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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.'"
>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
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".
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.
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'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.
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.)
Cogito, igitur comedam pizza.
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".
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.
One thing I know, and that is that I am ignorant...
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.
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.
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".
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 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.
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.
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...
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".
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)
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.
- Michael T. Babcock (Yes, I blog)
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.
My favourite audiophile rebuttal quote:
"If your hifi costs more than your music collection you have missed the point." - Unknown Source
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.
Kid-proof tablet..