmers ALAC is *much* better.

  58. Re:No by ozydingo · · Score: 2

    In medical tests, people are given a placebo and yet claim to feel better or feel the same effects as people who are given the real medication. These must be the same people who rail against mp3s.

    Don't dis the placebo effect, it works (for some limited benefits), even in cases where the subjects were aware that they were receiving a placebo
    The most similar analogy would be to say that someone can enjoy lossless music more than lossy music. This could be true even if they can't tell them apart in a blind study. Of course, under these assumptions, they'd also enjoy lossy music more than lossless music if the labels were switched and they believed the labels. It's enjoyed more simply because of what it is believed to be. That may be silly, but hey, who am I to crap on someone's enjoyment?
    On the other hand, making the claim that you can tell the difference, i.e. discriminate between then, is more directly challengeable and probably false in most cases.

  59. Re:Better question by ddd0004 · · Score: 2

    Thanks for the pointer, I've had my electrons swimming upstream all along. I also rewired my usb mouse after I discovered that it was wired the wrong way around at the factory. You won't believe the warmth of my lefts, the mellowness of my rights, the dynamic ups and well rounded downs.

  60. 9 out of 10 kids prefer... by Xenna · · Score: 2

    http://www.audioholics.com/news/industry-news/kids-prefer-poor-quality-mp3

    (and remember, kids are able to hear frequencies that you can't!)

  61. DSD is pretty awesome by lophophore · · Score: 2

    I have a DSD (SACD) Player. I have several discs of the same music in CD (red book 44.1 KHz 16-bit) and DSD. DSD is PWM at 2.8 MHz.

    I have done A/B tests with myself, and "blind" tests with friends. Everybody prefers the DSD playback. This is on higher end consumer gear, not high-end audiophile stuff by any means.

    I have no doubt the DSD versions were mastered more carefully. Perhaps that is the biggest difference. However, they do sound better than PCM CDs to my ears.

    --
    there are 3 kinds of people:
    * those who can count
    * those who can't
  62. Re:Better question by blind+monkey+3 · · Score: 2

    it doesn't matter how lossy or lossless the file is if you're listening with shitty white earbuds.

    Dude, use some alcohol wipes on them before you get an infected ear.

    "Modern music" is recorded with much higher gain than previously so having higher quality equipment probably won't make much of a difference if your taste is mainstream. A couple of links:

    The loudness war.
    This made me smile. Why? I was listening to a youtube clip on pc speakers to pick up the affect on sound quality after clipping occurs...... he does explain it well though.

    Disclaimer: Most of my digital music is in flac format - sounds brilliant through my main system at home, not so crash hot on my phone using black earbuds.

    --
    BM3
  63. Re:No by joelpt · · Score: 2

    '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.' ~ http://news.slashdot.org/story/12/03/06/0048259/why-distributing-music-as-24-bit192khz-downloads-is-pointless

    'For me, it is far better to grasp the Universe as it really is than to persist in delusion, however satisfying and reassuring.' ~ Carl Sagan

    Asking for people to behave rationally may not always be the easy way, but in my experience it is almost always worth doing. I think as a species we'd be a lot better off if everyone valued rationality highly, so I think we should encourage that in everyone.

  64. Re:No by Hatta · · Score: 2

    Tetrachromats can pass blind tests. Audiophiles cannot. That's the difference.

    --
    Give me Classic Slashdot or give me death!
  65. Re:Better question by Pieroxy · · Score: 2

    The linked article features $500 for some simple cables. But people can spend MUCH MORE MONEY than $500 on simple cables. For example:

    $699 for 3M of speaker cables: (look for STEREOVOX Firebird Speaker Cables, 3M): http://www.gcaudio.com/products/steals.html

    Ironically, the products are labelled "steals". Very true indeed.

    But there's more. Not all products are "steals". "The next step up is the LectraLine cables priced at $295 for the 1M" http://www.gcaudio.com/products/newArrivals.html

    But it gets better. At musicdirect.com you have power cords for $2,699.99 !!! Obviously it's "The Absolute Sound Golden Ear Award Winner!" Of course. http://www.musicdirect.com/c-650-power-cables.aspx

    But it gets better, again. At nordost they build power cables made out of "99.99999% oxygen free copper conductors." I let you imagine the cost of production. A mere 1.25M of power cord is 8,795.00 (and these are UK pounds, worth more than a dollar). For 5M count 20,495.00 pounds. Yes, that's about $31K !!! http://www.highendcable.co.uk/Nordost%20ODIN%20Power%20Cords.htm

    But it gets better, so much so that it gets boring. But still. Can you spend more than $31k on a simple pair of wires? Well, yes, you can. Look at the bottom of that page, 6M of speaker cable for only $50k. A bargain, really. http://www.audiofederation.com/dealership/prices/nordost/index.htm#prices

    It is astonishing to say the least. That said, it some people have the money...

  66. Re:Better question by rochrist · · Score: 2

    It doesn't take massively expensive equipment to hear differences. Just a decent amplifier and decent speakers.

  67. so record in high def, play back as CD-quality by Chirs · · Score: 2

    As others have said, there are valid reasons to record/mix in high def. But you should be able to downsample the final result to CD-quality with no audible loss in quality.

  68. Re:Better question by Anonymous Coward · · Score: 3, Insightful

    "I think the real point is that there are known limits to human hearing and many audiophiles fantasize about their hearing being superhuman"

    No. The difference between a live acoustic instrument or human voice and a recording is immediately obvious, even to people with significant hearing damage. Waving paper cones around in boxes is not a great way to reproduce sound, it's just all we have with today's technology.

    Audiophiles are not trying to get the last few percent of reproduction quality, they are trying to get some improvement on the terrible quality we have today.

    I say that as a studio engineer with 30 years experience. I do my best, but we are still in the very early days of recording and reproducing sound. Matters have not improved for so long that many people have forgotten how much of a compromise audio reproduction currently is.

    As ever, the hard part is the transducers. Wide bandwidth storage is practical now, but microphones and speakers generate huge amounts of distortion, and have bizarre phase responses and radiation patterns.

  69. Matter of principle by TeknoHog · · Score: 2

    I rip CDs to flac, because I don't want to keep worrying if I could have made a better rip.

    --
    Escher was the first MC and Giger invented the HR department.
  70. Re:Better question by Cillian · · Score: 2

    Genuinely not sure whether joking or audiophile.

    --
    -- All your booze are belong to us.
  71. Audiophily vs. Classical Music by billstewart · · Score: 2

    I was in college before CDs came out, so the audiophile types had vinyl, fancy-for-the-time turntables, high-quality cartridges and needles, etc. One of my housemates liked classical music, and said that once he had a medium-quality stereo system, it didn't make sense to spend more money upgrading the audio quality - it was a lot more important to get records from better orchestras with better conductors. His system was good enough that he could pretty much hear what they were playing, and if you were listening to Beethoven you wanted the Berlin Philharmonic, not the 101 Strings, and you probably had opinions about whether you wanted Furtwangler or von Karajan conducting, and getting rid of that next-to-last bit of distortion wasn't going to fix a lousy recording.

    I mostly listen to music in my car. A decent MP3 is close enough to CD quality when played over road noise, and it doesn't skip when you go over bumps.

    --

    Bill Stewart
    New Fast-Compression-only CPR http://preview.tinyurl.com/dy575ks
  72. Yes. by dradler · · Score: 2

    You can be trained to hear the lossy compression artifacts. But trust me, you don't want to be. Once you can hear them, you can't unhear them.