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24/192 Music Downloads Are Very Silly Indeed (2012)

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Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#31

Post by terinjokes »

pmalynin wrote: post #19

My go to comparison track for sound quality for a new device, headphones, DAC, amp, codec etc, is "Speak To Me" from Dark Side Of The Moon. I don't hear a difference between various rips that are 16/128 or 24/192, but I have noticed a difference listening to the Blu-ray version of it (which is many-many gigabytes in size). It is a definitely interesting experience, but the way I can describe it is as an absolute abse…

Which would be in line with the article: a SACD release pressed onto a CD-R sounded better than the CD release, because the SACD release had a better master.

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#32

Post by fusiongyro »

sitharus wrote: post #23
fusiongyro wrote: post #14

Earlier quoted context omitted.

Basically, the sampling theorem says you can reconstruct the exact waveform with a certain number of samples. Adding a bunch more samples bulks up the file, but you didn't need them to restore the exact waveform. However, the unnecessary samples are in the file between you and the next sample you do need. At high enough levels of waste this creates an I/O bottleneck that hampers performance. Another way to look at is…

> Increasing the data rate does not continue to make the waveform more and more detailed to human auditory perception in the way that raising the pixel density does for human visual perception. Early low data rate codecs - such as the one used for GSM mobiles - are obviously inferior, but still functional. I think a better analogy is that an iPhone 7 has a 1 megapixel screen, so there's no difference between a 1 mega…

I get what you're saying, that if you remove the ability to zoom you can equalize things, but without that twist I think this preserves the essential problem with the analogy that Monty works so hard to correct. If your data rate is too low, you don't have enough samples to recreate the waveform. So you create an approximate. But there is a finite number of samples that you need to recreate all of the information in the waveform, so once you have that, there isn't any additional information there for you to obtain if you continue to increase the sample rate.

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#33

Post by SmooL »

zkSNARK wrote: post #25
hamandcheese wrote: post #12

Earlier quoted context omitted.

24/192 can capture audio well beyond what we can hear. These sounds can, though, lead to distortion that is within our hearing range, depending on the equipment used. Therefore, rather than just being useless extra data, it can be actively harmful to the listening experience.

I doubt that is the case. Most systems which I have used actually sound better at the higher sampling rates. This is not because the sampling rates cause anything to sound better, but because the hardware works better at those rates. In a blind test, you wouldn't be able to decide which one you liked better, only that there is a difference. At least on the machines I have, there are some clearly audible differences.

"In a blind test, you wouldn't be able to decide which one you liked better, only that there is a difference"

That would have to be the noise. The math doesn't lie...

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#34

Post by trevyn »

anyfoo wrote: post #27
trevyn wrote: post #16

Earlier quoted context omitted.

I imagine it's because lossy codecs are tuned for human perceptual limitations, and when you process audio, it can "pull" areas of the sound that are otherwise hidden from your perception into perceptual ranges, analogous to how fiddling with the brightness/contrast of a highly compressed JPEG image can accentuate the artifacts.

We are not really talking about lossy codecs here, though, are we?

Ah, my mistake. But perhaps you could consider 24/192 -> 16/48 a lossy codec in this context; the same argument does apply, for the same reasons.

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#35

Post by zkSNARK »

sp332 wrote: post #22
squarefoot wrote: post #13

FTA: "192kHz digital music files offer no benefits. They're not quite neutral either; practical fidelity is slightly worse. The ultrasonics are a liability during playback." I would expect the samples to be interpolated before the final DA conversion takes place, so no extra ultrasonics should be involved. And there would be filters for them anyway, we still have to use them for 44.1, 48, 96 etc.

But if you're filtering out ultrasonics, then you've guaranteed that the extra samples will be wasted. There won't be any difference in the reconstructed signal at all.

As an engineer, those extra samples are never wasted, and as a listener, I could care less about the "wasted space." Drive space is meaningless at this point. When I buy a piece of audio, give me the highest quality available and I will make my own decisions regarding the frequency I want to listen to it at. The reasons behind not giving the user the highest quality possible all sound like a misguided excuse to prevent piracy.

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#36

Post by fusiongyro »

willglynn wrote: post #28
fusiongyro wrote: post #14

Earlier quoted context omitted.

Basically, the sampling theorem says you can reconstruct the exact waveform with a certain number of samples. Adding a bunch more samples bulks up the file, but you didn't need them to restore the exact waveform. However, the unnecessary samples are in the file between you and the next sample you do need. At high enough levels of waste this creates an I/O bottleneck that hampers performance. Another way to look at is…

Digital imaging definitely has sampling concerns too: https://en.wikipedia.org/wiki/Aliasing Even analog imaging has concerns that are better described in the frequency domain than the spatial domain: https://en.wikipedia.org/wiki/Airy_disk https://en.wikipedia.org/wiki/Optical_transfer_function

Interesting!

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#38

Post by kuschku »

This article makes the same mistake that is done frequently with video.

So, let’s look at a similar issue with video. Your display is likely only 720p, or 1080p, but a 4K video on youtube will still look a lot better, although technically it should have no visible difference.

But the reality is, we don’t get uncompressed video, or uncompressed audio.

We have a choice between audio or video compressed with lossy codecs, at 16bit/44kHz or 16bit/96kHz, or 4:2:0 video at 1080p or 4:2:0 video at 4K.

And just like you need 4K 4:2:0 mp4 video to get even close to the quality of uncompressed 1080p 4:4:4 video, you also need far higher sampling rate and depth of highly compressed audio to get a quality similar to 16bit/44.1kHz PCM.

That’s the real reason why 24bit/192kHz AAC downloads were even discussed.

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#39

Post by kobeya »

sp332 wrote: post #17
tomc1985 wrote: post #3

While the extra quality is lost on listeners, from experience I've found that super-HQ source material can change your results (for the better) when fed through distortion/compression/warming effects processors

Sure, when you're editing music it can be helpful to have super high sample rates and lots of bits to work with. But for listening, it's just a waste of space.

He was talking about listening.

Re: 24/192 Music Downloads Are Very Silly Indeed (2012)

#40

Post by sp332 »

zkSNARK wrote: post #35
sp332 wrote: post #22

Earlier quoted context omitted.

But if you're filtering out ultrasonics, then you've guaranteed that the extra samples will be wasted. There won't be any difference in the reconstructed signal at all.

As an engineer, those extra samples are never wasted, and as a listener, I could care less about the "wasted space." Drive space is meaningless at this point. When I buy a piece of audio, give me the highest quality available and I will make my own decisions regarding the frequency I want to listen to it at. The reasons behind not giving the user the highest quality possible all sound like a misguided excuse to preve…

Ok... but the point stands that the signal will be the same if you low-pass filter it no matter how many samples you take. You might as well distribute the file at 44.1 kHz and then interpolate to something higher yourself, if you want to play with it later.
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