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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.
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 find your well-reasoned and respectfully written response to be full of helpful counterpoints and useful references. I wish to subscribe to your newsletter.
>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.
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.
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.
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.
If you buy your music over the 'net, flac isn't an option, and CD stores are dying. One of the many reasons piracy is still so popular among audiophiles.
"People don't want to learn linux" hasn't been a valid excuse since '03.
"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/
Your cat is not "listening", it is simply tolerating that annoying racket that you call "music" in exchange for food, body heat, clean kitty litter, etc.
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".
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 ...
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.
For the rest of us on /. haven't we had all of our music in FLAC for a decade now? I don't even listen to music much and mine is.
My music is mostly stored in whatever the default is for YouTube videos that I've saved locally. I'm apparently even less of a music fan than you are.
Fun fact: I'm also an audio technician. Yes, I can hear the occasional damaged sound, but I'm not enough of an asshole to care.
You do not have a moral or legal right to do absolutely anything you want.
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.
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.
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.
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*
It's an option if you can manage to get gift cards to that Russian music store...
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 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...?
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.
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
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.
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)
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
True, some audio time stretching methods become algorithmically less complex with a high sample rate recording. But those can be produced from low sample rate recordings by interpolation.
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)
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.
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..."
And in that regard, there is a huge difference between 44.1kHz/48kHz/96kHz, but lesser of a difference between 96kHz and 192kHz.
Citation needed. Double blind if possible.
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.
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.
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.
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".
If you buy your music over the 'net, flac isn't an option
Guess you're not a Nine Inch Nails fan.
$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.
no it isn't. verisimilitude is, roughly, the quality of being believably realistic. truthiness is like "verisimilitudinous lying," i.e. the apparent realism is misleading, often toward the exact opposite of the truth.
"They were pure niggers." – Noam Chomsky
^ 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 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.
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
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."
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!
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!
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.
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.
The spectrum of a 22kHz sine wave is one peak at 22kHz.
The spectrum of a 22kHz square wave is peaks at 22kHz, 66kHz, 110kHz, 154kHz, 198kHz and so on.
So, if your sampling rate is 44.1kHz, you will only capture the 22kHz part and will get a sine wave.
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.
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.
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.
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.
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."
I agree, but it is not just the editing/mixing that benefits from full quality.
Doing the compression from 24/192 (or even my preferred 32/480) will be better than doing it from 16/48, even when compressed to the same bit level (though that difference will converge as you push the bit level down). The compression logic will have a cleaner source to work with if the high sample rate and resolution is handled properly. The end user will get slightly better audio in just the same space.
By all means do the studio mixing at as high a sample rate and resolution (uncompressed or non-lossy compressed) as you possibly can. Even video should be edited uncompressed or non-lossy compressed with the best video sourcing you can get, before you crunch it down to satellite, cable, and broadcast limitations.
now we need to go OSS in diesel cars
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.
And at that time, I don't want to have to redo all my music.
And at that time, I don't want to have to re-pirate all my music.
There, fixed it for ya.
now we need to go OSS in diesel cars
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.
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?
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
Check out Bandcamp. iTunes or Amazon aren't the do all and end all of online music stores.
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.
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
Human ability has limits, even taking standard deviations into account. Your chance of being able to hear this stuff is equal to your chance at being able to see microwaves.
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.
Then do some double blind tests that show that you can actually hear the difference.
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.
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!
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.
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.
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.
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!
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...
This article is bunk. I wasted 15 minutes reading through it.
So... you're arguing against maths. This will go well...
E.g. it can only support ONE sine wave at that point.
There is only ONE wave at a given frequency. Add together multiple waves at the same frequency and you get ONE wave eith a different phase.
SJW n. One who posts facts.
Then do some double blind tests that show that you can actually hear the difference.
the guy wants 480khz.
he'd show the difference with a oscilloscope. would be blatantly obvious there. couldn't hear it of course, but you could show a difference when pumping silence!
world was created 5 seconds before this post as it is.
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.
Wait a minute. You lead out by saying that the high resolution is a significant upgrade, but then say that increased resolution is pointless.
Which position are you taking?
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 ]
But looking it from a philosophical perspective, does master format even have to perfect? I believe that going over CD quality there would be diminishing returns of receiving more enjoyment (even if we are talking about a person who knows and cares about sound quality). At that point you could already improve the experience by having had a good night's sleep or high enough blood sugar. And of course there's the dynamic range compression thing which would be much more important to solve.
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.
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 ]
If you buy your music over the 'net, flac isn't an option
Guess you're not a Nine Inch Nails fan.
He said music. /me ducks
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 ]
Don't we already have the word "specious" for things that look true but are not? So might "intentionally specious" be a better definition?
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 ]
(+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.
BS. If the overtones of a flute high C and a piccolo high C are both under 22Khz, then sampling at twice that will catch all the overtones, and replaying the sample at the same rate will perfectly reproduce them.
And if the overtones are over 22Khz, but their lower-order harmonics aren't, the sampling will pick up the harmonics and reproduce them perfectly, even without the existence of the original overtone.
There is no subjectivity in that. An oscilliscope will show you that the overtones and/or their harmonics are all there.
The only step that decides whether or not the overtones have any influence is the quality of the low-pass filter. At 44Khz that can be a bit iffy, so using 48Khz to get a little more headroom is nice, but in practice you won't be able to hear a difference with anything above that.
"I know I will be modded down for this": where's the option '-1, Asking for it'?
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
What is an "end listener"? A person who doesn't like to sample, and never ever will in their life?
To me that's like not printing images with better than a certain resolution, because nobody will ever look at it with a magnifying glass. Heh! And if you slow something down or speed it up, the "hearable frequencies" end up being quite different. Just like with image data, sure, at a certain threshold it beomes cumbersome, but that's because space and processing power is limited -- not because you can ever have "too much resolution".
Sure, that's not the general use case, but still... this article strikes me as kinda shortsighted: Oh noes, the ultrasonics, which can be dealt with! Ignore the lost data, which cannot ever be put back once it's lost. It's not like data has any purpose other than to hear it as it is. And it's not like stuff will ever end up in the public domain, or as if sampling music is any fun...
Why does Photoshop have 16bit colour? [etc]
The parent already addressed and accepted use of higher sampling rates and resolutions for intermediate stages.
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.
What evidence do you have for this assertion?
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.
I'm no sure what you're trying to say here. If there are any "overtones" beyond the frequency range of human hearing, then humans aren't going to hear them. If this means that subtleties of the two instruments are indistinguishable at higher pitches, then this would be the case with someone listening to the original performance as well.
"Slashdot - News and Chat Sites Deviant". (Click "homepage" link above for details).
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'?
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.
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."
My music is mostly stored in whatever the default is for YouTube videos that I've saved locally. I'm apparently even less of a music fan than you are.
Fun fact: I'm also an audio technician. Yes, I can hear the occasional damaged sound, but I'm not enough of an asshole to care.
Call me an asshole then, because I care, and I'm not even an "audio technician". I did do a fair amount of audio engineering (studio and live) in a former "life", and I have always been fussy about the quality of recordings and their reproduction.
So off the top, let's agree that, regardless of digital distribution format, the production values of most recorded music these days is shit. (Note to all you "audio technicians" out there: compression has it's place, but trust me, it is not where you think it is.)
Now, if you don't value the difference between what comes down from YouTube and a well-crafted recording, digitized using equipment and techniques capable of faithfully transcribing all of the detail in the performance, and played back on equipment similarly capable, fine. That's a subjective judgement and it would be a fool's errand to suggest that you receive more or less enjoyment out of the pile of crap you've downloaded, just as it is foolish of you to suggest that anyone more discerning is an "asshole" for having different subjective values. The fact remains that there is a real and quantifiable difference between most common digital audio formats.
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
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
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.
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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.
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!
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.
Good example. Because sometimes people see those things. Human perception is psychological fundamentally. There is no reductionist science that will make is into something fully rational. Sorry, religious 'scientifical' types, uncertainty is the only certain in this universe.
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.
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?
The only step that decides whether or not the overtones have any influence is the quality of the low-pass filter.
And *here* is where the sample rate starts to make much more of a difference. The higher the sample rate, the better able you are to do filtering digitally (cheaply) instead of with a better (more expensive) analog filter circuit. You can't hear anything above 20 KHz or so, but having the extra data available sure makes it a lot more convenient for the DSP side of things.
Having said that, you can also take the existing low-rate signal, and repeat the samples as needed at a higher rate to get much of the same benefit.
Please stand clear of the doors, por favor mantenganse alejado de las puertas
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
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."
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.
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 ]
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.
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.
The "technician" term is also used in places where the "engineer" term legally requires having an engineering degree.
I prefer to use "technician" for myself because I work almost exclusively with live productions, mostly as a hobby, and on a system that's set up and ready to go. The engineering work is already done. I generally just sit there adjusting levels, and fiddling with EQ occasionally. Personally, I don't think I do enough mutilation to take the title "engineer."
You do not have a moral or legal right to do absolutely anything you want.
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.
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.
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.
"Sophistry" comes to mind.
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.
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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.
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.
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.
FYI, I don't think you know what you're talking about. "Better" is a very subjective term that is always relative to the situation at hand.
Please stand clear of the doors, por favor mantenganse alejado de las puertas
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.
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.
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"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.
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
Lety me add phase shift, frequency shift, stereo image blurring and oscillation, and beat interference to your list.
Michael J. Burns
Yes, but that's a case of dimishing returns. Monty does cover this in TFA, but I'll repeat it: if you're cutting off at 22Khz, then 44Khz sampling will give you little headroom for your lowpass filter, 48 will give you more, 96 is practically perfect, more is overkill.
Mart
"I know I will be modded down for this": where's the option '-1, Asking for it'?
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!
I can see beat interference from the sampling wave - whether point impulse or square - as, by itself, sufficient reason for wanting to double the sample rate.
Michael J. Burns
"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.
"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.
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.
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
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.
A good electret condenser microphone had an amazingly flat and phase-accurate response. Preamp equalization was the inverse of preemphasis and canceled in phase as well as frequency. Channel equalization could foul up phase response - but wasn't commonly used on sounds where this would be an issue.
As I pointed out, Time Windows use a hack to deal with room acoustics. The woofer is at the top of the tower, which is an accoustic transmission line driven by the backside wave (ala bass reflex). A pair of ports at the bottom are sized to correctly load it when letting the energy escape near the bottom, preventing the wave from bouncing back up and emerging through the driver. The transmission line is loaded with a fiberglass stuffing that attenuates higher frequencies in a way that models the diffraction of the driver's front wave around the tower on its way to bounce off the wall behind the speaker. When positioned according to the instructions, the unit is just far enough from the wall that the wave from the bottom ports cancels the wave that diffracts around the speaker and bounces from the wall. Thus the wall "disappears". You still have the acoustic effects of the side and back wall, furniture, and third bounce from those off the "disappeared" wall. But that actually helps, by making the rest of the room and its contents remain audible, while the speakers and the wall behind them become a window on the performance.
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!
Which is why you should buy your vinyl from Mobile Fidelity Sound or the like. Pressed into virgin vinyl (no ground up labels to make hisses and pops), their disk master cut using analog equipment at a reduced speed, no phase-trashing filtering, etc. You still need a pre-disk master good enough to take advangage of this or you still have issues. But many good masters do exist. And even when the original master isn't ideal the disk master cutting doesn't make it substantially worse, giving you something substantially better than an original commercial release.
Bantam Dominique roosters crow a four-note song. Once you've heard it as "Happy BIRTHday" you can't NOT hear it that way
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!