[LajostheHun]

How about is? It's just a big ass file, waste of storage.

 

[HAWX]

Hi people!

First of all, In my opinion, with todays technology, more than 16 bits and 44.1 kHz does not provide any descernable diffrence. My further understandings are only about the digital audio format, not about If "High Res" is necessary for music reproduction. You can think about the signal is produced by a computer, and decoded by other computer, no playback in real life.

What I learnt about more than CD quality, from this forum + a nice person in this forum:
1) With 24 bit, the minimum loudness step (or say loudness presicion) is increased, although we can not descern it.
2) 24 bit also allows more dynamic range, which I think unnecessary for music reproduction.


3) With higher sampling rates minimul frequency decimal we can go is less, in other words higher frequency presicion, altough standart CD quality already exceeds human hearing in reproduction.
4) Higher sampling rates also allows a file to contain more high pitched wave informations.

Ok, so the 2 and 4 are obvious ones, no need to argue about them.

About 1 and 3, please only contrubute If you have solid information. I can't say they are 100% true. But please don't tell me go watch xiph videos, or we don't need to talk about things we can't percieve with the audotory system, or any kind of side tracking, please. I have read this threads half of it and checked most of the external links given.

And my new question from thsi video;
Mark Waldrep, AIX records talks about High Res audio, and at some point ( 41:00 ) he says "192 kHz reduces the timing and delays are below the human treshold.."
5)So my question is Does high sampling rate also reduces the delays and improves the timing (not asking wheather we discern it or need it)?

Edited for the errors and clarity.

 

[gregorio]

how about 24bit vs 32bit? (32 bit 384khz)

If you've read the original post then you'd realise that however many more bits beyond 16 you go is just wasted. For the consumer distribution of music, even 16bits is more than enough.

Mark Waldrep, AIX records talks about High Res audio, and at some point ( 41:00 ) he says "192 kHz reduces the timing and delays are below the human treshold.."
5)So my question is Does high sampling rate also reduces the delays and improves the timing (not asking wheather we discern it or need it)?

Mark Waldrep does NOT in fact say that, he states that someone at Meridian told him that but that he doesn't agree!

5. It improves the timing and delays (and everything else) above the start of the filter of lower sample rates. At 44.1 for example, the filter typically starts around 20kHz to 21kHz, so a 96kS/s sample rate improves everything between about 20kHz-21kHz and say 45kHz (or wherever the start of the filter is). A 192kS/s sample rate should in theory improve everything above the filter start of 96kS/s, say between 45kHz and about 90kHz.

G

 

[castleofargh]

Hi people!

First of all, In my opinion, with todays technology, more than 16 bits and 44.1 kHz does not provide any descernable diffrence. My further understandings are only about the digital audio format, not about If "High Res" is necessary for music reproduction. You can think about the signal is produced by a computer, and decoded by other computer, no playback in real life.

What I learnt about more than CD quality, from this forum + a nice person in this forum:
1) With 24 bit, the minimum loudness step (or say loudness presicion) is increased, although we can not descern it.
2) 24 bit also allows more dynamic range, which I think unnecessary for music reproduction.


3) With higher sampling rates minimul frequency decimal we can go is less, in other words higher frequency presicion, altough standart CD quality already exceeds human hearing in reproduction.
4) Higher sampling rates also allows a file to contain more high pitched wave informations.

Ok, so the 2 and 4 are obvious ones, no need to argue about them.

About 1 and 3, please only contrubute If you have solid information. I can't say they are 100% true. But please don't tell me go watch xiph videos, or we don't need to talk about things we can't percieve with the audotory system, or any kind of side tracking, please. I have read this threads half of it and checked most of the external links given.

And my new question from thsi video;
Mark Waldrep, AIX records talks about High Res audio, and at some point ( 41:00 ) he says "192 kHz reduces the timing and delays are below the human treshold.."
5)So my question is Does high sampling rate also reduces the delays and improves the timing (not asking wheather we discern it or need it)?

Edited for the errors and clarity.

for you questions, there might be a need to precisely locate where in the playback chain you're looking at the data.
because if increase fidelity and just cover that added fidelity with noises or filters, then what the increased bit depth or sample rate does is irrelevant.

for 1) because of the nature of waves, a tone with less bits can always be looked at as being less precise in amplitude, or as always being correct+some noise. same idea as having 2 instruments playing at the same time. you can imagine the second ruining the fidelity of the first instrument, or you can imagine 2 perfectly fine instruments together. which is closer to the way we listen to sounds. so increased bit depth in that context doesn't change the instrument but simply reduces the extra noise that comes with it .
when you change the volume level digitally, each sample has a value and you change that value so that the DAC will now read everything as a lower amplitude(quieter). it means attributing a new value to each sample and in a discrete system you have to go for the closest approximation. in 24bit you'll have an available approximation that is closer than in a 16bit sample. I guess that is what you might mean by loudness precision?
of course any extra noise or distortion added by the DAC/amp/headphone end up being much louder that -144db below signal, so in practice you can say that those benefits are most likely buried under bigger errors anyway.

for 3) a 2khz wave absolutely does now need to be recorded at 192khz to be better. the erroneous intuition that adding more points equals to more precision is mostly irrelevant because that's not how we reconstruct a signal.
well it is if you use a NOS DAC with no filter, but fidelity on those things is bad as they disregard the very math that made digital audio possible so they become aliasing generators.

for 5) timing can mean a all lot of things. correct pitch, jitter, phase, how high in frequency we can go...
if you consider more than the audible range, then you can argue that high sample rate does a lot of things obviously. but if you stick to the audible range, stuff like the filters used, oversampling or reclocking might become as relevant or more relevant than highres music. you need to clearly define what you're talking about and where in the playback chain you're looking at it. by not doing that on purpose, a lot of unicorns are sold by talented marketing guys every year.

 

[protoss]

If you've read the original post then you'd realise that however many more bits beyond 16 you go is just wasted. For the consumer distribution of music, even 16bits is more than enough.



Mark Waldrep does NOT in fact say that, he states that someone at Meridian told him that but that he doesn't agree!

5. It improves the timing and delays (and everything else) above the start of the filter of lower sample rates. At 44.1 for example, the filter typically starts around 20kHz to 21kHz, so a 96kS/s sample rate improves everything between about 20kHz-21kHz and say 45kHz (or wherever the start of the filter is). A 192kS/s sample rate should in theory improve everything above the filter start of 96kS/s, say between 45kHz and about 90kHz.

G

Strange. I always thought and still do, that 32bit will have a lower distortion rate with perfect accuracy and higher bits of information store into the file than 24bit and 16bit.

I know that humans could only hear upto 124db. But 32bit has upto 200db into its files. Even thou we can not hear that 80db. I in theory will say 32bit will have a better studio quality than 16 and 24 bit.

I will also bet just like VHS to DVD to Bluray. We will eventually get that technology to decipher the audio out of that 32bit in our headphones. And later we will never look back

 

[castleofargh]

Strange. I always thought and still do, that 32bit will have a lower distortion rate with perfect accuracy and higher bits of information store into the file than 24bit and 16bit.

I know that humans could only hear upto 124db. But 32bit has upto 200db into its files. Even thou we can not hear that 80db. I in theory will say 32bit will have a better studio quality than 16 and 24 bit.

I will also bet just like VHS to DVD to Bluray. We will eventually get that technology to decipher the audio out of that 32bit in our headphones. And later we will never look back

now take your db talk and try imagining them being db spl. how loud do you figure the band will be playing when recording? 120db spl? certainly not.
what is the noise level in the studio? let's say they have a real quiet one and it's only 20db spl. anything below is ruined by that noise.
what is the noise level and dynamic of the microphones used? here is are a few examples but you can go look at the specs of many professional microphones. http://www.gras.dk/dynamic-range
now let's have fun, what is the dynamic range of your DAC? spoiler it isn't even 24bits and probably never will be.
what is the distortion level of your DAC, amp and headphone? no please, don't cry, it's going to be ok.
how loud do you listen to the song? 120db spl? I hope for you that you don't.

24bit files are already questionable for playback purpose. 32bit files are just stupid. there is nothing to gain from having them. not anything.


about "I know that humans could only hear upto 124db" this doesn't mean what you think it means. they take a person, put him in an anechoic chamber and test the quietest sound he can notice, which is very low only because he's in that anechoic chamber. then they crank up the volume level up to the point where it's physically painful for the guy and here you go 120 of possible dynamic.
but the instantaneous dynamic range of the human hear is on average closer to 60db. that's the range we can detect between the loudest and quietest sound at the same time in the song. once the sound reaches a given level, the protection mechanism dampens the eardrum making it less sensitive, so you can listen to louder sounds, but you no longer have the ability to notice the close to 0db spl you could do in total silence.
and of course it's not like you'll be listening to music in your own anechoic chamber, so don't dream about that 124db value as if it is what you'll notice in a song. for starters you'd have to listen at 124db+whatever noise level in your room to really get 124db of fidelity in the signal. good luck with that too.
humans are extraordinary creatures, but still very human. the more is better philosophy becomes irrelevant when you apply it only to 1 element of a long chain.

the race over video still kind of make sense on a huge screen with high contrast and gamut, because the eye still has the ability to notice the improvement if the screen can produce them. same thing can't be said for 16 vs 24 bit where people fail blind tests. if we rely on such tests, we can say with high confidence that 16bit is already transparent to the human hear when listening to music.

 

[bigshot]

In the recording studio, dynamic range and signal to noise are most important with vocals, because that's where the heavy duty compression is used. A flat recording has dynamics to spare, but when you start compressing vocals to make consonants clear over backing instrumentals, noise can get pulled up along with the subtleties of phrasing. That's why the cleanest performing amps and processors in a studio are the microphone pre's and the noise gates on the vocals. 24 bit allows a great deal of latitude to pull up sound buried in the mix. The sound files themselves aren't really the problem. The problem is how clean the signal coming in from the mic is.

 

[protoss]

now take your db talk and try imagining them being db spl. how loud do you figure the band will be playing when recording? 120db spl? certainly not.
what is the noise level in the studio? let's say they have a real quiet one and it's only 20db spl. anything below is ruined by that noise.
what is the noise level and dynamic of the microphones used? here is are a few examples but you can go look at the specs of many professional microphones. http://www.gras.dk/dynamic-range
now let's have fun, what is the dynamic range of your DAC? spoiler it isn't even 24bits and probably never will be.
what is the distortion level of your DAC, amp and headphone? no please, don't cry, it's going to be ok.
how loud do you listen to the song? 120db spl? I hope for you that you don't.

24bit files are already questionable for playback purpose. 32bit files are just stupid. there is nothing to gain from having them. not anything.


about "I know that humans could only hear upto 124db" this doesn't mean what you think it means. they take a person, put him in an anechoic chamber and test the quietest sound he can notice, which is very low only because he's in that anechoic chamber. then they crank up the volume level up to the point where it's physically painful for the guy and here you go 120 of possible dynamic.
but the instantaneous dynamic range of the human hear is on average closer to 60db. that's the range we can detect between the loudest and quietest sound at the same time in the song. once the sound reaches a given level, the protection mechanism dampens the eardrum making it less sensitive, so you can listen to louder sounds, but you no longer have the ability to notice the close to 0db spl you could do in total silence.
and of course it's not like you'll be listening to music in your own anechoic chamber, so don't dream about that 124db value as if it is what you'll notice in a song. for starters you'd have to listen at 124db+whatever noise level in your room to really get 124db of fidelity in the signal. good luck with that too.
humans are extraordinary creatures, but still very human. the more is better philosophy becomes irrelevant when you apply it only to 1 element of a long chain.

the race over video still kind of make sense on a huge screen with high contrast and gamut, because the eye still has the ability to notice the improvement if the screen can produce them. same thing can't be said for 16 vs 24 bit where people fail blind tests. if we rely on such tests, we can say with high confidence that 16bit is already transparent to the human hear when listening to music.

Well said. I will let others argue with you on this one.

cheers

 

[StanD]

Well said. I will let others argue with you on this one.

cheers

Why would we argue with him? We agree with him.

 

[protoss]

Why would we argue with him? We agree with him.

Just saying. Eventually someone will come and stir something up with different facts and ideas.Who knows.

 

[TheTrace]

Alternative facts, makes the forums go 'round.

 

[gregorio]

[1] Strange. I always thought and still do, that 32bit will have a lower distortion rate with perfect accuracy and higher bits of information store into the file than 24bit and 16bit.
[2] I know that humans could only hear upto 124db.
[3] But 32bit has upto 200db into its files. [3b] Even thou we can not hear that 80db.
[4] I in theory will say 32bit will have a better studio quality than 16 and 24 bit.
[5] I will also bet just like VHS to DVD to Bluray.
[6]We will eventually get that technology to decipher the audio out of that 32bit in our headphones. And later we will never look back.

1. No, perfect accuracy is possible with any bit depth. All you get with higher bit depths is a lower digital noise floor. You did read the OP?

2. Yes and a human can go for about 2 months or so without food. A human can also run the 100m in 9.58 secs but who would buy a running machine which could only operate at that speed and who would choose not to eat for two months just for the entertainment value? The limits of the human body are extraordinary but you seem to be forgetting that music products are entertainment, so we're not talking about the utter limits that the human body can endure, we're talking about the limits of what's comfortable, which are way, way lower! For this reason you'd be hard pushed to find any commercial recording with a dynamic range greater than about 60dB and most have several times less.

3. Theoretically 32bit has up to about 192dB. I say "theoretically" because in practice this number is utterly ridiculous. If it were possible to create and play back such a file, so you could actually hear what was stored in the lowest of those 32bits, then the peak level would produce roughly the same sound pressure level as the initial blast wave of an atomic bomb. It's pretty safe to say that being instantly vaporized is significantly beyond the comfort level of consumers!
3b. Let's look at it the other way and say that peak level of our theoretical 32bit audio file is within the upper limits of the comfort zone, so the question then becomes; what is 192dB quieter than that? The answer is; that level is significantly lower than the level produced by two hydrogen atoms colliding, a level which probably couldn't even move a single air molecule, let alone move the countless billions of air molecules necessary to actually propagate a sound wave. So, you cannot hear it because there is no sound wave to hear!

4. How do you arrive at that theory? You think we're setting-off and trying to record nuclear weapons in our studios or trying to record a couple of sub-atomic particles colliding?

5. Almost but only if you equate them accurately: VHS would be equivalent to an old cassette, DVD to a vinyl record and the latest 4k UHD, HDR Bluray would be somewhat less than CD (16/44.1).

6. That's indeed true ... I imagine it would be rather tricky to "look back" (or anywhere else), once your head has been vaporized!! Even if it were possible, headphones which could kill you instantly wouldn't exactly be the ideal Christmas gift. It would make an interesting bundle though, along with a TV capable of outputting so much UV light it instantly fries you to a crisp.

Crispy fried, vapourized audiophiles is quite an attractive proposition in some ways and if it were possible, I'd be sorely tempted to give some of them exactly what they're asking for! ... I can only assume that you've been completely suckered by marketing and/or audiophile myths/anecdotes and have no idea what you're actually suggesting?!

Eventually someone will come and stir something up with different facts and ideas.Who knows.

1. Science knows! Sure, some audiophiles (or more typically, those who sell equipment to audiophiles), come up with ridiculous ideas all the time and even sometimes present those ridiculous ideas as "different facts" but they're not really facts, they're just marketing bulls***. The basic facts of digital audio were invented 90 years ago, proven mathematically 70 years ago and no-one since has dis-proven them. In fact, doing so would invalidate the basis of all digital information theory and therefore demonstrate that no computer based technology works. This, along with most of the other facts in this post, were discussed in the OP, are you sure you've read it?

Hopefully, this is all starting to sound as laughably ridiculous to you as it does to us?

G

 

[HAWX]

Thanks @gregorio for answering no 5, what you say is logical. My English is not native thank you for clairty.

@castleofargh ..... I think I have already said "My further understandings are only about the digital audio format, not about If "High Res" is necessary for music reproduction. You can think about the signal is produced by a computer, and decoded by other computer, NO PLAYBACK IN REAL LIFE." regarding my understandings. You're still saying "spesify where you are looking at the playback chain" and limitations of dacs amps reconstruction and bla bla, I'm not even talking about the playback of the file.

I will repeat my understandings with asking questions and examples about 1 and 3.
1) For a given computer generated tone, (not captured with electronic devices) how presice the tone will be assigned to approximate dBFS without any dither comparing, 24 vs 16 bits? Let's say normally tone should be assigning about -50.5555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 dBFS. (100 digits after the decimal)
I think the 24 bit assignation will have more digit after the decimal point thus leading to more precise assignation where the precision of a number is the total number of significant decimal (or other) digits.

3) For a given computer generated continuous tone which has the frequency of 10,000.55555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 Hz (200 digits after the decimal), how presice this tone frequency value assignation will be, comparing sample rates of 44.1kHz and 192 kHz, without any dither. (Although I don't think the dithering will make a diffrence)
Again, I think the higher the sampling rate will be more presice to the original computer generated waveform.

I'm not asking this to get infromation about the playback fidelty of sampled file to human ears through dac/amp/headphones. My question was about math.

 

[gregorio]

1) For a given computer generated tone, (not captured with electronic devices) how presice the tone will be assigned to approximate dBFS without any dither comparing, 24 vs 16 bits?
3) For a given computer generated continuous tone which has the frequency of 10,000.55555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 Hz (200 digits after the decimal), how presice this tone frequency value assignation will be, comparing sample rates of 44.1kHz and 192 kHz, without any dither.

1. It doesn't matter whether it's 24bit, 16bit or just 2bit, the precision is infinite. IE. -50.555r dB (infinite number of decimal places). This is a basic, proven tenet of digital audio, as explained in the OP. Obviously though, it's going to be far more difficult to discern that precision above the digital noise floor with only 2bits rather than say 16bits. Secondly, in practise we're not going to see such accuracy out of a DAC, just the analogue section is not capable of anywhere near infinite accuracy. The actual level of accuracy varies from DAC to DAC.

3. Again, the accuracy is infinite at both 44.1 and 192. This is true of any frequency up to slightly below the Nyquist Point (half the sample rate). This is also covered by the proven basic tenet of digital audio (the Nyquist-Shannon Theorem). So, your example of 10,000.5r Hz can be perfectly encoded (with infinite precision) with any sample rate exceeding about 22kS/s. Also again, we won't get this infinite precision out of a DAC in practise.

G

 

[HAWX]

Dude, sorry but I think It is just impossible for any kind of finite file will be carriying infinite information. I have already looked the original post and many of the external links given in this thread. I need proof, or It would be great to make test in some kind of program wich showes me what I'm talking about.

You are right about 44.1 kHz and 16 bits are more than enough for digital audio playback, which I also agree. But that's not I'm asking for. Thanks.