24/96khz

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s_t wrote:
S0lo wrote:But isn't all the range from 20Khz and above are completely inaudible. The human ear simply can't notice it. All 96khz is doing is capturing an extra range from 22khz to 32Khz which is all practically silence to the ear.
IMO, benefits of higher sample rates doesn't come solely from increased frequency range. There is also increased precision for DSP and summing process in audible range. For example, with 44.1K sample rate you can have less than 6 samples available to form one cycle of 8000 Hz sine wave. With 96K you have 12 samples.
But according to Nyquist theorem (if I understand it well), you only need those 6 samples to recreate the original waveform EXACTLY as it was, no loss. Given offcours that you apply the brickwall filtering required at Nyquist limit which is already done by converters at both inputs and outputs of any audio interface

http://en.wikipedia.org/wiki/Nyquist%E2 ... ng_theorem

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S0lo wrote:
s_t wrote:
S0lo wrote:But isn't all the range from 20Khz and above are completely inaudible. The human ear simply can't notice it. All 96khz is doing is capturing an extra range from 22khz to 32Khz which is all practically silence to the ear.
IMO, benefits of higher sample rates doesn't come solely from increased frequency range. There is also increased precision for DSP and summing process in audible range. For example, with 44.1K sample rate you can have less than 6 samples available to form one cycle of 8000 Hz sine wave. With 96K you have 12 samples.
But according to Nyquist theorem (if I understand it well), you only need those 6 samples to recreate the original waveform EXACTLY as it was, no loss.
Exactly. Nonsense argument.

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S0lo wrote:
AudioGuy720 wrote:But isn't all the range from 20Khz and above are completely inaudible. The human ear simply can't notice it. All 96khz is doing is capturing an extra range from 22khz to 32Khz which is all practically silence to the ear.
Inaudible on its own for sure, however I think that if that above 20khz segment is harmonics of some high pitched instruments you can hear, then its presence or absence will change the perception of how those instruments sound. That is correct isn't it?

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eidenk wrote:
S0lo wrote:
AudioGuy720 wrote:But isn't all the range from 20Khz and above are completely inaudible. The human ear simply can't notice it. All 96khz is doing is capturing an extra range from 22khz to 32Khz which is all practically silence to the ear.
Inaudible on its own for sure, however I think that if that above 20khz segment is harmonics of some high pitched instruments you can hear, then its presence or absence will change the perception of how those instruments sound. That is correct isn't it?
There's some dispute over this but pretty much: no it is not correct.

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We need scientifical proof! :hihi:

But seriously I think it is correct and there is another example, which however does not prove this one, in which audio content which is not perceptible by itself, affects the quality/perception of the sound and that is the noise that is added when dithering down something to a lower bit depth.

Isn't that why mp3 encoding, which shaves off that ultrasonic content with it's low pass filter, isn't too good with maintaining fidelity of some highs?

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all placebo.

The higher number just makes it sound better.... :hihi:
Barry
If a billion people believe a stupid thing it is still a stupid thing

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jupiter8 wrote:
S0lo wrote:
s_t wrote:
S0lo wrote:But isn't all the range from 20Khz and above are completely inaudible. The human ear simply can't notice it. All 96khz is doing is capturing an extra range from 22khz to 32Khz which is all practically silence to the ear.
IMO, benefits of higher sample rates doesn't come solely from increased frequency range. There is also increased precision for DSP and summing process in audible range. For example, with 44.1K sample rate you can have less than 6 samples available to form one cycle of 8000 Hz sine wave. With 96K you have 12 samples.
But according to Nyquist theorem (if I understand it well), you only need those 6 samples to recreate the original waveform EXACTLY as it was, no loss.
Exactly. Nonsense argument.
Yes, in D/A converter the waveform is recreated. But consider that you do ALL your complex/multitude DSP and mixing with the low precision "6 sample" example instead of higher precision (unless DSP uses oversampling of course).
At D/A stage you can't recover the benefits of using higher precision in the first place.

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eidenk wrote:We need scientifical proof!
Sampling theory is better than that - it's mathematical proof. ;)

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hakey wrote:
eidenk wrote:We need scientifical proof!
Sampling theory is better than that - it's mathematical proof. ;)
A theory is just that, its not proof :wink:

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UltraJv wrote:
hakey wrote:
eidenk wrote:We need scientifical proof!
Sampling theory is better than that - it's mathematical proof. ;)
A theory is just that, its not proof :wink:
I think he meant to say theorem. ;)

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UltraJv wrote:
hakey wrote:
eidenk wrote:We need scientifical proof!
Sampling theory is better than that - it's mathematical proof. ;)
A theory is just that, its not proof :wink:
My bad - Nyquist-Shannon sampling theorem (which is a mathematical proof).

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UltraJv wrote:
hakey wrote:
eidenk wrote:We need scientifical proof!
Sampling theory is better than that - it's mathematical proof. ;)
A theory is just that, its not proof :wink:
A theorem is not a theory, and theorems are indeed proven. Its clear that all mentions of 'sampling theory' are actually references to Nyquist's sampling theorem.
An idiot on Set Theory:
"In some cases there is an object called red that contains everything that is red. In much the same way a pot is a plate."

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eidenk wrote:
Vectorman wrote:I think at this point the vast majority of the plugins I use are already internally oversampled.
Well perhaps but this means that two samplerate conversions per plugin are taking place in your audio processing chain and this isn't too good IMO as each of those conversions use extra CPU cycles in addition to degradate the sound quality because they are lossy.
You've only to listen to, for example, NI Massive, DCAM and DIVA at their highest quality settings vs. their lowest to hear that whatever losses occur due to sample rate conversion are outweighed by the sonic benefits. Using nothing but old, non-oversampled plugins and using modern plugs (those that even have the option) at their lowest-quality settings and then jacking up the global session rate just to try and avoid some extra sample rate conversions isn't something I can see as a viable solution.
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Vectorman wrote:
eidenk wrote:
Vectorman wrote:I think at this point the vast majority of the plugins I use are already internally oversampled.
Well perhaps but this means that two samplerate conversions per plugin are taking place in your audio processing chain and this isn't too good IMO as each of those conversions use extra CPU cycles in addition to degradate the sound quality because they are lossy.
You've only to listen to, for example, NI Massive, DCAM and DIVA at their highest quality settings vs. their lowest to hear that whatever losses occur due to sample rate conversion are outweighed by the sonic benefits. Using nothing but old, non-oversampled plugins and using modern plugs (those that even have the option) at their lowest-quality settings and then jacking up the global session rate just to try and avoid some extra sample rate conversions isn't something I can see as a viable solution.
I was thinking more about effects than instruments which is why I mentioned two samplerate conversions per plugin, and I sure know about DIVA high quality's settings (who doesn't?), and there is probably no doubt that running your DAW at 44.1 and using oversampled effects yelds better sound than using those effects without oversampling. But then why not running the DAW at 96khz with less or no need to use oversampling for those effects and thus save CPU and avoid as much as possible any kind of possible degradation of the sound due to multiple resamplings all over the processing chains? There is nothing wrong with that reasoning or is there?

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eidenk wrote: But then why not running the DAW at 96khz with less or no need to use oversampling for those effects and thus save CPU and avoid as much as possible any kind of possible degradation of the sound due to multiple resamplings all over the processing chains? There is nothing wrong with that reasoning or is there?
Is that a joke question?

Seriously?

Yes at 96khz oversampling is mostly not needed but:

You do realize that if you use plug at 96khz with no oversampling it will use same amount of cpu as it is using when you are driving it at 48khz + 2x oversampling ?!?!! So there isn't any saving of CPU or anything like that..

A bit offtopic but my humble prediction is that from now on and after three years 96khz will be pretty much non issue with future CPU in computer. i8 or even i9 will not have any problem with that. Just my guess though..but then i am guessing that everyone will run it at 96khz (or 88khz)..

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