There is no point to distributing music in 24-bit/192kHz format. • Hacker News Board

Re: There is no point to distributing music in 24-bit/192kHz format.

#161

Post by nullc » Mon, Mar 05, 2012, 11:48 PM UTC

mturmon wrote: post #133

nullc wrote: post #64

Earlier quoted context omitted.

No. This really is not the case. The article _specifically_ addresses this misconception. The signal reproduced from your 44.1kHz sampled digital input is not a stair-step like some broken waveform editor might display: On output it goes through a matched reconstruction filter (which may, in fact, be digital and involve an oversampled DAC or it could be analog though those are harder to build without compromise). Aft…

@nullc: of course you're right, and the commenter you're replying to does not understand the Nyquist-Shannon sampling theorem. Which is a shame, because the article specifically addressed this point. These discussions of audio standards always get sidetracked by people who don't understand or believe this result. (Have to admit, the result is surprising). I think there may be problems with the argument in TFA, which…

Indeed, but if there were non-linearies in the ear (there are many, of course) which allowed detection of ultrasonics (less likely, because the first stage of the ear is impressively linear) you'd expect them to show up in the actual listening tests.

The TFA does at least make this the-proof-is-in-the-pudding point somewhere in its depths. :)

Re: There is no point to distributing music in 24-bit/192kHz format.

#162

Post by juiceandjuice » Mon, Mar 05, 2012, 11:50 PM UTC

eatmyshorts wrote: post #129

Anechoic wrote: post #43

Earlier quoted context omitted.

For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz You are going to have to provide me with a citation to back that up because that goes against everything I've learned and experience in 17 years of working in acoustics.

Here is a Wikipedia article on the subject: http://en.wikipedia.org/wiki/Sound_from_ultrasound Basically, if you produce two ultrasonic frequencies, they will create an interference pattern at a much lower frequency than either of the individual frequencies. Modulate a signal on the difference between two signals, and you can create a directional speaker, since ultrasonic sounds tend to be highly directional (so long…

Recording music is supposed to be a snapshot (with room for interpretation) of the composition at play.

Trying to record an edge case like this is the same as recording in a room with bad acoustics. So you end up with some weird (but not faithful) representation of the sound which is a snapshot of the microphone's characteristics and directionality of the ultrasonic tones. It's not reasonable to assume any microphone will behave exactly like a human ear. Even if you could, you're going to have to mimic the tiny random movements a normal person would make listening to a sound, movements which would definitely impact the perception of the sound, because microphones are much more stationary than any human would be.

The "different sounding" argument two posts above is silly, because sound is almost never that monochromatic, and if it is, it's usually boring. Also I don't understand how missing out on an odd order harmonic would be a bad thing :) The reality is none of these arguments are based in a reality of what people would hear, and because of that, the arguments aren't practical.

In reality, 20 bits at 48kHz (or 64kHz) would be more than acceptable for even the most discerning of ears and probably the most practical in terms of space and fidelity, but it'd be a weird format to distribute in.

Re: There is no point to distributing music in 24-bit/192kHz format.

#163

Post by nitrogen » Mon, Mar 05, 2012, 11:50 PM UTC

bigiain wrote: post #156

marcusf wrote: post #127

Earlier quoted context omitted.

I might sound/be stupid for asking, but what's the actual physical response from something at 22 kHz+? I have a hard time picking up a pure sine > 17 kHz. I doubt I'd get any aural response from anything at 22 kHz, so what's the deal?

The deal is just that you're getting older. Your ears just don't work as well as a 12year old's. Neither do anybody else's your age (within the bounds of typical human variation - probably well over 95% of use _never_ heard 22kHz, not matter _how_ "young" our ears were).

I was once in a small, treated room working with some rather large PA speakers. I was curious how far my hearing range actually extended, and did something very unwise: I played a 20kHz tone and very briefly ramped the volume up and down. I definitely heard it, but I also induced quite a lot of pain. I learned two lessons: 1. my threshold of hearing at 20kHz is near or above the threshold of pain, and 2. don't do that ever again.

Re: There is no point to distributing music in 24-bit/192kHz format.

#164

Post by nullc » Mon, Mar 05, 2012, 11:53 PM UTC

cmer wrote: post #58

hackermom wrote: post #32

There is no point with going over 16 bits, but there is definitely a point with going over 44.1khz, as it allows you to actually reproduce waveforms more accurately than 44.1khz. Try reproducing f.e. a sinewave accurately over 4-5khz with a sample rate of just 44.1khz - it cannot be done, and at this point we haven't even taken into account the issue of varying slew-rate characteristics of the thousands or so differe…

Couldn't agree more with you! 192kHz is overkill as a "final" format. 16 bits is very limiting for music with lots of dynamics (ie: classical). Very quiet sounds sound quite bad at 16 bits, but since most pop music has about 6-12db of dynamic range, it doesn't make much of a difference. I always thought the sweet spot would be 96-24. But the truth is, the market wants smaller and portable digital files, not higher qu…

16 bits is not "very limiting" for anything, unless you think your ears themselves are very limiting.

Many things are mastered poorly— recording engineers crushing the dynamics in order to get the loudest possible signals— mostly a problem for pop music, but nothing is immune.

It's been observed that the various 'higher-definition' recordings have less brutal mastering— no doubt owing to the different audience they are marketed to. But this isn't a property of 24-bit vs 16-bit distribution.

Re: There is no point to distributing music in 24-bit/192kHz format.

#165

Post by eatmyshorts » Mon, Mar 05, 2012, 11:58 PM UTC

Anechoic wrote: post #155

eatmyshorts wrote: post #129

Earlier quoted context omitted.

Here is a Wikipedia article on the subject: http://en.wikipedia.org/wiki/Sound_from_ultrasound Basically, if you produce two ultrasonic frequencies, they will create an interference pattern at a much lower frequency than either of the individual frequencies. Modulate a signal on the difference between two signals, and you can create a directional speaker, since ultrasonic sounds tend to be highly directional (so long…

That article describes hetrodyning which happens because ultrasonic frequencies at high amplitudes interacts nonlinearly with air. You are not going to see that effect with sound waves generated near the audible spectrum, and normal loudspeakers are going to generate ultrasonic sound waves.

Yes, but the effects of interference patterns between multiple ultrasonic frequencies is the same, and definitely does affect the audible spectrum. This is why we must filter the square wave that comes out of a DAC. And the limitations of filters (phase shifts and roll-off) are why modern CD players oversample the signal--so that the filtering can be performed well beyond the audible spectrum.

Re: There is no point to distributing music in 24-bit/192kHz format.

#166

Post by sjwright » Mon, Mar 05, 2012, 11:59 PM UTC

davesims wrote: post #152

sjwright wrote: post #116

Earlier quoted context omitted.

I have also spent many hours spent in high-end recording and mastering environments, and it's my observation that most engineers suffer from confirmation bias just like everyone else on the planet. I've caught engineers using L1-Ultramaximizer (or similar) to bounce a recording down to 16-bit/44.1khz as part of the mastering process, and they're always surprised when they're completely unable to hear the difference e…

Audio, perhaps like the wine industry, is a vast bastion of confirmation bias and subjectivity, no argument there. But I know what my ears hear, and IMO there is absolutely a vast different between 44.1 and 192. I'm not sure how you can even question it. Someone else on the thread was saying it's impossible to hear the difference between 16bit and 24bit. I don't even know what to say to that. It's like telling me the…

  > I'm not sure how you can even question it.

With evidence.

Re: There is no point to distributing music in 24-bit/192kHz format.

#167

Post by kahawe » Tue, Mar 06, 2012, 12:00 AM UTC

cmer wrote: post #30

tompagenet2 wrote: post #6

Earlier quoted context omitted.

Hey cmer, thanks for posting I don't think I understand quite what you're saying and wondered if you could explain more. You and the article both say that humans can't hear above about 20kHz. If there are higher frequencies that create a harmonic at a lower frequency (e.g. a 33kHz harmonic that produces a sound at 16.5kHz) then surely that lower harmonic (16.5kHz in this case) will be recorded by the original recordi…

Let's make things super simple. Let's say you record 4 sine waves at a 192kHz sampling rate: 15kHz, 30kHz, 45kHz and 60kHz. All 4 frequencies will be captured and the 15kHz frequency will sound different to your hear because its harmonics. If you take this recording and master it for a CD (44.1kHz), you'll effectively get up to ~20kHz (since they're a low pass filter starting at around 16-18kHz). This means that only…

> The harmonics don't modify the fundamental frequency, they just trick the human hear. But when they're gone, they have no effect whatsoever.

This is the part I really do not understand... either my ear CAN pick up those frequencies, maybe the harmonics are "tickling" the little hairs inside my cochlea and ultimately the frequencies I can actually hear were altered in my perception that way - or I can not hear or sense the harmonics and they physically alter the "original" wave that I end up actually hearing.

Either way, pretty much the exact same thing should happen in a studio microphone. Those all do have frequency limitations and AKG, Royer, Rode, Shure, Sennheiser, Audio Tech, what-have-you pretty much all go up to 15kHz or 20kHz according to specs, if I understand them correctly, but not further than that. If it isn't even recorded, those frequencies I also cannot hear can NOT alter my perception so they HAVE to somehow change the frequencies I can hear and are being recorded... on top of that you are making "room" for frequencies up to, say, 60kHz but I very strongly doubt your mics can go even remotely that high.

Re: There is no point to distributing music in 24-bit/192kHz format.

#168

Post by xiphmont » Tue, Mar 06, 2012, 12:00 AM UTC

mistercow wrote: post #150

> Can you see the LED flash when you press a button? No? Not even the tiniest amount? I used to be able to see it when I was a kid (it looked very faintly red), but I just tried it and couldn't see it at all. That's actually a little bit disturbing.

I would guess that's because some earlier remotes used a higher frequency IR emitter that was in fact touching into the red.

These days, the various IR communication protocols have been standardized and virtually all use 920nm, 940nm or 980nm emitters, all of which will be invisible. I mentioned the Apple IR remote specifically because it's a remote most people reading TFA will have, and it's known to be a 980nm emitter.

Re: There is no point to distributing music in 24-bit/192kHz format.

#169

Post by tzs » Tue, Mar 06, 2012, 12:03 AM UTC

eatmyshorts wrote: post #145

cmer wrote: post #4

There's a lot of scientific-sounded content in this, but unfortunately most of it couldn't be further from the truth. I'm an ex-audio engineer and studied digital and analog audio engineering; this has been debated to death over the last 15 years. Digitally recording a triangle is the best example of why 48kHz is very limiting. The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 5…

Yes, but our ears only hear 20Hz-20KHz. So, according to Nyquist theory, you can recreate the entire signal that the human ear hears by recording those sonic artifacts that result from interference between supersonic harmonics. So while it's true that the human ear can't hear well above ~18KHz, and the interference between high order harmonics are audible, it's also true that a properly recorded signal, sampled at 44…

What about sound outside 20-20k that affects us via mechanisms other than being directly sensed in the air by our ears? For instance, consider frequencies below 20 Hz that we can feel with our feet as vibrations in the floor, instead of hear with our ears? Or what about the possibility of sound above 20k causing a vibration in something other than our ears, which could have a subharmonic in 20-20k that gets conducted to our ears via bone?

I'd prefer recording technology to err on the side of capturing what we need to reproduce all of that, even if we aren't sure that we need it.

Re: There is no point to distributing music in 24-bit/192kHz format.

#170

Post by wickedchicken » Tue, Mar 06, 2012, 12:05 AM UTC

rmcclellan wrote: post #109

aiscott wrote: post #45

Earlier quoted context omitted.

[edited: I was wrong]

This is completely incorrect, by shannon ( http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_samplin... ). The sampling frequency determines the maximum frequency that can be captured, not the temporal resolution. That said, a transient containing higher frequencies will be sharper than a transient that doesn't, but its onset time resolution will not be determined at all by the sample rate. Said another way, two b…

> two band limited pulse signals with different onset times, no matter how arbitrarily close, will result in different sampled signals.

This is true, but different than what I am arguing. You're saying that a listener over time will be able to tell that the two signals differ. I am saying that a listener will be able to determine this at fractional wavelengths.

It's similar to dithering a high dynamic range signal onto a lower bit depth: more than two samples are required for "evidence" of two different signals, while sampling at a high enough rate will tell you this almost instantly.

Again, I don't know if human ears are able to detect this, just that I haven't seen it addressed in these discussions.