7 ms·

Does pono's 24-bit audio matter? Hear for yourself.

21 points by hendriks 13y ago

tmikaeld 13y ago
It's not only the difference (or none) between 16 or 24-bit, but it's mostly about the quality of the DAC that is being used - in this case a very high-end one. Then again, it's been proven time and time again that the average person can't even hear the difference between mp3 or FLAC. Considering you'd also need proper high-end monitor headphones, i believe this is more about marketing, the brand and the pitch than it is about what you actually get or can actually use. + it can only hold about 2000 FLAC files even with expansion card.
- Shivetya 13y ago
  without the big name attached I doubt it would have had much traction. Its merely a very good marketing exercise. Snake oil for the 21st century. No better than all the joke techies used to make about HDMI cables. I always like Alan Parson's quote, audiophiles use your music to listen to their equipment
  - ANTSANTS 13y ago
    A 24-bit sample depth is not completely useless and silly in the way that gold plated HDMI cables are. You can get a real improvement in dynamic range with 24-bit samples, allowing a song to have quiet parts and loud parts with no loss in quality in either, which is hard to do with 16-bit samples without losing quality for the quiet parts or using dynamic range compression and having everything sound roughly the same. Of course, few people actually want impressive shifts in volume in their music, because prog is dead and no one wants their headphones to suddenly blow their eardrums out. Gotta compress the shit out of that dubstep for the kiddies! Somewhat ironically, greater bit depth in audio makes a bigger difference for film than most music because of this.
- nollidge 13y ago
  And it's a bulky weird triangle shape, which to me is the stupidest part of the whole endeavor.
  - arrakeen 13y ago
    It's not portable in the sense of an iPod or an iPhone where you carry it around with you in your pocket. It's portable in the sense that you keep it on your desk at home, grab it and take it to work, then sit it down on your desk at work. The design is not for everyone, but I think when viewing it through this use case, it has a much more successful design than a flat form-factor like most portable music players
    - nollidge 12y ago
      (i realize this is two days old, but i just saw your reply) But... they could have just made it flat, and then it fits the use case you laid out AND the pocket one. Like, there is zero advantage to the bulky shape.
- calinet6 13y ago
  Another problem is that "hear the difference" is so often defined as "hear the difference between 30 second samples repeated a dozen times." The testing impacts our experience immensely. It isn't equivalent enough to real music, and the human auditory sensory perception and recall systems just aren't perfect enough to make accurate judgements in those test cases. When testing you end up looking for discrepancies you can identify with words and identifiable momentary observations, things like "I can definitely hear this passage the notes are more muddled together in the compressed file." But you miss so many things that show up as minor feelings of unidentifiable hunches. Of course, you could change the format of the test, but even still, "identifying" is not "listening." Things you can not identify, or talk about, or remember, or form sentences about, can still impact your musical experience. Our ears and brains are complex, and the range of input that is perceived subconsciously is astounding. We accept this for vision, for taste, for emotions, for memory, but for some reason not for audio. The simple feelings that you can't put your finger on can be important to the experience even if they can't be identified. But if you start to talk about "realism" and "emotion" and "feeling," you are immediately blacklisted by the empirical measurement mafia. There is so much truth to the fact that scientific measurements and empiricism are important in determining the decisions you make about your audio storage and listening. You shouldn't pay money for things you can prove won't make a difference, and there are so many things out there that fit that bill. But we shouldn't throw out entire possibilities of discussion just because they influence parts of our experience that are subconscious or unidentifiable in an A/B test. Again, I'm not saying we should disregard scientific evidence, just that listening tests set a limit that might not be accurate. We're finding only things that can be tested, and many parts of the audio experience don't fit the test. In this case, there are more bits down there, higher resolution at lower levels. It basically means you can turn up the amplification and receive more resolution regardless of amplified volume. But it also means you have more information, and that's worth something.
  - calinet6 13y ago
    Of course, shut down the infidel. Believers in the church of the empirical only here.
JazCE 13y ago
Surely the test is slightly invalidated by converting an MP3 to wav? surely you would take a master/studio recording and convert that to wav/whatever
- jrabone 13y ago
  The test is most likely invalid - unless told otherwise SoX adds dither. Try adding -D, I expect the results will be different.
jrabone 13y ago
One thing that is alluded to but perhaps not explored fully is the use of noise-shaping dither techniques which rely on pushing the noise out into the ultrasonic region. Xiph mentions it in the context of 16 bit resolution, but I believe it interacts with the discussion of 192 kHz sampling. Significant noise power at ultrasonic frequencies (or even a skewing towards higher audible frequencies) represents a problem for tweeters & amplifiers both in terms of the intermodulation distortion mentioned in Xiph's article and in terms of power dissipation at the business end. IIRC Dell and VLC are having a falling-out because VLC's soft clipping is damaging the speakers in certain Dell laptops.
- mbell 13y ago
  > Significant noise power at ultrasonic frequencies (or even a skewing towards higher audible frequencies) represents a problem for tweeters & amplifiers both in terms of the intermodulation distortion mentioned in the article and in terms of power dissipation at the business end. IIRC Dell and VLC are having a falling-out because VLC's soft clipping is damaging the speakers in certain Dell laptops. Dell probably didn't bother putting an analog reconstruction filter on their DACs, assuming they used an average Realtek codec the digital filter in the DAC has a minimum cutoff point around 28khz but could be higher when operating with higher sampling rates. They probably also sent the signal into a filterless class D amp to drive the output. Those two things together add up to a lot of ultrasonic crap in the signal.
- ferongr 13y ago
  VLC doesn't soft-clip to my knowledge.
  - jrabone 13y ago
    From previous discussion, quoting 'jbk' VLC allows amplification of the INPUT above the sound that was decoded. This is just like replay gain, broken codecs, badly recorded files or post-amplification and can lead to saturation. ... snip ... At worse, this will reduce the dynamics and saturate a lot, but this is not going to break your hardware. Except it can - because the saturation skews the power distribution towards higher frequencies which weren't designed for. This is very, very well known - it's the reason why one always chooses an amplifier with higher output rating than the speakers, often by a factor of two. It's counter-intuitive but driving a low-rated amp into saturation can overheat and destroy tweeters. (Guitar amps get away with it by having massive voice coils on a speaker which will only ever need to reproduce up to about 5kHz.)
ohwp 13y ago
Newborns can hear a SNR of about 120dB which is around 21-bit. So 24-bit seems useless for playing back audio. But I always wonder how much of the sound we hear by feeling (skin, hair vibrations). Because maybe I can't hear 1Hz, but I can very well feel it.
- lmm 13y ago
  > Newborns can hear a SNR of about 120dB which is around 21-bit. So 24-bit seems useless for playing back audio. 21-bit would make for an awkward file format. If we can hear 17 bits of range that's enough to justify storing music as 24-bit.
- valdiorn 13y ago
  You can definitely "experience" the sub-audio range, but you don't need 24 bits to do that. I have very sensitive hearing, I hear things most others cannot. I can detect 192kbps vs 320kbps mp3 encoding. I cannot have any switch-mode AC/DC transformers in my bedroom because the switching noise actually keeps me awake (I charge my phone in the kitchen, most people think I'm mad when I complain about the "noise" :) Still, I've never been able to detect frequencies above 20Khz or hear the difference between 16 and 24 bit. I think anyone claiming to is full of crap.
  - TheLoneWolfling 13y ago
    My laptop whines when I scroll down image-heavy pages. Thus far, I've had a grand total of one other person comment on it.
- acjohnson55 13y ago
  A lot of music is felt through the body. That's simply a result of low frequency at high levels. That's just one reason why live performance is a more dynamic experience than listening over headphones, but adding dynamic range won't create this effect over headphones. 24-bit is pretty much useless for playback. While human hearing range may extend past 16-bits of dynamic range, that doesn't mean that full range is musically useful. It is, however, quite useful to record and mix in 24-bit for the headroom.
wardb 13y ago
Weird that the author uses a 'ordinary MP3' as source for the test. Garbage in, garbage out. Right? Would be a better test with a very high rez audio file.
- microcolonel 13y ago
  I suspect it's heavily compressed audio. His argument is that you'll have a hard time hearing it at all, so if anything's going to help you hear it in the first place, it's garbage in. garbage recordings are the only ones which will survive this test. Anyway, nobody here is both qualified and willing at the same time to tell you what you need to know about this topic. The article referred to, written by Monty at Xiph, should give you a very good overview of how this works . http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html
  - dankoss 13y ago
    > nobody here is both qualified and willing at the same time to tell you what you need to know about this topic. Maybe people who work in professional (not consumer) audio? Who design these systems for a living?
    - microcolonel 13y ago
      Hence "and willing".
pokes 13y ago
The author is likely using a 16-bit audio card to listen to the result, which is going to clip the least significant bits off anyway. If you undo the shifting with sox -v 4096, the quantization noise in the 16-bit version is audible, even on laptop speakers.
- Breakthrough 13y ago
  This is my only issue with the article - it's alluded that the -v and -b flags only perform multiplication and bit shifting. By default, this is not true because sox will use a higher bit-count and perform dithering when converting to a lower bit format (ref http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.html?showComment=1397570941495#c7516054401833345657 http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.ht... ) in order to remove the extra bits while reducing the impact on dynamic range. In all fairness though, this is the same argument used in the Xiph article ( http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html ) in an attempt to justify the use of 16-bit over 24-bit, the prior allowing for a comparable dynamic range with the use of appropriate dithering techniques. I just think it's important to note that the process used with sox is doing more than simple arithmetic bit-shifts/multiplication.
- danbee 13y ago
  It's probably worth pointing out that what you can hear there is dither noise. Quantization noise sounds far worse.
coldpie 13y ago
This test is pretty uninteresting, but I'm glad the author linked to the Xiph article because it's one of the best articles on the subject I've read. http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html
- twobeard 13y ago
  The author admits that "16 bits does not quite cover the entire theoretical dynamic range of the human ear in ideal conditions". But then the author concludes "let's not use 24 bits anyway because it doesn't really help anyway, and it wastes space" Wastes space? In an age where we stream gigabyte movies? Do we need to remind the author that we're no longer in 1982 and that we have evolved beyond floppy disks?
  - joshstrange 13y ago
    >> Wastes space? In an age where we stream gigabyte movies? Do we need to remind the author that we're no longer in 1982 and that we have evolved beyond floppy disks? Clever way of leaving out just how much space it takes up (takes up 6 times the space [0][1]) also as both this author and Xiph concluded you can't really hear the difference. When we stream HD (As you put it "Gigabyte movies") we are getting a clearly superior product. I can easily see the difference between 480/720/1080 whereas I seriously doubt I can hear the difference between 16 and 24bit audio. Yes space is cheaper than it used to be but it's neither free nor unlimited, not to mention the primary way people listen to music nowadays is probably on mobile devices where space is still at a premium or they enjoy their music via a streaming service a la Spotify/Rdio/Google Music where bandwidth is at a premium (And storage factors in here as well due to either being able to store less songs locally or in cache due to larger file size). [0] http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html [1] http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.html http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.ht...
    - ygra 13y ago
      The six times figure is for 192kbps and 24 bit. The move from 16 to 24 bit alone would just result in an increase of 50 %.
rrrooss 13y ago
This test is flawed in a few ways. The and the error lies in the source material. The why would there be a perciveable difference between the same inferior input shown to both encoding formats. Changing between encoding formats is a topic of its own but the problem here is similar to getting a 640x480 compressed jpeg and putting it into a 1920x1080 lossless format and the same source into a 1280x720 lossless file and saying. What's the difference? They both look crappy! Many people have an opinion on high res audio formats but they are so many places to go wrong. Audio should be given a little more credit for it immersive ability. When done right it can take you places. Something to consider is that our eyes can only see 1 octave of information our ears can hear 10 octaves. Whilst these high res formats aren't needed for every application they do make a big difference when the source material can take advantage of it. I feel in 2014 music should be released in the best format available from the studio and if u want a crappier version so u can shove 5000 songs on your iWhatever that's your choice.
- microcolonel 13y ago
  It's not the same as the same inferiour source material. When you apply negative gain to the 16-bit signal, it has less amplitude in absolute terms, but it should have all the resolution of a whole 16-bit sample in that space. The point is that whether that 16-bit signal is represented with the lower 4 bits of a 16-bit sample, or the lower 12 bits of a 24-bit sample, you can scarcely hear anything at all, let alone a qualitative difference between the signals. 192KHz is not a higher quality format, it is a production format. The purpose is to reduce the cost and finality of anti-aliasing filters when sampling the signal, it is just cheaper to manufacture. There's a good reason for this being the standard in professional audio, they need to buy a LOT of audio interfaces, many studios will have thousands of such inputs. Thanks to the basic laws governing signals, we know with near certainty that not only is 192KHz overkill, but so is 48KHz, and so is 44.1. Without significant new evidence showing humans hearing signals with frequencies greater than 24KHz, you will not make any convincing argument as to why we should go with any sample rate higher than 48KHz for human listening. As for 24-bit, it is another production interchange format. It's there so that you don't need to stand around adjusting gain knobs on audio interfaces so that you get decent fidelity but also don't clip. With 24-bit you can just sample your audio once, and assuming it's within a reasonable range, you can adjust the gain in the discrete signal. There is some indication that 16-bit sampling is less than completely ideal. The very best ears in humanity(newborn ears) distinguish about 21 bits in the safe ranges of amplitude, 24-bit may make sense in an audio system for newborn babies. I feel that in 2014, music should be released in the best format available from the studio, and if you want a crappier version so that you can shove only 100 songs on your iWhatever, that's your choice.
dankoss 13y ago
The methodology might be flawed, but the point stands: We can't use the full dynamic range of a 24 bit signal for audio. Great converters have 20-21 bits of range and if we fully use this range, hearing damage will start at maximum volume levels. If the volume is kept the same between the two files, there is a 48dB drop in level - not inaudible, but there are very few source materials that will actually use the extra dynamic range. On top of that due to fletcher munson curves we will only hear content in the 1-4kHz range at that low volume level.
- stinos 13y ago
  Great converters have 20-21 bits of range and if we fully use this range, hearing damage will start at maximum volume levels but that doesn't have anything to do with number of bits? I mean, an audio player system is basically a digital stream sent into a DA converter outputting an analog signal which is then fed into an amplifier where the signal is amplified and then sent into speakers of some sort. No matter if you have an 8bit audio stream, or a 128bit audio stream, the stages afterwards play a much larger role in the hearing damage caused. Also not all DA converters ouput the same level to begin with. Some are -5V to 5V while others (single power supply) can be 0V to 5V, etc etc.
  - dankoss 13y ago
    Hearing damage as a result of loudness; the electronics have little to do with it. If you set up a 96dB (16 bit) system such that you can hear the lowest bit toggling, your maximum volume will be 96dB above that. 96dB SPL is loud but not painful for average listening. If you set up a 20 bit system (120dB) so that you can hear the lowest bit toggling, 120dB above that will most certainly cause hearing damage if used for very long. 144 dB (24 bit) or more SPL will cause near instantaneous damage and pain.
S_A_P 13y ago
It would take highly dynamic music to really hear more bits. The quiet passages is where it makes the difference. Most music is compressed beyond the need for anything other than 40db of dynamic range. I barely can tell the difference between high sample rate 8 or 12 bit audio and 16 bit audio. In some circles I hear people talk about that "gritty" 12 bit sound, but its usually not properly dithered, heavily filtered, 20khz or less sample rate and generally destroyed on purpose. Audio is generally the best arena for snake oil since subjectivity is so high. I cannot believe that Focal Grande Utopia EM loudspeakers(180,000 usd MSRP) exist in quantity, but I know two people who own a set in their (comparatively) modest living room. I have heard them in action, they sound great, but Im not sure that I heard anything in any of the music that I hadnt heard with good headphones or speakers. At what point is it "good enough"? I own plenty of recordings that no amount of hifi audiophilia will help.(many that I recorded myself ;) )
hendriks 13y ago
For the test the quality of the input does not matter. You can conduct the same test with a beautiful 24-bit recording as source, you would still not be able to hear much in the end. And that's the point. When you print a photo in size 9x13cm, does it matter whether you print it in 10,000dpi or 20,000dpi? Hardly, unless you always run around with a microscope. And it does not matter, whether the photo was originally a good picture or not.
- PavlovsCat 13y ago
  > When you print a photo in size 9x13cm, does it matter whether you print it in 10,000dpi or 20,000dpi? Hardly, unless you always run around with a microscope. It can matter when you put it in a scanner though. I tried scanning some old photos, they look horrible compared to even my puny 8mp camera. Then I tried scanning a piece of cloth, and was blown away by the detail..
acomjean 13y ago
I have pretty good headphones (Sony MDR-V6). I can hear decently enough (although getting older). Like I noticed that the Deutche Gramaphone 111 years music collection I bought (256kbps mp3s) has taped performances from the 70 & 80s which have some tape his. Honestly as an end user its really good enough. If I was remixing this stuff and processing the audio I would want the non-compressed. (similar to shooting Raw photographs ). But at 256kbps honestly it really is amazingly good. thinking back to cassette tapes and records, its astonishing. As for the tape hiss on the classical music, the performances are so good I just ignore it. Sometimes I think people get worked up about the quality of the recording and ignore the quality of the performance.
- calinet6 13y ago
  > Sometimes I think people get worked up about the quality of the recording and ignore the quality of the performance. This is the main problem with audiophiles. Improve your equipment until it makes your music awesome, then just enjoy it.
  - danbee 13y ago
    "Normal people use their equipment to listen to music. Audiophiles use music to listen to their equipment."
PavlovsCat 13y ago
re: "It's useless because most people can't hear the difference." Sure, but when sampling and then processing things, you can consider it "headroom". Just like it's not hard to edit a photo in a few steps that the limitations of 8 bits per pixel begin to show. Though I suspect sampling frequency is a lot more important there, 24bit can't possibly be "too much". Yes, I understand that this is not a good argument for consumer products to be more expensive and need more storage, just so musicians of the future, or pirate musicians of today, can have a better time. So I think the best plan is to encourage everybody to become hobby musicians, then it would be an easy sell.
rrrooss 13y ago
There has been a lot of talk about quantative aspects of but little about the qualitative. What does a 18khz square wave look like wher recorded and reproduced in 44.1 kHz. How is phase affecte?. Phase of an audio signal reaching the ears helps you perceive distance and position. How many samples are used to represent the frequencies at the higher end of the audible range. What is the quantization error difference between 16 and 24. And if the author can't hear the difference between 4bit audio and 12bit audio I question what he/she is listening on. The aliasing would be HUGE.
- jrabone 13y ago
  An 18 kHz square wave sampled at 44 kHz looks like an 18 kHz sine wave, everything after the fundamental frequency is well outside the Nyquist limit and will have been thrown away by anti-aliasing filters. And furthermore you couldn't hear it even if it wasn't. Fourier decomposition of a square wave gives the sum of odd multiples of the fundamental, the next frequency is 3 x 18 kHz = 54 kHz.
  - rrrooss 13y ago
    Okay now actually look at the response on Matlab. As you get closer to the nyquist freq there are less samples to describe the wave. And while you get a bastadised 18k signal it's a far throw from what you put in. Anf while most are crying your splitting hairs the frequency response and dynamic range arguments pale in comparison to making the exact wave you put into the encoder come out. On the right system with the right recording the tiniest nuance in a room reverb helps trick your brain into believing the sound as actually happening. And this is not important to all listeners. But people saying that it makes no difference and is not important as a format to release music in are thinking only of their current needs and experiences. If you have actually heard a good recording on a good system in a good room you will know what I'm talking about. If listening experience has been laptop speakers, headphones and your mum and dads mini system then I totally agree any more than 320kbs mp3s are overkill. But to say that high res formats gave no place in consumer land is to show your lack of understanding of different people and different needs. In the age of iTunes you surely you buy the album and download it in whatever format u see fit. If u think audiophiles are wackos go get ya 320's. I would like to get it as I came fr the studio. The best it can be is as it came from the studio.
ntoll 13y ago
"My God! What has sound got to do with music!" ~ Charles E.Ives (https://en.wikipedia.org/wiki/Charles_Ives https://en.wikipedia.org/wiki/Charles_Ives) (and I'm inclined to agree)
raverbashing 13y ago
At the last year's IFA (http://b2b.ifa-berlin.com/en/Home.html http://b2b.ifa-berlin.com/en/Home.html) there was a demonstration of a 24-bit audio player with corresponding high quality audio files and headphones. The results, even though it was a single test, were very good. Of course the issue is, the whole chain has to be 24-bit, if you're going to pump your 128kbps MP3 there the difference is probably negligible (not to mention the 1-bit DACs commonly used, that's one of the elephants in the room) About the MP3 files, yes, the difference is audible. But of course you need good headphones.
danbee 13y ago
So I converted a few tracks to 8-bit with noise shaped dither. Have a listen and remember that the noise floor is now 48dB higher (8-bits) than it would be with 16-bit. https://www.dropbox.com/sh/1h4z0quoybo60c0/O1SwT35Nve https://www.dropbox.com/sh/1h4z0quoybo60c0/O1SwT35Nve
rrrooss 13y ago
More to it than frequency response and dynamic range. The goal should be for the most accurate reproduction of the input waveform. Here is a pic of transient response for different sample rates. http://www.eirec.com/DPimages/digisqwvtest.jpg http://www.eirec.com/DPimages/digisqwvtest.jpg
rrrooss 13y ago
Frequency response and dynamic range are low hanging fruit in high res audio debates. They are not the only factors in accurate reproduction of an input wave. What is rarely considered is the temporal resolution of the format. Which is the formats ability to describe change over time. To plot a graph but depth is the y axis resolution and sample rate the x axis. Again I'm talking to temporal resolution. Transient response is directly effect by this. The link below shows the advantages of hi res audio formats inside the human hearing range. So if accuracy is of concern high res formats DO hold more information about the original wave form. If file size is of concern hi res is not applicable. http://www.eirec.com/DPimages/digisqwvtest.jpg http://www.eirec.com/DPimages/digisqwvtest.jpg This is an example if transient response of different sampling rates.
- dankoss 12y ago
  I tried to find the context for that image but there wasn't anything on the site. The image is most certainly incorrect -- or more likely, looking at the output of the analog electronics after the DAC. Sampling theory says that a perfect square wave can be represented at any frequency below Nyquist. That doesn't mean that the codec or the analog electronics are capable of responding instantly at those frequencies, but that has nothing to do with the fact that a 1kHz square wave can be perfectly sampled with a 44.1kHz sample rate. The image is simply incorrect. Transient response in the real world is generally limited only by the acoustic transducer response of the system, because everything else has the ability to respond much faster than audio rates. With Pono, this means that the earbuds or headphones you use with the player will have a greater affect on the transient response than the electronics inside.
  - rrrooss 12y ago
    This image is from RME a sound card maufaturer. And the image is of the analog output after dac. So still in the electrical domain. Short comings of a playback system has what to do with the accuracy of a recoding medium? There are SO many factors that stop and audio signal reproduction from being perfect. But the signal domain is the easiest to make better. Transducers that have to fight the law of physics for accuracy are the obvious weak point. But as with anything rubish in = rubbish out. Before you even get to the weakest point of an audio system the transducer (the speaker) transistors and amplification have their own short comings the accuracy of an amplifiers transient response is measured by its slew rate. A square wave is impossible for a speaker cone to reproduce in theory the rising and falling edge is describing an instantaneous movement. A speaker cone can't do this, inertia says no. And this is called distortion. As every part of the system introduces its own distortion the accuracy gets less and less. To accept distortion at the signal level in the persuit of accuracy is counterintuive. But if your system can't take advantage of this extra accuracy, go ahead and use a lesser format, no sense in the extra file size. If you want a reference quality signal, use the high res formats. To be clear here I'm not saying I want to listen to digitally encoded square waves, a square wave is a tool for showing the accuracy of input=output. A square wave being the hardest analog waveform to capture and reproduce. I am objecting to the idea that hi res formats have no advantage over what we are used to as consumers. It is the content distributors that are taking advantage saying that 24bit/96k is more expensive. The bandwidth and storage does not make this a more premium product you should be paying for the album not the format. But why limit choice because the masses can't see the benefits. If you buy the album you as the consumer can choose the format. I'm not sure if all this arguing stands up to pick whatever format suits your needs. This argument should be if I bought the album why can I choose the format most applicable to my playback needs/desires. And if your concern is that company's charge more for the high res, that refeclts poorly of the distribution company. If you think it's a waste of space just download the lesser format. If I had a choice I'd download it as it left the studio/mastering house.
sp332 13y ago
Did anyone else actually try this? I thought the differences were quite noticeable. I ran it through Foobar 2000's ABX Comparator and I got the right answer 5 out of 5 times.

21 points by hendriks 13y ago

It's not only the difference (or none) between 16 or 24-bit, but it's mostly about the quality of the DAC that is being used - in this case a very high-end one. Then again, it's been proven time and time again that the average person can't even hear the difference between mp3 or FLAC. Considering you'd also need proper high-end monitor headphones, i believe this is more about marketing, the brand and the pitch than it is about what you actually get or can actually use. + it can only hold about 2000 FLAC files even with expansion card.

Shivetya 13y ago
without the big name attached I doubt it would have had much traction. Its merely a very good marketing exercise. Snake oil for the 21st century. No better than all the joke techies used to make about HDMI cables. I always like Alan Parson's quote, audiophiles use your music to listen to their equipment
- ANTSANTS 13y ago
  A 24-bit sample depth is not completely useless and silly in the way that gold plated HDMI cables are. You can get a real improvement in dynamic range with 24-bit samples, allowing a song to have quiet parts and loud parts with no loss in quality in either, which is hard to do with 16-bit samples without losing quality for the quiet parts or using dynamic range compression and having everything sound roughly the same. Of course, few people actually want impressive shifts in volume in their music, because prog is dead and no one wants their headphones to suddenly blow their eardrums out. Gotta compress the shit out of that dubstep for the kiddies! Somewhat ironically, greater bit depth in audio makes a bigger difference for film than most music because of this.
nollidge 13y ago
And it's a bulky weird triangle shape, which to me is the stupidest part of the whole endeavor.
- arrakeen 13y ago
  It's not portable in the sense of an iPod or an iPhone where you carry it around with you in your pocket. It's portable in the sense that you keep it on your desk at home, grab it and take it to work, then sit it down on your desk at work. The design is not for everyone, but I think when viewing it through this use case, it has a much more successful design than a flat form-factor like most portable music players
  - nollidge 12y ago
    (i realize this is two days old, but i just saw your reply) But... they could have just made it flat, and then it fits the use case you laid out AND the pocket one. Like, there is zero advantage to the bulky shape.
calinet6 13y ago
Another problem is that "hear the difference" is so often defined as "hear the difference between 30 second samples repeated a dozen times." The testing impacts our experience immensely. It isn't equivalent enough to real music, and the human auditory sensory perception and recall systems just aren't perfect enough to make accurate judgements in those test cases. When testing you end up looking for discrepancies you can identify with words and identifiable momentary observations, things like "I can definitely hear this passage the notes are more muddled together in the compressed file." But you miss so many things that show up as minor feelings of unidentifiable hunches. Of course, you could change the format of the test, but even still, "identifying" is not "listening." Things you can not identify, or talk about, or remember, or form sentences about, can still impact your musical experience. Our ears and brains are complex, and the range of input that is perceived subconsciously is astounding. We accept this for vision, for taste, for emotions, for memory, but for some reason not for audio. The simple feelings that you can't put your finger on can be important to the experience even if they can't be identified. But if you start to talk about "realism" and "emotion" and "feeling," you are immediately blacklisted by the empirical measurement mafia. There is so much truth to the fact that scientific measurements and empiricism are important in determining the decisions you make about your audio storage and listening. You shouldn't pay money for things you can prove won't make a difference, and there are so many things out there that fit that bill. But we shouldn't throw out entire possibilities of discussion just because they influence parts of our experience that are subconscious or unidentifiable in an A/B test. Again, I'm not saying we should disregard scientific evidence, just that listening tests set a limit that might not be accurate. We're finding only things that can be tested, and many parts of the audio experience don't fit the test. In this case, there are more bits down there, higher resolution at lower levels. It basically means you can turn up the amplification and receive more resolution regardless of amplified volume. But it also means you have more information, and that's worth something.
- calinet6 13y ago
  Of course, shut down the infidel. Believers in the church of the empirical only here.

JazCE 13y ago

Surely the test is slightly invalidated by converting an MP3 to wav? surely you would take a master/studio recording and convert that to wav/whatever

jrabone 13y ago
The test is most likely invalid - unless told otherwise SoX adds dither. Try adding -D, I expect the results will be different.

jrabone 13y ago

One thing that is alluded to but perhaps not explored fully is the use of noise-shaping dither techniques which rely on pushing the noise out into the ultrasonic region. Xiph mentions it in the context of 16 bit resolution, but I believe it interacts with the discussion of 192 kHz sampling. Significant noise power at ultrasonic frequencies (or even a skewing towards higher audible frequencies) represents a problem for tweeters & amplifiers both in terms of the intermodulation distortion mentioned in Xiph's article and in terms of power dissipation at the business end. IIRC Dell and VLC are having a falling-out because VLC's soft clipping is damaging the speakers in certain Dell laptops.

mbell 13y ago
> Significant noise power at ultrasonic frequencies (or even a skewing towards higher audible frequencies) represents a problem for tweeters & amplifiers both in terms of the intermodulation distortion mentioned in the article and in terms of power dissipation at the business end. IIRC Dell and VLC are having a falling-out because VLC's soft clipping is damaging the speakers in certain Dell laptops. Dell probably didn't bother putting an analog reconstruction filter on their DACs, assuming they used an average Realtek codec the digital filter in the DAC has a minimum cutoff point around 28khz but could be higher when operating with higher sampling rates. They probably also sent the signal into a filterless class D amp to drive the output. Those two things together add up to a lot of ultrasonic crap in the signal.
ferongr 13y ago
VLC doesn't soft-clip to my knowledge.
- jrabone 13y ago
  From previous discussion, quoting 'jbk' VLC allows amplification of the INPUT above the sound that was decoded. This is just like replay gain, broken codecs, badly recorded files or post-amplification and can lead to saturation. ... snip ... At worse, this will reduce the dynamics and saturate a lot, but this is not going to break your hardware. Except it can - because the saturation skews the power distribution towards higher frequencies which weren't designed for. This is very, very well known - it's the reason why one always chooses an amplifier with higher output rating than the speakers, often by a factor of two. It's counter-intuitive but driving a low-rated amp into saturation can overheat and destroy tweeters. (Guitar amps get away with it by having massive voice coils on a speaker which will only ever need to reproduce up to about 5kHz.)

ohwp 13y ago

Newborns can hear a SNR of about 120dB which is around 21-bit. So 24-bit seems useless for playing back audio. But I always wonder how much of the sound we hear by feeling (skin, hair vibrations). Because maybe I can't hear 1Hz, but I can very well feel it.

lmm 13y ago
> Newborns can hear a SNR of about 120dB which is around 21-bit. So 24-bit seems useless for playing back audio. 21-bit would make for an awkward file format. If we can hear 17 bits of range that's enough to justify storing music as 24-bit.
valdiorn 13y ago
You can definitely "experience" the sub-audio range, but you don't need 24 bits to do that. I have very sensitive hearing, I hear things most others cannot. I can detect 192kbps vs 320kbps mp3 encoding. I cannot have any switch-mode AC/DC transformers in my bedroom because the switching noise actually keeps me awake (I charge my phone in the kitchen, most people think I'm mad when I complain about the "noise" :) Still, I've never been able to detect frequencies above 20Khz or hear the difference between 16 and 24 bit. I think anyone claiming to is full of crap.
- TheLoneWolfling 13y ago
  My laptop whines when I scroll down image-heavy pages. Thus far, I've had a grand total of one other person comment on it.
acjohnson55 13y ago
A lot of music is felt through the body. That's simply a result of low frequency at high levels. That's just one reason why live performance is a more dynamic experience than listening over headphones, but adding dynamic range won't create this effect over headphones. 24-bit is pretty much useless for playback. While human hearing range may extend past 16-bits of dynamic range, that doesn't mean that full range is musically useful. It is, however, quite useful to record and mix in 24-bit for the headroom.

wardb 13y ago

Weird that the author uses a 'ordinary MP3' as source for the test. Garbage in, garbage out. Right? Would be a better test with a very high rez audio file.

microcolonel 13y ago
I suspect it's heavily compressed audio. His argument is that you'll have a hard time hearing it at all, so if anything's going to help you hear it in the first place, it's garbage in. garbage recordings are the only ones which will survive this test. Anyway, nobody here is both qualified and willing at the same time to tell you what you need to know about this topic. The article referred to, written by Monty at Xiph, should give you a very good overview of how this works . http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html
- dankoss 13y ago
  > nobody here is both qualified and willing at the same time to tell you what you need to know about this topic. Maybe people who work in professional (not consumer) audio? Who design these systems for a living?
  - microcolonel 13y ago
    Hence "and willing".

pokes 13y ago

The author is likely using a 16-bit audio card to listen to the result, which is going to clip the least significant bits off anyway. If you undo the shifting with sox -v 4096, the quantization noise in the 16-bit version is audible, even on laptop speakers.

Breakthrough 13y ago
This is my only issue with the article - it's alluded that the -v and -b flags only perform multiplication and bit shifting. By default, this is not true because sox will use a higher bit-count and perform dithering when converting to a lower bit format (ref http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.html?showComment=1397570941495#c7516054401833345657 http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.ht... ) in order to remove the extra bits while reducing the impact on dynamic range. In all fairness though, this is the same argument used in the Xiph article ( http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html ) in an attempt to justify the use of 16-bit over 24-bit, the prior allowing for a comparable dynamic range with the use of appropriate dithering techniques. I just think it's important to note that the process used with sox is doing more than simple arithmetic bit-shifts/multiplication.
danbee 13y ago
It's probably worth pointing out that what you can hear there is dither noise. Quantization noise sounds far worse.

coldpie 13y ago

This test is pretty uninteresting, but I'm glad the author linked to the Xiph article because it's one of the best articles on the subject I've read. http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html

twobeard 13y ago
The author admits that "16 bits does not quite cover the entire theoretical dynamic range of the human ear in ideal conditions". But then the author concludes "let's not use 24 bits anyway because it doesn't really help anyway, and it wastes space" Wastes space? In an age where we stream gigabyte movies? Do we need to remind the author that we're no longer in 1982 and that we have evolved beyond floppy disks?
- joshstrange 13y ago
  >> Wastes space? In an age where we stream gigabyte movies? Do we need to remind the author that we're no longer in 1982 and that we have evolved beyond floppy disks? Clever way of leaving out just how much space it takes up (takes up 6 times the space [0][1]) also as both this author and Xiph concluded you can't really hear the difference. When we stream HD (As you put it "Gigabyte movies") we are getting a clearly superior product. I can easily see the difference between 480/720/1080 whereas I seriously doubt I can hear the difference between 16 and 24bit audio. Yes space is cheaper than it used to be but it's neither free nor unlimited, not to mention the primary way people listen to music nowadays is probably on mobile devices where space is still at a premium or they enjoy their music via a streaming service a la Spotify/Rdio/Google Music where bandwidth is at a premium (And storage factors in here as well due to either being able to store less songs locally or in cache due to larger file size). [0] http://people.xiph.org/~xiphmont/demo/neil-young.html http://people.xiph.org/~xiphmont/demo/neil-young.html [1] http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.html http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.ht...
  - ygra 13y ago
    The six times figure is for 192kbps and 24 bit. The move from 16 to 24 bit alone would just result in an increase of 50 %.

rrrooss 13y ago

This test is flawed in a few ways. The and the error lies in the source material. The why would there be a perciveable difference between the same inferior input shown to both encoding formats. Changing between encoding formats is a topic of its own but the problem here is similar to getting a 640x480 compressed jpeg and putting it into a 1920x1080 lossless format and the same source into a 1280x720 lossless file and saying. What's the difference? They both look crappy! Many people have an opinion on high res audio formats but they are so many places to go wrong. Audio should be given a little more credit for it immersive ability. When done right it can take you places. Something to consider is that our eyes can only see 1 octave of information our ears can hear 10 octaves. Whilst these high res formats aren't needed for every application they do make a big difference when the source material can take advantage of it. I feel in 2014 music should be released in the best format available from the studio and if u want a crappier version so u can shove 5000 songs on your iWhatever that's your choice.

microcolonel 13y ago
It's not the same as the same inferiour source material. When you apply negative gain to the 16-bit signal, it has less amplitude in absolute terms, but it should have all the resolution of a whole 16-bit sample in that space. The point is that whether that 16-bit signal is represented with the lower 4 bits of a 16-bit sample, or the lower 12 bits of a 24-bit sample, you can scarcely hear anything at all, let alone a qualitative difference between the signals. 192KHz is not a higher quality format, it is a production format. The purpose is to reduce the cost and finality of anti-aliasing filters when sampling the signal, it is just cheaper to manufacture. There's a good reason for this being the standard in professional audio, they need to buy a LOT of audio interfaces, many studios will have thousands of such inputs. Thanks to the basic laws governing signals, we know with near certainty that not only is 192KHz overkill, but so is 48KHz, and so is 44.1. Without significant new evidence showing humans hearing signals with frequencies greater than 24KHz, you will not make any convincing argument as to why we should go with any sample rate higher than 48KHz for human listening. As for 24-bit, it is another production interchange format. It's there so that you don't need to stand around adjusting gain knobs on audio interfaces so that you get decent fidelity but also don't clip. With 24-bit you can just sample your audio once, and assuming it's within a reasonable range, you can adjust the gain in the discrete signal. There is some indication that 16-bit sampling is less than completely ideal. The very best ears in humanity(newborn ears) distinguish about 21 bits in the safe ranges of amplitude, 24-bit may make sense in an audio system for newborn babies. I feel that in 2014, music should be released in the best format available from the studio, and if you want a crappier version so that you can shove only 100 songs on your iWhatever, that's your choice.

dankoss 13y ago

The methodology might be flawed, but the point stands: We can't use the full dynamic range of a 24 bit signal for audio. Great converters have 20-21 bits of range and if we fully use this range, hearing damage will start at maximum volume levels. If the volume is kept the same between the two files, there is a 48dB drop in level - not inaudible, but there are very few source materials that will actually use the extra dynamic range. On top of that due to fletcher munson curves we will only hear content in the 1-4kHz range at that low volume level.

stinos 13y ago
Great converters have 20-21 bits of range and if we fully use this range, hearing damage will start at maximum volume levels but that doesn't have anything to do with number of bits? I mean, an audio player system is basically a digital stream sent into a DA converter outputting an analog signal which is then fed into an amplifier where the signal is amplified and then sent into speakers of some sort. No matter if you have an 8bit audio stream, or a 128bit audio stream, the stages afterwards play a much larger role in the hearing damage caused. Also not all DA converters ouput the same level to begin with. Some are -5V to 5V while others (single power supply) can be 0V to 5V, etc etc.
- dankoss 13y ago
  Hearing damage as a result of loudness; the electronics have little to do with it. If you set up a 96dB (16 bit) system such that you can hear the lowest bit toggling, your maximum volume will be 96dB above that. 96dB SPL is loud but not painful for average listening. If you set up a 20 bit system (120dB) so that you can hear the lowest bit toggling, 120dB above that will most certainly cause hearing damage if used for very long. 144 dB (24 bit) or more SPL will cause near instantaneous damage and pain.

S_A_P 13y ago

It would take highly dynamic music to really hear more bits. The quiet passages is where it makes the difference. Most music is compressed beyond the need for anything other than 40db of dynamic range. I barely can tell the difference between high sample rate 8 or 12 bit audio and 16 bit audio. In some circles I hear people talk about that "gritty" 12 bit sound, but its usually not properly dithered, heavily filtered, 20khz or less sample rate and generally destroyed on purpose. Audio is generally the best arena for snake oil since subjectivity is so high. I cannot believe that Focal Grande Utopia EM loudspeakers(180,000 usd MSRP) exist in quantity, but I know two people who own a set in their (comparatively) modest living room. I have heard them in action, they sound great, but Im not sure that I heard anything in any of the music that I hadnt heard with good headphones or speakers. At what point is it "good enough"? I own plenty of recordings that no amount of hifi audiophilia will help.(many that I recorded myself ;) )

hendriks 13y ago

For the test the quality of the input does not matter. You can conduct the same test with a beautiful 24-bit recording as source, you would still not be able to hear much in the end. And that's the point. When you print a photo in size 9x13cm, does it matter whether you print it in 10,000dpi or 20,000dpi? Hardly, unless you always run around with a microscope. And it does not matter, whether the photo was originally a good picture or not.

PavlovsCat 13y ago
> When you print a photo in size 9x13cm, does it matter whether you print it in 10,000dpi or 20,000dpi? Hardly, unless you always run around with a microscope. It can matter when you put it in a scanner though. I tried scanning some old photos, they look horrible compared to even my puny 8mp camera. Then I tried scanning a piece of cloth, and was blown away by the detail..

acomjean 13y ago

I have pretty good headphones (Sony MDR-V6). I can hear decently enough (although getting older). Like I noticed that the Deutche Gramaphone 111 years music collection I bought (256kbps mp3s) has taped performances from the 70 & 80s which have some tape his. Honestly as an end user its really good enough. If I was remixing this stuff and processing the audio I would want the non-compressed. (similar to shooting Raw photographs ). But at 256kbps honestly it really is amazingly good. thinking back to cassette tapes and records, its astonishing. As for the tape hiss on the classical music, the performances are so good I just ignore it. Sometimes I think people get worked up about the quality of the recording and ignore the quality of the performance.

calinet6 13y ago
> Sometimes I think people get worked up about the quality of the recording and ignore the quality of the performance. This is the main problem with audiophiles. Improve your equipment until it makes your music awesome, then just enjoy it.
- danbee 13y ago
  "Normal people use their equipment to listen to music. Audiophiles use music to listen to their equipment."

PavlovsCat 13y ago

re: "It's useless because most people can't hear the difference." Sure, but when sampling and then processing things, you can consider it "headroom". Just like it's not hard to edit a photo in a few steps that the limitations of 8 bits per pixel begin to show. Though I suspect sampling frequency is a lot more important there, 24bit can't possibly be "too much". Yes, I understand that this is not a good argument for consumer products to be more expensive and need more storage, just so musicians of the future, or pirate musicians of today, can have a better time. So I think the best plan is to encourage everybody to become hobby musicians, then it would be an easy sell.

rrrooss 13y ago

There has been a lot of talk about quantative aspects of but little about the qualitative. What does a 18khz square wave look like wher recorded and reproduced in 44.1 kHz. How is phase affecte?. Phase of an audio signal reaching the ears helps you perceive distance and position. How many samples are used to represent the frequencies at the higher end of the audible range. What is the quantization error difference between 16 and 24. And if the author can't hear the difference between 4bit audio and 12bit audio I question what he/she is listening on. The aliasing would be HUGE.

jrabone 13y ago
An 18 kHz square wave sampled at 44 kHz looks like an 18 kHz sine wave, everything after the fundamental frequency is well outside the Nyquist limit and will have been thrown away by anti-aliasing filters. And furthermore you couldn't hear it even if it wasn't. Fourier decomposition of a square wave gives the sum of odd multiples of the fundamental, the next frequency is 3 x 18 kHz = 54 kHz.
- rrrooss 13y ago
  Okay now actually look at the response on Matlab. As you get closer to the nyquist freq there are less samples to describe the wave. And while you get a bastadised 18k signal it's a far throw from what you put in. Anf while most are crying your splitting hairs the frequency response and dynamic range arguments pale in comparison to making the exact wave you put into the encoder come out. On the right system with the right recording the tiniest nuance in a room reverb helps trick your brain into believing the sound as actually happening. And this is not important to all listeners. But people saying that it makes no difference and is not important as a format to release music in are thinking only of their current needs and experiences. If you have actually heard a good recording on a good system in a good room you will know what I'm talking about. If listening experience has been laptop speakers, headphones and your mum and dads mini system then I totally agree any more than 320kbs mp3s are overkill. But to say that high res formats gave no place in consumer land is to show your lack of understanding of different people and different needs. In the age of iTunes you surely you buy the album and download it in whatever format u see fit. If u think audiophiles are wackos go get ya 320's. I would like to get it as I came fr the studio. The best it can be is as it came from the studio.

ntoll 13y ago

"My God! What has sound got to do with music!" ~ Charles E.Ives (https://en.wikipedia.org/wiki/Charles_Ives https://en.wikipedia.org/wiki/Charles_Ives) (and I'm inclined to agree)

raverbashing 13y ago

At the last year's IFA (http://b2b.ifa-berlin.com/en/Home.html http://b2b.ifa-berlin.com/en/Home.html) there was a demonstration of a 24-bit audio player with corresponding high quality audio files and headphones. The results, even though it was a single test, were very good. Of course the issue is, the whole chain has to be 24-bit, if you're going to pump your 128kbps MP3 there the difference is probably negligible (not to mention the 1-bit DACs commonly used, that's one of the elephants in the room) About the MP3 files, yes, the difference is audible. But of course you need good headphones.

danbee 13y ago

So I converted a few tracks to 8-bit with noise shaped dither. Have a listen and remember that the noise floor is now 48dB higher (8-bits) than it would be with 16-bit. https://www.dropbox.com/sh/1h4z0quoybo60c0/O1SwT35Nve https://www.dropbox.com/sh/1h4z0quoybo60c0/O1SwT35Nve

rrrooss 13y ago

More to it than frequency response and dynamic range. The goal should be for the most accurate reproduction of the input waveform. Here is a pic of transient response for different sample rates. http://www.eirec.com/DPimages/digisqwvtest.jpg http://www.eirec.com/DPimages/digisqwvtest.jpg

rrrooss 13y ago

Frequency response and dynamic range are low hanging fruit in high res audio debates. They are not the only factors in accurate reproduction of an input wave. What is rarely considered is the temporal resolution of the format. Which is the formats ability to describe change over time. To plot a graph but depth is the y axis resolution and sample rate the x axis. Again I'm talking to temporal resolution. Transient response is directly effect by this. The link below shows the advantages of hi res audio formats inside the human hearing range. So if accuracy is of concern high res formats DO hold more information about the original wave form. If file size is of concern hi res is not applicable. http://www.eirec.com/DPimages/digisqwvtest.jpg http://www.eirec.com/DPimages/digisqwvtest.jpg This is an example if transient response of different sampling rates.

dankoss 12y ago
I tried to find the context for that image but there wasn't anything on the site. The image is most certainly incorrect -- or more likely, looking at the output of the analog electronics after the DAC. Sampling theory says that a perfect square wave can be represented at any frequency below Nyquist. That doesn't mean that the codec or the analog electronics are capable of responding instantly at those frequencies, but that has nothing to do with the fact that a 1kHz square wave can be perfectly sampled with a 44.1kHz sample rate. The image is simply incorrect. Transient response in the real world is generally limited only by the acoustic transducer response of the system, because everything else has the ability to respond much faster than audio rates. With Pono, this means that the earbuds or headphones you use with the player will have a greater affect on the transient response than the electronics inside.
- rrrooss 12y ago
  This image is from RME a sound card maufaturer. And the image is of the analog output after dac. So still in the electrical domain. Short comings of a playback system has what to do with the accuracy of a recoding medium? There are SO many factors that stop and audio signal reproduction from being perfect. But the signal domain is the easiest to make better. Transducers that have to fight the law of physics for accuracy are the obvious weak point. But as with anything rubish in = rubbish out. Before you even get to the weakest point of an audio system the transducer (the speaker) transistors and amplification have their own short comings the accuracy of an amplifiers transient response is measured by its slew rate. A square wave is impossible for a speaker cone to reproduce in theory the rising and falling edge is describing an instantaneous movement. A speaker cone can't do this, inertia says no. And this is called distortion. As every part of the system introduces its own distortion the accuracy gets less and less. To accept distortion at the signal level in the persuit of accuracy is counterintuive. But if your system can't take advantage of this extra accuracy, go ahead and use a lesser format, no sense in the extra file size. If you want a reference quality signal, use the high res formats. To be clear here I'm not saying I want to listen to digitally encoded square waves, a square wave is a tool for showing the accuracy of input=output. A square wave being the hardest analog waveform to capture and reproduce. I am objecting to the idea that hi res formats have no advantage over what we are used to as consumers. It is the content distributors that are taking advantage saying that 24bit/96k is more expensive. The bandwidth and storage does not make this a more premium product you should be paying for the album not the format. But why limit choice because the masses can't see the benefits. If you buy the album you as the consumer can choose the format. I'm not sure if all this arguing stands up to pick whatever format suits your needs. This argument should be if I bought the album why can I choose the format most applicable to my playback needs/desires. And if your concern is that company's charge more for the high res, that refeclts poorly of the distribution company. If you think it's a waste of space just download the lesser format. If I had a choice I'd download it as it left the studio/mastering house.

sp332 13y ago

Did anyone else actually try this? I thought the differences were quite noticeable. I ran it through Foobar 2000's ABX Comparator and I got the right answer 5 out of 5 times.