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88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)
Old 9 hours ago
  #1141
Gear Addict
 

Quote:
Originally Posted by Tom Barnaby View Post
Mathematics are useful only if they are a good representation of reality. It looks like that the sampling theorem is not a very good model for the recording of acoustic instruments. We will have the opportunity to discuss this later.

.
This is equivalent to saying you're not trusting the superposition principle. Just to let you know in what realm of absurdity you're lurking when you say this stuff..
Old 6 hours ago
  #1142
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monomer's Avatar
 

Quote:
Originally Posted by Tom Barnaby View Post
I think He would do this if some clients insisted to receive a 44.1/16 master,
I think that if He insisted on using 44.1/16 He would be out of a job and would not be taken seriously by clients.

Mark my words, soon all these high-profile people will switch to 32 bit conversion. They can't afford not to.
Old 6 hours ago
  #1143
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monomer's Avatar
 

Quote:
Originally Posted by Tom Barnaby View Post
Mathematics are useful only if they are a good representation of reality. It looks like that the sampling theorem is not a very good model for the recording of acoustic instruments.
In what way are acoustic instruments different from all other sound sources we know of?

Do you have an example that highlights this unsuitability of sampling theorem with regards to acoustical instruments?
Old 5 hours ago
  #1144
Gear Maniac
 

Quote:
Originally Posted by sax512 View Post
This is equivalent to saying you're not trusting the superposition principle. Just to let you know in what realm of absurdity you're lurking when you say this stuff..
There is no absurdity here. The main question consist in understanding how superposition works in real world audio. It looks like Fourier series and the Fourier transformation, although they are a good mathematical model, don't take in account all the audio phenomenons that occur in the physical space.
We have to consider phase questions with more precision.
Old 4 hours ago
  #1145
Gear Guru
 

Quote:
Originally Posted by sax512 View Post
Quote:
Originally Posted by Tom Barnaby View Post

Thanks.

This is a very good article with a lot details. I took the time to read it thoroughly and I must admit that I was wrong.
Please, forget all what I said about ultrasonics lately.
The audio band we have to consider actually extents from 20 Hz to 20 kHz.
Quote:
Originally Posted by Tom Barnaby View Post

192 kHz is the way to go if we are looking for high fidelity in digital audio.
Dude!
Anyone still in doubt this is trolling?...

Why are you people feeding it?
Old 4 hours ago
  #1146
Gear Maniac
 

Quote:
Originally Posted by mattiasnyc View Post
Anyone still in doubt this is trolling?...

Why are you people feeding it?
Do you mean that Gavin Lurssen is a troll, too ?
Please, get some information about him or go to theaudiophileman.com and read the article of june 2018 by Paul Rigby. It is called : George Harrison Vinyl Box Set, the ultimate review.
You will learn very interesting things there.

.
Old 3 hours ago
  #1147
Guess it's about time to close the thread.
Old 3 hours ago
  #1148
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monomer's Avatar
 

Quote:
Originally Posted by Tom Barnaby View Post
It is called : George Harrison Vinyl Box Set, the ultimate review.
You mean that interview where the mastering engineer states that he got the job because he previously worked for harrisons kid?
Old 3 hours ago
  #1149
Gear Maniac
 

Quote:
Originally Posted by FabienTDR View Post
Guess it's about time to close the thread.
Why do you want to close this thread ?
You are free to give your point of view about sampling frequency and explain us how you see things.

Old 3 hours ago
  #1150
Lives for gear
Quote:
Originally Posted by Tom Barnaby View Post
Why do you want to close this thread ?
You are free to give your point of view about sampling frequency and explain us how you see things.

Please, stop your trolling.
Old 2 hours ago
  #1151
That last little bit -- you can't have it.

Quote:
Originally Posted by monomer View Post
Mark my words, soon all these high-profile people will switch to 32 bit conversion. They can't afford not to.
Yes, they'll do whatever people pay them to do, whether it makes any sense or not. People are still squashing the $#it out of streaming content in the age of loudness normalization. It's stupid, and it sounds terrible, but customers want it so professionals do it.

On the other hand, most mastering engineers are already using 32-bit (or higher) processing, and have been for well over a decade. It's 32-bit floating point, simply for mathematical precision. But there are no 32-bit A/D or D/A converters -- they do not exist!

Yes, I know there are some products that output in 32 bit format. For instance there are some prosumer recorders from Sound Devices that use a gain-ranging design and output in 32-bit float. They've apparently decided that their target customers are too dumb to set gain properly. But their latest pro recorders don't do that; they output 24 bit fixed point.

You can also find some commercial "32-bit" converter chips. But look at SNR, SFDR, ENOB, or whatever spec you want to: it's all bull$#it.

Some friends of mine designed a DC measurement product with about 27 bits of resolution. It remains pretty-much unchallenged 30 years later.

David L. Rick
Old 2 hours ago
  #1152
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Monkey Man's Avatar
 

Tom, you're still hung-up on the idea that things happen in the acoustic domain that conventional-digital capture doesn't account for.

As pointed out much earlier, the mic/s convey whatever is captured as a single stream of varying voltage (2 streams if stereo), hence the math and current methodology are perfectly-sufficient and not somehow missing something or taking some esoteric BS into account.
Old 1 hour ago
  #1153
Gear Maniac
 

Quote:
Originally Posted by Monkey Man View Post
Tom, you're still hung-up on the idea that things happen in the acoustic domain that conventional-digital capture doesn't account for.

As pointed out much earlier, the mic/s convey whatever is captured as a single stream of varying voltage (2 streams if stereo), hence the math and current methodology are perfectly-sufficient and not somehow missing something or taking some esoteric BS into account.
I think you did not understand what I mean. We perfectly agree about signal capturing as a stream of varying voltage. The problem lies before capture.
The point we don't agree about it how signals superposition works in the physical space. I'm talking about acoustic instruments, as I can guess it's a bit more simple with synthesizers.
There is a lot of microreverberation phenomenons, including variations inside the envelope of a sound, that make things much much more complicated than the model described by Fourier Analysis. These phenomenons produce phase problems that require a frequency as high as 192 kHz to dealed with properly.

You may think that my theory is esoteric, but think of what Gavin Lurssen said in his interview to Paul rigby. This guy is among the best mastering engineers in the USA and what he says about digital audio should not be neglected. If you don't believe me, visit his website.

.
Old 1 hour ago
  #1154
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Monkey Man's Avatar
 

Thanks for answering, man.
Old 22 minutes ago
  #1155
Gear Guru
Quote:
Originally Posted by Tom Barnaby View Post
192 kHz is not about recording ultrasonics. We will have the opportunity to discuss this later.

Good luck if you want to achieve high fidelity with 48 kHz. It will be a tough job !

Seriously that is a sample rate for sfx so you can slow sound to half speed. I'm not going to argue "high fidelity" with you but point you to 60+ years of recorded music that sounds pretty damned good......Ultrasonics doesn't really translate into enjoyable listening for humans FWIW.....
Old 7 minutes ago
  #1156
Gear Maniac
 

Quote:
Originally Posted by Monkey Man View Post
Thanks for answering, man.
You are welcome. I will try to give more detailed explanations about my theory later, but I'm busy with something else now, and I don't want to give more information until I'm able to present a rigorous demonstration. All I want to say right now is that a higher sample rate allows a better representation of phase, which enables a much more realistic sound.
According to me, with CD we are facing a kind of blurr at the phase level that can be drastically reduced when using a higher sampling rate. I want to stress that 192 khz is not about capturing ultrasonics, as frequencies above 25 kHz are filtered out.

.
Old 3 minutes ago
  #1157
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bogosort's Avatar
Quote:
Originally Posted by Tom Barnaby View Post
There is a lot of microreverberation phenomenons, including variations inside the envelope of a sound, that make things much much more complicated than the model described by Fourier Analysis. These phenomenons produce phase problems that require a frequency as high as 192 kHz to dealed with properly.
Fourier analysis accounts for all possible phase variations: every frequency's magnitude and phase contribution to the signal is encoded in a complex number. Indeed, without the phase component, the signal's envelope is undefined.

But this is all irrelevant to an ADC, which knows nothing about Fourier transforms or complex numbers. As has been shown in this thread, the phase resolution of an ADC is independent of its sample rate. As long as those micro phase variations are being encoded by the microphone, they will be encoded by the ADC just as well at 44.1 kHz as they would at 192 kHz.

Quote:
You may think that my theory is esoteric, but think of what Gavin Lurssen said in his interview to Paul rigby. This guy is among the best mastering engineers in the USA and what he says about digital audio should not be neglected. If you don't believe me, visit his website.
One can be a great mastering engineer while also having a layman's understanding of digital theory. As joeq often says, a race car driver might know a thing or two about engines, but being a race car driver doesn't make you an expert on engine design.
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