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Why Distributing Music As 24-bit/192kHz Downloads Is Pointless
An anonymous reader writes "A recent post at Xiph.org provides a long and incredibly detailed explanation of why 24-bit/192kHz music downloads — touted as being of 'uncompromised studio quality' — don't make any sense. The post walks us through some of the basics of ear anatomy, sampling rates, and listening tests, finally concluding that lossless formats and a decent pair of headphones will do a lot more for your audio enjoyment than 24/192 recordings. 'Why push back against 24/192? Because it's a solution to a problem that doesn't exist, a business model based on willful ignorance and scamming people. The more that pseudoscience goes unchecked in the world at large, the harder it is for truth to overcome truthiness... even if this is a small and relatively insignificant example.'"
I know, Stephen Colbert is Reddit's hero and they're starting to infiltrate this site as well, but seriously. Call them lies. That's what they are, that's what they -deserve- to be called. Are people really that passive-aggressive and afraid of expressing themselves that they won't call someone who lies a liar any more?
Does it matter when the dynamic range is shot to hell?
There is a huge problem with file sizes (so both hard drive space and download bandwidth) with lossless files, so no, it's not entirely without problems.
lossless formats and a decent pair of headphones and a set of really expensive MONSTER CABLES will do a lot more for your audio enjoyment than 24/192 recordings.
There, ftfy.
Lossless formats (eg. Flac) by definition have no loss. You must be confused.
Different people have different cognitive abilities - this extends to our senses. The average person lacks perfect pitch, cannot tell the difference between SD and HD unless they're side by side, thinks their 128kbps MP3s sound alright, doesn't notice 60Hz jitter on their LCD, and so on.
It's the people on the fringes with superior senses who notice this stuff. But for the rest, this is all outside of their senses, so they're going to rubbish the quality paranoias of so-called audiophiles and videophiles.
I record my performances at 96 kHz sample rate, I have to say that the music sounds much better at 96 kHz than 48 kHz I think (feel?) because the higher sample rate gives audio effects like reverb a lush, deeper sound.
The more sample units per second give the effects more to work with, in addition, even though you can't hear above and below certain frequencies recording those inaudible frequencies has an effect on the final product.
You may be able to find some scientific proof of this but for me it's an ear thing, higher sample rates sound better.
"If any question why we died, Tell them because our fathers lied."
"I can't hear your rational argument over the impeccably better-than-perfect sound from my 83 trillion dollar sound system. Thank you, Monster!"
/. haven't we had all of our music in FLAC for a decade now? I don't even listen to music much and mine is.
For the rest of us on
I'm not sure why this particular technology is so bizarrely specious in claims. I'm sure in fifty years we'll argue over the best neural interface with its platinum, massaged, better than reality addition is.
Yep, not to mention the audio effects like reverb improve dramatically at higher sample rates (if they are written to take advantage of them)
"If any question why we died, Tell them because our fathers lied."
I find your well-reasoned and respectfully written response to be full of helpful counterpoints and useful references. I wish to subscribe to your newsletter.
Is is not possible that one day 'upgraded' sensory implants could be the norm for humans? Cybernetic generations to come may lament all of the lost audio information in recordings of our era.
>There is a huge problem with file sizes
Not any more, pumpkin.
We hit the terabyte size in drives a couple of years ago. There's no reason to be buying this format vs "archive quality" cd-audio or other lossless.
Buy/rip lossless. Transcode to lossy as needed. Anything else and you're being ripped off.
I listen to real music with real instruments. The "swish" you get in high-frequency percussion with lossy algorithms is annoying as fuck.
--
BMO
You are missing the point of the article. 192KHz is not 192kbps.
Yeah. Anyone who can't hear a tone at 90kHz is deaf.
Yawn. The point of higher sampling and bit rates is to have the most accurate representation of the original source material. Using a greater number of bits improves the dynamic range and reduces quantization noise. Some real instruments have spectral information above 22kHz, and most stuff created digitally in studios uses a sampling rate much higher than 44.1kHz.
If you want to do more with the music you purchase than just listen (and no doubt some people have the ears the appreciate the higher fidelity anyway) and do things like remix and reprocess, you want the better versions.
Double blind test results or I will continue to believe that you are suffering from Illusory superiority.
-1 overrated isn't the same thing as "I disagree".
Can someone explain to me what KHz "sampling rate" has to do with the frequency range you can sample?
How many more years will slashdot have an off-by-one error on your Score in your profile?
I have a PhD in Digital Music Conservation from the University of Florida. I have to stress that the phenomenon known as "digital dust" is the real problem regarding conservation of music, and any other type of digital file. Digital files are stored in digital filing cabinets called "directories" which are prone to "digital dust" - slight bit alterations that happen now or then. Now, admittedly, in its ideal, pristine condition, a piece of musical work encoded in FLAC format contains more information than the same piece encoded in MP3, however, as the FLAC file is bigger, it accumulates, in fact, MORE digital dust than the MP3 file. Now you might say that the density of dust is the same. That would be a naive view. Since MP3 files are smaller, they can be much more easily stacked together and held in "drawers" called archive files (Zip, Rar, Lha, etc.) ; in such a configuration, their surface-to-volume ratio is minimized. Thus, they accumulate LESS digital dust and thus decay at a much slower rate than FLACs. All this is well-known in academia, alas the ignorant hordes just think that because it's bigger, it must be better.
So over the past months there's been some discussion about the merits of lossy compression and the rotational velocidensity issue. I'm an audiophile myself and posses a vast collection of uncompressed audio files, but I do want to assure the casual low-bitrate users that their music library is quite safe.
Being an audio engineer for over 21 years, I'm going to let you in on a little secret. While rotational velocidensity is indeed responsible for some deterioration of an unanchored file, there's a simple way of preventing this. Better still, there have been some reported cases of damaged files repairing themselves, although marginally so (about 1.7 percent for the .ogg format).
The procedure is, although effective, rather unorthodox. Rotational velocidensity, as known only affects compressed files, i.e. files who's anchoring has been damaged during compression procedures. Simply mounting your hard disk upside down enables centripetal forces to cancel out the rotational ruptures in the disk. As I said, unorthodox, and mainstream manufactures will not approve as it hurts sales (less rotational velocidensity damage means a slighter chance of disk failure.)
I'd still go with uncompressed .wav myself, but there's nothing wrong with compressed formats like flac or mp3 when you treat your hardware right
--
BMO
> I listen to real music with real instruments. The "swish" you get in high-frequency percussion with lossy algorithms is annoying as fuck
Seconded. Many things sound fine (not great, but OK) in medium to low bitrate MP3 or OGG or AAC or whatever. Some things sound terrible, and when they do, it sucks to listen to.
As a former audio engineer with some ranking success, I can tell you that it's true -- delivering high-sample rate audio as an end format is really pointless. It hardly makes sense in a studio, and definitely is illogical for the distribution of a final mix.
However, there is an increase in quality using 24 bit. Most people just assume increasing the bit depth is the same as increasing the sample rate, but this is incorrect and short-sided. With higher bit depths, you can get your analog components operating a little further away from the noise floor. This also makes dithering much less noticeable (the noise you hear when you crank the volume up as a song fades out). Why? There are more "levels" for each sample to be recorded into. It's like going from 16 to 24 bit color. You would notice this.
For the 192 KHz fans out there, there is direct and proven mathematical reasoning for why 44 KHz audio is plenty. That, and your equipment probably can't produce it. Your converters probably suck at this frequency, and your ears definitely can't vibrate that quickly. More samples doesn't "smooth out" the waveform.
Given how common time-stretched audio is these days — for DJing, looping, etc, high sample rate music files are ideal.
I love the word "truthiness"
Go listen to Stuart Copeland tap on his hi-hats with FLAC, shn, cd-audio, or apple lossless, and then at 192.
Then get back to me.
--
BMO - One world is enough, for all of us --The Police
my cat is doing the listening? Would 24-bit/192kHz music be better? Seriously. Not kidding.
Except that the article refers to 24-bit linear PCM audio files that are encoded at a sampling rate of 192 kHz (equivalent to 9216 kB/sec compared to the MP3's 192 kB/sec)
Hertz versus kB/sec... totally different units.
For what it's worth, most audiophile sites like HDTracks sell high-resolution files that are 24-bit / 96 kHz. (4608 kB/sec)
Very few people (if any) besides fanatical audio buffs would deal with anything above that. DSD (SACD) is different enough that it's hard to compare to this.
So I don't subscribe to the $1500 per power cord group that some people do (usually the same folks who claim a $2400 USB cable increases the separation of instruments within a digitally encoded file).
However, I do own some good equipment- not the best, but pretty decent as far as studio setups go. ATM I'm rocking an Apogee Symphony I/O over Apogee's proprietary PCI-e interlink card (a Symphony 64). Yes, Apogee's driver support and customer support is shit, but when their equipment works it works pretty damned well. On the other end of that is a 5.1 setup consisting of four ADAM S2X speakers and a SUB12. The speakers were around $2500/pop and they're self powered (that is, they have the amplifiers built-in) and run over balanced XLR.
I didn't buy this equipment because it sounded "good" or "colourful" or "warm" or any of that bullshit. I bought it, because, when I want to listen to stuff that's in either 24-bit/96kHz or 24-bit/192kHz (which is a bit excessive, I'll admit)- I know that what I'm being audibly blasted with is as accurate as it will ever be. I don't care if the precision is sharp on the ears or unpleasant to some people. If I want to listen to music (when I'm not busy making it), I want to hear it exactly as it was recorded.
And in that regard, there is a huge difference between 44.1kHz/48kHz/96kHz, but lesser of a difference between 96kHz and 192kHz.
The thing about 192kHz is that it's such a high sample rate (a lot of people tend to work at 96kHz professionally), you need the equipment to handle it. Lots of interfaces will happily handle a couple of channels at 192kHz, but forget about streaming 16 channels at that same sample rate over anything that hasn't cost you a few thousand bucks and hooks up to your DAW/recorder over a proprietary high-speed interface.
So there's a lot of junk floating around out there that claims to be 192kHz, but with the right tools (I can't personally tell the difference with my ears) you can quite clearly see that only part of it (or none of it) was recorded at 192kHz. The studio gear used simply didn't support that sample rate, or they didn't opt to use it, or some outboard gear didn't jive well with it, or whatever.
My point here is that a lot of people will try to screw you out of money for 24-bit/192kHz music when in fact you're not getting anything anywhere near that. And a lot of people don't even know what the hell that means- so you get the kind of people trying to listen to that crap through a bog standard HTIB system in a box where the quality is such shit coming out of the speakers that you wouldn't be able to tell the difference between a CD and that stuff anyways.
So yeah, for the majority of people out there- 192kHz/24-bit is pointless unless: A) the entire audio pipeline that produced that tune was running at 192kHz/24-bit, and B) you have actual hardware capable of playing that back properly, and not some HTIB thing you bought from Futureshop that sounds good "because it's really loud".
Frankly, I find it hard to believe that enough people out there want 192kHz/24-bit for legitimate reasons (owning proper hardware for reasonable playback) that there's actually a market for this stuff. So it makes me think that this stuff is being targeted at people with iPods and shitty desktop speakers on their iMac computer. In which case, yeah, it really doesn't matter. You're not going to hear any difference between a lossless FLAC file at CD quality or a 192kHz/24-bit file freshly bounced from the studio masters.
-AC
I see no rational basis for limiting myself to audio intended for those with hearing worse than average. (Nor do I limit what I read because of the poor reading skills of others; limit my choice of where to walk because too many have lost the skill in their desire to drive everywhere; limit who I know because politicians like to divide humanity into them and us; etc)
Limit yourself by personal ethics or by personal physiology, not by pseudoscientific efforts to brand "standard deviations" as deviants.
When you can tell the difference between 44.1/16 and 192/24 in a double blind trial, come back and we'll talk.
Subjective opinions about audio quality, particularly those accompanied by words like "deaf" or "idiot", are worse than useless. Subjective listening is deeply suggestible and unreliable. Claimed differences among any acceptably well designed audio electronics virtually always disappears under rigorous and controlled testing.
To give just one example, listeners reliably prefer the louder source in subjective testing, even if the difference is not consciously perceptible. If a 192/24 D/A is just 0.1db louder than a 44.1/16 source, listeners will tend to describe it in all sorts of subjective terms... "edgier," "richer," "more forward," "cleaner impact," "deeper soundstage" etc when in fact it is simply a little louder.
"Because it's a solution to a problem that doesn't exist, a business model based on willful ignorance and scamming people."
which happens to be a business model that works, unfortunately
intellectual property law is philosophically incoherent. it is your moral duty to ignore it or sabotage it
Ask any GeekSquad or Best Buy salesmen and they will tell you that you need full gold plated $2,000 HDMI cables for professional audio quality and $110 Monster ones for basic audio and video. They are not highly compensated so well for nothing you know
http://saveie6.com/
If you're sure you can hear a difference, why don't you ABX and prove it (or give strong evidence for it)? It's easy to hear a difference if you think you're supposed to, or if you paid a lot of money for speakers, etc. But its a lot harder to hear differences if you're doing a double blind test.
It's certainly OK to allow your emotions to take over if it makes you feel better to know you're listening to 24/192, but that's different than there actually being a perceived difference. You feeling better listening to 24/192 is an opinion, but whether you can actually perceive a difference is fact; lots of people confuse the two, so don't feel too bad.
Not nearly as much, no, but then that applies to very little of the music I buy. (And when it is true of it, it's usually for effect -- e.g., Daft Punk). Mass market music may be mixed for shit, but then I don't think 24-bit/192kHz is being aimed at the group of people.
Really, though, the article is pretty convincing bunk. I love his argument that sampling over 48kHz makes the audio more distorted and worse; it's a stroke of genius to turn reality on its head, like something you would find in a political campaign.
(Disclaimer: I write digital audio software for a living and have kept limited the sampling rates to 44.1 kHz and below, because it's appropriate for the type of use it sees. It also uses 32-bit audio where appropriate.)
Did you listen to it double blinded? No? Then I don't care what your confirmation bias tells you that you heard. The difference is beyond your ability to hear, but not beyond your ability to deceive yourself into believing what you want to believe.
-1 overrated isn't the same thing as "I disagree".
People that have crappy sound systems do not realize the difference, or they just don't care. I don't mean that in a negative way, they like volume, not quality. Nothing wrong with that. I wish I was that way. Regular non remastered PF from the 70's is very noticeable to me when it is compressed and I can even tell when I'm driving 55 down the highway with a moderately priced car stereo. While at home on my couch listening to my home stereo, I'd rather listen to AM radio talk shows then music I am familiar with in a compressed format. It's just not the same and not enjoyable. Some people get into music more than others. Nothing wrong with that.
Bad boys rape our young girls but Violet gives willingly.
Well, since you can't tell the difference, why would anybody else?
A triangle or bell should ring, not crackle.
A snare brush rustles at 192/24 instead of sounding like rustling paper.
Go listen to some LIVE music to hear what REAL instruments sound like instead of judging based on your years of bias listening to compressed and crappy CDs.
Of course if your music consists of synth beats, vocoder samplings, and other such drek, you've never HEARD a real instrument before in your life to know what one SHOULD sound like.
I do not fail; I succeed at finding out what does not work.
A group of sixty audio professionals and audiophiles did a series of controlled double blind trials published in the Journal of the Audio Engineering Society. They found no perceptible degradation caused by a 16-bit/44.1kHz A/D/A.
http://www.aes.org/e-lib/browse.cfm?elib=14195
No loss from the original sampling, i.e. they didn't loose any information in the compression. Most music is sampled at (correct me if I'm wrong someone?) 44kHz, I forget how many bits, I think 16. The thing being touted is sampling it at 192kHz with 24bit resolution, which is much higher on both counts, and therefore, in theory, should produce better quality reproduction of the sound based on oversampling and reduction of the signal to quantization noise rate. The point the TFA makes is that human ears can't hear the difference, although I think that some audiophiles may beg to differ.
FWIW, I have quite bad ears, a recording needs to be quite bad before I notice it. I'm an electronic engineer though, so I know all the theory...
One thing I know, and that is that I am ignorant...
You mean like, honkies, spics, niggers, dune coons, prairie niggers, kykes, faggots, chinks, canucks, wops, guineas, krauts, and polocks? I think that's everybody anyway, my apologies if I left out any group, I try to be an equal opportunity offender, challenging people to be adults and get over their group identitied. Criticism welcome. Cowardly disapproval spurned.
I know, Stephen Colbert is Reddit's hero and they're starting to infiltrate this site as well, but seriously. Call them lies. That's what they are, that's what they -deserve- to be called. Are people really that passive-aggressive and afraid of expressing themselves that they won't call someone who lies a liar any more?
Okay, everybody, listen up: Anonymous Coward is having a rough day so let's all be extra nice to him!
"I like to lick butts!" by MobileTatsu-NJG (#32700246) (Score:5, Informative)
There are many examples. I doubt many people care about the difference (I certainly don't), but that doesn't mean it can't be detected.
If you can't hear the difference between cymbals, bells, brass, and other "edgy" instruments at 44KHz/16bit "lossless" and 192KHz/24-bit, you're either deaf or using earbuds.
Idiot.
Im going to have to agree there. Since the industry moved away from vinyl and analog recording equipment the quality of audio has gone down. However since recently receiving several old analog recorded albums remastered to super audio cd which varies in their methods of transfer, however all are 24bit and above 86khz. Do represent to my ears a serious improvement in quality. I could go on and on about resonance and timbre, but suffice it to say these qualitys are difficult for digital equipment to deliver, something only a truly a discerning ear can notice. And it has nothing to do with your capability to hear high frequency sounds (my ears top out just under 18khz). Just ask Stevie Wonder.
> If you can't hear the difference ...
I certainly can. I'm glad to hear others say that, too. I thought it was just me.
We have an analogous problem in broadcasting -- everyone wants to use compressed formats to save space and upload/download time. Files are thrown all over the Web now. (I haven't seen a reel tape in years, though I think we still have an old reel-to-reel somewhere just in case. Political season coming up, after all.)
The problem is REALLY bad when you repeatedly encode. For example, our digital automation systems wants to compress files. Our studio to transmitter links (STLs) want to compress to save bandwidth. HD Radio compresses the SNOT out of the audio. Honestly ... some of the crap that I hear on the radio now is so bad I don't know how anyone can listen to it. It swishes, it glitches, it swarms, it sounds brittle, it's awful.
I made a rule in our facilities a few years ago that if it wasn't at least 256 Kilobits, we wouldn't air it. This annoyed some people -- one guy had to dump and entire music library that he'd spent a week putting into the system -- but it was awful.
Maybe there's no point in 192/24 for kids listening to pirated music on $20 MP3 players, but I refuse to believe that most people can't hear the difference. Heck, I'm getting old and I'm half deaf nowadays, and I can immediately hear the difference. There's just no comparison.
Cogito, igitur comedam pizza.
Is it just me or is this article just a bunch neo whiny cry babies with crappy 16/48 audio cards trying to talk down about well built 24/192 cards
It sure ain't the SCIENCE cause 24/192 is both more samples and more bits than 16/44, so the article is anti-scientific to claim it's a better signal at 16/44 or 16/48.
Finally we get to the meat, distributing music, okay correct for a download distro 24/192 is a stupid format to download, personally I'd rather have 320k mp3, at some point it is easier to just mail a DVD's with those tracks to someone who must be working with 24 bit tracks. It's more of a production format than a buying a CD (in this case DVD) to listen to your favorite band. Most people don't sell this format, just like they don't sell WMA, or .mod files instead of mp3's, or wav ~cdda so the argument's a moot point, and the few people that do sell this format, who gives a shit, is it really bothering you so much you have to tear it down, why don't you take a deep breath and check that your mortgage paperwork isn't signed by linda green?
Creative Audigy 2 ZS platinum pro is now eight years old and on XP taking up one PCI slot, still kicks most of your ass to this very date. Two out of the two I bought, still work and they both have been through several motherboards which fried.
Truthiness refers to a specific kind of lie-- a lie that sounds true, and that a large segment of people really want to be true. The kind of thing that's close enough to true for AM radio talk show hosts.
And now... I'll get off your damned lawn. Don't forget to take your teeth out before falling asleep.
Not if you don't know any better. ;-)
Seriously, its been so long since I've seen a live band I don't know what a drum is supposed to sound like.
At my age my ears are not so hot.
Sig Battery depleted. Reverting to safe mode.
The point is that the 44KHz 16bit track has already been compressed from the original recording. However you rip that track, lossless or lossy, it doesn't matter; you're still not getting the original track.
Knowing this, it doesn't mean that the tracks some sites are selling as 192KHz 24bit are from the original sources, or will even sound better, either. The original track could have been recorded with bad equipment or settings. In other cases, when doing comparisons on CD tracks vs high resolution tracks from sites like HDTracks, you can sometimes find that the HDTracks track is just the CD track with increased reported resolution/file size - possibly due to the inability to acquire the original material, though it could also be as simple as pure greed and laziness. Not that all of the albums on those sites are fakes, but a few of them have been found to be ripoffs.
There's also the fact that it's extremely unlikely anyone can tell the difference between an encode at 96KHz vs 192KHz. If they are both properly encoded from the same source, it's unlikely there will be any audible difference between them.
Duh! It's called political correctness. And if you even dare show a pair, others will kick them till you're blue in the face.
Life is not for the lazy.
Indeed. One of the overlooked but highly important issues with sampling rates is that although you can represent up to Nyquist in a periodically sampled signal, that is the limit for infinite length recordings. For finite-length recordings, it isn't all or nothing, represented perfectly or not at all -- instead the uncertainty (read: representation error) increases as you approach Nyquist.
Too bad Shannon and Nyquist are dead. It seems they've completely misunderstood the math. How embarrassing they passed on before you could correct their mistake. Now they'll never know.
Oops, using KHz instead of kHz.
than their CD transfers. In the cases where I recall being most disappointed (I've thrown out almost all my vinyl records), it was the dynamic contrast that was missing in the CD versions, for example a pianist striking chords from dropping his hands a foot above the keyboard.
Maybe these were just bad transfers... I don't know.
The Nyquist limit is the highest frequency that can be represented, yes.
But at Nyquist, only one shape of waveform can be represented. Depending on the design of the DAC, it could be a square wave, triangle wave, or sine wave. But only one of those.
With this in mind, I don't understand why Monty says that beneath Nyquist, everything is captured perfectly and completely. That seems plainly untrue to me.
The value of higher sampling frequencies isn't to reproduce frequencies above 20kHz. The value is to preserve the characteristics of waveforms within the range of human hearing, pushing aliasing artifacts into the ultrasonic, where they can be gently filtered out between 20kHz and 30kHz.
That said, to me that means there is some value in 96 kHz distribution.. 192 kHz does seem like vast overkill.
On tracks that you have listened to for many years, you know what to expect because you remember it. You've heard it 100's of times on many different systems over the years. When the music is compressed, you can hear the difference almost immediately, specially on the higher frequencies. I've personally never used 192/24 but I have used various forms of vbr/cbr at different rates and different encoders over the years. I've settled on a rate that balances space and quality. I still notice the difference though. Same going the other way, I've listened and "learned" tracks that were compressed and finally got an uncompressed version. I notice the difference that way as well. Was I happy with the original compressed version? Yes, it was all I had and it sounded as good as I had ever heard to that point.
Using your own argument, why not just use 128/16 or 96/24? There is obviously a difference right? What some people notice or not does not mean others do not.
Your claim about loudness being perceived as better is well known and no secret. Why do you think masters are mixed with such high average levels these days? Just because 95% percent of the population thinks louder is better does not mean everyone does. I am not some crazed audiophile with strange beliefs, rituals, and exotic equipment and it doesnt take that to hear differences.
Bad boys rape our young girls but Violet gives willingly.
You my friend, are a stranger to the truth.
I used to think like you. Spent thousands on audio equipment.
Now that I'm deaf in one ear I listen to MP3s through $24 headphones.
Being deaf saves a lot of money.
This space available.
If George Carlin were still alive, he would mod you up right now.
for future DMCA kruft
Xiph.org must be talking about elevator Muzak because:
1. High-Frequency Sound Above the Audible Range Affects Brain Electric Activity and Sound Perception
2. High-Frequency Sound Above the Audible Range Affects Sounds Within the Audible Range
http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
That copypasta hasn't been funny for at least five years if ever.
If you wanna troll, let's go... I'll take your side, you take mine and no one under the age of thirty will have any freaking clue what just happened. /g/
>>>
yarbles*
Double blind test or gtfo. The peer reviewed research says you can't hear it. Talk is cheap, show us some data.
-1 overrated isn't the same thing as "I disagree".
I know I have a terrible ear. I can not make the difference between 16bit/128Khz and 24bit/196Khz. However, I performed double blind test using 24bit/196Khz and lossless flac on friends of mine that claimed to hear the difference.
Actually about 3/4 of the ones that claimed to hear the difference could not: They got it right close to half the time, so pure luck.
Yet a couple of them could make the difference very clearly (close to 100% of the times).
There is a huge problem with file sizes (so both hard drive space and download bandwidth) with lossless files, so no, it's not entirely without problems.
I own (legally, even) somewhere on the order of 2500 CDs.
I have ripped all of them to FLAC (lossless).
Total size, under 600GB. I could easily fit my entire collection on a single HDD five years ago. Today, they don't even count as the biggest single directory on my home file server (hell, not even third place - Though in fairness, I do collect historically-significant Linux distro ISOs).
FWIW, even ripped raw rather than compressed as FLAC, they would still fit on a single 2TB drive. Audio really doesn't present all that much of a problem these days.
The article only debunks putting 24/192 and 24/96 audio as the audio that goes on your iPod, because the equipment (eg cheap speakers and headphones) is more likely to damage your hearing from lacking the required fidelity to reproduce the full spectrum but not filter out the dangerous naive mastering.
The problem is not with 24-bit/192kHz music downloads. The problem is some idiot is touting them "as being of 'uncompromised studio quality'". Who said they're even close to lossless...?
There is NO problem with the technology.
Back to content...
You don't need to spend thousands anymore. A single K will get you close enough these days unless you want to fill 20+ by 20+ rooms with full bass (bass is where you'll wind up spending the most in large rooms despite what the audiophiles might want you to believe)
MP3s suck as soon as you turn them up even a little, amplifying their shortcomings. At low volumes, things still sound hollow, except for non-harmonic electronica, bleah. I personally prefer to listen to a lot of things at relatively low volumes through ear buds, but much louder when pumped through decent speakers. And yes, you can tell the differences in both cases.
Being deaf would suck, although we're all heading that way through time unless science saves us.
The cesspool just got a check and balance.
Sure, but with the loudness war, they're not really using the 16 bits they have, so what's the point?
Educating people is fine, but the elitists will always say swear that x is better than y, even if it is provably otherwise. Just like some people will swear they saw Elvis working as a hooker at the Rt. 97 truck stop blowing Jesus.
Silence is a state of mime.
I think you're full of crap. Prove me wrong. Or at least cite me wrong.
I think Truthiness covers half truths too. A half truth is that 24-bit/192kHz audio is higher quality than 24-bit/96KHz audio.
The whole truth is that only your house cat would be annoyed at 96KHz, or an audiophile dog.
There is a huge problem with file sizes (so both hard drive space and download bandwidth) with lossless files
Not any more, pumpkin.
We hit the terabyte size in drives a couple of years ago.
Grandparent said "and download bandwidth". Filling one of those drives takes four months or more over an urban home Internet connection capped at 250 GB per month, and that's if you don't do any Facebook, Slashdot, Cracked, deviantART, Netflix, YouTube, or everyone else's favorite bandwidth hogs. Over rural broadband, it takes 16 years due to 5 GB/mo caps imposed by satellite providers.
Replying to remove bad mod
The point is that the 44KHz 16bit track has already been compressed from the original recording. However you rip that track, lossless or lossy, it doesn't matter; you're still not getting the original track.
Actually - it wasn't compressed - it was the limits of the recording equipment at the time. 192KHz/24 bit wasn't common in the 80s.
Knowing this, it doesn't mean that the tracks some sites are selling as 192KHz 24bit are from the original sources, or will even sound better, either. The original track could have been recorded with bad equipment or settings. In other cases, when doing comparisons on CD tracks vs high resolution tracks from sites like HDTracks, you can sometimes find that the HDTracks track is just the CD track with increased reported resolution/file size - possibly due to the inability to acquire the original material, though it could also be as simple as pure greed and laziness. Not that all of the albums on those sites are fakes, but a few of them have been found to be ripoffs.
Unless the track are genuine 192KHz/24 bit tracks, that is true. CD tracks can sound as good or better than 192KHz/24 bit tracks, it all depends upon settings. CD tracks can also sound worse than 92KHz encoded MP3s, again, it depends upon settings.
There's also the fact that it's extremely unlikely anyone can tell the difference between an encode at 96KHz vs 192KHz. If they are both properly encoded from the same source, it's unlikely there will be any audible difference between them.
This, however, is patently false. Given appropriate equipment and a person with a reasonable ear (mine aren't even that great and they suffice) and you can definitely tell the difference between 92KHz and 192Khz, and even straight CD tracks they were encoded from. It does require that the original source have enough depth that something is lost, however. Simple electronica, or other music that samples heavily from trivial sources will not provide enough depth to tell.
At this point my entire collection is lossless (CD quality at a minimum), and yes, it even makes a difference in my car, which has a halfway decent audio system. The other vehicle needs new speakers and an amplifier, the former sound blown and the latter was never clean to begin with, enough so that I pretty much haven't listened to music in it in years, just haven't gotten around to replacing it as it was only short trips anyways.
The cesspool just got a check and balance.
I don't care how highly you think of yourself, until you show me some data you are a worthless troll.
-1 overrated isn't the same thing as "I disagree".
Illusory Superiority. It's good to have a name for the condition. A disease that those suffering from desperately want to avoid being cured of.
Unfortunately, it has been proven time and again for decades that if you're not exposed to sounds early in your life, you may never be able to hear them because your neurons never develop the pathways to recognize those sounds.
I wasn't going to respond, but then you hit bullshit central.
You can't hear sounds if you haven't heard them before? Seriously? Do you really believe that?
They won't believe you. They believe their ears must be superior to those pseudo-audiophiles. Your post should have ended all discussion, but *sigh* it won't.
A couple of my own notes about ruining the dynamic range...
- today it's taken to a new level, I see more and more songs where not only DR is destroyed but sound is also harshly clipped
- not only mastering, but the mixing step is also to blame, where every track volume might be cranked up or, too many tracks recorded where the song becomes an (easily-compressible) mush
- it sometimes appears that a band might have the first one or two records with good sound quality, but when they make it big, they "bend over and take it up the ass", thus the rest ones being crap
He mentions in the article that 24/192 source would be a benefit to those that need to reprocess the audio, such as in remixing/effects/etc.
Therefore, using his own logic, you can deduce that being able to download and/or purchase music in those formats would be a benefit to anyone wishing to use it as a source in their mixes.
So he proves himself wrong that distributing 24/192 is pointless, as im sure there are plenty of remix artists out there looking for high quality samples, and many of whom use sound processing hardware designed specifically for a 24/192 source.
-HasHie
Fair point. The people who go on about 24/192 probably don't really listen to the kind of music which is affected by the loudness war. Most audiophiles I know are heavily into jazz or classical music, the recordings of those usually try to be quite faithful to the original.
One thing I know, and that is that I am ignorant...
I made a rule in our facilities a few years ago that if it wasn't at least 256 Kilobits, we wouldn't air it.
That'd make it impossible to play mono recordings, such as 8-bit game soundtracks or old Beatles songs, because in many codecs, mono typically maxes out at half the bitrate of stereo. MP3, for example, doesn't go over 160 kbps per channel if I remember correctly. Or by "256 kbps" did you mean "128 kbps per channel, which for most recordings that people in the studio will deal with means 256 kbps"?
If you turn the samples up until you can hear the noise floor, you can easily hear the difference. Of course, at those levels, a full range signal would launch your speaker cones out of the cabinets. So is that a fair comparison of 16 vs 24?
There are any number of ways to cheat an ABX test to your own satisfaction. If the goal is to delude yourself, you'll probably succeed.
1. Find post asking for results of a properly conducted double blind test.
2. Ramble on about your various stereo equipment for a couple paragraphs, show a complete ignorance of confirmation bias.
3. Completely fail to provide the requested evidence, wasting every ones time.
4. ???
5. Profit!
-1 overrated isn't the same thing as "I disagree".
I would fall into this group. My hearing is not good enough at this resolution, and the 16bit/44.1kHz rate was chosen because it allowed accurate enough replication of all frequencies within the 99 plus hearing percentile that it was deemed good enough.
The 192kHz/24bit applies to multi-channel sound, where it can make a difference, but I can't speak to the specifics why that is as that's not my area of expertise. I'd guess it's because effectively you'll drop below those key values and it becomes noticeable. Hearing is notoriously sensitive to direction, so the diffraction patterns have to make sense to your ears, or so I hear, at least when I was configuring the surround sound on my receiver.
The cesspool just got a check and balance.
Inaudible (super hi/low) resonant harmonic frequencies sustain a waveform over time = less volume needed - fidelity improved. But A/D D/A complicates things
Even TFA states that mastering comes into play very much, and I've noticed that 24/192 music usually is way better mastered than 16/44. It kind of makes sense, since why would you bother releasing 24/192 through a crappy analog chain, while 16/44 is so ubiquitous that resulting CDs run the gamut on mastering quality. While I agree you will not hear the difference between _perfectly mastered_ 16/44 and 24/192, I think there is a greater point which is missed, and that is mastering tends to run better with higher fidelity formats since crappy mastering is more obvious with 24/192. Maybe 24 bits will lower the noise floor more so high dynamic instruments (drums, etc) will come across a bit better due to less compression usually applied. Not sure.
Lemmings reliably prefer a cliff. Does that make it right or a sound choice?
The cesspool just got a check and balance.
Of course this is ludicrous.
No one can see X-rays (or infrared, or ultraviolet, or microwaves). It doesn't matter how much a person believes he can. Retinas simply don't have the sensory hardware.
I wouldn't be so sure... $10 IR filter goggles. The human senses do have limits, but they're rather soft and fuzzy. First, there's genetic variation in the exact sensitivity range (e.g. some people can perceive further into the "infrared" spectrum than others, it's a common high school & college lab experiment). Plus, pedantically, everyone can detect IR up to 3,000 nm at least, cooking would be highly impractical otherwise, and Beethoven felt for vibrations so he could continue composing/performing despite his deafness (IOW, our senses overlap, very important for concert goers that like to feel the bass).
Second, and more importantly, the raw signals are integrated by the brain in a semi-predictable pattern (obviously it's a self-teaching neural network, so people process things differently, although there are common trends). An insect has a compound eye with dozens or hundreds of photoreceptor units. Individually, they're not terribly sensitive, but when integrated provide a much clearer picture. It's akin to how photographers can merge multiple overlapping images to create gigapixel-level quality.
Given harmonics, pinna distortion and such, it wouldn't surprise me if hair cells do not impose an absolute limit on hearing, as the article states. OTOH, I doubt that 192 kHz offers any real sound improvement, but I don't think you can argue that with just biology, as there are few, if any, definites in that subject.
Go listen to Stuart Copeland tap on his hi-hats with FLAC, shn, cd-audio, or apple lossless, and then at 192.
Yup, the cymbals suffer the worst, though I'm not sure how much of it is due to the sample rates and how much is due to the psychoacoustic modeling (or which particular suband coder is being used).
My God, it's Full of Source!
OUTSIDE_IP=$(dig +short my.ip @outsideip.net)
Ehm have you ever read the Nyquist theorem? They clearly state that it is only valid under conditions. It is other peoples fault for using it to cover situations it wasn't proofed for.
I'll buy that a higher sampling rate makes certain reverb techniques algorithmically simpler. But if the original signal doesn't have any audible energy over 20 kHz, why store energy over 20 kHz in the mastered file? You can downsample from 176 or 192 kHz to 44 kHz during mastering. Then on playback, the sound card resamples it back up to 88, 96, 176, or 192 kHz to filter out ultrasonic images before handing it to the DAC.
Part of the problem is that you can't amplify the lower signals cleanly.
I certainly can't hear the difference between 44.1/16 and 192/24 with headphones, but when I'm cranking 1000W through a set of speakers & sub you do notice crappy MP3s or encodings
"...you can definitely tell the difference between 92KHz and 192Khz, and even straight CD tracks they were encoded from."
Right, because ultrasonics distort more at 192kHz, thus degrading the quality of the audio reproduction as it reaches your ears.
If you remove the ultrasonics, then you likely cannot. And even if you can, I don't care, because I can't. Feel free to disagree with science to justify your hefty investment and your belief that your ears and equipment are somehow better, that's cool.
TossableDigits.com: Temporary Phone Numb
Did you listen to it double blinded? No?
Just because I'm lazy about organizing my files I have some music tracks in both mp3 and FLAC. If I'm listening with good speakers and something with good sound comes on (e.g. Miles Davis - In a Silent Way) half the time I'll think, "oh, the cymbals are dead, I need to skip to the FLAC track."
Since the music player is randomly selecting the file I hear and I don't know which one is coming up, I think that satisfies double-blind criteria.
It doesn't eliminate a poor quality encoding algorithm, though.
My God, it's Full of Source!
OUTSIDE_IP=$(dig +short my.ip @outsideip.net)
There are entities I'm sure who could do marvelous things with the bandwidth above 44KHz.
These same entities also have plans for the extra 8 bits of color in your 32-bit colordepth capable devices.
A snare brush rustles at 192/24 instead of sounding like rustling paper.
While that's true, it would definitely also be true at 64/24, and likely at 64/20 I think. While 44/16 is a marginal format that with good D/A conversion can merely deliver what most equipment is able to reproduce, 192/24 is *way* beyond what anyone can hear.
Just curious - in the same way higher quality imaging allows for larger scaling, does higher quality audio allow, for instance, louder music to be heard more clearly?
I'm not deaf, but I've never spent more than $10 on headphones.
This signature has Super Cow Powers
No, they don't.
-1 overrated isn't the same thing as "I disagree".
It's true. I'm not really sure why the audiophiles are so obsessed with this.
This signature has Super Cow Powers
Too bad Shannon and Nyquist are dead. It seems they've completely misunderstood the math. How embarrassing they passed on before you could correct their mistake. Now they'll never know.
You completely missed the point--Nyquist of course understood the math perfectly, but most people who talk about it do not. If he were alive he would not be embarrassed, just terribly frustrated about his work being abused.
Than an "audiophile"? I think not.
Yup, it's one in the same.
Life is not for the lazy.
The 192 is the red-herring. 44/24 would be fine (we don't need more than 44kHz sampling once processing has been done, but having the recording mastered to a 24bit format would change the requirements for compressing the dynamic range. Also, in the case of hi-hats being tapped, they are quite quiet, and so don't use all of the 16bit dynamic range of CD. You'd be lucky to be hearing 9bits of it unless the mastering engineer has overdone the compression. The effect, then is one of bitcrushing to 8-9bits (vs 16-17bits dynamic range left in a 24bit recording) which one can learn to hear even when subtle.
-- The Grand Teddy Bear has Spoken: "Windows 8 Source Code Available NOW! more disgusting than your pr..."
This article is of particular interest to me because just recently I dropped over $1000 on a pair of high-end Sennheiser HD-800 headphones, but now I'm finding the amplifier's background noise is a lot more noticeable. Before, with cheap headphones that didn't have the same dynamic range, it didn't matter, but now it's the limiting factor.
I've got a reasonably decent FLAC collection, some of it classical music in 24/96Khz format, but the background hiss is detracting quite a bit from the potential quality.
What's a good amplifier for headphones that's optimised for a low noise floor instead of power, but isn't over-priced? I don't need a receiver with hundreds of inputs, I need something that takes a single digital input, and outputs the highest possible quality headphone output. Is there anything like that out there?
I'm not deaf, but I've never spent more than $10 on headphones.
You'll be in for one heck of a shock the day you hear what music actually sounds like.
That might hurt if it didn't come from a guy who is angrily whining about a pop-culture reference on Slashdot.
"I like to lick butts!" by MobileTatsu-NJG (#32700246) (Score:5, Informative)
Good God. Sampling frequency is not the same as bitrate.
forgot "abo".
The correct word is "verisimilitude."
English is already diverse enough that we don't need to invent stupid synonyms for useful words that already exist.
Sigh. Why is it that if people haven't heard something before, they immediately jump to the conclusion that the speaker is a lying bullshit artist?
Here's a Wikipedia article that discusses just one of the sound differences between English and other languages, a difference of inflection which English and Chinese speakers literally cannot hear when the other speaks.
http://en.wikipedia.org/wiki/Unreleased_stop
This difference was explained in my sociology classes in first year university as an example of how learned behaviours can prevent the ability to even perceive alternatives later in life, much less learn them.
As a personal example, I had a friend named Xu. He kept complaining that I mispronounced his name. But it wasn't intentional -- I literally could not hear the difference when he tried to compare his pronunciation with mine. It sound like he was saying the same thing twice. To this day, I've never been able to hear the difference between the "correct" pronunciation and what I used to say.
I do not fail; I succeed at finding out what does not work.
There are lots of double blind tests. Most that mean anything are between CD quality and above. No difference found after a year plus of testing. If you want to hear some differences in what's left out when items are compressed A refutation of the validity of double-blind audio tests The main point would be that a well mastered CD is better than a poorly mastered 192kHz/24 bit recording, and the same goes for a poorly mastered CD vs a 192 encoded well mastered piece. However, when the original quality material is of like quality, many can tell the differences until they get to CD quality. After that, a smaller segment can tell. What's been destroying music is the large group of folks who've never heard anything that wasn't put through a pipe filled with a wet sponge first. If that's all you've been exposed to, even the clear trill of a bird might sound unpleasantly harsh in its clarity.
The cesspool just got a check and balance.
Your data is irrelevant.
WRONG. The AES performed proper tests with audio professionals and audiophiles both, and neither could tell the difference after something was put through a 16b/44.1KHz digital stage. There is no degradation that can be detected by the human ear. The data conclusively says so, end of debate.
I enjoy listening to youtube videos even though the audio quality is most often crap. I know the artifacting is there, but unless it gets real godawful (like if an organist hits 12 keys/pedals at once and the mp3 just has no chance with its available bandwidth) the human brain - still one of the most amazing signal processors ever known - is remarkably able to tune it out. The really funny part is that you act as if this is a bad thing.
Sure, but encoding at lossless (which is what I do for albums that are important to me, rest are 192 kbps iTunes purchases) is entirely different than just wasting space. Lossless has a tangible benefit, whereas as the article points out, outside production, stuff like 24 bit audio does not.
It's the equivalent of encoding beyond lossless, just adding extra bits on top of a lossless encode that you'll never hear ever.
Don't forget the room, when you're talking about costs. The room you listen in is at least as important as the gear you buy--something audiophiles often overlook.
The "perceptual narrowing" you're referring to which occurs with speech does not affect the ability to distinguish frequencies of general sound or music. People are constantly exposed to large amounts of sound covering the entire frequency range from birth onwards. This has nothing to do with specific instruments.
This article argues with the use of scientific reasoning that when mastered correctly, there is no perceptible difference in audio signals and refers to research which would appear to prove this. This has nothing to do with individual instruments or "golden ears", as the author explains. He also explains many of the false reasons that people believe 192kHz to be of perceptibly superior quality.
Difficult to explain, but it reminds me of how some people say that there's no point having a frame rate higher than 30 fps. No, your eyes can't actually see the screen flickering above that frame rate, but that doesn't mean it looks perfectly fluid. The author is assuming the point of diminishing returns is actually a point of no returns, which may be far from the truth.
It sounds so good when you listen to it... But you won't hear it in a double blind. Smug superiority and ignorance of sampling theory is not something to be proud of. OTOH, can I interest you in a wooden knob for your amplifier? Only an idiot wouldn't hear the difference between this precision made piece of state of art sound tech and a cheap plastic piece of shit. You're no idiot, are you?
My "hefty" investment was only a few hundred dollars, because of dropping costs and, sadly, I can't really tell the difference anymore in higher level equipment. This is probably no more than your investment, unless you're listening to $50 commodity junk. The real problem with compression is the dropping of harmonics and other effects that add depth, including wave shapes that are not possible to replicate during compression modes, at least at those low resolutions.
Truth be told, the cost for amplifiers is THD at a certain output level, the lower the THD at the higher the level, across a broader spectrum, the higher the cost. In plain english, this means less distortion of the originating signal as it is amplified. And yes, this is something almost anyone can hear.
The cesspool just got a check and balance.
Given appropriate equipment and a person with a reasonable ear (mine aren't even that great and they suffice) and you can definitely tell the difference between 92KHz and 192Khz,
So you didn't read the fine article, I gather.
Did you run your ABX testing? No? Thought not.
Sig Battery depleted. Reverting to safe mode.
I contend that such tests are an indictment of blind listening tests in general
I stopped reading after that sentence and the seemingly endless stream of strawman arguments. The guy is a pontificating moron who wouldn't know good science if it bit him in the ass.
-1 overrated isn't the same thing as "I disagree".
Another (almost racist) example is the infamous inability of the Chinese to pronounce "L" and "R" properly.
You may have noticed that recent immigrants from China don't have this problem with their English. That's because modern Chinese students are exposed to and practice speaking English, so they've learned to recognize the sounds and pronounce the difference.
But 20-30 years ago, immigrants saying things like "Flied Lice" instead of "Fried Rice" was quite common, because they were literally unable to hear the difference.
I do not fail; I succeed at finding out what does not work.
$24 earphones?! You lucky devil.
When I was a wee lad, we had to listen to music through paper cones pressed to our ears. And they weren't real paper, mind you, but a great bloody lot of wasps nests glued together with our own spit.
Youngsters just have no idea.
So perhaps I should have correctly stated that "they migrate in large groups without apparently considering the consequences and sometimes dying en masse", but it's much easier to just say "cliff".
The cesspool just got a check and balance.
If you want to see illusory sup[eriority, read any large homebrewer forum.
Given appropriate equipment and a person with a reasonable ear (mine aren't even that great and they suffice) and you can definitely tell the difference between 92KHz and 192Khz,
So you didn't read the fine article, I gather.
Did you run your ABX testing? No? Thought not.
You apparently didn't read my post nor the one I responded to. Nope, not a word.
The cesspool just got a check and balance.
the trick is getting noise from the real world to sit quietly below the 7 dB loudness that a 16 bit noise floor gives us with an ideal listening environment (ie 83 dB SPL when presented with pink noise at -20dBFS in digital land).
i really hope EBU R-128 gains more momentum. it's been adopted in the broadcast industry very fast, but that's preaching to the choir. i don't think it'll ever make headway in the music industry unless apple rename it "iLevel" and insist on it - rejecting any music submitted to their store that doesn't meet the spec that they totally invented.
If it weren't for the fact that all popular music has its dynamic range compressed to provide maximum loudness for the entire song, dynamic range would be be a problem.
The problem is that, on soft passages, where the high 8 or 10 bits are zero, you're listening to 8 or 6 bit audio. That quantization can be heard. This is a problem for classical recordings made without any dynamic range compression. Of which there are very few.
This is an issue only if you listen to classical music in a very quiet environment. It doesn't matter for car audio. It doesn't matter for Apple's trendy crap earbuds. So almost nobody cares.
^ So says the article...too bad MPEG audio (including MP3) wasn't finalized until November 1992, with a public release in 1993, and formal specification in 1994...(first software mp3 encoder wasn't released until July 1994) :P
Unfortunately, such a gross overstatement kinda makes me doubt everything else in the article.
"A Goddess rarely smiles for she is forced by others to be an island unto herself." - Zephiris
I think I can find a compromise that should work for everyone: Why not just run the needlessly good 24 bit 192 hHz music file though a lossy compressor that does psychoacoustics well - something like AAC or maybe even OGG? Everyone agrees that the vast majority of the data in 24/192 can be thrown away with zero perceptible loss. Fine, let's do it. But let's do the bit discarding in some principled way, guided by a reasonable psychoacoustic model. Isn't that a lot better than indiscriminately downsampling to 16/44.1? By anyone's lights, a 16/44.1 FLAC at 1100 kbps will not sound better than a 24/192 OGG at 1100 kbps - or even 700 kbps, for that matter. The nice thing about this plan is that we have good models for the human threshhold of detection. Scientists claim that 16/44.1 is so good that any improvements on it will not be detected. Maybe, but what if they're wrong? Why not start with the data rich source and apply our acoustic models to throw out only the data that we know is FAR FAR FAR BEYOND our threshhold of detection? It would still be most of it, but at least we'd know we're throwing out the RIGHT data.
The Nyquist limit only applies with a perfect deadwall filter at half the sampling frequency before the digitizer, and an infinite-order reconstruction filter afterwards. Neither of which is realizable because both have infinite group delay.
In reality, with piecewise-constant or -linear reconstruction filters, you want to sample at at least 5 times the maximum input frequency if you want to get back something that fairly faithfully resembles your input. This is why digital oscilloscopes routinely have "100MHz; 1Gsps" written above their faceplate.
The most compressed examples of the loudness war usually still sound pretty much the same when downsampled to 8 or even 7 bits.
However, thankfully, classical music is not affected by the loudness war.
I've done double blind studies on my dog and he can tell the difference.
The quantization noise introduced by using 16bit rather than higher precision samples is 10 * 16 log 2/log 10 = 48 dB below peak. Can you hear this? Maybe -- maybe in low-level signals anyway.
From experience with Impulse Tracker back in the day, I can definitely tell the difference between 8 bit and 16 bit samples played in a quiet room (noise 24 dB below peak). However, this is with samples that weren't properly dithered before downsampling; I imagine the quantization noise would be less onerous if they were.
Classical music has a notoriously wide dynamic range; it's not inconceivable for there to be plenty of passages in a Romantic orchestral work that are themselves 24 dB below peak, and then the SNR is only 24 dB -- somewhat perceptibly not transparent, but the noise is probably nothing more than a slight hiss unless there's no dithering. (Of course, there is probably more than -24dB of noise in the analog original, anyway, if it's an orchestral recording.)
As for 192kHz -- it's not going to make anything worse, but it's not going to make anything better either unless you're trying to call dogs, thanks to Mr. Nyquist.
Yeah, what would a guy named xiphmont know about signal processing?!
Your data is irrelevant.
And the luddites score another convert.
Jesus was all right but his disciples were thick and ordinary. -John Lennon
Article is full of crap that is just plain wrong and misguided and the analogies suck. I'm not sayin 192KHz 24 bit is needed but the article is weird and says things that just are not true.
For example, saying that a stairstepped sine wave is mathematically the same is wrong --- stairsteps are impulses convolved with a square wave "impulse". This creates a roll-off at high frequencies. Basic signal processing. If you don't understand it, don't worry. Many don't. But the resulting sine wave will be the wrong amplitude. Sampling theory is based on infinitely fast impulses at each sample point, not stairsteps. A subtle point, but he misses many subtle points.
As for 192K 24Bit, there are reasons it is useful as opposed to 48Khz or especially 44.1KHz 16Bit:
1. Dynamic range. 20 bits gives 120dB +_ a few. But 16 bits (96dB) is not enough. 24 bits is way overkill, but doesn't hurt anything except storage space. Home theatre systems with 16 bits make audible noise when you turn them up. Put you ear next to the speaker when it is quiet and you will hear hiss. It may hurt your ears if they are there when when some sound comes through, but that depth is audible. His assertion that 16 bits is enough is not science, it's his opinion. (maybe even 18 bits is enough, but 18 is on the edge)
2. Simplicity of DAC - 192KHz means that dac filters can easily remove images. 96Khz is high enough to make the filter job simple, but 192Khz is simpler yet. His assertion that doing steep filters in digital is no issue means he doesn't really understand digital filters. Steeper slopes means higher lobes and more passband ripple.
3. All his talk about ultra sonics is laughable. Design a bad amp and it will sound bad. So what? Oh --- put in a bad signal so it will sound better?
4. My only point of full agreement is that you need good equipment first, 192/24 second. And I partially agree in that 192/24 is overkill.
There's a whole lot of snake oil in the audio business that needed some serious debunking.
-jcr
The only title of honor that a tyrant can grant is "Enemy of the State."
This article is bunk. I wasted 15 minutes reading through it. And they didn't even cover multi-tone and complex waveforms (which would have shown it to be bunk). Pure sine waves actually do well with digital sampling. But as you reach the edge of the Nyquist limit, you reach a point where the number of waveform states (how many sines waves of various values can be mixed) that can be rendered by the sample converges to unity. E.g. it can only support ONE sine wave at that point. Raise the sample rate and then you have the capacity to render multiple sine waves at the same frequency and many others.
A higher sample rate at say 192kHz is NOT done for the purpose of being able to encode sinusoid components up to 96kHz. It can do that (with that one sine wave limit that point). But is is appropriate to sample after a low pass filter (for example at 18 kHz) that limits the signals to only what you want. And then after conversion back to analog, clean it up with the low pass filter (again, at 18 kHz).
Listen to speech filtered with a 4kHz lowpass filter in an all-analog path. You will be able to tell it is filtered if your hearing is normal. Now digitize that filtered speech with an 8kHz sample rate. Convert it back to analog, and filter it again. The highs (up to 4kHz) will still be there (Nyquist says so, and this is valid). But there will also be new intermodulation products all over the place, especially among the high frequency components. It will give the audio a tinny or metallic sound quality.
Looking at it as combinations, a 44100 Hz sample rate at 16 bits is enough to render a 22050 Hz tone at any of 32768 intensity levels. However, if you have a 2nd tone of 22000 Hz, with each at 16384 intensity levels to avoid an overload, there are now 268435456 level combinations to be encoded. Now the 16 bits isn't enough. You need to double it. That can be done by either 32 bit sampling (hard to do) or doubling the sample rate (still 32 bits but now done as a pair of 16 bit samples). Fortunately you won't have mixed signals that high very often. However, you can easily have many signal components at lower frequencies. You will need plenty of bits for each. Even 192 kHz sampling is not enough to render 4 full range sine components at around 4 kHz. One or even a few levels of inaccuracy won't be heard. But these combinations rise very rapdily with just a few components.
For wider band audio with a higher sample rate, because most people hear weakly at higher frequencies, the effects will be less perceived, if at all. But they will be there, and a small portion of the population (including myself) can hear it.
Personally I'd rather they would go with 32 bits and 480kHz sampling.
now we need to go OSS in diesel cars
Unfortunately, we have to buy HDDs in pairs, since one of them must be mounted upside-down for the RAID set to cancel out digital dust.
At my age my ears are not so hot.
Seconded.
And this kiddies is why you should take warnings regarding exposure to loud noises seriously. Hearing loss is a real problem if you ignore those warnings. I did. Got myself a real nice case of tinnitus now.
All this talk about 44 and 48 and 192 is interesting.
But the question is, does it go to 11?
The article does not mention a digital source sweeping from 5Hz to 20Khz on a typical consumer grade CD player. I've looked at a few sweeps. Forget the lack of ultrasonic material recorded above 20Khz. The real aliasing between the sample rate and sampled music is the biggest reason for dirty sound in samples with higher frequency content. Only a higher Sample Rate will fix that. The Denon technical audio CD is a good source to test this yourself. It is digitaly mastered from a digital source for all test signals without any analog resampling. Good luck finding one. They are getting rare and fetch high prices.
http://en.wikipedia.org/wiki/Aliasing
http://www.amazon.com/Denon-Audio-Technical-Various-Artists/dp/B0000034ME
The truth shall set you free!
Based upon this comment, I'll assume you've been listening to music since before CDs were common. That means you also remember when AM radio stations actually played music. From your own argument, this means your ears are trained to ignore anything over about 5kHz. You CAN'T hear what you claim to hear.
As someone of kraut, dago, limey, paddy, and prairie-nigger descent, I'm highly offended. You forgot almost half my family. Thanks a lot.
My last hearing test has shown that I can hear up to 21khz. I play Tin Whistle, Great Highland Bagpipe, Ceilidh Pipe, and Guitar. I have heard the rattle of a live sax. I have heard a delicate triangle ringing out over a live orchestra. I have heard live trumpet. I've spent quite a bit of time training my ears to hear those sounds.
.wav .wav .wav and FLAC, encoded with the FLAC reference encoder
I have consistently failed to find a difference between the following in ABX tests I have run:
192/24 and 44/16
96/24 and 44/16
44/16
My reference tracks have been Pink Floyd's "Time", Sirenia's "Meridian", Bach's "Herz und Mund und Tat und Leben" part 7 conducted by Nikolaus Harnoncourt.
The reference system was a PC with an Asus Xonar Essence sound card, a Rogue audio Perseus pre-amp, a pair of Rogue M-180 monoblock power amps, and Vandersteen Signature 2ce speakers. (My father's sound system and my PC).
Of course, msobkow will claim that since I like Highland Bagpipes my hearing is inferior, and I can't hear the differences because he's better than me.
That said, I do like having music in 192/24. Why? Because I can play with it. I can edit it, there's more headroom. If I feel that "Another Brick in the Wall" just needs a tin whistle part, well, I'll have an easier time editing it in without distortion. But for listening? Nope.
Not a sentence!
This is what you would see from the apple store:
Crappy album, $8.99
Crappy album lossless audio $14.99
Crappy Album 24/192 $22.99
Even though the album was probably recorded at 24bit 96k or less.
People would buy the 24/192 version, play it on their computer who's audio driver is set at 16bit 44.1 over some cheap speakers who have a range of 60hz-12khz, that cant even be stressed by a low bitrate lossy audio files or on their 6.99 ear buds from their iphone.
While we could use an improvement in audio files for the end user, let keep it somewhat realistic.
While a higher sampling frequency and bit depth is a big help during recording and editing, much of the sound quality can be kept during down sampling when done correctly before distribution to the consumers. If people were downloading 24/192 files, they would use some free audio convertor to encode them to mp3 or m4a to put on their portable device, then we would be right back where we are at now.
And you expect us to take your opinions on sound quality seriously?
we're talking about sample rates (kHz). you seem to be talking about bit rates (kbps).
I once had the same idea that 192kHz is overkill, but I've been following a course on digital audio processing and I'm not so sure any more: it's not about the frequencies per se, but about the shape of the waveform. This has its influence on the timbre of the sound. Not that I'm convinced that 192k quality can be heard either, but it's not simply a matter of the Fourier spectrum and frequency response of the ear. That said, a lot of it is probably just nitpicking, if you want better sound I suggest investing in better speakers just like the article said. When playing high quality music through laptop speakers, the sampling rate isn't the reason why it sounds shitty.
"It's too bad that stupidity isn't painful." - Anton LaVey
The best part is that when people are arguing whether 192 is too much, he went over the top with 256 and 320!!! Another good reason to always put units after some arbitrary numbers.
Democracy is for the people; you only vote once per season and we'll do the rest of the work for you don't have to.
If you record sound with the Bass guitar at live gig levels, the bass goes squeaky with most forms of compression. Severe wave forms break during most forms of compression. With a straight WAVE file it sounds fine. That's what I have found.
The purpose of existence is to make money.
Because we should be long-sighted. Sound technology is evolving just like all other technology - in a decade, we may have drastically better speakers, for example. And at that time, I don't want to have to redo all my music.
If we can stay ahead of the curve, we'll be better off (just look at the betamax>blu-ray progression, didn't anticipate better quality TV's)
training doesn't make one's senses better. it trains the observer's brain to relay the appropriate signals, rather than ignoring them.
i can spot a boom mic in shot almost subliminally. i can spot jitter of all kinds, motion-compensation artifacts, compression artefacts, spots on film (white and black), and can even tell if a cameraman was running out of film, and when the roll was likely to end by looking at the subtle increase in spottiness. other people can't spot these things.
that said, my eyes are pretty poor. my ears are pretty poor, but i can spot when a (perceptibly) lossy source has been used in a master well before i whip out the spectral view. other people can't.
that said, decent mp3 (lame preset standard, or even medium) flies by undetected. ditto the equivalent transparent settings in all audio encoders. ditto a decent h.264 compared to the film scans it came off, when viewed with the same chroma sampling (otherwise it'd be cheating to compare 4:4:4 with 4:2:0).
my wife can tell you every ingredient that goes into a tiny sample of food. i need twice as large a sample to correctly identify only half as many ingredients. my senses are trained (though not as well), but not as sensitive. good thing considering i work in media production, not food.
my point - you're fooling yourself if you think you have better senses than an average joe - you've just trained you brain to pick different things. they probably enjoy the movie more than you...
Why bring up MP3! This article is about 44.1/16 vs 192/24. Use lossless comparisons, damn it!
Democracy is for the people; you only vote once per season and we'll do the rest of the work for you don't have to.
The reason the studios use it is because then often modify the music/samples. For an end listener its pointless.
Right, I'm sure that the problem is just that all those young peoples' music sucks, and literally no one appreciates classical and other forms of acoustic music any more. That's why no one can distinguish the difference, and it has nothing at all to do with the difference lying entirely beyond your ear's range of physical perception.
On warm summer nights I enjoy sitting on my front porch, with a dry gin made from hand-picked juniper berries, some artisan cheese and bread made out of flour that has been milled before sunrise. And if I am in the mood for it, I also enjoy 192kHz music with my bat friends. For us discerning people this is just a standard of living.
You make one good point: the proliferation of poorly mastered and encoded recordings has probably distorted the average person's perception of what sounds good.
Unfortunately everything else you say is pure bullshit. The author of the article got it right: there are advantages to high bitrates for recording and editing purposes, but for playback, anything higher than 48/16 is a waste. I don't care if you think your magical ears can detect that 192/24 is "sharper" or has a better "brassy rattle" or shoots literal rainbows out of the speakers. The science (both theoretical and experimental) simply doesn't support what you're saying, no matter how many insults and shiny adjectives you throw in.
Oh, and I'm a classically-trained clarinetist (in the last semester of my doctorate in performance) and a recording engineer. Literally 100% of the work I do is with live instruments. I think it's fair to say that I do "know the joy of hearing real music", don't you? I know you played in your middle school band or something, but maybe you could calm down and listen to the knowledge of people with advanced experience in the relevant fields.
when was the last gig you went to?
the BG noise level in any venue will be well into the 80dB area, even when everyone's been hushed. even at a quiet gig for quiet music (like a chamber orchestra in a polite suburb).
to get clear of that, the band need to play well into hearingdamageville.
if you wear plugs, you'll be getting at very best 20dB attenuation, and it'll be a non flat response - you'll definitely lose all the high end from about 10K up. also, you'll be hearing your own body at deafening levels.
if you don't wear plugs, then you'll be stripping your ears bare, and that triangle will indeed sound like a crackle.
if the band is not that loud, you'll suffer the noise floor of your surroundings and bang goes those extra 8 bits plus a lot more.
you really don't know what you're talking about.
Yep, and that tale is bullshit too. The captain's log didn't say anything of the sort; he just made the observation that the natives ignored the ships as if they weren't there. There is nothing to indicate that the natives couldn't see the ships.
there are many flavours of mp3.
i don't think you'll get much disagreement if you were to frame your argument "FLAC sounds better than your average mp3", though you'd still need to qualify that with "by average, i assume a mean bitrate of 128kbps and Xing as a reference encoder".
you'll never do an ABX though - it might lead to disturbing conclusions about the cables your stereo uses and the money spent on them.
here's a tip - use pro gear. it's cheaper and sounds better than the upper tier of the hi-fi market.
The ability of the wealthy to afford large hard drives does not mean file sizes aren't an issue for other less fortunate people. My hard drive is 75 GB and most of that is taken with important stuff, as is my external drive, so there's not much room for music and compression matters quite a lot.
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i've heard of straw men, but i love where you've taken it! straw generations!
i'm going to start using this term.
i'm turning 30 soon, so i presume i'm in your straw generation.
i've heard a lot of music. live, recorded, on good gear, on bad gear, in well tuned rooms, in poorly tuned rooms, in bars, in weird gypsy caves in ancient cities, in stadiums, or right into my ear from a cute, naked singer, chelsea hotel #2 style (this is the best way to listen to music).
i don't just pump the top 40 into my cloth-ears through white buds of mediocrity, though i look around and am tempted to believe some of my peers do. but to take myself as an average, i can't possibly reach that conclusion.
perhaps you fancy yourself to be markedly above average?
What? Dogs can't enjoy music now?
-- no sig today
They only determined there's no immediately detectable conscious difference. Now consider this research: http://jn.physiology.org/content/83/6/3548.full So frequencies we don't consciously notice affect brain activity. Thus your reference is not as conclusive as you imply; still need studies to eliminate the possibility that inaudible frequencies do not impact the brain's perception of audible frequencies in a subtle manner over long listening. I've been suggesting we need long-term listening blind tests with psychological assays for about a decade, but haven't found volunteers that want to go through the trouble.
"Politicians and diapers must be changed often, and for the same reason."
your methodology is wrong, or your soundcard is doing it wrong.
read the aes article, and try to reproduce their experiment and try it on yourself. if you still pick a difference, then you get to be king of the digital england.
Bit rate is different from sampling rate. But thanks for showing your ignorance.
44.1 was chosen to fit reasonably well in an NTSC video signal... there's some antique A/D converters out there that output composite and intended to use VHS tapes as media.
48 would have been better, and this was rectified with DVD, but the music industry lags behind...
Blind tests show that we perceive ultrasound: http://jn.physiology.org/content/83/6/3548.full So I suggest you GTFO. Albeit the effect is not conscious, no one has ruled out that it cannot subtly affect the perception of audible sound over long periods of time to the point where a conscious preference may develop in long term listening, without subjects of a study being able to describe the specific difference. In fact, this is more than plausible, given the reference I posted and others like it.
"Politicians and diapers must be changed often, and for the same reason."
I've got one better than blind tests, which are still based on introspection: _measure_ the effect precisely. And when you do, it turns out that the brain can perceive even ultrasound: http://jn.physiology.org/content/83/6/3548.full
"Politicians and diapers must be changed often, and for the same reason."
Though in fairness, I do collect historically-significant Linux distro ISOs.
Wow, I'm really impressed by that. Do you have the Linux disto that Jefferson wrote the constitution on or the one Hitler used to build the V2 rockets?
Big apple, new Yorik, undig it, something's unrotting in Edenmark.
are you michael kristopiet or something?
why don't you put us all out of our misery and ABX yourself - you clearly have time to do it.
Not wanting to go deaf, I use high quality devices with low THD percentages so I can listen at lower volume with maximum impact. Most people don't realize that high volumes are much less necessary as noise is removed and SNR goes up. With a very low noise level, you can play music at relatively low volumes that sounds incredibly good, whereas the high THD injection from a pair of crappy headphones or terrible stereo will cause you to turn up the volume repeatedly to counteract the noise.
- Michael T. Babcock (Yes, I blog)
Wealthy? 500GB is the smallest retail hard drive size worth purchasing these days, even with the stupid ramped-up pricing these last months.
- Michael T. Babcock (Yes, I blog)
Linear quantization never made sense to me as far as encoding audio. Human ears, like our other senses, are logarithmic. The difference in linear intensity between two soft sounds is far more detectable than the same difference between two loud sounds. Linear quantization is thus wasteful in one end of the absolute intensity scale, and possibly insufficient in the other end. Why use an encoding so far from the optimal? Hardware considerations are not a good excuse because the same digital processing circuitry that the average delta-sigma DAC chip in every piece of consumer gear uses to convert the audio into a high bitrate/low bit depth stream before actual conversion to an analog signal can be trivially modified to handle nonlinearly quantized inputs.
"Politicians and diapers must be changed often, and for the same reason."
And don't forget the interference effects you get when you have different sample rates. 192kHz is not dividable by 44kHz.
If builders built buildings the way programmers wrote programs, then the first woodpecker would destroy civilization.
You got marked flamebait and yet I can prove the same thing double-blind using Blu-Rays and uncompressed audio as well (cf. http://www.blu-raystats.com/Stats/Stats.php).
I've flipped between audio inputs for several people while watching movies without telling them; often starting the movie at the lower audio quality, sometimes at the higher -- and they have all said, even totally non audiophile normal people, "what happened?" or "oh wow that sounds much better, what did you do?"
To be fair, this is usually 24bit 96kHz audio, not 192, but it really does make a difference -- everyone claiming otherwise probably has terrible speakers or a horrifyingly high THD rating on their stereo equipment (you should check).
My Yamaha has 0.02% THD and I use 14AWG plain copper speaker wire fyi -- headphone listening is with a beautiful pair of DT770s.
- Michael T. Babcock (Yes, I blog)
The ability of the wealthy to afford large hard drives does not mean file sizes aren't an issue for other less fortunate people. My hard drive is 75 GB and most of that is taken with important stuff, as is my external drive, so there's not much room for music and compression matters quite a lot.
I think it's time for you to reacquaint yourself with current disk drive pricing. About six months ago, I got some 2TB drives at about $200 each. The 1TB models were half that and the 500GB even less. And, it you can't retrofit internal SATA drives, they have equivalent [self-powered] USB ones. So, I'm guessing $75 would allow you to upgrade your present system.
Like a good neighbor, fsck is there
Excuse me, sir, I don't believe you did a peer-reviewed study to determine if he was a troll or not. Until you can show me some data in a proper scientific journal that he is a worthless troll, I think it's an open question still.
Plus, if he is a troll they have those big pointy ears, so that's clearly how he got his great hearing. You know they live under bridges for the acoustics, right?
Big apple, new Yorik, undig it, something's unrotting in Edenmark.
Brick compression is the bane of my existence -- luckily it hasn't happened to quality movies yet.
- Michael T. Babcock (Yes, I blog)
Amen. I own some really good Jazz and Swing recordings, and the difference between the well-encoded and poorly-recorded variety is night and day. Sadly I love live recordings, but they're often terrible.
- Michael T. Babcock (Yes, I blog)
The last live gig I went to (I'm not the parent your replied to) was at Hugh's Room in Toronto, and you could hear someone put their glass down. The room was nearly silent aside from the band, because they were there to hear the band.
Pick your venues better.
- Michael T. Babcock (Yes, I blog)
I could maybe save you an additional 50%. I have a friend who is also deaf in one ear. You could go halfsies and spend only $12 on a headphone. Which one of your ears works?
Why are people marking every post by those with both taste in music and proper hearing as trolls? Its not trolling to post an opinion.
- Michael T. Babcock (Yes, I blog)
No amount of evidence will convince people like you -- every time they end with "that's fine for you, but most people ..." even if the evidence /is/ provided.
Get over yourself.
- Michael T. Babcock (Yes, I blog)
No offense, but what was the THD rating on the equipment you used for listening? It really does make a difference. If you listened with a sound card in a PC, you probably lost most of the difference to EM noise.
- Michael T. Babcock (Yes, I blog)
What you're talking about is a different sort of process from what the article is discussing.
With respect to language, any given language involves mapping sounds to syntax, in a process which simplifies what's heard for the purposes of language processing. Two sounds that are slightly different are both mapped to the same syntactic unit, like an "L" sound. Different accents, dialects, different languages that are closely related, can have slightly different maps, so that one person hears an "L" when another hears an "R". And, no language attached syntactic significance to every sound. Those that are not mapped to a syntactic unit are, for the purpose of language processing, ignored. This is why it can be difficult for learners of a new language to reproduce certain sounds: sometimes it's obvious that a speaker is making a specific sound that is a syntactic unit, but it falls between the sounds for two syntactic units with which you're familiar; or, it's a sound you're not used to having any syntactic meaning at all.
That's very different from the issue the article discusses, however. Language sounds are all well within the range of human hearing, whether you attach syntactic significance to a sound or not. The article was discussing the range of sounds that it is physically possible for a human being to hear, because of the physical characteristics of hairs attached to neurons in a human ear: about 20 Hz to 20 KHz. There's some individual variation: one person in this thread said he was tested as able to hear 21 KHz. But no one can hear 192 KHz.
24 bit is a potentially significant upgrade. 192khz is not. What really makes a difference is surround sound. 5.1 music sounds amazing and I'd take a 44khz 5.1 channel recording over a stereo (2.0) 192khz recording any day. Grab an SACD or DVD-A disc if you can find one and check it out (surround sound only, as I think most agree the increased resolution is pointless). It's the best sound upgrade you can make (if you can find something you like in the pathetically small amount of surround sound releases). There is a version of Blu-ray (3.0) that is audio only that I hope takes off as SACD and DVD-A are all but dead.
Definition of troll: person who doesn't really think whatever he's claiming to but fakes it for lulz.
Thus, by using the term you actually just demonstrated that you do care about it after all.
You're like a woman at the Olympics claiming that men aren't naturally stronger than women. Of course she's stronger than most men, but the statistic is still true on average.
You may be one of the few, but how many people of your age group honestly value even a CD quality track on a real stereo system over an MP3 with high distortion earbuds? No matter your personal experience, I think the poster's point was valid in general, don't you?
- Michael T. Babcock (Yes, I blog)
Are you sure you're not confusing 192 kb/s with 192 KHz?
You misquoted the article. But thanks for the link.
You seem to have purposely left out "and students" in the test group, that only someof the testing was done on high end equipment, and that the noise *was* perceptible but normally only at very high volumes.
Your evidence actually speaks against you, especially when you lie about it.
- Michael T. Babcock (Yes, I blog)
They don't care. Ironically the idiots pushing for lower standards don't understand psychoacoustics at all. For those with proper hearing, do a double-blind test of the same music samples in a variety of encoding qualities and rate them on a scale of both how good they sounded and how they made you feel. When you've finished, you now have proven only one thing -- your own preference. Enjoy that. I have mine.
- Michael T. Babcock (Yes, I blog)
As I posted earlier, that's a false summary of that document. Feel free to re-read it. The difference is audible, just not at what was considered normal volumes.
- Michael T. Babcock (Yes, I blog)
80dB of background noise at a classical music concert? I believe you may have confused this with techno concerts or Andrew W.K.. They've all very similar, I can see how the mistake was made.
Now, if we can get one of the latter two to conduct the first, we'd be in Epic territory.
Unlike the commenters to your post, I'm impressed. What do you do .iso collection is worth preserving and passing on to your heirs, if you have any.
to back up your data? I think both your music and Linux
You know, if I listen real intently to the hum of my computer's fans I start to hear all kinds of brief whispered sounds and bits of music and voices I've heard before. That doesn't mean they're actually there, it means my ears are feeding me what I want to hear.
The difference is, I actually recognize that this is a case of deluding myself and regard it as a momentary amusement, you think you're actually hearing the magical superiority of audio encoded at several times the minimum rate (good in principle, meh in practice) and 24 bits (pointless for listening). Did you know that you never had 24 significant bits in the first place since opamps pretty much never have better than 20-21 bits of linearity, and that at unity gain?
The mighty wiki disagrees: "The reported completion date of the MPEG-1 standard, varies greatly: a largely complete draft standard was produced in September 1990, and from that point on, only minor changes were introduced.[2] The draft standard was publicly available for purchase.[14]"
Analogies don't equal equalities, they are merely somewhat analogous.
There is one aspect that my rusty EE signal theory can no longer reproduce: what about phase accuracy of frequencies near the sampling frequency? Isn't it true that for a simple S&H quantizer, the amplitude information captured by the filter varies with cos(d), where d is the phase difference between the sampler and the signal?
I understand that higher-order filters are less susceptible to this quantization loss, but don't they do so at the expense of phase accuracy? Even if you sample at 192kHz or higher, how do you maintain both phase and amplitude accuracy if the reconstruction rate is only 44kHz?
That study is not without controversy. Cf. http://en.wikipedia.org/wiki/Hypersonic_effect
Though in fairness, I do collect historically-significant Linux distro ISOs).
You must be great at parties. ;)
Oh if I had mod points I'd mod you up. Yours is probably the most helpful comment. I was going through the comments trying to make sense of all of this as I couldn't work out why there's a difference between my older mp3s at 96k and my more recent ones at 192k (or more) if there isn't supposed to be. I hadn't realised the article was about kHz and mp3s are generally rated in quality by kbps (and after checking they all seem to be 44kHz).
Suddenly the article makes a whole lot more sense. Thanks!
Many people think a "factoid" is a small fact. Actually a factoid is something that sounds true, but is actually false.
You were mistaken. Which is odd, since memory shouldn't be a problem for you
Though in fairness, I do collect historically-significant Linux distro ISOs.
Wow, I'm really impressed by that. Do you have the Linux disto that Jefferson wrote the constitution on or the one Hitler used to build the V2 rockets?
Oh come on, everyone knows that Jefferson ran BSD and Hitler insisted on OS/2.
XML is a known as a key material required to create SMD: Software of Mass Destruction
http://en.wikipedia.org/wiki/Hypersonic_effect
You mean like, honkies, spics, niggers, dune coons, prairie niggers, kykes, faggots, chinks, canucks, wops, guineas, krauts, and polocks? I think that's everybody anyway, my apologies if I left out any group, I try to be an equal opportunity offender, challenging people to be adults and get over their group identitied. Criticism welcome. Cowardly disapproval spurned.
No no no... Porch Monkey. It's okay, we're taking it back.
You missed the "soulless" AKA Gingers.
The average person lacks perfect pitch, cannot tell the difference between SD and HD unless they're side by side, thinks their 128kbps MP3s sound alright, doesn't notice 60Hz jitter on their LCD, and so on.
As a music video junkie, I have noticed that with visual content added, I can more easily tolerate a bit crappier sound quality. You can chuck 128kbps down my throat if there's pretty pictures aside (it still doesn't make it hi-fi of course). Listened separately, it sounds dull. After all that's pretty obvious psychological note (as the senses are more saturated), but still interesting.
If george carlin were still alive, i bet he'd want out of that grave and not really care about someones post on an internet forum.
Audiophiles are some of the most amazing people I've ever seen. I've seen some buy $5000 power cords. Yes, that's five thousand dollars.
These guys should be left alone. Just shield any cable with gold and sell them for a couple of thousand bucks, making a 98% margin. That's what they want!
Write boring code, not shiny code!
This article blows. Its rife with errors and assumptions the author doesn't understand. I'm going to go kill myself now.
Sincerely,
Every AES Member
Can't read the article without paying...
Write boring code, not shiny code!
Not everything you think you perceive is stuff you actually do hear. Did you know that?
Write boring code, not shiny code!
I wouldn't take anything this guy would say he did ! There are methods to cheat on an ABX test. This guy is so sure he'll hear the difference that he'll cheat to convince us he can hear the difference !
Write boring code, not shiny code!
No professionally conducted double blind test has found any difference above 16/44. None. Even including people that claimed they could tell the difference before the test weren't able to differentiate anything above 16/44. The only ones that claim that are people that have never taken a properly conducted AB double blind testing.
Don't you find it intriguing? It's a bit like telepathy. Some claim they are able to do it. But it has never been proven and boy, have there been a number of tests on this subject! This doesn't prevent some mono zygotic twins to claim they could feel their sibling's accident from 1000km away.
You sound just like them.
Write boring code, not shiny code!
Why stop at 192khz/24bits (remember, it's 192KHz, not 192KBits/s). If you refuse most people can hear the difference btw 44.1KHz and 192KHz, why not crank it up to 10MHz? 10GHz? And why stop at 24bits? Why not 1024bits? 1Mbit? Did you do some tests or is it just some kind of gut feeling?
Write boring code, not shiny code!
96KHz isn't the audio frequency. It doesn't mean that the audio contains a 90Khz tone. It's the sampling rate. The higher the sampling rate smoother the signal.
Human perception wise a audio signal recorded at 96KHz sampling rate might well be indistinguishable from one sampled at 192Khz, but so is the file size between these files for practical purposes. I don't deceive my self thinking that I'm hearing better sound from a 192Khz file, specially considering that I'm using a basic pair of headphones on a my basic phone to listen to them. But my thinking is that future technologies might let you do interesting things with the extra bit of data which is useless to us right now. So given the choice I opt to get the higher sampled versions. Kind of like with digital pictures which are too noisy or blurred, but which might be cleaned up with future algorithms to give us a slightly more useful picture.
I may not be able to hear the difference between the quality increase, but, since the overall size of music files is so small relative to bandwidth and storage, I prefer FLAC et al simply for the sake of having a true archival version.
Well, technically speaking, finite-length signals can't be band-limited due to the uncertainty principle, and a band-limited signal which has been windowed in time will have some spill-over, causing small amounts of aliasing. Of course, in theory, this effect is really minuscule if you have a long enough signal, a good windowing function and/or not setting your sampling rate at exactly twice the bandwidth of the original unwindowed signal. The engineering rule of thumb pz came up with for oversampling would only be useful for ADCs and DACs due to limitations and difficulty in designing good analog filters. The intermediate storage format for the signal digitally would not really benefit much from such a high sampling rate.
You forgot tundra nigger. Mustn't ignore the northern peoples.
It is also the lone insult in a hilariously offensive five minute tirade that got a guy beat down by a cop up here. Everything else rolled right off, until he came out with that one.
Oversampling (i.e. 192kHz) allows much more room to develop a good anti-aliasing filter. [... filters] designed for the higher sampling rate can have more linear phase.
This is an especially serious issue for percussive sounds, which have both a very broad spectrum and a strong sensitivity to phase errors in reconstruction.
The broad spectrum means there's a lot of energy in the high frequencies that map into the audible range due to sampling aliases. Oversampling lets filters have greater image attenuation.
Percussive sounds are very short and the phase relationship between the harmonics must be maintained to keep them short when reconstructed. So a non-flat phase response in the antialiasing filters lengthens the time of the reconstructed sound. This is VERY audible, making the sounds "muddy" rather than "crisp". Phase distortion also interferes with reconstructing the apparent location of the sound source in stereo and other multi-channel audio systems. So the flatter phase response of the anti-aliasing filters that are possible with higher sampling rates produces a very noticeable improvement in the sound quality.
= = = =
I learned that last from Steve Eberbach, designer of the DCM Time Window loudspeakers. These had a very flat phase response, good enough to allow a listener to track thechanging location of the "veep" sound of a recorded accoustic guitarist's fingers sliding on the wound strings. In addition to not distorting the sound (thus not producing an acoustic image of the enclosure), the speakers also had a hack to cancel the reflection from the wall behind them, resulting in the acoustic effiect of the room's wall going away, becoming a window on the recorded performance. Thus the name: Time (because the response was flat in the time domain (phase) as well as frequency), Window (for the "window on the performance" effect).
CDs began to come out shortly after the introduction of these superb loudspeakers. And Steve had a lot to say about them. The low sampling rate chosen and resulting rotten filter phase response wiped out much of his speakers' advantage over the competition. (The choice of a linear, rather than a floating-point-like compressed, encoding also limited dynamic range, making quantization error audible as noise and intermodulation distortion in quiet passages.) Only listeners playing vinyl disks or dolby tapes could really appreciate the difference between his product and other high-end speakers.
Bantam Dominique roosters crow a four-note song. Once you've heard it as "Happy BIRTHday" you can't NOT hear it that way
The only reason you are noticing a difference in your 192kHz tracks is because the master is different. Different doesn't always have to be good, either. Yes, it would be nice to actually have the original material, that was recorded at the highest quality, and was edited in the highest quality, sent down to us at the highest quality, but that's not what actually happens. Maybe it will start to happen in the future, or we'll just keep up the Loudness War. It's possible that a very small amount of music is being released in high quality all the way down the line (Linn Records, Trent Reznor), but it's not what's happening with the majority. If you're trying to say that you're noticing a huge improvement in old music that's been remastered and released in 192kHz, it's due to been remastered, and very likely not the 192kHz.
I have a good amount of tracks from HDTracks (new and old), and a bit of it sounds better, but it doesn't sound better due to the resolution, it's because the actual tracks have been mastered to sound better - moving instruments around and doing a better editing job in general. Listening to tracks from the same album, the 88/96kHz encodes vs the 192kHz encodes, there's no difference.
In the grand scheme of things, we're all pretty much blind and deaf.
Ydco co
It is if you're trying to disprove facts with one.
Dilbert RSS feed
Is there a way to read this while at the same time not paying them $40?
Write boring code, not shiny code!
Hey Mike, how are those $1000 speaker cables working for you?
Your comment points out a huge issue with some sites that release high resolution audio, especially if it's older music.
For the last decade, people have been upmixing regular stereo CDs to 5.1, and doing it extremely well. There have been many cases where a few years later the studio releases its own 5.1 version, using the original material (supposedly) and it comes out sounding worse than a stereo upmix that some guy made in his basement. You can search Demonoid for classic examples of this happening, or just to get your hands on some of the upmixes, if you're interested (you'll have to be able to play DTS files).
Back to the point, I wouldn't be surprised if a large amount of the "classic" albums that are released with higher resolutions are just upmixes, which account for situations like HDTracks' Rolling Stones collection being released in multiples of 44. At the very least, it looks like the source material wasn't recorded in the highest quality possible, or maybe at the time the highest possible just wasn't where we are now.
"It's true enough that a properly encoded Ogg file (or MP3, or AAC file) will be indistinguishable from the original at a moderate bitrate." Rubbish. Any lossy format but particularly mp3 sounds GRUESOME to anyone with a trained ear. And untrained ears can certainly tell the difference once it's pointed out, usually on a good system. If you want to know how to get 11:1 compression ratio on a pseudorandom source like sound, it's simple - they throw away most of the information ,particularly spatial information in the upper frequency ranges. You can't "hear" some of those frequencies, but you can certainly perceive when they are absent. I personally cannot stand mp3s and never use them. FLAC all the way.
Ah, the bottom of the page has a description of how they remastered it. So, it looks like the originals were recorded in a multiple of 44 of some sort, otherwise they'd have introduced interference effects into the final product.
The Nyquist limit only applies with a perfect deadwall filter at half the sampling frequency before the digitizer, and an infinite-order reconstruction filter afterwards. Neither of which is realizable because both have infinite group delay.
In reality, with piecewise-constant or -linear reconstruction filters, you want to sample at at least 5 times the maximum input frequency if you want to get back something that fairly faithfully resembles your input. This is why digital oscilloscopes routinely have "100MHz; 1Gsps" written above their faceplate.
But to get around the problem with filters, only the A/D/A hardware needs to operate at higher sample rates. The actual bandlimited data can be stored in what Nyquist says, after it's been put through a nice long lowpass FIR filter.
There was already a perfectly good word for that.
while I might agree with you that 'real' instruments are harder to sample, your superiority complex doesn't help your case with those who do like electronic stuff. guess waht? we like good sound too, and it's even harder to get this stuff decently mastered, not like your jazz...
192kHz is not 192kbps. Sampling rate is not bitrate. A 96/24 track will have a bitrate of around 2500-3200kbps. 192/24 will be around 5000kbps. MP3 a 44/16 will be where ever you encode it to, capping out at 320kbps.
And yes, of course some frequency headroom in data storage is required for practical signals, to avoid aliasing with realizable filters. But if you have proper hardware for the conversions, it's nowhere near 5x.
Looking at Gr8Apes' other replies, he was definitely talking about 192kbps MP3s. Completely different thing from a track encoded at 192kHz/24bit and about 5000kbps!
That's not equal opportunity offence: anyone who matches two of the labels is offended twice as much as someone who only matches one. You should apologise to canuck faggots for double-counting them as well as to all those (e.g. limeys, frogs, lipstick-wearing pigs) you forgot.
No amount of evidence will convince people like you
Please get back to us when you find some within the realm of statistical significance.
Protip: There isn't any.
I gotta go with the AC on this one, as that stupid ass word is okay in politics where every damned thing is just a different degree of spin so "truthiness' can be appropriate but here I doubt its being done for some sort of spin, more likely its a classic case of "biggerer is betterer" and Lord knows we've seen that enough in society, everything from SUVs to Hollywood boomfests, so I have to say that stupid ass word just don't fit in this situation.
As for TFA? Meh I suppose its all pretty much relative and how much abuse your ears have taken on what sounds good or not to you anyway. Most here would probably gag if they picked up my MP3 player as its all 64k but after 30 years of playing rock bass with big ass amps combined with all the outside noise frankly when i'm out and about I can't tell any difference. Now of course inside is a different matter but i don't have a bunch of noise but even then anything from 192k through 320k sounds fine to me and i'm sure if you gave me a blind listening test i doubt i could tell the difference between lossless and 192k.
So why not just let folks choose from whatever size they want? Its not like the old days when we had to squeeze every bit of room out of our 10Gb HDDs, just give us 192k, 320k, and the 24bit 192khz and let us listen and decide for ourselves.
ACs don't waste your time replying, your posts are never seen by me.
You can't hear sounds if you haven't heard them before? Seriously? Do you really believe that?
It's true! Some sounds have to be 'learned' - for exactly the same reasons that a duck's quack doesn't echo.
No sig today...
192, 256 & 320? I think you may be slightly confused here. TFA isn't about the BIT sampling rate (Kbps), it's about THE sampling rate (kHz). 192Kbps != 192kHz. Your 256 & 320 encodes are probably 44.1kHz or 48kHz (98kHz at best). 256kHz-320kHz would be a weapon of some sort.
Suppose you two sound sources producing a 94KHz sound, and a 90KHz sound (say a sine wave). Individually the human ear cannot hear them. But played together interference will create harmonics, one of which is at an audible 4KHz.
196KHz is unnecessary as a final product, but if the sources are to be mixed then the extra range could create audible sounds.
The point of 24bit/96khz is that it allows for better processing of the data.
It's meant for music post-production (applying effects/plugins, remixing, etc).
Who says that 24bit/96khz downloads are made just for listening?
There are no "final mixes".
Those audio qualities are meant to allow remixing of the songs.
The problem with everybody commenting in this topic/news/thread is the premise that 24bit/96khz recordings are meant only to be heard. Yeah, there is no point in hearing music at that quality. But there is a lot of people that enjoy remixing music, and that is definitely what they need.
Having uncompressed, lossless audio of YOUR music (yes, you buy it, it's yours, and you own it, and you can do what you want with it, et cetera, et cetera, et cetera) allows you to do post-processing that you otherwise would not be able to do with a shitty compressed AAC.
Let's say I wanted to dub a song I own over a home video I took of my kid sledding. Let's say I wanted to add some effects to it. I could do this if I had high-quality sampling of the original. It would sound like shit if my source was a 128kbit MP3.
The higher the sampling rate smoother the signal.
Well... no. There's enough information in a low sampled curve. As TFA explains it, the output isn't "jagged" when played back in analog.
Human perception wise a audio signal recorded at 96KHz sampling rate might well be indistinguishable from one sampled at 192Khz
as explained in the article:
- Yup the human ear won't hear anything aboe 20kHz sounds, because it doesn't have any receptors for that.
But there are some real-world problems that come into the mix. No audio installation is perfect. You always get distortions.
- Thus, a 192kHz sampled file could contain frequencies up to 96kHz. These are sound which can't be heard in theory. In practice if you throw 96kHz frequencies to a sub-optimal speaker, the speaker can barf a lot of distortions, including distortion below the the 20kHz. So not only are you trying to output a sound that can be heard, but you force the speaker to produce bad noise *which* is audible.
But my thinking is that future technologies might let you do interesting things with the extra bit of data which is useless to us right now.
Hard to do anything with those bits at all. We simply lack the anatomic feature to do anything with them. Unless you do something like transpose everything at lower frequencie (slow down everything 2x = move everything 1 octave lower). At which point you aren't really outputing the original sound anymore. You're simply using the data to produce new sounds that weren't here to begin with.
The only practical use-case for this would be zoologist studying animals whose sound are beyond the human hear range. In that case "moving everything a couple of octave down" would help the scientist have an approximation with which he can work (to find rythms or other variation that are inaudible in the original frequency range). But that has nothing to do with hearing music made by human, for humans, with instruments designed for human hearing ranges.
Kind of like with digital pictures which are too noisy or blurred, but which might be cleaned up with future algorithms to give us a slightly more useful picture.
The situation with pictures is slightly different. What you're speaking about is spacial frequency. I.e.: resolution.
And human eyes can percieve way much more than some blurry low-res pictures. And in addition to that, there's this thing called zooming which makes perfectly sense to record picture at higher resolution. Because looking at details is simply looking at the same picture at another scale.
The "visual equivalent" to 192kHz sounds would be recording colours outside the human range. Like recording also infra-reds, microwaves, ultraviolets, and X-Rays.
Things that can't never been seen, because human lack the corresponding apparatus. The only way to get someting out of this extra data would be to transpose it into the visible domain. Thus use pseudo-colours to display levels of low infrared (heat), etc.
Just like the "zoologist" use-case above, there are a lot of scientific use-case where that could actually make sense (as an exemple, think about all the data collected by astronomers).
But in no way is it useful to record X-Rays to enjoy a painting by some known artist. The painting was done by a human painter, for human public, using colours chosen for their effect on an un-aided human visual system, disposed on a canvas in a way which is pleasing to the eyes.
(Well, okay. I know that some scientist use infra-red or X-ray image of paintings to analyse how they were done, what are the layers underneath or if there's even another picture over which the current one was painted. But these are scientist analysing the paint, so we're agin on the "scientific analysis" use-case).
24/192 makes sense as an intermediate format to avoid rounding errors, aliasing during filtering, etc.
There could be also some scientific value to keeping
"Sufficiently advanced satire is indistinguishable from reality." - [Tips: 1DrYakQDKCQ6y52z6QbnkxHXAocMZJE61o ]
What's pointless is any further debate about moving to 20MHz samples at 64 bits when music distribution has a much more serious (and actually real) problem. So much of our music is being destroyed beyond recovery before it even leaves the production desk.
No music I produce will succumb to this trick, ever. Perhaps that's why I don't get as much radio play these days.
"Nine times out of ten, starting a fire is not the best way to solve the problem." - my wife
Well lies are not the opposite of truth. But truth is the opposite of lies.
Lies are intentional falsification of the truth. But you can stick to believing non truths as truths and not really know that we are spreading falsehoods thus we are not lieing.
If something is so important that you feel the need to post it on the internet... It probably isn't that important.
In a perfect world, 16bit/44.1kHz might be enough. But we are living in a real world, that means that we and electronic devices are not perfect in implementing 16bit/44.1kHz audio. For example, we do not have a perfect electronic brick wall low pass filter. For mixing and mastering a recording, it is much better to have higher resolution materials than 16bit/44.1kHz. In fact, almost every audio studio does that in higher resolution all the time.
Out of interest, how do you argue against the original article, which addresses all of your points and reaches the conclusion that several decades of peer-reviewed research has failed to find any audophiles, let alone average people, who can tell the difference between 44.1/16 and 192/24? I mean this genuinely, not in a snide way.
Live music is clearly different in quality from recorded music, however I'd attribute this to the spacial and environmental limitations of recording (such that binaural techniques seek to eliminate, although I have personally not heard any), not frequency.
As for 192kHz -- it's not going to make anything worse, but it's not going to make anything better either
According to the article, it can make things worse.
Real-world playback hardware can make distortions.
A 96kHz sound is inaudible (for humans, at least).
But a 96kHz signal thrown on a real-world speaker might get distorted. And some of the these distortions can end up in the audible spectrum.
So instead of hearing nothing, you end-up hearing noises caused by something which shouldn't be heared and thus has nothing to do here in the first place.
"Sufficiently advanced satire is indistinguishable from reality." - [Tips: 1DrYakQDKCQ6y52z6QbnkxHXAocMZJE61o ]
When human perception is fine with 8bits per channel?
Because when you fiddle with the signal you need that invisible signal to retain your fidelity.
Reasons for audio fiddling include:
1) Sound balance within the room
2) Sound quality for listener presence
3) Sound quality for room acoustics
4) Sound quality for different speaker systems (including headphone)
And that sampling frequency only gives you the correct frequency replication up to the Nyquist limit. It doesn't replicate phase or amplitude correctly, you need oversampled source for that. To get the high C of a flute to sound different from the high C of a piccolo, you need to include more than just a sample at twice the frequency, since the overtones are at different apmlitudes compared to the main note.
So you do need 92kHz sampling. Or limit your ability to distinguish real-life instruments with a main frequency over 11kHz.
You got marked flamebait and yet I can prove the same thing double-blind using Blu-Rays and uncompressed audio as wel {...} I've flipped between audio inputs for several people while watching movies without telling them
No sorry. That's single-blind. They don't know it (they are blind), but you (the experimenter) are doing the flipping so you know (you're not blind).
Double blind would be giving both sample to a machine choosing randomly which signal to produce (A-B-X tests for example. You, the experimenter, give 2 samples to a machine. The machine plays A, then B, then chooses one of the two randomly and the audience has to pick up if it was A or B. Neither you or they know it).
Also, you're home made experiment fail to take into account:
- The switching between the 2 source is audible because the equipment switchs modes.
- There's no guarantee that the sound recorded in the 2 sources is exactly the same. Specially regarding the volume. Our brains are wired in a way that we think that anything louder is always better. If the 24/96 track is a few fractions of dB louder, the audience will find it inherently better.
"Sufficiently advanced satire is indistinguishable from reality." - [Tips: 1DrYakQDKCQ6y52z6QbnkxHXAocMZJE61o ]
Have fun spending money in useless moronphile material.
192 encoded samples are definitely poor audio representations of actual music
wow! you seem to know a lot about digital signal processing (sarcasm)
No no no, you're doing it wrong. You're supposed to say it like
I used to be an audiophile like you, then I took an arrow in the ear.
I would fall into this group. My hearing is not good enough at this resolution, and the 16bit/44.1kHz rate was chosen because it allowed accurate enough replication of all frequencies within the 99 plus hearing percentile that it was deemed good enough.
The 192kHz/24bit applies to multi-channel sound, where it can make a difference, but I can't speak to the specifics why that is as that's not my area of expertise. I'd guess it's because effectively you'll drop below those key values and it becomes noticeable. Hearing is notoriously sensitive to direction, so the diffraction patterns have to make sense to your ears, or so I hear, at least when I was configuring the surround sound on my receiver.
16bit/44.1kHz rate? Since when 16bit is the sampling rate? Of course you cannot speak specifics, you are mumbling about your own ignorance and making yourself a fool.
Now this one is so dumb that is even funny!
Hearing is notoriously sensitive to direction, so the diffraction patterns have to make sense to your ears
WTF are you talking about????????????
Lags behind? Meh. The music industry succeeded while the film industry was still pushing analog.
We didn't get digital 48KHz film soundtracks (let alone digital soundtracks of any sort) for movies in the home for another decade or more after CDs had become routine and commonplace.
Kid-proof tablet..
No, there is no audible difference between 44.1kHz and 192kHz if all you want to do is listen. However, if the intent is to do any post-production work, re-mixing, mash-ups, whatever - then the quality makes a big difference.
Try running time-shifting or pitch-bending (not dumb-resample where time and pitch both change), and I assure you, you'll get much cleaner results starting with the 192kHz file.
I own an $8,000.00 CD player, and a $3995.95 receiver, and I must say: I agree. "High-end" power cords are a fallacy built upon a whim built upon a notion to make money.
But there's a little bit to say about power: When the windows and the walls themselves are rattling, the overhead lights are dimming on every bass note, and the power supplies in the amplifiers are struggling to keep up with the diminished current availability, one does what one must.
Does that mean $5,000 power cords are cost-effective? No, of course not. But it might mean bigger power cords, more branch circuits, and perhaps a service upgrade to the house are worthwhile. Copper is copper at 60Hz@120VAC, and more of it is better.
(Please note that I have very little time/money in this ~$12k worth of silly high-end gear, so my confirmation bias may be lacking compared to someone who actually had something significant "invested" in such pricey kit.)
Kid-proof tablet..
A single photo receptor might not be able to see a transition shorter than X ms.
BUT
Your eyes and your head move around. Or objects themeselves can move around.
- Have a laser pointer.
- Have the laser light blinking, even at some ridiculously fast rate (200Hz).
- Move the laser point around, fast enough.
- You'll get the impression of a dottet line, not the impression of a moving point.
Your retina can notice things blinking at more than 200fps, even if single receptors can, just due to the relative momtion of the object inside the field of view.
Hearing frenquency range is fixed (well, mostly. I know /.ers can think of corner case, like when doppler effect comes into play). Your ear hears noises up to ~20kHz and nothing beyond due to physics and mechanical constrains. A 30kHz sound will always be a 30kHz sound (well minus the doppler corner case) and will never be heard. A 5kHz is a 5kHz sound no matter what and should be heard by anyone with an ear still able to detect 5kHz noises.
The video equivalent of this isn't the FPS question, but the wavelenght. An eye can only see visible light. You cannot see deep IR or microwave, nor can you see high -UV or X-rays (well again, corner cases: the repectors in the retina *should* be able to detect some near UV light, but the eye len blocks this light. And rightly so, because otherwise the UV will fry the retina. But some people with replaced artificial lens could see a little bit of UV).
insisting that 192kHz sampling is better, is like insisting that you need to be able to record from microwaves all the way up to X-rays in order to enjoy classical paints. sorry, no. You won't be able to see any difference in a reproduction of Monnet with and without the x-rays.
The fps situation is closer to the problem of number of speakers in a positionnal audio system.
In theory we have only 2 ears and can should only need 2 channels.
In practice humans move their head around. For a 2 channel audio to be positionnally perfect, you would need to track the motion of the head and vary the channels accordingly.
It's simply cheaper and easier to put a greater number of speaker and channels, and let the ears hear the difference caused by the motion of the head.
even if it's an technnical overkill, it's simpler that way.
For the same reason (specially with older CRT which could actually output it) its simpler to output at 150fps, rather than try to deal with and compensate for artifacts due to thing moving in the field of view at 30fps.
"Sufficiently advanced satire is indistinguishable from reality." - [Tips: 1DrYakQDKCQ6y52z6QbnkxHXAocMZJE61o ]
The interesting thing about that study is that they did find the high resolution masters sounded better than the CD versions. This was because they were different, better quality masters, and people had taken more care in making them. The improvement was still audible after the 16/44 AD/DA conversion, so it could not have been just the sample rate/bit depth.
Of course this is not an argument for higher sample rates, but for better quality master recordings!
Wealthy? 500GB is the smallest retail hard drive size worth purchasing these days, even with the stupid ramped-up pricing these last months.
Ooo... such a hot shot... how does it feel to be able to buy a new hard drive every year? I wouldn't know.
What are the bitrates/quality of the audiotracks that are sold online today? thnx
afaik, with my midclass Sennheiser phones I've heard quite a difference between 192kbps and 320kbs (i know you talk extreme quality at the moment, not the strong compression on CD quality)
(+1) I marked you informative and then... waded in to ask for more info please on such a pair of headphones as I'm looking to upgrade
sag
professional recording engineers, students in a university recording program, and dedicated audiophiles.
Yeah those sure sound like people who haven't trained to tell the difference. *rolls eyes*
You're a delusional moron, accept it and move on.
Heck, I'm getting old and I'm half deaf nowadays, and I can immediately hear the difference. There's just no comparison.
No you can't, your brain is lying to you as TFA said. Of course, it also explicitly said that this is for the end consumer and that higher quality was useful in the production pipeline. Needless to say a bad encoding would also violate the assumptions in TFA.
So no, you can't tell the difference between a proper 44/16 encoding and a 192/24 recording assuming the volume of both is identical (down to the 0.1db).
Only if your definition of "perfectly good" is "so convoluted that nobody EVER uses it". ;)
Let's be honest here, verisimilitude exhibits a superlative and ostentatious preponderance of syllables.
"Mind, as manifested by the capacity to make choices, is to some extent present in every electron." -Freeman Dyson
As a child, I was able to reliably hear 38KHz signals from an piezoelectric TV remote control.
So, that's >76KHz (Nyquist) just to satisfy my own childhood ears. 96KHz would do fine.
But I'm not so special (and wasn't than, either), and both storage and bandwidth are cheap these days.
So why 192KHz? I ask: Why not?
Kid-proof tablet..
Seems you got lucky with your onboard audio. My experience with onboard audio over the last three mainboards is as follows:
-Abit IC-7 from 2004: Lots of background noise. Scrolling the screen was audible as crosstalk on the headphones. Buying a 20 Euro Soundblaster Live (PCI) was quite an improvement.
-Asus M2N from 2007: Supposedly 24 bit high definition, which I don't quite buy in terms of actual quality. But good enough that I didn't bother to get a discrete sound card for this PC.
-Asus M4A78LT from 2011: OK (but not great) with walkman headphones at low volume. Unable to provide more than low volume to said headphones without clipping. Upgraded that one with an old Soundblaster Audigy I picked from someone else's discarded PC. Sound quality improved at all volumes and high volumes were now possible, as opposed to the onboard audio.
C - the footgun of programming languages
The way to go is to use lossy compression formats based on 24 bit raw data with at least 96kHz sample rate.
Reducing the file size drastically from that starting point is possible without any reduction in perceived quality. But doing that by the way the CD does (e.g. removing half the samples and cutting of the lower bits) does a really bad job of distributing the error.
Especially a dynamic of more than 16 bits is important for classical music or movie audio tracks. If you have a 60dB dynamic in a track, the silent parts will be quantized to 6 bits on a CD. A dolby audio stream will at a medium data rate will have much better signal quality than the CD in cases like that.
Of course the worst thing to do is to convert it to CD format first and then add lossy compression later, as you get the worst of both worlds.
He claims: "No peer-reviewed paper that has stood the test of time disagrees substantially with these results."
How about this one?
http://www.physics.sc.edu/~kunchur/Acoustics-papers.htm
My feeling is, I *know* I hate the sound of dither. And I *know* I hate the flat sound of stuff missing. And I *know* I hate the audiophile super-precise 16- and 32-bit mods/chiptunes and so on, even when they're made by producers with huge experience in audio processing and studio work and... musical theory, and so on. And those digital songs are created by people who should, optimally, be producing the best possible stuff to listen to. But instead the only people who like it are apparently called Seapunks.
Where do I stand? I don't really know. I don't care so much as long as the song I'm listening to sounds as good as an analog recording. In my experience, that happens around 24 bit, 192khz. I don't know *precisely* where it happens, but I know the next step down the digital compression staircase, (164 isn't it? I don't remember) has noticeable losses, and 128 is intolerable for most music. And we're talking about, hmm, almost doubling in size. And we're still talking about megabytes, not anything huge. So I make the sacrifice, and I don't hear any of the things I hate: *dither*; digital conch-shell effect (great now I sound like a Seapunk); "something's not there"; no bass; none of the high-end distortions or hisses I know should be there from experience listening to that synthesizer; etc.
The author gripes a little about "training" the ear making people think they have better hearing. He also goes on about how the wider range is needed in the studio to have more room to work in, but from experience I know if you screw up a recording, once two layers of sound are mixed you aren't going to take that mix and magically move one of them around without also moving the other. But he's talking about side-effects and so on. What it sounds like to me is he's saying "well, if everybody was using transparent oversample filters both in the analog-to-digital and digital-to-analog transition, and if everybody had really fine and precise playback and speaker equipment, and if everybody was a perfect sound engineer and producer and everybody was a perfectly trained listener, there'd be no reason to go to 24 bit 192khz."
And yet there are all these little indications along the way of how the wider range and higher frequency are useful for correcting errors. So it sort of dawns on me, he's asking for *more* effort out of the world in order to justify staying at a width and frequency that have *less* to offer, and his major argument is the amount of space it will all take up. So it sort of fails Occam's razor in a way.
So am I wrong about my reasons to keep using 24 bit 192 khz? I've been doing that for years, and I only go into all this because people are starting to ask questions. Like the other day I was reading an article that bewailed our fates at the hands of "all these people who are producing music for the iPod-headphone crowd".
I had to stop, like, wtf? What's an iPod headphone got to do with it? Then I realized, I make music for the JVC marshmallow earbud. The original ones that still cost around $20, not the new trashy model (which I have, now, and which I hate) that only cost $14. Am I some kind of culprit of some kind of some shit or other? What am I doing wrong? I mean, arguably, my digital tracks are equally for people who buy really low-range response giant speakers for their cars, and I do that on purpose because it's funny. So I have a reason.
But where are all these people suddenly coming from who have these really huge bones to pick with entire industries and crap? What does it all MEAN?
((If you wonder what I'm talking about when I mention 16/32-bit mod music.... fine, if you want to force me to do it, there's a bunch of stuff you could dredge up from the 90s but here's basically THE top result for searching for such stuff: http://modarchive.org/index.php?request=view_by_moduleid&query=34414 ... if you want me to rip my own dick off, force me to listen to that cymbal crash on constant repeat))
"Stratigraphically the origin of agriculture and thermonuclear destruction will appear essentially simultaneous" -- Lee
No, better DACs will fix it. A typical consumer grade player may well have a lousy cheap DAC to eke out a few more microcents of profit for the manufacturer.
I've done comparison listening using FLAC on mobile media players, and the quality of the DACs used is the distinguishing characteristic, closely followed by the quality of the amplifier. The winner is still Cowon, whose iAudio range is well-known for high-quality DACs, and still my favourite to carry classical music on.
Mart
"I know I will be modded down for this": where's the option '-1, Asking for it'?
Even the best D/A converters are inaccurate beyond 11 or 12 bits. That bascially means that 16 bit lossless is really 11-16 bit lossless.
Just go to a good headphone forum and you can quickly grab a $80 pair of headphones that are insanely good. Just don't buy those POS Dr Dre things that are worth $20.
1) 192 khz sampling -> up to 96 khz frequency.
2) Subliminal advertising for cats !!
3) PROFIT !!!
Well said. Let me pitch in: I have a background of 11 years of classical guitar, and I like to listen to classical music. I can spot a lossily encoded file at bitrates that create a significantly better than 50% compression over FLAC, which is why I carry my classical stuff as FLAC.
I cannot, however, hear a quality difference on the same equipment using better than 48/16 sampling.
Mart
"I know I will be modded down for this": where's the option '-1, Asking for it'?
Sorry dude, but if you can't afford a new HDD once per 2 years, then you probably don't have as much music as the wealthy guy does. Those pirated FLACs are a different story.
FYI: Taking the highest price per GB for storage would bring you to $3.5(Enterprise Class 512GB SSD). Those $3.5 let you store 4 FLAC albums that would cost $9.99 each. Thus your costs for owning and storing 4 albums is $43.47. Per GB cost of a HDD is below $0.10 these days. I really don't know what are you talking here about.
You must be a nigger.
"on my home file server"
"Total size, under 600GB."
This is where you left reality. Believe it or not, most people don't have a file server setup in their home. At best, they have a 2nd HD in the 500GB to 1Tb range, the former of which would be entirely encompassed by that music...
one person hears an "L" when another hears an "R"
Can't you fix that by swapping the red and white cables?
How about this one? http://www.physics.sc.edu/~kunchur/Acoustics-papers.htm
Abstract is:
"Many misconceptions and mysteries surround the perception and reproduction of musical sounds. Specifications such as frequency response and certain common distortions provide an inadequate indication of the sound quality, whereas accuracy in the time domain is known to significantly influence audio transparency. While the upper frequency cutoff of human hearing is around 18 kHz (or even lower in older individuals) a much higher bandwidth and temporal resolution can influence the perception of sound. Non-linearities and temporal complexities in the auditory system negate the simple f ~ 1/t reciprocal relationship between frequency and time. In our group's research -- which lies at the intersection of psychophysics, human hearing, and high-end audio -- we measure the limits of human hearing and relate them to the neurophysiology of the auditory system. These experiments also help to define the criteria for perfect fidelity in a sound-reproduction system. Our recent behavioral studies on human subjects proved that humans can discern timing alterations on a 5 microsecond time scale, indicating that that digital sampling rates used in common consumer audio (such as CD) are insufficient for fully preserving transparency."
That said, I do like having music in 192/24. Why? Because I can play with it. I can edit it, there's more headroom.
Right, and this is the point that the article entirely ignored. I'm usually listening to a lot of live stuff, and often encode to 44.1/16 (lossless) for listening, which works fine. But so much work goes into many of the recordings, if the source is 192/24, that's what gets archived and maintained.
I don't buy TFA's claim about 192kHz introducing distortion effects, from my experience that is totally false.
"Somebody has to do something. It's just incredibly pathetic it has to be us."
--- Jerry Garcia
Hey! You missed 'mick'. This is discrimination! Also I think you meant 'polack', you merkin wanker.
Because taste in music is irrelevant and claims to superior hearing are unsubstantiated.
Even if the human ear could not tell the difference at normal playback, a higher rate will allow it to be played back slower with a quality the human ear still can't detect. This is important if you do a lot of editing.
You can see this in high speed video, when it gets played back at a slower rate than recorded, it still seems very smooth, but if you slow down something recorded at the normal rate, it is clearly not as smooth; Audio works the same way.
If you are plannign to do any kind of audio editing it is even more important to get it in a higher rate format.
44/16 .wav and FLAC, encoded with the FLAC reference encoder
You do realize this was a completely pointless test? That (regardless of what flac encoder/decoder and regardless of its settings) the decoded flac file will always be bitwise identical to the original wav file *by definition*?
You willfully leave out nerds, geeks, dorks, and spazzes? Obvious /. bias! ;)
I8-D
lemmings don't prefer anything. lemmings just walk forward. and occasionally, SHOULD I DECIDE! they will all pull their heads off and pop like champagne bottles.
world was created 5 seconds before this post as it is.
Old meme is old, but credit for the creative twist. I chuckled.
Hail Eris, full of mischief...
E pluribus sanguinem
it's quite possible that you were just hearing 38/2 khz. hearing sound produced by that 38khz sound rattling the device.
Jefferson didn't write the constitution, idiot.
Hitler wouldn't have been a Linux user because he detested communists. :P
Hail Eris, full of mischief...
E pluribus sanguinem
Damn, I hate getting to these threads late, especially when it's a subject that interest me so much. Always some clown with an offtopic first post (modded up of course) followed by an answer to the offtopic post that's modded offtopic when it isn't. I'd have to wade through hundreds of responses to find any real insight or information.
TFA is exactly right and exactly wrong.
If you're listening to modern, popular music, a 16 bit sample is more than sufficient, because popular music has no dynamics. Even when they digitize the old analog music that was engineered to give the best dynamics physics would allow the medium to have (think Boston's first album) they compress the dynamics to make it "loud." I mention Boston because the band's leader was really pissed off at how bad the CD sounded.
But if you're listening to classical, with its very soft passages, loud passages, and especially when there are cannons in the recording, you want as large a dynamic range as you can get -- and with digital sampling, that means as high a bit rate as you can get. The very soft (compared to the loudest) sounds will have the same as an eight bit rate or lower -- the highest crest of these waves will take fewer than eight bits to render.
As to sampling rate, that depends on your output transducers, whether speakers or headphones. If you have a boom-box type setup with a four inch midrange and a subwoofer (most common these days), the sampling rate doesn't matter much because your speakers aren't going to be able to accurately reproduce the 15+kHz tones accurately anyway. However, if you have good (read: expensive) speakers, with each one having say an eighteen inch woofer, two midrange drivers (squawkers) of different sizes, a good tweeeter that will go up to nearly 20 kHz and what they used to call a "supertweeter" with a range of 17-30kHz, those expensive speakers are wasted on a 44k sample rate.
At that sample rate a 15kHz tone has only three samples. With only three samples there's no way to accurately draw the waveform. With three samples there's no way to discern between a sine wave, a square wave, or a sawtooth wave.
We now return you to your regularly scheduled offtopic jokefest.
Free Martian Whores!
I will take truthiness over the mental masturbation that is this article. The sampling rate should be adjusted for each and every track. But putting the idea out there that its a crappy choice is a lie too. I will dump compression any day for the original WAV. Then this argument truly is utterly pointless.
After reading a considerable amount of this growing debate, I have this to address to the people who staunchly support the article's premise:
I get tired of all this "probably" assumption. How probable is it that everybody in the world is going to grab the best possible equipment for recording, conversion, amplification, reconversion and playback and make sure the entire chain from creation in studio to recording to distribution to downloading decompressing and playback is going to involve all of this fucking equipment and that everybody's going to use it properly? Give! Fucking! Up! You fucking... all you autistic chart-wizards make LESS sense than the people you accuse of being fucking "audiophiles"! Your ear, for example, isn't a fucking test tube with a formula written on it! People like you remind me of this one "mentally superior" moron who really did think that a circle was just a 360-sided polygon. You'll cite all your expertise, but just listen to the shit music that gets recorded in 16 bit and 44.1 khz: it's a bunch of fucking chiptunes and weird ass math-audiophile .MOD tunes from the 90s, that sound like exquisite dogturd. Frankly, I'd rather have this hugely "unnecessary" range, frequency and sampling rate that do nothing but TAKE UP A FEW MORE MEGABYTES, and listen to the world's IMPERFECTLY recorded music produced on ANALOG instruments and catch all those imperfections than worry about the seemingly autistic insistences of a handful of overanalysers like your camp.
"Stratigraphically the origin of agriculture and thermonuclear destruction will appear essentially simultaneous" -- Lee
I wish you guys would get this right. There is absolutely no way you can tell the difference between a 15kHz sine wave, square wave, or sawtooth wave (apart from amplitude, perhaps).
Sawtooth waves have even and odd harmonics, and square waves only have odd ones. This means that the first harmonic of a 15kHz sawtooth wave would be at 30kHz, and the square's 3rd harmonic would be at 45kHz. As you pointed out, even if you could hear them, you'd have to have damn good speakers to reproduce.
Three samples is enough to reproduce the 15kHz fundamental per Nyquist.
I do hope you're maintaining proper .cue sheets for those CDs. I always find it funny when people rip CDs to individual .flacs per track and throw away metadata, like lossless is only important for the audio.
Get free bitcoins: http://freebitco.in
The main article cites this: "Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback," as a representative confirming source that there is essentially no perceptual audio difference between CD and SACD bitrates. However, it is clearly not a scientifically done test nor are the authors in any way scientifically trained. Moreover, they ignored several important factors in doing listening tests.
First, they do not define what the expected outcome should be, that is, they seem to state it as 50% from
"were the same as chance:49.82%"
but then in the next paragraph they start making comments like
"Females got 18 in 48, for 37.5% correct"
If women are getting 37.5% correct when the statistical expected outcome is 50%, then there is a correlation.
If the expected outcome is 37% then the authors should have explained the reasoning for different criteria.
Second, one of the most important factors in listening tests is whether or not the music was very familiar to the human testers. For example, I might be able to pick a particular version of the "1812 overture" from 16 vs 24 bit but I would not be able to do that for "Eminem" - When I know a particular piece well, my ability to discern differences increases. The article did not mention this at all.
Lots more anyway the upshot is just that it is not a scientifically done test nor writeup.
Even if not consciously audible, the higher frequencies have effects upon the perception of audible ones.
This has been scientifically tested, even going to the level of measuring brain waves.
Converting from one sample rate to another, provided it's done using a proper asynchronous sample rate conversion algorithm, will be just as acoustically transparent as converting between two rates that are multiples of each other.
Having the two sample rates you're converting between be multiples of each other, or rational, does help with the computational efficiency somewhat. But other than that, it's mathematically the same process.
The worst assumption you can make is that since one audio sampling rate is a multiple of the other, it's an easy process of just "adding and dropping samples". It's not; any rate conversion process has to be combined with a filtering process in order to prevent high frequencies aliasing to low frequencies (if lowering the sample rate) or low frequencies being 'duplicated' up into higher frequencies (if raising it).
(DSP engineer here, I've been writing audio processing code for almost 10 years..)
For most people, there is no place where sounds above 20 kHz will irritate a nerve ending enough to send an impulse to your brain. Thus, no sound higher than 20 kHz is audible, and 20 kHz corresponds to a 40 kHz sampling rate. (One sample at the low point on the wave, the next sample at the next high point, etc.
The problem in your analysis is that a "sound higher than 20 kHz" may be inaudible, in the sense that you don't detect a sustained sine wave at such a high frequency. But the Nyquist theorem applies to Fourier components---infinitely long unmodulated sine waves---rather than intuitive "sounds." Modulated sine waves at audible frequencies have Fourier components above audible frequencies with audible effects on the modulation.
Mike O'Donnell http://people.cs.uchicago.edu/~odonnell/
It almost feels too easy doing this, like beating a 5 year old at chess but..
U mad bro?
Even if not consciously audible, the higher frequencies have effects upon the perception of audible ones.
This has been scientifically tested, even going to the level of measuring brain waves.
It has it's uses. None of which have anything to do with listening :P
I use my field recorder at 96khz a lot... because if I play it back at half-speed, there's double the information in the high end you can get to. This is especially cool with sounds from birds and insects. Things you can't hear normally, and still couldn't hear if I had recorded at 44khz and slowed that down.
For large sets, this will be our guide even unto death, for the LORD will work for each type of data it is applied to...
And neurophysicists conclude that while the higher frequencies might not be consciously percepable that does not stop them having effects upon the perception of the audible ones.
They went to the level of measuring brain waves.
http://en.wikipedia.org/wiki/Factoid
"A factoid is a questionable or spurious (unverified, false, or fabricated) statement presented as a fact, but with no veracity."
Give me a link to a .flac or similar lossless that you think proves the point - I'll mp3 it at 192Kbps and abx test. I'd love to be proved wrong, but I've not yet been able to distinguish the two with any of my music.
Get free bitcoins: http://freebitco.in
we had some costumer that we had to put on satellite but that is extreme cases (people living over 8KM cable distance from the co servicing the line or with extremely bad cables)
Long distances to the DSLAM and undermaintained cables are the reality in the more thinly populated parts of the United States.
Is the 5GB/month the absolute max or the one 80+% of people chose as the next tier is significantly more expensive?
The latter. Providers of big downloads or streams have to plan for the tier that customers actually have. But because agricultural technology has shrunk the fraction of people who need to live in rural areas to grow food, providers of big downloads or streams appear to ignore satellite users and target urban and suburban demographics, optimizing their PC- and TV-targeted offerings for DSL, cable, and fiber.
Years ago, SUN microsystems promoted a nonlinear quantization, called "mu-law." A key problem is that the nonlinear function has to be applied to the sum of many frequency components, so it causes cross-modulations between them. A particular example: a low amplitude high frequency signal component may appear and disappear as a high amplitude low frequency component varies between 0 and its maximum. Since a high frequency component is much louder than a low frequency component of the same amplitude, the effect can be quite dramatic.
Mike O'Donnell http://people.cs.uchicago.edu/~odonnell/
Maybe its just me or perhaps the Canadian disposition, but I don't think Canuck is really all that offensive (compared to some listed).
I can think of a few more not worth mentioning. There is a few on the list I have heard of, but really don't know what they mean, which I am OK with really. Some of the war ones, seem quite mundane, though perhaps they started out as code or something, like Jerry or Charlie, etc... Which actually reminds me of The Cryptonomicon and using the word nip, as a shortened Nipppon.
It seems many slurs probably came out of wars, I wonder how many were specifically contrived purposely to try and dehumanize a group simply to make it easier psychologically for soldiers to kill them. Which really if you think about it, makes it even more offensive to use such language. Anyway as my grandma told me, sticks and stones may break my bones, but names can never hurt me.
The other point is that even listening to 192k/24b properly means you need to send the data bit perfect to a 24bit / 192k capable DAC. People playing these new high res files through plain old software players on their computer and then out their sound card as analog to go to their preamp are kidding themselves. That kind of audio chain isn't going to be good enough to benefit from 24b/192k and pretty much explains the "I don't hear any improvement" result.
With digital EQing and convolution $24 headphones or canalbuds can sound just fine. Frequency response is the most important factor affecting quality of sound for both headphones and speakers, and this is exactly what you can fix with a good equalizer. I love the PortaPros I have that cost 20€ on sale after just a very crude measurement of impulse response with the free and excellent DRC and a convolver audio effect. On my Sansa Clip+ with Rockbox I use the 5 band parametric EQ to fix the sound of my Sony EX50LPs, which are my most used headphones despite me owning full size headphones and canalbuds 5 times the price (which are great, too, and will no doubt last me longer, but are not as tiny, convenient and care free).
Of course there are many factors you cannot fix with EQ - distortion being a big problem with many types of headphones, quickness (as measured by waterfall plots), sensitivity and impedance (you want these to be a good match with your source), noise isolation, repeatability of seal, not to forget the inaudible but important factors such as comfort, build quality and style.
The ideal frequency response of headphones is still open for debate - most headphones shoot for a diffuse field response. Regardless of ideal most headphones have obvious flaws in their frequency response that can be fixed with the tools available for free.
While 44/16 is a marginal format that with good D/A conversion can merely deliver what most equipment is able to reproduce, 192/24 is *way* beyond what anyone can hear.
That's true, but irrelevant. The point is not whether some alleged audiophile can hear a 96kHz tone (because they can't), but whether it's easier and cheaper to design a filter that has no phase distortion at 20kHz, but is down 48dB by (1) 22.05kHz (e.g. ~-200dB/octave); or (2) 96kHz (e.g. ~20 dB/octave). The answer, objectively, without any audiophile or golden ears claims, is the latter.
You must not be using Monster uranium tipped, cables with platinum mesh shielding. The casing is made up ground up of unicorn hooves, and leprechaun tears. A Native American Indian shaman then did a special secret ceremony than imbues the cable with special supernatural powers.
I can go way beyond the mere mortals 192khz, 320khz is the absolute lowest that I use. Only my cables let me fit that large a sound file down it, as the fatter the file, the more cable you need!
Most of the comments here are covered in the fine article. It makes a hell of a lot more sense than most of the magical thinking being espoused in these comments. Most of you are regurgitating the same myths that the article dispels in detail. You just end up looking stupid posting comments on an article you clearly did not read.
The conclusion "lossless formats and a decent pair of headphones will do a lot more for your audio enjoyment than 24/192 recordings" does not prove the headline "24-bit/192kHz Downloads Is Pointless"
The article points to the visible light spectrum as analogous to the audio spectrum. This makes more clear the faulty reasoning. Light is not sound. Light is quantum at its source not analog. The best analogy for quantum (to explain that mysterious atomic effect in human perception terms) is.... digital! Analog does not resemble it so much.
What is "Fluorescence"? How do overtones and ultrasonic noises interact with audible noises? The answer is simply: They are. Do we understand it fully? No. Sorry, but that is not what Science means. If you think that Science means we know all these things for certain then you don't understand the word science. For you science has become a religion of certainty and false security
What organ causes hearing and where is "sound" created? The Brain.
To me this argument is like a teenager trying to say that only certain drugs will get you legitimately high. That someone 'couldn't really" have narcotic effects in their brain because they weren't using the "real" stuff. (Near beer vs real beer, or lotsa vodka vs only beer...) But one is conflating the mechanism with the final effect. If the final effect is psychological then ALL TRICKS TO ACHIEVE THAT END ARE VALID. Yes, there a physical limits that are known and are important and it is important to debunk pseudoscience that is glossing over that stuff. However, sometimes that is simply not the important question. Sometimes the important human effect is not in the realm of the known or is in the realm of the psychologically subjective. In that case, don't discredit the whole branch of human knowledge called science by applying it to something for which it is not suited. Like the question of "what kind of art is scientifically best" you are getting into angels on a pinhead territory and then look to the company you keep.
Stupidity is its own reward.
The author may be correct that 24/192 offers no advantages, he is wrong in saying that it is slightly worse.
While he's correct that frequencies much greater than >20khz can cause problems downstream, the problem is at least as bad with 16/44. The sampling theorem says that 44khz sampling is enough to correctly reproduce frequencies in the audio range (half the sample frequency). However, 16/44 reconstruction requires prefiltering and I believe can also introduce spurious high frequency components (the Nyquist theorem says nothing about frequencies higher than half the sample rate), so a brick wall filter is needed to remove any frequencies over 22khz, so you need to filter out high frequencies in either case. At least with a higher sample frequency you can use a more gradual filter, which is better in theory (though probably no different in practice). In particular, 24/192 will not sound any worse than 16/44
Such an elementary error calls the value of the whole article into doubt
There are reasons why this bitrate and sampling frequency are used and it can be heard. It is not futile and it is not just big numbers for the sake of it. I speak as a technical director for an AV company and as man who built several home and project studios and was part of 3 major studios migration from analog to digital technologies. I once was a teacher and technical supervisor in a sound design school.
24bits:
-you can and you WILL hear the difference with 16bits. It basically record finer amplitude variations than 16bits, therefore the dynamic range is increased and there is less approximation of values when the sound is digitized. the end result is that stereo spacialization is usually better as the right and left channel amplitude differences are closer to reality and very fine variation will lead to audible different result. Less quatization noise is heard (the low amplitude pop corn noise and "8bit feel" you get when listening to low amplitude digital recording) as lower amplitude values are represented with more bits and therefore are less coarse. More importantly any processing playing with amplitude is rendered much more accurately with finer detail. A digital compressor/limiter won't screw you stereo image, an expander won't bring more distortion to your mix by amplifying quantization noize for example. Echos and especially reverb will be MUCH finer and accurate, the tails won't cut off and won't sound like white noise.
192KHz:
- this one is tricky as it has a lot of use in the studio but it can barely be heard even on the best systems. Basically pretty much all AD/DA system uses brickwall filters to filter frequencies above 20KHz, the limit of a very healthy ear, so as to prevent foldback frequencies. The higher the sampling frequency the softer the slope of that filter is because the foldback won't happen until 96KHz is reached compared to 22KHz on 44.1KHz. the brickwall filter at 44.1KHz is harsh and many people with good sound system were complaining (me included) that it could be heard and was annoying. At 192KHz it is softer, enough to not be a disturbance. On the other hand the most important reason and use for 192KHz is latency. When recording someone in the digital world you have to deal with the fact that as a certain number of samples will have to be created before what goes in, goes out, at 44.1KHz there was an audible, annoying delay, if audio was processed it was unlivable for most musician. at 192KHz this delay is essentially eliminated and only the most discerning musician will be annoyed by it. So in that sense 192KHz is not really needed for most people and indeed very few people have systems that will indeed let them hear the difference with 44.1KHz but it is there.
I guess we all like to believe there is a big evil industry in all domain that make us buy stuff we don't need, I like to believe my i5 750 is as good a an i7 960 for what I do but the reality is the i7 960 IS better and with the right application the difference means a lot. Same goes for cars, a 2001 Toyota echo will get me around but a more expensive cars will get me around in more comfort will less issues. Same goes for audio, most people using gaming headphone or 5.1 gaming audio setup and cheap all-in-one sound system will never ever hear the difference between 16bit 44.1KHz and 24bit 192KHz but it doesn't mean it is not there and it doesn't mean it is not significant and that it's a lie. It might not be a necessity but for professionals like me (as in "it is how a make a living" not as in "I am an expert, listen to me") it is significant. For audiophile it is significant also and for people who listen to music all day (ear fatigue will come in much later with 24bit 192KHz than with 16bits 44.1KHz).
That's true, but irrelevant. The point is not whether some alleged audiophile can hear a 96kHz tone (because they can't), but whether it's easier and cheaper to design a filter that has no phase distortion at 20kHz, but is down 48dB by (1) 22.05kHz (e.g. ~-200dB/octave); or (2) 96kHz (e.g. ~20 dB/octave). The answer, objectively, without any audiophile or golden ears claims, is the latter.
Yes, but it's also pretty damn easy at 96kHz... Or even 88.2 or 64. It's just tricky at 44...
BTW, despite all the loud claims here, there was a double-blind study long ago that found that some people can hear the difference caused by harmonics up to about 30kHz. Nobody can hear pure tones at those frequencies of course, but the interference patterns with the base frequencies affects the "tone" of the sound. Unfortunately, that was so long ago that I wouldn't even know where to start to look for a reference. Anyways, give that the upper range of what might affect quality, slightly, for some people, is around 30kHz, then you could argue against 64. But 88.2 & 96 are still not hard to construct appropriate filters.
And yeah, I know, nobody's equipment is going to reproduce those 30kHz 3rd harmonics anyway. But if we're talking about a format delivering everything that could matter musically, rather than throwing away what most people won't notice, then 44.1kHz is inadequate, but 192kHz is still overkill.
I imagine that the whole reason for 192/24 is analogous to 12-16 bits per channel for color images--not that anybody can see that, but that in digital processing you're going to get rounding & truncation, and by the time you're done processing, you have effectively "re-quantized" to a lower resolution, so that if you start with only the resolution humanly perceptible, you end up with perceptible degradation. So for the original master format, you need a few bits more than for any final product.
You do realise that the phase is already smeared by the microphone, the preamplifier and probably a channel EQ? To say nothing of room acoustics in the playback environment.
The majority of vinyl cut since the mid '70s utilised a digital delay. I'm sure the surface noise and scratches really benefitted from these speakers though, especially records cut on lathes using non-oversampling 32kHz 12 bit delays. As for the phase accuracy of consumer grade dolby decoding... HA!
I sent this article to my friend, who owns a recording studio, and has been doing Audio Engineering for 15+ years. This is his response by text:
I find a couple problems with this article. For one: hardly any studio records at 24/192. 24/48 is more common along with 24/96. Recording at a higher sample rate decreases the risk of aliasing as they have pointed out but then they say to use over sampling to fix this problem. Which is somewhat right, but a lot of plugins being made namely compressors introduce a shit ton of aliasing that if recorded at a higher sample rate wouldn't affect it as bad. Over sampling helps this if the plugin is designed to have over sampling. Most do not. The benefits of recording at 24/192 is mainly for people who work in film. When using pitch and time manipulation the quality of the sound has less artifacts when using lower sample rates. Still, it would be better to not have to convert sample rate and bit rate down to consumer level. Think of the sound of a WAV compared to an mp3. But also brings me to my last gripe: Consumer products such as cd players, iPhones, stereos and such have very crappy Digital-to-Analog converters which would defeat the whole purpose of having higher quality audio anyway. The converters in these devices are probably worth a dollar or so. It would be like running a blu-ray movie thru a 1960s television. No point. Although the article brought up good points, overall I would have to disagree with it and find their argument full of shit. My 2 cents. Lol
Careful with that strawman. I never asked for a description of the difference, or if other people can hear a difference. I asked for any indication that the GP can tell a difference and isn't simply talking out of his ass.
-1 overrated isn't the same thing as "I disagree".
Really? Because I'm pretty sure I stated upfront what evidence I would need to be convinced. I don't think it's unreasonable either. I've never ended a conversation with a statement like that because the evidence is never presented and instead I get lame descriptions and obvious confirmation bias. Don't let that get in the way of your trolling though.
-1 overrated isn't the same thing as "I disagree".
Like FoolishOwl said that's connected with the processing of language; as per Wikipedia: Japanese speakers are, however, able to perceive the difference between English /r/ and /l/ when these sounds are not mentally processed as speech sounds.
That's true, but irrelevant. The point is not whether some alleged audiophile can hear a 96kHz tone (because they can't), but whether it's easier and cheaper to design a filter that has no phase distortion at 20kHz, but is down 48dB by (1) 22.05kHz (e.g. ~-200dB/octave); or (2) 96kHz (e.g. ~20 dB/octave). The answer, objectively, without any audiophile or golden ears claims, is the latter.
Yes, but it's also pretty damn easy at 96kHz... Or even 88.2 or 64. It's just tricky at 44...
32kHz would require a roughly 90dB/octave filter. That's not so easy.
BTW, despite all the loud claims here, there was a double-blind study long ago that found that some people can hear the difference caused by harmonics up to about 30kHz. Nobody can hear pure tones at those frequencies of course, but the interference patterns with the base frequencies affects the "tone" of the sound. Unfortunately, that was so long ago that I wouldn't even know where to start to look for a reference.
No need, I know it well.
Anyways, give that the upper range of what might affect quality, slightly, for some people, is around 30kHz, then you could argue against 64.
No, again, I'm arguing against 64 based purely on the low pass filter you want to design that is flat at 20kHz with no phase distortion, and down 48 dB by the Nyquist frequency. Forget what some alleged person can hypothetically hear - I'm talking about 20 kHz... and really, I'm even talking about phase distortion at even lower frequencies.
But 88.2 & 96 are still not hard to construct appropriate filters.
And at 192kHz, it's even easier.
And yeah, I know, nobody's equipment is going to reproduce those 30kHz 3rd harmonics anyway. But if we're talking about a format delivering everything that could matter musically, rather than throwing away what most people won't notice, then 44.1kHz is inadequate, but 192kHz is still overkill.
Again, you're focusing on the highest frequencies people can hear. That's irrelevant to my point, which is that, the higher your sample frequency, the smoother and gentler your low pass filter can be, without any effects lower than 20kHz.
I imagine that the whole reason for 192/24 is analogous to 12-16 bits per channel for color images--not that anybody can see that, but that in digital processing you're going to get rounding & truncation, and by the time you're done processing, you have effectively "re-quantized" to a lower resolution, so that if you start with only the resolution humanly perceptible, you end up with perceptible degradation. So for the original master format, you need a few bits more than for any final product.
Mostly correct... You're absolutely right for 24 bit - and in fact, most high end digital audio processors work at 32 bits internally. But that's just bit depth... sample frequency is unrelated to that. Where sample frequencies matter is where I said - the antialiasing filters.
Yeah great let's crank it up so we can hear the glorious audiodouche quality for about 20 minutes before our ears start bleeding. What a fantastic idea.
-1 overrated isn't the same thing as "I disagree".
Ok, I'll link.
http://isohunt.com/torrent_details/371429905/the+police+flac?tab=summary
Just about any Police song with Stuart Copeland on the drums, which is nearly every Police song, which is why I referenced The Police in my earlier message. The song I quoted at the end is a good example.
Any time you've got a percussionist like Stuart who is in love with clanging metal (hi-hat, cymbals, glockenspiel, triangles, chimes, etc), you're going to have a lot of high-frequency harmonics that MP3 encoders fuck up every time. Indeed, I cannot point to a single song that I have that has high-frequency stuff in it that the encoder has not fucked up at 256Kbps and below.
I cannot describe the distortion outside of using the word "swishy."
A curious song that does not have clanging metal that MP3 encoders fuck up is "Sad To See The Season Go" by Cowboy Junkies. Encoders have problems with Margo Timmins' and her backup singers' voices on this song as they are nearly in phase and on the same frequency. I have yet to see an MP3 that has not fucked up the harmony at 192. The MP3 algorithm was tuned to the human voice (in particular Suzanne Vega's voice). There is something about this song that plays havoc with the algorithm.
--
BMO
Those 'real' instruments you speak of are based on mathematical principles in a very similar way to synthesizer instruments. In fact, synths can go one step further and make ANY sound imaginable, allowing for potentially much better sounds than what the restricted real world can dish out.
Besides, melody, harmony, intricacy, orchestration, variety and other factors are what makes music great or not, not some false conception of how 'real' the instruments are.
Why OpalCalc is the best Windows calc
Judging by the modding I guess Louis C.K got mod points.
"If you are going through hell, keep going." - Winston Churchill
I'm a proud Hillbilly, lived my whole life in the WVa hills, even managed to have a career as a software developer, only moved out while I was in the service/drafted.
Not that we don't like NYC, Caribbean Islands, the different hills and mountains in Colorado, WY, AZ, NM, etc.
But you can call me Hillbilly and be accurate. I think it's illegal to discriminate against Hillbillies in Cincinnati, where lots of us have gone looking for good jobs.
Think of the Irony!
Go watch some live music! This is how real musicians make a living, by coming to your town and showing you a good time. Take advantage of it.
And when you do, bring some ear plugs. Some $12 earplugs from etymotic research can change your life. They attenuate the sound, without muffling it. I go to concerts about twice a month if there's anything good, and honestly they sound better with the ear plugs. If the music's so loud it's beyond the linear range of your ears, it's no fun anyway.
Give me Classic Slashdot or give me death!
Guess how long ago 1992 was? That's not exactly a gross overstatement - rounding off by 2 years?
Nah, it just shows where his confusion is. He's thinking 192, 256, or 320kbps, not kHz. The fact that 192kbps is the range where all but a *very* few people stop being able to distinguish the MP3 from the CD or FLAC recording has him thinking that with his 'golden ears' he'll be able to hear the difference at at least half again that data rate, but not recognizing that kbps and kHz discussing different things.
Definitely an audiophile.
Unless you watch it on a TV with the tv commercial loudness filter turned on. Basically a dynamic range compressor. And out of the box, it's usually turned on and stupid people like it that way.
Rather than insult you, I just ask that you at least try to perform a blind trial on yourself. I know that in the past sometimes, I've been very surprised at what I really, truly think to be better, only to be confounded when I mix them up without knowing what I'm listening to.
It's an INCREDIBLY easy mistake to make. You owe it to yourself to at least do some self-research.
Why OpalCalc is the best Windows calc
missed crackers
The post walks us through some of the basics of ear anatomy, sampling rates, and listening tests, finally concluding that lossless formats and a decent pair of headphones will do a lot more for your audio enjoyment than 24/192 recordings.
Right--because if *you* can't see a use for it at the moment, there must not be one... ...or maybe you're just plain wrong.
There's no place like
Thank you. You took the words right out of my mouth. That statement (about 15 KHz sine, square, sawtooth) perfectly summed up how poorly the poster understands digital signal theory.
As a DSP guy, you're probably one of the few that would really, truly appreciate this book.
Please stand clear of the doors, por favor mantenganse alejado de las puertas
I misunderstood the 192/24. I thought they were talking about compressed MP3s at 192. I should have read the links before I posted. I have no experience with comparing uncompressed rates that high. Everything I said about comparing compressed files to uncompressed 16/44 is true for me though.
Bad boys rape our young girls but Violet gives willingly.
BTW, despite all the loud claims here, there was a double-blind study long ago that found that some people can hear the difference caused by harmonics up to about 30kHz. Nobody can hear pure tones at those frequencies of course, but the interference patterns with the base frequencies affects the "tone" of the sound. Unfortunately, that was so long ago that I wouldn't even know where to start to look for a reference.
No need, I know it well.
Can you provide a reference?
No loss from the original sampling, i.e. they didn't loose any information in the compression. Most music is sampled at (correct me if I'm wrong someone?) 44kHz, I forget how many bits, I think 16. The thing being touted is sampling it at 192kHz with 24bit resolution, which is much higher on both counts, and therefore, in theory, should produce better quality reproduction of the sound based on oversampling and reduction of the signal to quantization noise rate. The point the TFA makes is that human ears can't hear the difference, although I think that some audiophiles may beg to differ.
FWIW, I have quite bad ears, a recording needs to be quite bad before I notice it. I'm an electronic engineer though, so I know all the theory...
The human ear can hear the difference between 44kHz and 88kHz or at least I can. But when you are talking about 192kHz that is dog ear territory and no human no matter how good their ear can hear frequencies that high.
88kHz is as high as the human ear can handle and thats if you have a very good ear.
I looooove the melodious distortion I get from using single ended WE 300Bs as my output stage. Ella never sounded better!
Your hard drives last longer than a year?
Just because it CAN be done, doesn't mean it should!
BTW, despite all the loud claims here, there was a double-blind study long ago that found that some people can hear the difference caused by harmonics up to about 30kHz. Nobody can hear pure tones at those frequencies of course, but the interference patterns with the base frequencies affects the "tone" of the sound. Unfortunately, that was so long ago that I wouldn't even know where to start to look for a reference.
No need, I know it well.
Can you provide a reference?
Here's one, and here's another.
Basically, the idea is that ultrasonic tones (say, 30kHz and 29kHz) may be inaudible, but generate a difference tone that is audible (at 1kHz in that example).
Mike, the point is both sides are wrong--the audiophiles and. I'm right smack in the middle, because I have both experience as a musician and as an engineer who's built and designed audio equipment. While I've been criticizing those opposing your side for the most part in this story, it's because of the preponderance of skeptics on slashdot. On the other hand, many audiophiles clearly don't understand that blind testing is critically important, and that yes, it is possible to carry out blind testing in a valid way that is beyond reproach, and that indeed many things that audiophiles love do not make a difference (speaker cable floor stand-offs? shakti stones?). The best case is when people who are both scientifically minded and rigorous in their approach, yet into audio and understand audiophile concerns, perform research. Then you get stuff like Geddes and Lee's blind tests showing that THD correlates very poorly with perception of distortion, but that specifically weighted metrics can in fact correlate well with perception (due to the ear masking some types of distortions and being very sensitive to others). In other cases, you see people like the uber audiophile skeptic engineer Douglas Self come around on some points and recognize that some things he did not think make a difference in fact do, after discussions on diyaudio.org and measurements he further performed as a result.
"Politicians and diapers must be changed often, and for the same reason."
It's never that simple, though. There's clear intent in a post such as yours to imply a certain generalization, given the overall subject of the article.
"Politicians and diapers must be changed often, and for the same reason."
Thanks for the link.
"Politicians and diapers must be changed often, and for the same reason."
Wouldn't this only be an issue with processing? I was talking about encoding for storage and transmission only.
"Politicians and diapers must be changed often, and for the same reason."
Umm, TFA talks about 16 vs 24 bit and addresses the question of dynamic range. It's the mastering that's the problem, not inherent issues with 16 bit audio. Somebody else already addressed your comment about sample rates. Did you actually read TFA?
and a "regular" CD.
But we have paragon amps and proper monitors.
And a proper listening space.
The assumption that it's just your ears that contribute to perception of sound, the assumption that people only perceive sound up to 20kHz, these and other assumptions and statements made by so many self proclaimed experts here are demonstrably incorrect.
You'll never really experience the kind of audio reproduction that is possible with $15k worth of high end audio equipment, and that's fine, it's not something that's "worth it" to you. I'm sure most posters don't even have a proper place TO listen to high-end audio. It's not for everybody. It isn't being a snob, it's just an interest you don't share, or really know very much about.
the interwebs are srs bsns
Uhh, yeah, almost any sound card and most motherboards are equiped with S/PDIF. I doubt that was an issue.
Well... no. There's enough information in a low sampled curve. As TFA explains it, the output isn't "jagged" when played back in analog.
No, but it is *aliased*. The waveform between two samples is a simple interpolation. It is probably pretty close to the original sound, but there will always be some error too.
Simple math problem:
- take this *aliased* waveform. (the result of a join-the-dot interpolation).
- compute the "error" (i.e.: substract the original perfect waveform from what you consider an aliased thing)
- do a fourrier transform on this error (i.e: look at the harmonics).
- all the frequencies which compose the error will be above the audible frequency range
- i.e.: you won't be able to hear the difference. i.e.: the aliasing isn't audible
that means your ears don't give a damn fuck about the aliasing.
And that's using the "join-the-dot" misconception, which doesn't even exist when playing back on real-world equipment.
Linear interpolation (actual "join-the-dot") did make problems back in the module-tracker era, when 8kHz instruments samples were interpolated into a 44.1kHz soundcard output.
because then, some of the "error" was in the audible range (4kHz to 20kHz).
"Sufficiently advanced satire is indistinguishable from reality." - [Tips: 1DrYakQDKCQ6y52z6QbnkxHXAocMZJE61o ]
The only ones that claim that are people that have never taken a properly conducted AB double blind testing.
And these same people have the power of invisibililty too. But only when no one else is looking.
I think most of the folks in this argument aren't realizing they are arguing about the wrong thing.
My "CD Quality" 192 kbps MP3 rips are ripped at 44kHz. 192 kHz in this context IS overkill for any human.
Wouldn't this only be an issue with processing? I was talking about encoding for storage and transmission only.
No, this is precisely a problem for playback. With sublinear encodings, there is no way to present a low amplitude component of a sound accurately in the presence of a high amplitude component. Since perception is sensitive to components at different frequencies fairly independently, the loss of accuracy in the smaller component can be quite perceptible.
In fact, nonlinear representations, such as floating point and phasor representations, are often good for certain parts of processing, but not for playback.
Mike O'Donnell http://people.cs.uchicago.edu/~odonnell/
Oh good lord, why didn't I think to check wikipedia. Well, anyway, thanks for those pointers.
You're actually wrong. Human ears are relatively good at hearing phase relationships and volume relationships between sounds, as these are key components in determining a sound's direction. Thus, even though you cannot hear the fact that it has turned into a sawtooth wave, you can at least potentially hear that the peak is at the wrong point in time, and you can almost certainly hear that the amplitude is reduced inconsistently from wave to wave.
This paper is also wrong in its claim that 20 kHz is "generous". It isn't. I've done listening tests and have successfully heard high-pitched whines up to... it was either 22 or 23 kHz (which was where I stopped trying, not where I stopped being able to hear), and I'm not even all that young. Admittedly, this is at relatively high amplitude, but the notion that most people can't hear 20 kHz is just plain wrong, and if you start out with that fundamentally wrong premise, you pretty much have to question all the other assumptions, too.
They also make the fundamentally incorrect claim that everything below the nyquist limit is sampled perfectly. This is also provably and trivially false. The Nyquist theorem says no such thing. It merely says that signals above that limit will result in "folding", causing aliased frequencies below the limit, which means that any frequency below the Nyquist limit can be captured without aliasing. However, music is not a single frequency in isolation; it is a bunch of frequencies interacting in complex ways. The Nyquist theorem says nothing about the phase of a signal near the Nyquist limit being consistent relative to other signals at lower frequencies, and in fact, it is not. Nor does the Nyquist theorem state that the frequency will be captured in a way that maintains consistent amplitude as you approach the limit; indeed, it isn't.
Read the Wikipedia article about the Kell factor in display technology, and you'll understand why this is a problem. Notice that with display technology, there is no anti-aliasing filtering involved (because the signal is a known signal that is entirely below the Nyquist limit), so this roughly maps onto what would happen if you could magically create a perfect anti-aliasing filter on the input side. You don't become nearly artifact-free until the frequency you are sampling is about 2/3rds of the Nyquist limit. This is an indisputable fact.
Admittedly, these artifacts are less objectionable in audio because of the anti-aliasing filtering that occurs (both on input and output), but no filter can magically "fix" that inconsistent amplitude. It represents actual information loss—the signal is equally likely to be a constant 15 kHz tone with constant amplitude as it is to be a signal that varies on either side of 15 kHz with a variable amplitude—and once that precise phase and amplitude information is lost, it is impossible to definitively reconstruct it.
In other words, this article is just plain wrong, almost top to bottom.
Besides, the real question is not whether 44.1 kHz is "good enough". It provably isn't, if you care about faithful reproduction over the entire human hearing range. The question is whether the information in the top octave of human hearing is in any way useful or important, to which the answer is "probably not". That's not the same thing as saying that 44.1 kHz or even 48 kHz sampling rate faithfully reproduces the entire range of human hearing, though, but rather it is merely saying that most people don't care about its deficiencies. A 48 kHz sampling rate is "close enough" up to about 16 kHz, which is a broad enough frequency range to be "good enough" for all practical purposes.
Check out my sci-fi/humor trilogy at PatriotsBooks.
Truthiness... Like needing a gold plated HDMI cable because it is better than one without. People want to believe it, it sounds good, but it's absolutely false. Anyone that understands that HDMI is digital, it transmits 1's and 0's, will know that it isn’t dependent on the conductiveness or noncorrosive of a material. RF on the other hand flows over the surface of the metal without penetrating deep into the cable. Hence, you want a metal that is non-corrosive as well as good conductor. If you used something other than gold for RF you can run into a problem of the RF signal reflecting from impurities in the connector and having part of that signal travel back towards the cable. This in turn will cause interference in the wave and cause problems. This has absolutely nothing to do with how HDMI transmits data, which is in the cable. But people are still thinking that to watch TV you must need gold plated connectors because gold plated connectors were used before... It has to be worth that extra $20 to $1000, right?! Lol! Truthiness....
Well, get yourself into a lab since you're evidently a freak of nature, the first human in history with ears that can detect such high frequencies.
Dogs hear up to 40kHz-50kHz.
Miraculous, unprecedented ear, you mean.
What was your source material? Encoding at a higher sampling rate is irrelevant if you're starting out with a signal that was already sampled at 44.1 kHz (e.g. a CD). The information loss occurs during the recording/encoding process, not during the playback process.
I have no problem whatsoever hearing the difference between tracks recorded at 44.1 kHz and 96 kHz in my home studio. The 96 kHz tracks preserve the upper harmonics better. The difference is particularly obvious with complex sound sources like crash cymbals. If you're doing the same tests and can't hear a difference, your signal chain is probably rolling off the top end. Either that or you don't record enough rock music. :-)
Check out my sci-fi/humor trilogy at PatriotsBooks.
...I can see at least one bogosity and a couple of omissions. The author claims that the "phase doesn't matter" with the Nyquist criterion, when it can easily be shown that, for instance, sampling a 20KHz sign wave at exactly at 40KHz can result in a zero signal if the input and the sampling are synchronized such that the sampling points all occur as the input waveform crosses zero. If they're slightly out of sync, something will get through but it'll be greatly attenuated. More importantly is the issue of "aliasing"--if there's any component to the input that's of a higher frequency than the sampler, the digital result will contain a "difference" component somewhere in the audible spectrum. For an idea of what this might sound like, listen to Don Ellis playing his trumpet through a ring modulator at the beginning of "Hey Jude" from the "Live At Fillmore" album. In practice, the sampling rate is placed somewhat higher than the maximum input frequency, to compensate for the analog input filter's cut-off being less than perfect. The 44.1 KHz rate for CD audio was the lowest rate at the time that allowed the recording industry to be able to claim "high-fidelity" i. e. reproduction of a 20-20KHz bandwidth. 48KHz is probably safer. Admittedly 192KHz is overkill, but perhaps not for mastering, assuming the amount of post-processing that's likely to happen between the original recording and the listener. Typical "webcasting" software, for example, contains multiple layers of digital filters, compression and whatnot, so it helps to start with something that's not already compromised.
I have not done double-blind tests, but I have recorded at 44.1 kHz and at 96 kHz, and the difference in the sound of individual tracks while tracking is quite audible. Thus, I'm inclined to believe that the failure to detect the difference had more to do with the original source material than with the limits of human hearing....
Because the article is paywalled, I'm curious what the original signal was, as that makes a big difference. Psychoacoustics teaches us that one sound can mask another. Thus, a recording of a symphony orchestra concert might be complex enough that your brain can't perceive the difference in high frequency content between 44.1 kHz and higher rates. A recording of a single solo instrument, by contrast, might result in an easily perceptible difference, depending on the instrument.
And the microphone choice makes a difference, too. The mass of the diaphragm (and anything that the diaphragm moves, in the case of a moving coil dynamic mic) makes a big difference in high frequency response. It could very well be the case that there was no difference in perception because the signal contained almost no high frequency content to begin with.
Without a very broad range of tests, all this test proves is that given that particular set of source material, nobody in their test group could tell the difference. This suggests that nobody can tell the difference for that particular set of source material. It does not answer the more general question of whether sound quality is reduced by sampling at 44.1 kHz instead of a higher rate.
Either way, I have personally tested my hearing and can hear beyond 22 kHz, which means that there are sounds that I can hear that are provably not reproducible at 44.1 kHz. Therefore, the claim that no one can hear the difference between 44.1 kHz and 96 kHz is preposterous on the surface, even if you had a theoretically perfect antialiasing filter and a theoretically perfect reconstruction filter.
Check out my sci-fi/humor trilogy at PatriotsBooks.
The sampling rate doesn't mean the signal you hear is "smoother." That claim is total garbage and shows immediately that you don't understand what analog-digital or digital-analog conversion is about.
A signal of finite duration can be expressed as the sum of sinusoids ("pure tones"). It takes only two pieces of information to reproduce a perfectly smooth sinusoid: its amplitude and its frequency. Sampling at discrete intervals gives us enough information to reproduce exactly all the sinusoids present in the signal up to half the sampling frequency. That's called the sampling theorem.
Furthermore, you quite assuredly do quite literally deceive yourself thinking you're hearing better sound from a 192 kHz file. This is no insult to you, nor am I saying you're being disingenuous with your claim; it's just that part of being human is that our cognitive biases are often stronger than our sensory perception. Do an ABX double-blind test between your 192kHz file and a version correctly downsampled to 44.1kHz and there's no way you'll tell the difference. Your ears are physically incapable of hearing any frequency anywhere close to the missing frequencies.
Please read the linked article, as Monty does a great job of explaining all of this and more.
Over the years, I've done listening tests at 96 and 192kHz in a few studio control rooms with various convertors (apogee, lavry, lynx) and various monitors (ADAMs, ATC, PMC, Quested). Nobody can reliably identify high sample rate recordings in double blind tests.
Dan Lavry debunked this 192kHz bullshit years ago, I'd suggest you go and read his sampling theory paper.
ok
My biggest problem with this is the fact that if i want to get a digital file from the Itunes store, I'm not going to get 16/44, I'm getting a lossy format, and If I rip a CD, I'll rip to mp3 at 320kbps. why? because I have the CD, I can put it in anytime I want, but my Iphone doesn't do FLAC
So anybody telling you that you don't need 24/96 or higher because we already have 16/44 is clearly in the way of people getting even that. I bet if audio DVD was ever the standard, we wouldn't get more music per disc, but higher quality music, and along with it higher quality equipment needed to play that music. I wouldn't be surprised if in this ideal scenario, 16/44 would be considered "good enough" for people.
So I'm all for progress, I'm sure today's music wont benefit from higher bit/sampling rates, but you never know what people in the future can do with it. there's been many other times when people have stood in the way of progress and declared the status quo "good enough"
They did this for a while. Maybe still? About 10-12 years ago. I forget what the market-speak trade name for it was. But they sampled at 1 Ghz. The trade magazines were divided in their opinions, and it must have died a fine death in the marketplace since no one here has referenced it This was definitely a recording format. .
I think we have to differentiate between mobile and home/studio listening. Considering the average playback hardware, for listening, 16/44 is fine for the 99.999 percent of listeners. For mobile listening 192kbps MP3 will exceed the needs of most. I prefer 320 kbps because it makes percussion sound better.
Most people listen in a car, bus, at work, on the job, etc. Low noise floor and dynamic range are moot. the reproduction amplifiers in cheap phone/pods aren't up to the task anyway, much less the average headphone/earbud.
I hesitate to use the term "audiophile" because of its pejorative connotation, but for people with above average sensitivity in hearing and training in sound artifacts, I think high resolution files are a good thing. Not only for private listening, but for a possible future when we regain a public domain and the remixing/sampling world takes off again.
For an analogy, think of DVD compilations of old TV shows that were encoded from tapes of television broadcasts. They look...ok...but when they go back to the original masters and re-release them there is an appreciable difference. Strangely though, consumer television/video playback formats are increasing in resolution, while common audio formats have been regressing.
The author of the article completely misses the secondary benefits of 24/192 delivery: Such fidelity (if delivered uncompressed) allows the recording to be considered archival and can be used for audio research, or as a historical record of the production methods used at the time the music was mastered. The format is indistinguishable from 24/48 *for most people*, obviously, but that doesn't make it worthless. If I could own 24/192 downmixes from the studio masters, I would consider it a unique opportunity.
"If the overtones of a flute high C and a piccolo high C are both under 22Khz, then sampling at twice that will catch all the overtones, and replaying the sample at the same rate will perfectly reproduce them."
Only possibly if the flutist plays forever - if the flutist ever stops playing, then the signal can't be both time-limited and band-limited.
I have a small correction. dB is a ratio of power, so it should be 2*48 or 96 dB.
What, you've never heard of KelvinHertz?
How do you know how hot the color of the frequency is?
I see even classic Slashdot is now pretty much unusable on dial up anymore.
Actually later tests revealed that the 1.5 KHz tone that the expert heard was an artifact from his own tape player, and not the compression codec.
Y so srs?
"Another Brick In The Wall" needs kazoos. Lots of kazoos.
Yes, you can hear the difference in amplitude, but AFAIK that won't affect the music in a qualitative way.
And IIRC, human ears are not good at hearing phase differences unless the phase is changing. Again, you won't hear a qualitative difference between the fundamental of a square wave or sawtooth if you can't hear the harmonics.
"Time" is a great song but a horrible recording, in fact all of Dark Side is really noisy, rolled off, and has other technical faults. of course you can't hear a difference between 192/24 and 44/16 on that piece, the original on LP is equivalent to like 35/12.
what about it? wanna suck my big black cock?
You have a few inaccuracies in your post. No sampling rate will ever perfectly capture a square wave or sawtooth wave, unless you use the exact frequency of that wave (or multiples of that frequency), and happen to match the phase of the wave with the sampling point. So given your example of a 44k sample rate capturing a 15kHz square or sawtooth wave, you are correct that you can't reproduce the waveform from these samples. You can't perfectly reproduce those waveforms even if you used a 196kHz, or 196000kHz, sampling rate.
According to the Nyquist-Shannon sampling theorem, you can perfectly reproduce a sinusoidal waveform (phase and amplitude included), if the frequency components of that waveform are less than half the sampling rate. Therefore when we talk about sampling signals, we are always talking about sampling sinusoidal waveforms. Square or sawtooth waves cannot be sampled, because their sinusoidal frequency components extend to the infinite.
Hope that makes sense to you.
I can't comment too much on the sufficiency of 16-bit levels to a sample, but the article does say that the noise introduced at this level is below human hearing, so if correct, seems to me it'll do the job. That's 65536 different levels of amplitude. Should be enough to capture the quietest oboe and loudest trumpet, at the same time. If pop music recording studios are compressing the dynamics to the upper range of the bit level, that doesn't stop a classical recording studio from using the whole 16-bit range.
You're welcome!
The students were in audio related fields. They were included because otherwise the data would be age biased. Younger ears can hear higher frequencies. I had no desire to obscure my point by explaining the finer design points of the study to those who couldn't figure it out for themselves. If you want to call that "lying," so be it.
If you take an extremely quiet passage on a properly mastered and dithered CD and amplify it to levels where the quantization noise is audible, a 0db passage with the same gain will either destroy your test equipment or your ears, whichever comes first. This is not a matter of opinion, it is a matter of physics.
My usual AC response issues aside, you don't understand audio signal interference, do you? You're better off using an external USB-attached quarter inch audio box than using *any* internal sound card due to the high levels of EM interference in the PC.
- Michael T. Babcock (Yes, I blog)
I paid $0.75/m for my cables at Home Depot. Read my post again AC.
- Michael T. Babcock (Yes, I blog)
Indeed, when you start discussing actual psychoacoustics research with either group they get all upset it seems. Arguing against high fidelity audio for a niche market seems utterly stupid, and arguing against testing is stupid too. Of course, neither group admits that we know very little about the brain's processing of input data.
- Michael T. Babcock (Yes, I blog)
I have a pair of Beyer Dynamic DT770 studio headphones that I use with my Yamaha receivers. They also work very well attached to my PSP when gaming at night due to their screw-in 1/8" to 1/4" adapter.
I stood in a local music shop for about half an hour listening to music on each set of headphones there until I found a set that sounded both incredible and that I could afford, and this was them. YMMV.
- Michael T. Babcock (Yes, I blog)
Its misrepresentation plain and simple, any reporter knows this when covering a story about something with multiple angles, and we geeks should at least try to be unbiased in our representations of facts if we're going to be taken at face value.
When you leave out details that matter and pretend the article claims something it does not, you're just throwing away any credibility you had.
- Michael T. Babcock (Yes, I blog)
Sorry for the double-reply, but since when are "studying audio" and "having excellent hearing" not orthogonal?
- Michael T. Babcock (Yes, I blog)
DAT recorders... a staple of film sound acquisition until just recently when flashcards finally replaced them. 48k.
all digital video formats that have ever existed have had 48k audio (except for some misguided prosumer "long play" modes that were 32k).
35mm soundtracks were dependent on multiplexes accepting enough downtime to install the new heads... but they actually used a type of QR code for dolby since it's inception (between the perf holes in the left side of the print, you'll see dense clouds of points with a tiny dolby logo in the middle).
Laserdiscs have carried ac-3 audio at... 48k
48k was ubiquitous from the moment there was media to record to... 44.1k was a compromise because the media didn't exist yet.
what i mean by lagging behind is that given the fall of CDs, there's no reason for the music industry to not standardize on something that actually requires cheaper electronics for a better sound (i'm talking about the brickwall filter). it's also a neat multiple of 96k, 192k, etc, meaning that only trivial adjustments are needed rather than complex filters to convert between them... again at higher quality (as in bad filters will affect the audible passband, not just the ultrasonic fairyland well above it).
while the music industry is catching up, they really need to consider including loudness as a requirement for mastering (where the volume slamming happens is actually premastering, though it's referred as mastering, to the chagrin of the people that run the replication plants).
Well, get yourself into a lab since you're evidently a freak of nature, the first human in history with ears that can detect such high frequencies.
Dogs hear up to 40kHz-50kHz.
Miraculous, unprecedented ear, you mean.
I may be a freak of nature but I don't think I'm so far outside of the range of human hearing. I think there is a bellcurve and most people cannot hear beyond 44kHz but some people can hear beyond this. I can hear the difference clearly. You give me two songs one in 44kHz and one in 88kHz and I can hear a difference for certain provided that the original master was in 88kHz or higher and not at 44 or some trickery like that.
How I can hear the difference I'm not entirely sure. I know other people hear a difference as well but for the most part the difference between 48 and 88 isn't as big as the difference between 44 and 48. Also 88kHz hurts my ears if I listen to it for too long while 44kHz does not. 48 seems just right.
Ah - I hadn't realised he was the drummer for The Police. Turns out I have "The singles collection" on CD, so I've used track 1 - roxanne, for my test.
I used foobar2k's ABX component on my standard setup, and I used lame V3.98 for mp3 conversions. I concentrated hard on the hi-hats in particular.
I failed the ABX test against lame at 192Kbs constant and q4 VBR (average 137kbps)
I had a little success at q6 VBR (average 112kbps), but not conclusive success.
I had no trouble at all at q7 VBR, but then lame resamples at 32kHz for that, and it was very noticeable.
So, maybe I'm just deaf, or my equipment sucks. But for my purposes, q4 VBR is definitely sufficient for playback, and frankly q6 VBR is good enough for me, which is why I use it on my portable player to fit more music on. Although I do keep all my CDs ripped in lossless archives anyway.
Get free bitcoins: http://freebitco.in
Hence, you want a metal that is non-corrosive
That is why your cheap "digital" USB cables have gold plated connectors.
Great spirits have always encountered violent opposition from mediocre minds -- Albert Einstein
There's no reason to be buying this format vs "archive quality"
I'm totally down with your sentiment but Google want to compress headers for a good reason.
Great spirits have always encountered violent opposition from mediocre minds -- Albert Einstein
When two or more instruments play a loud chord, the interference of the inaudible overtones from each instrument produce a distinct "ring" of audible difference tones, audible only at live gigs and on well reproduced SACD recordings. I've seldom heard the same effect to the same degree from a CD. Don't be fooled, this is a real and reliable enough effect for us classical musicians to use it to tune chords. This "ring" should be reproducible in 24/192 when these HF overtones in the stereo or surround channels interfere, which a CD cannot reproduce since there's nothing > 20kHz.
Granted, as mentioned in TA, the amp and speakers need to not be so rubbish as to introduce distortion > 20kHz.
Whilst I can tell in a blind test between the CD and SACD mode of the same disc of a recent BIS recording of Carmina Burana, it's only during certain passages of music where I am listening out for the difference tone "ring". Most of the rest of the time, I can't tell, and 16/44 CDs sound great. I don't think the fact that I am a classically trained musician matters.
That said, I think it's important NOT to be under the illusion that, just because you can't hear anything over 20 kHz (actually, ~16 kHz for most people), that there are no audible consequences when there is more than one channel.
In fact, given that well mastered vinyl played on good cartridges can reproduce fequencies to 60 kHz and beyond, this live "ring" may help explain why some folks still prefer vinyl recordings of classical music to the CD.
44kHz is about twice as high as people with good hearing can hear, and 88kHz is way beyond the range of human hearing.
Are you talking about sample rates, not audio frequencies? The sample rate needs to be double the highest audio frequency in the signal.
If you say you can hear a 44kHz audio frequency, that would obviously be bullshit.
Well, since there are loads of ABX tests available which show people do not hear a difference you can conclude that whatever this research tested for is not present in the music tested with in the ABX tests.
So whatever they found is not related to music and should be considered off topic.
You can't perfectly sample any waveform at any frequency, but the more samples per crest, the more accurately the waveform will be reproduced. At CD sampling rates you can indeed reproduce a 300 Hz waveform of any shape very accurately; there are 146 samples in its crest. That's plenty to accurately describe a sawtooth or square wave with subaudible aliasing. Not so at 15kHz with only three samples.
According to the Nyquist-Shannon sampling theorem, you can perfectly reproduce a sinusoidal waveform (phase and amplitude included), if the frequency components of that waveform are less than half the sampling rate.
Remove the word "perfectly" and that is accurate.
If pop music recording studios are compressing the dynamics to the upper range of the bit level, that doesn't stop a classical recording studio from using the whole 16-bit range.
I didn't say that was the case. I said with pop music it doesn't matter since dynamics don't seem to matter any more in pop music. But if your cannon in the 1812 Overture are at the highest level and the soft flute is 1/100th of that, your flute only has a range of 0 to 500. That's only a few bits.
Free Martian Whores!
Yes, diameter is king.
"Blind tests show that we perceive ultrasound: http://jn.physiology.org/content/83/6/3548.full [physiology.org] "
Since the body of ABX tests that shows people do not hear any difference between present and filtered ultrasound in music is much much larger that the body of theses guys we can safely assume that ultrasound frequencies, albeit maybe perceptable, have no significance whatsoever on listening to music.
"As a personal example, I had a friend named Xu. He kept complaining that I mispronounced his name."
I, for one, am able to perceive such small intonation differences in foreign languages due to a bug in my brain.
All this is besides the auditory system and is much more related to understanding the intent of the sound.
In your example, you are simply not looking for the right king of difference.
If you were to take a recording of your pronounciation and compare that to a recording of your friends voice you would find numerous differences.
There is a difference of pitch, there is a difference in how the sound resonates in your mouths etc,etc,etc.
Now there is a slight difference somewhere and your friend puts some significance to that difference.
You hear the same difference, you even perceive it but you do not understand that there is some significance to it.
So this is completely about putting significance in sounds and is not about perceiving sound as such.
It is about understanding speech.
Quoting from TFA:
In our hypothetical Wide Spectrum Video craze, consider a fervent group of Spectrophiles who believe these limits aren't generous enough. They propose that video represent not only the visible spectrum, but also infrared and ultraviolet. Continuing the comparison, there's an even more hardcore [and proud of it!] faction that insists this expanded range is yet insufficient, and that video feels so much more natural when it also includes microwaves and some of the X-ray spectrum. To a Golden Eye, they insist, the difference is night and day!
Of course this is ludicrous.
No one can see X-rays (or infrared, or ultraviolet, or microwaves). It doesn't matter how much a person believes he can. Retinas simply don't have the sensory hardware.
I beg to differ.
annoying as fuck
You're doing it wrong.
Different people have different cognitive abilities - this extends to our senses. The average person lacks perfect pitch, cannot tell the difference between SD and HD unless they're side by side, thinks their 128kbps MP3s sound alright, doesn't notice 60Hz jitter on their LCD, and so on.
It's the people on the fringes with superior senses who notice this stuff. But for the rest, this is all outside of their senses, so they're going to rubbish the quality paranoias of so-called audiophiles and videophiles.
A long time ago, in a galaxy far far away, CRT monitors running at a refresh rate of less than 75Hz used to bother the hell out of me while none of my coworkers seemed to mind. Due to that, I was the 1st person in the office to get a fairly decent (in comparison) 15" monitor while everybody else, management included, used cheap 14" ones.
I never considered my vision to be "superior" in any sense, it might be just that my brain does not do visual interpolation very well.
Rereading it and my response - I did goof on the MP3 bit rates (too late I guess) Thanks for catching that. Besides making statements about 192kHz / 24 bit vs 44.1 kHz/ 16 bit sampling rates / depth, I also made a statement about comparisons to MP3s and slipped on kbps. (damn the lack of an edit feature, even though it wouldn't have helped in this case.)
The cesspool just got a check and balance.
I have heard the rattle of a live sax. I have heard a delicate triangle ringing out over a live orchestra. I have heard live trumpet. I've spent quite a bit of time training my ears to hear those sounds.
I've seen things you people wouldn't believe. Attack ships on fire off the shoulder of Orion. I watched c-beams glitter in the dark near the Tanhauser Gate. All those moments will be lost in time, like tears in rain.
As opposed to a lie that sounds like a lie? :/
"I want more resolution... fucker."
I sell short stories in audio form (mp3) and I never sample over a configuration of a mono channel at 96 kbps with a sample rate of 44100 Hz. It works superb for me. I can't afford to waste web hosting space with 128 kbps.
For example, this one: http://sathyaish.net/stories/thelastleaf.aspx is at that configuration. It sounds great!
Also, all of these are mostly at 96 kbps with some exceptions at 128 kbps. http://sathyaish.net/voice
I don't see the point for general distribution. However, just because a human cannot hear the differences in an audio sample like this doesn't mean its not useful. If you process audio a lot, having more headroom results in fewer errors and side effects.
The same is true for my photographs. I record at far higher resolutions and colors than I really need, because I lose less when manipulating the images. Its not my final output that needs the headroom, its the steps before.
I expect that probably far fewer people manipulate audio, and that's the more accurate reason why a format like this has little value in the distribution case. There is a sweet spot somewhere that allows some fiddling without artifacts without also being too large to be generally useful, or so it seems to me.
I don't buy music online because the quality is so bad. If you know what it sounds like on CD and then you listen to in courtesy of an iTunes download, whole parts of the range and timber are just awol and it's all you can think about.
noise canceling headphones are a horror if what you're interested in is getting as close as you can to "being there". Ditto stuff like DOLBY. All those switches stay in the OFF position.
I have to RTFA, but in general i will testify that there are a lot of us out here who love CDs because of extremely high fidelity of the music and loathe iTunes and Amazon downloads because it sounds like shit. We have money to spend, but we're not spending it on that.
So if someone is trying to rectify this and sell to this part of the missing consumer base , all I can say is "of course I want all my music to be in lossless digital format, stored on some device that fits in my pocket and with my entire collection readily available to me at any time.
Why?
Is this going to happen any time soon? "
The point of this post is that distributing audio at 192/24 is pointless. This is correct. There are lots of reasons to record audio at high bit rates/depths. Among these is the fact the higher harmonics greatly affect processing and is big part f the reason many engineers will still run their audio through analog equipment for "that sound". In post production for tv and film high sample rates allow for greater flexibility in time stretching and pitch shifting. Also for archival purposes it is always better to have more and be able to give out only what is necessary. There are various tests with trained listeners being able to discern between high sample rate (96,192) and regular(44,48) sample rates. This was in a controlled in environment on proper equipment able to reproduce the high frequencies up to to 100k. As per the nyquist theorem digital audio limits the highest frequency to half of the sampling rate, thus the highest reproducible frequency with 192khz audio is a 96khz tone. This is nearly 4 times the highest frequency we can hear but the harmonics ainteract with the frequencies we can hear.
OK I see what he's saying and he's right. This is similar to the claims made for monstrously thick cables years ago. The claim then was the impedance of VERY thick cable , the resistance, was lower so more hi fi sound made it to the speaker.
The only problem was- no one could identify the difference in double blind studies. So much for that you might think but never think reality will interfere with marketing, and these fat cable companies are all doing OK even today.
Still doesn't make the schlock that Amazon and iTunes give you any better.
For anyone who isn't aware, the Hi Fi world is chock full of people who are basically insane and who not only will, but LOVE paying astronomical sums for any technology that promises higher hi fi. Thus the $150,000 home speakers and the $2500 cables and the ads that claim that their master craftsmen know *just exactly* how many times to wind some solid gold wire around some speaker part in order to get the highest high fidelity.
A similar situation exists with wines. Astronomical prices for nothing but the marketing around a bottle.
What can anyone say? Stupid people's money eventually drains away and the money of vain glorious stupid snobs gets hoovered out with an elephants trunk.
One problem I've come across as a mastering engineer is that a lot of the tracks I get aren't written with any dynamic range to start out with. Compressing it to brick-clip status or leaving it uncompressed and dynamic doesn't change the fact that whoever arranged the song turned everything up to "play all parts loud at the same time for the whole song." And while that's fine for like, a 2:30 punk song, for an 8-minute track? Buh.
----
"I used to listen to Null Device before they sold out."
You'll be in for a bigger shock when your cat chews through your $120 headphone cables.
Trust me.
----
"I used to listen to Null Device before they sold out."
This is why I don't have cats...
Who said humans were the only listeners? There's dolphins and dogs .. and of course, digital editing requires gross oversampling for frequency shifting, shortening or any other resampling technique. The fact is the sample rate is probably too low for that.
The assumption of the article is rubbish. It assumes the only processes involved are converting digital data to analogue and then human listening of the analog.
I heartily agree and have done the same. It was a revelation when I first got my Carver power amp and discovered with its power meters that most of the time it was pulling less than one watt per channel. Of course, I had reasonably sensitive speakers too, but nothing extravagant.
Read the Nyquist-Shannon sampling theorem. It's perfect reproduction. Mind you, there are caveats. Sample length has to be infinite. So, it's not practical, but what I said was actually perfectly accurate.
In any case, what I was trying to get at, is that sampling a sinusoidal waveform, even a 15kHz wave at a 44kHz sampling rate, reproduction is going to be very accurate. The mathematics show it. Certainly orders of magnitude more accurate than capturing a square or sawtooth.
If a soft flute has a range up to 500, that's still quite an accurate capture of amplitude. It's greater than 10, and even 11, so Nigel would approve. Besides, the practicality of listening to a cannon and soft flute, at their natural volume level, in the same piece, is rather bewildering.
But but stegasaurous err... stenography depends on it.
No one can hear 24 bit audio so the lower bits can be
written almost in the clear to pack a message down the
road.
The low bits are also magical and can be used for keys and other cryptography
values hither and thither....
He's talking about mp3 compression rates. So he's totally wrong, but not for the reason he thinks.
me too. spent a lot of cash on midrange system. I am quite deaf in one ear and other one isnt perfect, _but_ I can still tell a difference in quality when playing CD's on my fantastic Pioneer DV-656 player DVD/sacd/dvd-audio that is 12 yrs old compared to the crappy (music wise) Sony BDP bluray player I bought 6 months ago.
Pioneer must have much better dac, and I use analogue cables to the receiver whereas Sony player is using hdmi cable.
But honestly, biggest problem i have is room acoustics. Got all this great gear, and it's probably reaching 20% of it's potential in my living room.
"Everyone knows that vi vi vi is the number of the beast" -- Richard Stallman
Mind you, there are caveats. Sample length has to be infinite.
Exactly! Note that the closer you get to "infinite" the closer you get to "perfect". The higher the sampling rate, the closer to infinite and the closer to perfect.
In any case, what I was trying to get at, is that sampling a sinusoidal waveform, even a 15kHz wave at a 44kHz sampling rate, reproduction is going to be very accurate. The mathematics show it. Certainly orders of magnitude more accurate than capturing a square or sawtooth.
Yet there are more than sine waves in sound. A rock guitar fuzzbox changes the guitar's sine wave to a square or sawtooth (most fuzzboxes and wah wah pedals have a switch to select between square and sawtooth). With three samples (do the math!) it is impossible to discern those three entirely different waveforms. They will be distorted into a sine wave.
Have you ever studied sound with an oscilloscope? One of my undergrad physics classes was about this very thing, although it was in the late '70s and there were no digital samples back then.
Have you seen rock or blues bands with the guitar feeding into a small tube amp, with a mic in fron of it feeding a transistor amp? That's because if you overdrive a transistor amp to clipping levels, you get a perfectly square square wave, but with a tube amp the wave's corners are rounded (as seen in an oscilloscope) at clipping levels.
It is mathematically impossible to do that with three samples.
Besides, the practicality of listening to a cannon and soft flute, at their natural volume level, in the same piece, is rather bewildering.
Those pieces are ancient, and are usually performed outdoors. You would have to have an incredibly good setup to get anywhere close to accuracy with those pieces.
Free Martian Whores!
I beg to differ in this regard. "A Fourier analysis of a sine wave is the sine wave itself. A Fourier analysis of a square wave or saw-tooth wave shows harmonics and subharmonics." We can hear those subharmonics.
So, then the next thing to look at are the Fletcher - Munson curves. These curves are averages over a population, without stipulating, by frequency the standard deviation. While I can hear to 15780hz, my wife can hear to 16,200hz, and my father, to 12,500hz. What I have as a threshold, such as hearing the ticking of my watch, my father is unable to hear his watch, even when pressed against his ear.
FM curves should be broken down by age groups, with groups being 1 year apart for people over age 60.
No, I believe that anything over 60hz sampling is a waste of bandwidth. My high quality earphone diaphrams are resonant at 20 cycles, so in theory they should vibrate at 20khz, and they do, but with tremendous mechanical loss. And my ears as well have tremendous loss above 15khz. If I need to hear above 15khz, I should have electrical connections directly to nerve endings in my body, with the belief that nerves transmit messages at the speed of light.
Leslie Satenstein Montreal Quebec Canada
Sample _length_, not sample rate. The longer you sample a sine wave, the closer to perfect you will be able to reproduce it, provided the sample rate is over twice as high as the frequency.
While I'm no expert on audio, only having studied undergrad signal theory for an electrical engineering degree, seems like the question here is: can the human ear discern between hearing a square or sawtooth wave, compared to hearing their sinusoidal waveforms bandwidth limited to the audible frequency range. If the answer is no, then distorting a square or sawtooth wave into sinusoidal components is not a problem. Hence there is no need to perfectly reproduce a square or sawtooth wave, because our ears would not be able to tell the difference.
Sample _length_, not sample rate.
Exactly what do you mean by "sample length?" If by it you mean that there are three samples in a 15kHz tone and hundreds in a 300Hz tone, then that is accurate. Your 300 Hz tone wil be more accurate than the sample of a 15kHz tone. But its is because of the number of samples collected per wavecrest.
Nyquist can be overly simplified to say that you need more than two samples to reproduce a wave.
can the human ear discern between hearing a square or sawtooth wave, compared to hearing their sinusoidal waveforms bandwidth limited to the audible frequency range.
That is exectly the right question, and the answer is a clear "yes". If you can hear a tone you can discern different wave shapes for that tone. It's the main reason people say that LPs sound "warmer" than CDs; it has to do with CD's aliasing distortion, which analog recordings don't have (even though there are other forms of distortion).
Raise the sample rate where there are enough samples to accurately render a 20kHz waveform of any shape and your digital sample will sound "warmer" than the LP while lacking the LP's inherent noise problems.
Nyquist doesn't apply to analog recordings because there are no samples per se, it is continuous. LPs had a fantastic frequency range. The way quadraphonic LPs worked was the rear channels were modulated with a 40kHz tone and added to the front channels, then subtracted on playback by phasing. That 40kHz tone that held the rear channels is twice as high as the best human ear can discern.
Free Martian Whores!
"The noise of the CD-quality loop was audible only at very elevated levels."
So how do you get "They found no perceptible degradation caused by a 16-bit/44.1kHz A/D/A." ?
Remember that many double blind tests have errors in the setup that can remove the possibility of a positive result. For instance, if the switching system degrades the signal significantly, the possible further degradation of the ADA can be masked.
Having said that, I have done my own tests with SACDs.
When I compare the sound of the red book (CD) layer to the SACD layer, I rarely hear a difference. But when I do, It may be that the red book layer is not really a direct down sample of the DSD encoding, so inconclusive, but it showed to me that red book is better than I thought.
Sample _length_, not sample rate.
Exactly what do you mean by "sample length?" If by it you mean that there are three samples in a 15kHz tone and hundreds in a 300Hz tone, then that is accurate. Your 300 Hz tone wil be more accurate than the sample of a 15kHz tone. But its is because of the number of samples collected per wavecrest.
No, I think he's talking about the window of time over which the system is evaluated. IIRC, there's some stuff in Shannon-Nyquist about theoretically perfect reconstruction requiring an infinite time window (regardless of the number of samples taken per cycle).
That is exectly the right question, and the answer is a clear "yes". If you can hear a tone you can discern different wave shapes for that tone. It's the main reason people say that LPs sound "warmer" than CDs; it has to do with CD's aliasing distortion, which analog recordings don't have (even though there are other forms of distortion).
The reason (some) people like the vinyl sound is that it actually distorts the signal much, much more than a CD does, in a way which some find pleasant (it's responsible for the "warmth" you describe). The reason for this is easy to understand, if you spend some time investigating how LP recording actually works. Look up RIAA equalization sometime. The TLDR version is that just to play back a LP without having it sound like ass, the playback system has to implement circuitry which un-does some signal mangling done during LP mastering, and the un-mangle process is never perfect (and can't recover everything which was lost in mastering anyways).
Also, from your comments about CD distortion, you seem to actually believe in the "stairstep" mental model of sampling systems. They don't work like that. The stairsteps, or quantization noise, do not actually appear in the final output. Proper sampling and playback requires two key "brick wall" low-pass filters in the analog domain. One goes before the ADC, the other after the DAC. The purpose of the first filter is to remove all analog signal components which could cause "aliasing", i.e. those above 1/2 the sampling frequency (AKA the Nyquist frequency). The second filter is called a "reconstruction" filter, because it literally reconstructs the original bandlimited analog waveform from the stairstepped waveform. Basically, all the quantization noise is composed of frequencies greater than 1/2 the sampling frequency, so if you once again filter out everything above the Nyquist frequency, you're left with the original analog signal.
The big deal about Shannon-Nyquist is that they Did The Math, and proved (beyond a shadow of a doubt) that a sampling system consisting of an input filter, a sampler, a 'desampler', and an output filter really does more or less perfectly reconstruct the original analog waveform, minus all frequency components above the Nyquist frequency. And yes, this really does mean you can perfectly reconstruct a sine wave with only slightly more than 2 samples per cycle, no matter how impossible that might seem by intuition.
Raise the sample rate where there are enough samples to accurately render a 20kHz waveform of any shape and your digital sample will sound "warmer" than the LP while lacking the LP's inherent noise problems.
Nope.
This "20 KHz waveform of any shape" stuff (especially with respect to triangle waves) is a classic way that people fool themselves about how this works. Here's a neat web page showing what's actually going on with a triangle wave:
http://www.bsharp.org/physics/guitar
Basically, in signal theory, all waves of any shape are composed by summing sinusoids. A 20 KHz triangle wave is actually a 20 KHz sinusoid fundamental plus an infinite series of lower-magnitude sinusoidal harmonics at multiples of 20K.
The thing is, your ear is basically an array of small amplitude sensors, each of which has a narrow bandpass filter so it only responds to
"The fact that 192kbps is the range where all but a *very* few people stop being able to distinguish the MP3 from the CD"
I can hear the different for any bitrate of MP3 for jazz and percussion. Once OGG hits about 192Kbit, I can't notice the difference from the wav.
High intensity sounds like percussion actually hurt my ears if at low bitrates/sampling. MP3 compression murders percussion, so it almost always hurts my ears. I literally feel pressure against my ears like an ear infection. Even once the sound is stopped, I typically still have ringing and a general headache that may last for a 30-60min. So, I can literally feel the difference between high and low quality encoding, immediately... for higher frequencies anyway. Most pop music isn't an issue, but get some classical or jazz, my head wants to explode.
I do tend to hear when someone turns on a CRT. I'm so glad those have been mostly phased out. One time at a bar, there were a few TVs above the booth my friends were at. I asked them if they could hear the really loud sound those TVs were putting off, no one claimed they could. By the time I left, my ears were ringing and were quite painful. I had a hard time hearing my friends over the high pitch squeal those TVs were making. I distinctly remember feeling like I had swimmer's ear after that experience.
You're wasting your time. He's been told over and over again, even studied the subject, and still comes away clueless. He doesn't get it, and he never will.
mcgrew thinks "waveforms" are magical strings of sound that sail through the air clinging together in a "square shape" or "sawtooth shape."
He can't grasp that it's possible to not hear part of a square wave because the harmonics are too high for the eardrum or hair cells to respond to - or comprehend that the harmonics above 20kHz have almost no energy (0.02% for a high piano note) and won't budge your eardrum anyway.
Instead, he prefers to think that 20kHz roll-off filters "distort square waves into sine waves." If you could just keep that darn wave square, by God, you'd hear what a different quality it has from a 20kHz sine wave!
S/PDIF removes the internal DAC/ADC from the picture. You still need a DAC/ADC some where, but it's going to be on the other end of the fiber.
Want to know what's funny? The author of the article claims the maths are being misunderstood. LOL!!!!!!!!!!!!!!!!
Does the author understand the fact the Nyquist theorem only applies to continuous time signals, not discrete time signals? No.
Does the author undestand the difference between a continuous time signal and a discrete time signal?
Probably no.
Does the author understand the fact the continuous time signal in the theorem is represented as a Real Function (http://www.proofwiki.org/wiki/Definition:Real_Function)?
Probably no.
Does the author know what the Domain of a Real Function is if it isn't specified (this is explained by the article I linked above)?
Probably no.
Does the author have any idea what a Real Number (http://www.proofwiki.org/wiki/Definition:Real_Number) actually is?
I guess that's another no.
Does he know something about interpolation errors and quantization errors?
Does he know something about audiophile equipment?
Let me think very, very hard now...
No.
Wait a minute, please...
Still no.
about flac for years
So what do you call it when people refuse to believe the truth because it doesn't fit their perception of reality?
There are damn good reasons why studio work is done in 24/192, and while I agree that most playback devices cannot produce the uber-high frequencies, nor would we want them to, I think the arguments against distributing 24/192 are pretty weak.
For one, his argument about digital sampling is bullshit, and demonstrates a poor understanding of the Nyquist-Shannon theorem. In his idealized case of a pure sine wave, yes you only need to sample at the Nyquist frequency. Once you start mixing different sounds together, that all falls apart since the sound wave is no longer a stable shape but rather an additive-subtractive mess of several frequencies, which do reconstruct in such a predictable fashion. Heck, even a simple square wave at f/2 will result in audible distortion on the DAC side, because it simply cannot recreate the infinite "slope". It's not even a matter of hearing up to 22khz, I know I can't anymore, but the harmonics of a true square wave cover the entire range down to 0hz, and that's what you can actually hear. If that square wave gets tapered or rounded by inadequate sampling, you end up with a triangle or sine wave which sounds radically different.
Does the average ear need 24/192 to be satisfied ? No. Does it mean we should entirely stop distributing such content ? Fuck no. I have a pretty decent studio setup, cheap but decent, and good enough ears with the technical training to notice those distortion artifacts. Okay, I'm a freak of hearing with perfect pitch and damn near digital memory for audio - hell I can identify a few dozen vocal mics just by listening to a mixed and mastered CD. Those high resolution recordings are for ME! They provide me with some geeky audio entertainment, which makes it worth the extra download time and minor expense of 24/192 capable equipment. My wife, who is a trained opera singer, cannot hear those details; she doesn't listen in such analytical fashion. Hell she can't even tell if I subtly pitch or time-stretch a track for DJ mixing... For her uses, 44khz is more than enough. So what's so wrong in providing different files for different listeners, and why does this Monty guy think his opinion trumps anyone elses ?
-Billco, Fnarg.com
24-bit is a vast improvement over 16-bit, both technically, and audibly. Also 48k is a significant improvement over that, just ask any experienced audio engineer. Anything above 48k/24bit is a marginal improvement in most cases, if any improvement can be detected at all. But do not confuse the issue by giving the impression that the current audio fidelity is sufficient, it's pretty atrocious, and it is a problem, despite what you may think. Most notably lossy data compression is proven to cause ear fatigue, so the most important step is providing lossless audio, which is available in many cases. The next step is to improve fidelity over the archaic 44.1k/16bit standard, to at least 48k/24bit. If you doubt this, ask any professional that works with audio and prepare to be schooled.
More than he shows in that post, I would hope. Explains a lot, though...
Actually, you can't discern a wave shape at all. If you have a wave with the exact same harmonic ratios as a square wave, but the higher harmonics are out of phase, the wave will not look like a square wave, but it WILL sound like one.
It's only when the phases are changing that you can hear the difference - with a constant tone, a square wave is indistinguishable from an infinite number of waveshapes with the same harmonic content.
The hair cells in your ears respond to ranges of frequencies, and they ignore anything outside their range ("ignore" is probably a bad way of putting it - they are unresponsive).
Only if your definition of "perfectly good" is "so convoluted that nobody EVER uses it". ;)
Let's be honest here, verisimilitude exhibits a superlative and ostentatious preponderance of syllables.
Indubitably.
What about playing that music on a very large sound system at some sort of concert or outdoor festival by, I don't know, a DJ. Just because you won't need it, doesn't mean others won't.
Never say never. Ah!! I did it again!