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.

#131

Post by polshaw » Mon, Mar 05, 2012, 11:22 PM UTC

Anechoic wrote: post #121

brigade wrote: post #119

Earlier quoted context omitted.

I'd disagree, at least among the people I know: they all have cheap HTIB systems and the single biggest, most cost-effective improvement you can make from there is to buy better speakers. Then you can worry about your room.

No, even in that case, the room can still overpower the speaker. A $200 HTIB system in a properly treated room will sound a lot better than $28,000 Wilson Watt Puppies in a bathroom for example.

Sure, but that's a rather contrived example-- most people have a fairly normal room, and the average joe would be best served by getting a good pair of speakers, and a reasonable amplifier and DAC, before worrying seriously about room acoustics.

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

#132

Post by sirwitti » Mon, Mar 05, 2012, 11:24 PM UTC

aiscott wrote: post #45

jaylevitt wrote: post #29

Earlier quoted context omitted.

I'm pretty sure that #2 isn't true; signal processing folks will be able to phrase this better than I can, but I think that if you have enough information to capture the waveform at a given frequency, you also have enough information to precisely place it in time - phasing errors are more likely due to quantization error, which is about bit depth, not sample rate. No?

[edited: I was wrong]

i think what jaylevitt is referencing to is that there is interpolation going on in the dac. that could mean (i'm no dac expert, so not sure) that the dac could guess more granular than the sampling rate would allow the start points (of transient e.g.)

but the question for me is how exact that guessing is. correct me if i'm wrong but, that interpolation happens twice: when recording by the adc and on playback by the dac.

so a lot of that whole discussion (yeah, finally something about acousticts :) depends on how accurate interpolation works in adcs and dacs.

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

#133

Post by mturmon » Mon, Mar 05, 2012, 11:26 PM UTC

nullc wrote: post #64

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…

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 is based exclusively on standard linear systems theory.

Of course, the ear and some of its perceptual components may be significantly nonlinear, and thus not covered by the frequency response graphs of TFA.

These graphs assume linear systems, in which you put two frequencies in, and the same frequencies pop out in scaled form. Nonlinear systems can produce new frequencies in response, and this possibility is not discussed in TFA. Probably these effects are quite minor, but may be audible to some listeners on some equipment for some choices of source material.

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

#134

Post by shabble » Mon, Mar 05, 2012, 11:26 PM UTC

Anechoic wrote: post #82

cynicalkane wrote: post #68

Earlier quoted context omitted.

The second half of the statement is wrong, but the first half is right. Harmonics in real-world instruments are not usually exact multiples of the fundamental. A simple diffeq model of a rigid oscillator will show you this mathematically. An extreme example is present on modern pianos, where the high rigidity of the loud, heavy piano strings can cause tuners to stretch the lowest and highest notes as much as a half-s…

Harmonics in real-world instruments are not usually exact multiples of the fundamental. A simple diffeq model of a rigid oscillator will show you this mathematically. That is absolutely incorrect, mathematically speaking, harmonics are by definition "integral multiples of the fundamental." (Fundamentals of Acoustics, Kinsler & Frey).

People from a musical, non-signals background tend to use 'harmonics' as a synonym for 'overtones' or 'partial tones', which is where the confusion arises, I suspect.

There's a measure -- inharmonicity[1] -- of how far the actual overtones of a particular instrument differ from their theoretical fundamental multiples.

[I suspect you already know this. This reply is probably for others' benefit]

[1] https://en.wikipedia.org/wiki/Inharmonicity

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

#135

Post by glassx » Mon, Mar 05, 2012, 11:27 PM UTC

amouat wrote: post #16

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…

I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim. Just because it is difficult to record a triangle does not necessarily mean it is impossible to accurately recreate the sound (to human ears) u…

> I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim.

Yes, you're right.

In fact, some of the section X references don't even mention hearing, they talk about "alpha-EEG rhytms" (in this case "listeners explicitly denied that the reproduced sound was affected by the ultra-tweeter") and "bone-conducted ultrasonic hearing" trough the "saccule" ("organ that responds to acceleration and gravity and may be responsible for transduction of sound after destruction of the cochlea").

--

In fact, most of the claims of the article are around the fact that there is energy over 20khz and how it can affect recording process.

This is a well known fact, and this is exactly why engineers filter out sub-sonic and super-sonic frequencies, especially today: stuff that you can't hear (or feel) will just suck your headroom and make you lose the loudness war.

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

#136

Post by eatmyshorts » Mon, Mar 05, 2012, 11:27 PM UTC

Anechoic wrote: post #62

cmer wrote: post #49

Earlier quoted context omitted.

Humans will hear the impact > 20kHz frequency has on the lower frequencies, not the 30kHz frequency itself. That's been proven a million times.

If you're not going to hear the frequency, then there's no reason to record it, so I don't see what you're objection is.

Yes, but if you sample the frequency to create a step wave, then neglect to filter the results, you will end up reproducing tons of high frequencies. That is why we need to filter the output for signals >20KHz...to remove these harmonics that result from reproducing the square wave.

Of course, filters aren't perfect, and result in phase shift and roll-off. So we over-sample the signal to create a signal with a much higher frequency than 20KHz, so that the filtering occurs well outside the audible band, allowing us to filter out all of these harmonics without affecting the desired signal.

Basically, the end result is that by sampling the signal, you are introducing high frequency content that must be removed prior to playback. This high frequency content is one of the reasons old CD players from the 80s and 90s cause "listener fatigue", although I have no sources to back up that last statement.

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

#137

Post by cynicalkane » Mon, Mar 05, 2012, 11:27 PM UTC

Anechoic wrote: post #82

cynicalkane wrote: post #68

Earlier quoted context omitted.

The second half of the statement is wrong, but the first half is right. Harmonics in real-world instruments are not usually exact multiples of the fundamental. A simple diffeq model of a rigid oscillator will show you this mathematically. An extreme example is present on modern pianos, where the high rigidity of the loud, heavy piano strings can cause tuners to stretch the lowest and highest notes as much as a half-s…

Harmonics in real-world instruments are not usually exact multiples of the fundamental. A simple diffeq model of a rigid oscillator will show you this mathematically. That is absolutely incorrect, mathematically speaking, harmonics are by definition "integral multiples of the fundamental." (Fundamentals of Acoustics, Kinsler & Frey).

Then one would be forced to conclude that many instruments have no harmonics at all, which is obviously not what 'harmonic' is referring to in this thread of discussion. Why be pedantic when it's obvious what everyone is talking about?

Anyway, it's not as though mathematical literature requires you to use a term exactly one way. I had a diff eqs textbook that used the word 'harmonic' in exactly the way I used it above when I made reference to diff eqs...

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

#138

Post by pragmar » Mon, Mar 05, 2012, 11:29 PM UTC

surrealize wrote: post #61

cmer wrote: post #49

Earlier quoted context omitted.

Humans will hear the impact > 20kHz frequency has on the lower frequencies, not the 30kHz frequency itself. That's been proven a million times.

Note the lack of citation

http://en.wikipedia.org/wiki/Overtones

I don't know about the physics of the speaker itself generating the overtone (in cabinet), but it could certainly resonate a wine glass in the room, for example.

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

#139

Post by marcusf » Mon, Mar 05, 2012, 11:30 PM UTC

marcusf wrote: post #127

Anechoic wrote: post #99

Earlier quoted context omitted.

Agree one-hundred percent about the room (although the prescription isn't always as simple as "get a thicker rug" etc). The other issue regarding high-frequency sound reproduction is that in most cases, the loudspeaker won't be outputting much beyond 22-25 kHz (assuming very good quality loudspeakers, cheap consumer grade units might struggle to hit a -6 dB point at 18 kHz) and even for the speakers that have usable…

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?

...of course I mean audible. Or spectral? Not aural, anyway. English is not my first language. Sorry.

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

#140

Post by sjwright » Mon, Mar 05, 2012, 11:31 PM UTC

sjwright wrote: post #78

I must say I get rather irritated when people spend time worrying about dubious 'tweaker' methods to improve their audio, when the most under-performing component of most people's sound equipment also has the lowest-hanging fruit: The room itself. When people ask me where they should spend money to improve the quality of their hi-fi or home theater system, in nearly every case my response will be something like "get…

I should point out that after room treatment, my next recommendation tends to be an amplifier with Audyssey MultEQ in-room calibration. I've never heard a listening environment that didn't sound unambiguously better with it enabled.