Why Distributing Music As 24-bit/192kHz Downloads Is Pointless 841
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.'"
Can we stop using the word "truthiness," please? (Score:5, Interesting)
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?
Re:Can we stop using the word "truthiness," please (Score:5, Funny)
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!
Re:"Truthiness" is a dumb word (Score:5, Interesting)
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.
Re:Can we stop using the word "truthiness," please (Score:5, Informative)
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.
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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.
No smooth (Score:5, Informative)
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
Re:No smooth (Score:5, Interesting)
Re:Why does Photoshop have 16bit colour? (Score:5, Informative)
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.
Re:No smooth (Score:4, Informative)
You may not be able to hear the higher frequencies, but when they're sampled with a too low sample rate, you'll be converting waveforms you can hear.
Nyquist assumes that the signal to be sampled does not contain any frequencies higher than half the sampling rate. Any that exist thus *are* expected to be filtered out beforehand, otherwise aliasing will occur.
Try it for yourself on paper.
The "samples" do *not* represent the final "reconstructed" wave (are you suggesting the same "join the dots reconstruction" misconception that most people have about Nyquist?). My understanding of Nyquist (probably incomplete and far from perfect, but still miles better than most people's fundamental misunderstanding) is that this sample output has to be filtered so that all the harmonics above half the sampling rate are removed. Since Nyquist only says you get perfect reconstruction for frequencies up to that limit, there's no contradiction there.
A "perfect" square wave (which can never actually be created in the real world) has harmonics of infinite frequency, and even a "real-world" as-near-square-as-makes-no-difference-wave will contain very high harmonics. If one was to do a Fourier transform on a square wave, filter out all the frequencies above the human range of hearing, then convert it back to the familiar (spatial domain) wave form, it wouldn't be square any more.
Therefore, you can't sample a square wave using standard techniques anyway.
Its also called a factoid (Score:5, Informative)
Many people think a "factoid" is a small fact. Actually a factoid is something that sounds true, but is actually false.
Re:Can we stop using the word "truthiness," please (Score:4, Informative)
There was already a perfectly good word [wiktionary.org] for that.
Re:Can we stop using the word "truthiness," please (Score:5, Funny)
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.
Re:Can we stop using the word "truthiness," please (Score:5, Funny)
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Re:Can we stop using the word "truthiness," please (Score:5, Funny)
You willfully leave out nerds, geeks, dorks, and spazzes? Obvious /. bias! ;)
Re:Can we stop using the word "truthiness," please (Score:5, Informative)
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.
yeah, just use monster cables. (Score:5, Funny)
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.
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Warm. Sparkling. Punchy. Silken. Pristine. Thumping. Brilliant. Dynamic. Crystalline.
And the best modifier of them all: Audiophile-quality.
How's that?
Pfft. (Score:5, Funny)
I have a PhD in Digital Music Conservation from the University of Florida. I have to stress that the phenomenon known as "digital dust" is the real problem regarding conservation of music, and any other type of digital file. Digital files are stored in digital filing cabinets called "directories" which are prone to "digital dust" - slight bit alterations that happen now or then. Now, admittedly, in its ideal, pristine condition, a piece of musical work encoded in FLAC format contains more information than the same piece encoded in MP3, however, as the FLAC file is bigger, it accumulates, in fact, MORE digital dust than the MP3 file. Now you might say that the density of dust is the same. That would be a naive view. Since MP3 files are smaller, they can be much more easily stacked together and held in "drawers" called archive files (Zip, Rar, Lha, etc.) ; in such a configuration, their surface-to-volume ratio is minimized. Thus, they accumulate LESS digital dust and thus decay at a much slower rate than FLACs. All this is well-known in academia, alas the ignorant hordes just think that because it's bigger, it must be better.
So over the past months there's been some discussion about the merits of lossy compression and the rotational velocidensity issue. I'm an audiophile myself and posses a vast collection of uncompressed audio files, but I do want to assure the casual low-bitrate users that their music library is quite safe.
Being an audio engineer for over 21 years, I'm going to let you in on a little secret. While rotational velocidensity is indeed responsible for some deterioration of an unanchored file, there's a simple way of preventing this. Better still, there have been some reported cases of damaged files repairing themselves, although marginally so (about 1.7 percent for the .ogg format).
The procedure is, although effective, rather unorthodox. Rotational velocidensity, as known only affects compressed files, i.e. files who's anchoring has been damaged during compression procedures. Simply mounting your hard disk upside down enables centripetal forces to cancel out the rotational ruptures in the disk. As I said, unorthodox, and mainstream manufactures will not approve as it hurts sales (less rotational velocidensity damage means a slighter chance of disk failure.)
I'd still go with uncompressed .wav myself, but there's nothing wrong with compressed formats like flac or mp3 when you treat your hardware right
--
BMO
Re:Pfft. (Score:4, Insightful)
Doood ... just, dood. You originally posted this, word for word, elsewhere (http://www.investorvillage.com/smbd.asp?mb=1911&mid=10609989&pt=msg). Either you are a bug-eyed alien, a prankster, or a combination of the two.
For those who aren't in on the secret, you can look up "rotational velocidensity" -- on the Urban Dictionary. It is the supposed loss of bits in a file over a time, which is absolutely ludicrous. Digital is digital. It's ones and zeroes. Files stored digitally don't degrade, unless you're talking about media degradation (ex., CDs and DVDs can possibly suffer from loss of data over time).
Dood also talks about files "repairing themselves," which is somewhere south of ridiculous.
But enough of this. I fell for it and actually answered it.
("Digital dust." Heh.)
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"For those who aren't in on the secret"
I think you were the only one.
Re:Pfft. (Score:5, Funny)
No one never told you about backups and hashes?
I think the parent knows all about hashish.
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>My understanding is that it's better to use CDs with gold reflective layers, rather than silver, as silver is prone to tarnishing. Is that correct?
Taking this question seriously because there is an actual serious answer to this.
No. A bit is a bit is a bit. Gold reflects infrared better, but not enough better that it makes a difference in the end.
The biggest risk to CDs is voids in the lacquer on the top. Any scratches or holes in the lacquer top, the aluminum layer underneath oxidizes and vanishes.
The bit depth does matter (Score:5, Insightful)
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.
Re:The bit depth does matter (Score:4, Insightful)
I would say that theoretically, 44 kHz is enough, but in practice the filtering is a bit of a PITA. WIth 48 kHz, you can use shorter filters and it's much easier to convert to-from other widely used sampling rates (e.g. 8 kHz and 16 kHz for telephony/VoIP). Otherwise, I fully agree that 192 kHz is totally stupid.
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Re:The bit depth does matter (Score:4, Insightful)
First, most studio masters are 48kHz. Finding 96kHz or even 192kHz mastered audio is HARD. The range and selection of media capable of those sample rates is extremely low. Maybe under 100 Blu-Rays have 96kHz audio tracks, and far fewer have 192kHz tracks. And 96kHz has been around since the DVD days, and we still get audio mixed at 48kHz.
They do, however use 24bit sampling.
As for why go 96kHz or 192kHz, it's quite minor. For this, we need to explore sampling theory.
First, you have an analog signal. Then you MUST pass it through a low-pass filter (called an anti-aliasing filter) that bandlimits the input signal so it doesn't exceed the Nyquist limit (which will cause aliasing in the sampled waveform).
The trouble spot is the analog filter. If we assume that human hearing stops precisely at 20kHz, at 44.1kHz, we have to have a filter that basically has a stop band from 20kHz to 22.05kHz. It takes a lot of work to do this and the filters tend to be pretty big if you want to achieve filters that have flat passbands and low phase-distortion.
At 48kHz, you have a stop band of 20kHz through 24kHz, which makes for a much easier design. At 96kHz, you have a LOT of stop band. Enough so that you can perhaps set the passband higher (you have to block frequencies above 48kHz, so you can start your stop band somewhere between 24-25kHz which should cover the majority of people's hearing. And you'll have a whopping 24kHz or so for the stop band, making for a very clean filter with gentle rolloff (which generally gives you better passband performance - flatter response on the pass band, and very low phase distortion).
At 192kHz, that's really getting excessive - even if you set the pass band at a ridiculous 48kHz to cover every possible human and dog, there's a pile of bandwidth available for the stop band.
96kHz audio may sound better if you're young (or a dog), but a good chunk of the older population has hearing that rolls off starting around 16kHz or so.
Hence why the vast majority of works are sampled at 48kHz - it really is good enough and those that can hear ultrasonic will lose the ability in a few years.
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It's not the music that matters so much as the mixer/DAC.
High bit-depth playback matters greatly for any system that controls volume at the mixer stage rather than the amp stage. This is very important for PCs, where it is common to keep the amp (speakers) at a fixed volume and control the actual listening volume from the operating system's mixer.
If you keep the volume all the way up on your mixer, controlling your listening only at the amp stage, then a 16-bit pipeline is plenty. Highly integrated hardware
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Which is why sound cards that do mixing/volume control in the digital domain have an upscaling step before their final mixing stage. So the source material still doesn't need to have more bits.
Re:The bit depth does matter (Score:4, Insightful)
as a current audio engineer (doing dac's and spdif circuits), let me inform you that 88.2, 96, 176.4 and 192 are well alive and working well and showing some really impressive test measurements.
I can't hear any better than cd redbook (even then its better than my aging hearing) but I sure can see it on the test gear I use to design my own gear with.
its cheap, too. wolfson dac chips are $10 or so, give or take. that's a current high-end pick and it tests very very well. so well that most analog buffers can't keep up and many power supplies are not low noise enough.
I do agree tha 192k is overkill for final delivery. I also shoot photos and I downmix to 8bit jpg but I insist on getting 24bit raw images, doing all my processing at 24bit color, then finally going down to 8 again for jpg saving. audio is exactly like that, too.
but in photo, you are either slim (8bit jpg) or really a pig and taking up far too much room. in audio, there are many grades. you can be 88.2 (relative of 44.1 cd) or 96k (never intending to go down to cd silver disc format). you can record at a multiple of 96k (first multiple is 192k) and then downconvert to 96k for user distribution.
dacs at 24/96 are a GOOD break point for performance (chip and circuit) and cost. files are not that big at 2496 either, really. 192 is nuts for end users but 2496 is quite good.
Re:The bit depth does matter (Score:4, Informative)
Maybe you should insist more, because they're no such thing as a 24bit/channel camera.
AFAIK, the highest bit depth you can get is 16bit/channel on high end medium-format sensors.
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and your ears definitely can't vibrate that quickly
Your ear drums top out at 20KHz, but some of the small bones in your ear will vibrate up into the 60s' and that passes on auditory information. This can help provide clues for positioning, at least.
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As a former audio engineer with some ranking success, I can tell you that it's true -- delivering high-sample rate audio as an end format is really pointless. It hardly makes sense in a studio, and definitely is illogical for the distribution of a final mix.
Maybe this is true for people who just want to listen - but what about non-studio music nerds that want to play around and sample and remix tracks? Amateur musicians would like as high-sample rate audio as possibly, so that any down-mixing artefacts don't accumulate.
The only argument for not distributing the full sample rate audio in the current environment of high bandwidth and high disk space is if you believe that music creation should start and end in studios. I can't express how much I disagree with
Re:The bit depth does matter (Score:4, Insightful)
However I've always doubted it as, it's can be defeated with a pen and paper.
Basically, what you're saying is that you have no background in maths, but you can disprove a very well known and thoroughly proven theorem by sketching lines on a peice of paper.
You can also draw a triangle with bent edges to disprove Pythagoras too, if you like.
You can also disprove Fermat's Last theorem by showing 1782^12 + 1841^12 = 1922^12 on many common calculators, too.
It is well known that the Nyquist frequency (and that frequency only) cannot have the phase or amplitude reconstructed correctly. *every* *single* frequency below that can, no matter what you think your bar graphs look like.
22kHZ will not be reconstructed correctly. 21.9999kHz will be, and that's still above the threshold of hearing.
Since you've gone to the effort of drawing them, now draw them after they've run through an analog filter. That's a little bit harder...
The sampling errors you refer to add noise over the entire frequency spectrum. This is well known and the article even addresses it very obliquely (noise floor).
the poster at xiph never heard of Monster Cable (Score:5, Funny)
"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
It really doesn't matter (Score:4, Insightful)
Re:It really doesn't matter (Score:4, Insightful)
16 bits isn't enough dyanamic range, sort of. (Score:3, Insightful)
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.
Why not lossy-compress 24bit/192kHz? (Score:3)
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Does it matter when the dynamic range is shot to hell?
Re:The article writer is a deaf idiot (Score:5, Funny)
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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.
Re:The article writer is a deaf idiot (Score:5, Insightful)
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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 list
Re:The article writer is a deaf idiot (Score:5, Insightful)
Re:The article writer is a deaf idiot (Score:4, Insightful)
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!
Re:The article writer is a deaf idiot (Score:5, Informative)
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.
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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.
Re:The article writer is a deaf idiot (Score:5, Informative)
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
Re:The article writer is a deaf idiot (Score:5, Insightful)
Re:The article writer is a deaf idiot (Score:4, Funny)
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.
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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...
Re:The article writer is a deaf idiot (Score:5, Insightful)
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Re:The article writer is a deaf idiot (Score:4, Informative)
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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!
Re:The article writer is a deaf idiot (Score:5, Insightful)
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.
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> 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 d
Re:The article writer is a deaf idiot (Score:5, Insightful)
>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
Comment removed (Score:5, Funny)
Re:The article writer is a deaf idiot (Score:4, Funny)
$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.
Re:The article writer is a deaf idiot (Score:5, Insightful)
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.
Re:The article writer is a deaf idiot (Score:5, Funny)
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?
Re:The article writer is a deaf idiot (Score:5, Funny)
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.
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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.
Re:The article writer is a deaf idiot (Score:5, Funny)
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.
Re:The article writer is a deaf idiot (Score:5, Interesting)
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.
Re: (Score:3)
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.
Re:The article writer is a deaf idiot (Score:5, Interesting)
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...
Re: (Score:3)
Sure, but with the loudness war, they're not really using the 16 bits they have, so what's the point?
Re:The article writer is a deaf idiot (Score:5, Insightful)
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.
Re:The article writer is a deaf idiot (Score:4, Interesting)
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.
Re: (Score:3)
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
Re: (Score:3)
we're talking about sample rates (kHz). you seem to be talking about bit rates (kbps).
Re:The article writer is a deaf idiot (Score:5, Insightful)
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.
Re:The article writer is a deaf idiot (Score:4, Funny)
Yeah, what would a guy named xiphmont know about signal processing?!
Re:The article writer is a deaf idiot (Score:5, Interesting)
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...
Re:I can tell the difference (Score:5, Informative)
Re: (Score:3)
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.
Re:Audiophiles (Score:5, Insightful)
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.
Re:44KHz (Score:4, Informative)
http://en.wikipedia.org/wiki/Nyquist_frequency [wikipedia.org]
Re: (Score:3)
Either Harry Nyquist or Claude Shannon [wikipedia.org] probably could have. But they are both dead now. So we will have to take Monster Cable marketing department's word for it now.
Re:44KHz (Score:4, Informative)
The Nyquist-Shannon Sampling Theorem [wikipedia.org] basically shows that if an analogue signal contains no frequency higher than B Hz then sampling at any rate greater than 2B Hz is adequate to reproduce the signal without aliasing. In the case of audio recording intended for the human ear, the highest audible frequency is about 20kHz and the minimum sampling rate to cover that should be 40kHz. This is (partly) where the 44100 HZ sampling rate of CD audio comes from. In practice sampling is usually performed faster than required by the theorem (though not four times faster). The theorem is not sufficient in itself to guarantee perfect reproduction and is limited by the ability of real systems to match the mathematical ideals during sampling and reproduction. Reproduction is, however, typically very close.
The 192kHz sampling that is the subject of this thread is capable of capturing frequencies well beyond the capability of a human ear to hear, or any typical speaker system to reproduce.
Re: (Score:3)
There may be no theoretical benefit, but since there's no such thing as an ideal sampler or filter or quantiser, it has many practical benefits.
Re:44KHz (Score:5, Informative)
There may be no theoretical benefit, but since there's no such thing as an ideal sampler or filter or quantiser, it has many practical benefits.
Here is a quick example. You sample at 44 kHz. The first Nyquist zone is from 0 to 22 kHz, the second one is from 22 to 44 kHz (with flipped spectrum.)
Now, say that some [mechanical] harmonic from some instrument has frequency of 33 kHz. We don't hear those with our ears (parts of the ear are too massive to vibrate fast enough) so no harm done. The orchestra is playing as usual.
But now record this orchestra with an imperfect antialiasing filter (there are reasons why a perfect one wouldn't do you much good anyway.) The 33 kHz harmonic falls into the 2nd Nyquist zone. It will be played back as if it was (22 kHz - 11 kHz = 11 kHz.) Can you hear 11 kHz? Most people hear it just fine. Think about it for a moment. There was no 11 kHz signal in the original spectrum; there was 33 kHz, an inaudible one. The artifact showed up because a [lossy] mathematical operation was performed on the data that describes the signal. The resulting distortion produced an audible tone where none was present originally.
However if you encode at, say, 128 kHz sampling rate, things change. First, the antialiasing filter - even if it is of the same architecture - will have its cutoff way below the Fs/2. This means that signals of the second Nyquist zone will be attenuated by many tens of dB - essentially they can be completely eliminated because nobody cares what you do to ripple and phase above 30 or 40 kHz. Second, for the alias to show up it has to be in LF radio band now, starting at 128 kHz. Microphones aren't even mechanically capable of picking up those frequencies. And finally, if that 33 kHz harmonic passes through the filter (with the same mediocre attenuation as in the first example) ... it will be played back as 33 kHz, and it won't go anywhere. The amplifier will filter it, and the speakers will attenuate it greatly. In other words, a serious distortion that was present when you are sampling at 44 kHz disappears when you are sampling at a much higher rate.
Re:44KHz (Score:5, Interesting)
That's a profound misrepresentation of how hearing works.
Here's an oversimplified and inaccurate explanation. The ear's mechanism relies on different frequencies providing the highest level of excitation at different places. Your trained nervous system recognizes each different place as a different tone.
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.
Re:What if... (Score:5, Funny)
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.
Re:Pro recording (Score:4, Insightful)
44.1kHz will be able to capture the basic information of the signal, as the human ear can hear to 20kHz in some cases, and Nyquist's theorem says that to recover the information you need to sample at least double the highest frequency. Oversampling (i.e. 192kHz) allows much more room to develop a good anti-aliasing filter. It may be that the reverb is phase-shifted somewhat with standard AA-filters, but ones designed for the higher sampling rate can have more linear phase. Also, higher sampling rates allow for better reconstruction of the actual wave form, if you're interested in music rather than just information. So yes, sampling a telephone call at 192kHz would be stupid, but if you're an audiophile, doing it for music is quite reasonable.
Re: (Score:3, Informative)
Oversampling (i.e. 192kHz) allows much more room to develop a good anti-aliasing filter.
*whoosh*
As the whole point of the article goes right over your head! You do not need any anti-aliasing. If you sample at 40 kHz with a decent equipment and a good 20 kHz low-pass filter then you can completely and faithfully recover a signal of less than 20 kHz by applying the Whittaker-Shannon interpolation formula.
Now we generally sample at 44.1 kHz in order to have some oversampling to take care of non-ideal filters and such. This is 10% oversampling and it's far more than you need with modern equipment
Re:Pro recording (Score:4, Informative)
Re:Pro recording (Score:5, Insightful)
The problem with low-pass filtering was resolved eons ago with a concept called "oversampling."
Only the earliest and ruddiest of CD players (and a lot of computer sound cards) had a brick-wall filter at ~22.5 KHz. The rest of them resampled the input by 4x or 8x, or converted the original signal to PWM, and then applied the anti-aliasing filter at a frequency several octaves above the range of human hearing.
This hypothetically pushed the nastiness inherent of a steep filter to a realm well outside such that humans could hear, and at least far beyond the limited confines of a CD.
Welcome to 1985, where your stated concerns are both accurate and already solved.
Re:Pro recording (Score:5, Informative)
Some of the issues pointed to in this and other posts regarding oversampling and AA filters are not really relevant to the subject at hand, given the technology currently in use. A statement like 'oversampling at 192 kHz' shows a lack of knowledge regarding the kinds of audio converters that have been in use for a good while now. A Delta Sigma ADC running with an Fs of 48 kHz might often be oversampling at 3.072 MHz or 6.144 MHz. Anti aliasing filters that many people have mentioned are implemented digitally inside the converter (no need for external analog filters, which may well exhibit many of the problems mentioned), and actually have extremely good pass band ripple.
Look at datasheets for converters from manufacturers such as TI (burr brown) [ti.com], cirrus [cirrus.com] [page 36 here has detailed plots of 48, 96, and 192 kHz pass pand characterisitcs for the device, highlighting the fact that increasing the sampling rate does not improve pass band ripple for this device (also note the scale is 0.02 dB/div)], AKM [asahi-kasei.co.jp], Wolfson micro [wolfsonmicro.com] You will find pass band pass responses that are flat to within less than +/- 0.05 dB over the audible range, and stop band attenuation in excess of 100 dB, whether sampling at 48 kHz or 192 kHz. If you can find anything in actual converter datasheets that points to better converter performance from selecting a higher sampling rate, I would be interested to see it.
All in all, the basics of sampling theory don't really help people to understant the real world issues in designing a moden high end audio device. And in the end, surely the proof of the pudding is in the blind tests, that never seem to show that anybody can tell any difference when moving to higher rates? Even if there were a few people who could hear this difference in some perfect listening envirmonment, would it really make sense for everyone else to go out and buy 192 kHz equipment?
Re:Pro recording (Score:5, Informative)
Re:Pro recording (Score:4, Informative)
Recording a signal with high fidelty is NOT a matter of just taking samples at defined intervals. If you do that you will get aliasing (higher frequencies getting converted to lower frequencies by the sampling process). So before you sample you need an "anti-aliasing filter" to remove signal components above the nyquist point.
However filter design is a compromise, a filter with a steep response in the frequency domain will have a long impulse response in the time domain. A filter that doesn't cause phase distortion will cause pre-echo when fed with an impulse signal. Further making high order analog filters reliable and well behaved is difficult.
Similarlly at output many digital to analog conversion methods will produce unwanted copies of the signal beyond the nyquist point, again a filter (known as a reconstrution filter) is needed to remove these.
96KHz gives you a much bigger "gaurd band" between the audio signal and the nyquist frequency so your anti-aliasing and reconstruction filters can be much less aggressive.
Using oversampling (running your recording/playback devices at higher than the sample rate you are storing the music at) and doing most of the filtereing digitally can remove the issues with high order analog filters being unstable but it can't change the fundamental issue that a filter with a sharp response in the frequency domain will have a long impulse response in the time domain or that a filter with no phase distortion in the frequency domain will have pre-echo in the time domain.
Re:Pro recording (Score:5, Informative)
I recently remixed a classic recording for sony records. The files where rolled off of tape at 24bit/96k. 48k I can understand but 96k is pointless. WAAAAAAY beyond the range of human hearing. In the old days, things like cymbals and brass could really stick out because the encoders and decoders where just not where they are today.
Anyone that tells you they can hear the difference between 48k and 96k is dreaming. Its the quality of the recording that counts more than anything these days.
The difference is that the antialiasing filters are much simpler and have a gentler roll-off when sampling at 96kHz. The high-order filters necessary to ensure adequate attenuation at Nyquist and above when sampling at the lower rates have this tendency to ring.
Re: (Score:3)
My point exactly. Storage is not a factor so when you have the choice in a studio setting why not keep the best resolution you can?
Now for playback you obviously wouldn't want to log 2GB files for a 3min hiphop song, in that case you can just downsample to
flac/48khz or even 256kbps mp3. Hell, most consumers are happy with the crappy 112/128kbps rips they get extracted from youtube videos.
Re:Pro recording (Score:5, Insightful)
When I listen to music, its not for the data -- its for the feeling. You should try listening to music for the feeling too ;-)
My opinion.
Re:Pro recording (Score:5, Insightful)
My favourite audiophile rebuttal quote:
"If your hifi costs more than your music collection you have missed the point." - Unknown Source
Re: (Score:3)