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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)

#311

Post by leephillips »

Mediterraneo10 wrote: post #302
5zBFyURxgY wrote: post #300

Earlier quoted context omitted.

Still very silly IMAO. Very, very few recordings are done in 24/192 because of the many implications this has on your entire studio setup. You'll need an exceptional good clock to start with, and all other equipment needs to align to that clock. Then all plugins/processing you use needs to be in the same 24/192 domain, otherwise your signal is reduced to the limit of that plugin/processing and all previous efforts ar…

You ought to have read the rest of this thread before leaving this comment. I’d be very happy with 24/96 or 24/48 if I could get recordings at that bitrate that aren't given a loudness wars treatment. Since I often can't, I have to go for 24/192 or SACD even if all that extra room is completely superfluous, just because that format was decently mastered. > Most music producers use samples... Most people interested in…

"some classical labels are now giving loudness-wars treatment"

That's depressing.

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

#312

Post by emerged »

titzer wrote: post #289
Mediterraneo10 wrote: post #241

Earlier quoted context omitted.

It has nothing to do with the 24/192 values themselves, but rather the treatment given to the recording during the mastering. Because 24/192 and SACD are "audiophile formats", the engineers who master the recordings expect them to be listened to in serious listening environments that are relatively noise-insulated so that very soft parts can be heard clearly, and there are no complaining neighbours so the loud parts…

With the higher 24 bit resolution, would it not be possible for someone to "decompress" the dynamic range with less distortion than 16 bit?

No, compression can't be undone. It destroys information. Especially with a limiter. The mastering chain will have typically done other things to the signal as well.

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

#313

Post by mixedCase »

Fifer82 wrote: post #287
z3t4 wrote: post #269

Earlier quoted context omitted.

Sadly most people have never experienced good sound. When you hear it though. But taste can be very different.

I have! I really have! My friend spends all his money on audio. I sadly just don't understand. You are correct. I was going to say that he pays infinite money for what to me is a 10% increase in enjoyment, but perhaps for you and him, your brain is better with sound, and you guys might get a 1000% improvement for your dollar. Although in any case, nothing justifies the pricetag dished out to audio enthusiasts.

An 100-150 dollar headset of the right type is hardly "infinite money" and I would call it paradigm-changing compared to what most people seem to be using, at least over here.

It's only at higher tiers that the difference might really be imperceptible to many ears.

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

#314

Post by leni536 »

kazinator wrote: post #147
Const-me wrote: post #134

If the only thing you do with your music is listen, then yes, 24/192 delivers questionable value compared to the more popular 16/44.1 or 16/48 formats. However, all musicians I know use these high-rez formats internally. The reason for that, when you apply audio effects, especially complex VST ones, these discretization artifacts noticeably decrease the result quality. Maybe, the musicians who distribute their music…

I do not believe it. 24 bits is definitely needed for processing (better yet, use floating-point). Not 192 kHz; no friggin' way. Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works.

What if I simply want to slow down the signal 2 times (without pitch correction)? Then 44.1 kHz is obviously not enough. Maybe 192 kHz is way overkill though, but I would argue that that's the point. You don't want it to become a bottleneck ever for any effects.

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

#315

Post by finnthehuman »

Blahah wrote: post #298
copperx wrote: post #257

Earlier quoted context omitted.

As Barbie would say: Audio engineering is hard! Let's play with the calculator.

Barbie has piloted commercial airliners, been on a scientific space mission, and has an M.D., so I don't think she'd be likely to say that.

https://en.wikipedia.org/wiki/Teen_Talk_Barbie

The meme "Math class is hard, let's go shopping" is only slightly apocryphal. Two of the voicebox lines were "Math class is tough" and "Want to go shopping?"

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

#316

Post by CodesInChaos »

lz400 wrote: post #214
em3rgent0rdr wrote: post #210

Earlier quoted context omitted.

> "If you download mp3/spotify" You are misconstruing Monty's argument here. He is very much against mp3...in fact he says he could tell the difference between high bitrate mp3 and 16 bit 44k wav. The real point of the video is that 16 bit 44k wav is beyond sufficient...don't need to go beyond that to 24-bit 192kHz.

As far as I know, people consistently fail to tell the difference in blind A/B tests in listening between (any decent) FLAC and a well ripped mp3 from the same source. I know I can't. I can't link to proper double blind studies but that's the general consensus in the non-BS-audiophile community as far as I know.

You can't tell the difference on the vast majority of music. But encoders occasionally mess up, producing an audible difference. In the case of MP3 this often takes the form of a pre-echo. For example encoding cymbals and hapsichords seems to be difficult, though much less of a problem than it used to be. My understanding is that such killer samples exist for good MP3 encoders, even at 320kbit/s, but it has been a couple of years since I last looked into the topic.

(Personally I downloaded some of those killer samples and couldn't tell the difference, but other people reliably tell them apart in an ABX test.)

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

#317

Post by dom0 »

hashmal wrote: post #309
akvadrako wrote: post #291

Earlier quoted context omitted.

No, I don't have it wrong. Compression means it's effectively using less of the range and the dynamics could be represented with fewer bits in an optimal encoding. All the levels may be used in track, but it's more biased towards the high end. To help you understand, imagine 1-bit audio. What would it sound like? For each frequency, you can only have a single volume, i.e. it's maximally compressed.

I can confirm, you have it wrong.

But also correct here:

> Compression means it's effectively using less of the [dynamic] range and the dynamics [range] could be represented with fewer bits in an optimal encoding.

Lower DNR can indeed be encoded with fewer bits per sample.

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

#318

Post by anentropic »

Mediterraneo10 wrote: post #241
hashmal wrote: post #236

Earlier quoted context omitted.

How does 24/192 help with dynamic range?

It has nothing to do with the 24/192 values themselves, but rather the treatment given to the recording during the mastering. Because 24/192 and SACD are "audiophile formats", the engineers who master the recordings expect them to be listened to in serious listening environments that are relatively noise-insulated so that very soft parts can be heard clearly, and there are no complaining neighbours so the loud parts…

this is roughly what the "Mastered for iTunes" badge means on iTunes downloads

the mastering engineer has to be approved and there are some minimum dynamic range standards

also 24 bit (but not 192khz) master files have to be supplied

reportedly some of the streaming services (Spotify, YouTube) are now applying some 'loudness normalisation' which will bring some of the bad loud masters into line (it won't restore their dynamic range but will make them similar loudness to other tracks)

the loudness wars were never about what's good for listeners, but rather a competition for tracks to sound louder than other people's tracks when played consecutively on radio or in your music player

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

#319

Post by dom0 »

mixedCase wrote: post #313
Fifer82 wrote: post #287

Earlier quoted context omitted.

I have! I really have! My friend spends all his money on audio. I sadly just don't understand. You are correct. I was going to say that he pays infinite money for what to me is a 10% increase in enjoyment, but perhaps for you and him, your brain is better with sound, and you guys might get a 1000% improvement for your dollar. Although in any case, nothing justifies the pricetag dished out to audio enthusiasts.

An 100-150 dollar headset of the right type is hardly "infinite money" and I would call it paradigm-changing compared to what most people seem to be using, at least over here. It's only at higher tiers that the difference might really be imperceptible to many ears.

For all its flaws even a Beyerdynamic DT 770 Pro is a very good headset that's head and shoulders above...

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

#320

Post by Applejinx »

Applejinx wrote: post #307

I work in this industry, and I have produced what is almost certainly the most high-performance dither to date, which works through noise-shaped Benford Realness calculations: http://www.airwindows.com/not-just-another-dither/ I mention this to say that I can absolutely make 16 bit 'acceptable' or 'listenable', even for audiophiles. I do that for a living. And yet… Monty is wrong. To cover the range of human listener…

I will add that the concerns of transient timing are actually a fallacy: given correct reconstruction, sampling is more than capable of producing a high-frequency transient that crosses a given point at a given moment in time that's NOT simply defined by discrete samples. Reconstruction is key here, and no special technique is required: sampling and reconstruction alone will produce this 'analog' positioning of the transient anywhere along a stretch of time.

The accuracy is limited by the combination of sample rate AND word length: any alteration of the sample's value will also shift the position of the transient in time.

But since the 'timing' issue is a factor of reconstruction, you can improve the 'timing' of transients at 44.1K by moving from 16 to 24 bit. The positioning of samples will be a tiny bit more accurate, and that means the location of the reconstructed wave will be that much more time-accurate, since it's calculated using the known sample positions as signposts.

Positioning of high frequency transients does not occur only at sample boundaries, so that alone isn't an argument for high sample rates. You can already place a transient anywhere between the sample boundaries, in any PCM digital audio system. The argument for higher sample rates is use of less annoying filters, and to some extent the better handling of borderline-supersonic frequencies. For me, the gentler filters is by far more important, and I can take or leave the 'bug killing' super-highs. I don't find 'em that musical as a rule.

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