And:::::
 
If plugins are already oversampling, what happens to the sound quality at higher that 44 sample rates?
 
Do the plugins work better?   Are they optimized for a particular sample rate? 
 
Will any of the designers tell us?

so, ...I'm wondering what's the bottom line?  is 44/24 or 48/24 as good (maybe 'good' isn't the word...) as 88.2/24 or 96/24?  I think I read all of this thread and the articles associated with it.  I watched the video with the dude and the goatie who seemed to know a lot about this.  Maybe if we want to get closer to sounding like tape go with "lesser values"?  Frankly, I can't say I know.  I've listened to stuff I did years ago cut at 44/24 and stuff I'm doing now cut at 96/24 and sometimes I actually think back then was every bit as good.  I wouldn't mind hearing anyone's succinct response.

mettelus

MarioD
So my question would be why record at a higher kHz if you can't tell the difference?

I think the bottom line for this is not the capture or final medium but the processing that is done in the middle. Some VSTs can/do take advantage of higher sampling rates which can be audible. Even with that being fact, the "how audible" continues to fuel debates. The end result really boils down to personal preference.

Yes.
Depending on what you're doing, a lot of stuff sounds great recorded/ mixed 44.1/  24.
44.1/ 16 even.
 
But I can hear a little more air/ depth to some plugins @ 96/24 (this depends a bit on the program material, a reverb tail during a rest in the music, for example) so, even though I have to make adjustments to keep this hot rod from leaving the road, I'm making my new recordings/ mixes 96/24 for the time being.

Just sayin....
 
I have some commercial FLAKS at 96K and 192K... I can NOT hear the difference between the two but the comparison to 96K and the same CD.... I can hear the difference... to me, instruments are more defined and placement in stereo field more accurate.
 
But, I do record everything at 24/44 with SONAR.  But if I thought my stuff might be marketable for FLAKs. I would record at the higher sample rate as well.

MarioD
Thanx Drewfx1 for your response.

I mostly use Amplitude 3 for my guitars and the look-a-head Waves CDA drum processor.  That may be why I didn't hear any difference but I am going to try again.

Go to AmpliTube's preferences and you can choose oversampling. I found no audible advantage to running it at 96 kHz.
 
What some people have a hard time wrapping their head around is why material that's recorded at 96 kHz doesn't have to play back at 96 kHz. That's because the improvement is in the audio range. When you convert down to 44.1 kHz, it has no trouble reproducing the audio range so any "goodness" is preserved. I don't recall anyone not noticing a difference in the audio example I posted way back when of then z3ta+ (set specifically NOT to oversample, that way I could compare) recorded at 44.1 and recorded at 96 then converted back down to 44.1. It was pretty obvious the highs were cleaner and more accurate when the instrument was recorded at 96.

Well, despite what anyone says, I started recording at 96kHz - because the highs and smoothness of the recording and other things I can only relate to when I hear it - sounds better. And anyway, my Hardware synthesizer only goes up to 20kHz, so here I am going to assume since I do not use them yet, that any soft synths will also probably only go so high in frequency. Therefore to me the debate about intermodulation noise, distortions, or beats from high frequencies recording which is not done anyway - 20kHz is the limit on my hardware synth, I would think that any such synths or sampled rom or ram was done by professionals that decided that it was better to limit the frequency range in the first place, and I do not think that I have to debate that with anyone else.
 
It should sound the same at a lower sample rate recording, but I found out as for me, it does not.
I don't that mixing down to a lower sample rate at the end is going to leave out that openness or extra space I hear in the high frequencies at all when all done. And since there are digital downloads that can be played on WMP at 48kHz and 24-bit, even if it were on a CD, then recording at a higher sample rate than what mixed down to - leaves it better in the end result.
 
And then there is the electronics of the units, the way the samples were created and the whole entire line of production of the sounds used to make the music whereas to me professionals determined this sound production in the first place, and then the managers came along and decided to sell it for whatever price and determined if any productions values were changed or not, and decided on the price they could sell it at. Then there is the ethics and other business considerations also in the end.
If all of us had plenty of money, I sure all people would have better more expensive equipment in the end and the studio to have it all in - for production of sounds.
 

Another relevant article, on oversampling interpolating DACs:
 
http://www.analog.com/sta...s/tutorials/MT-017.pdf
 

Anderton
 
I don't recall anyone not noticing a difference in the audio example I posted way back when of then z3ta+ (set specifically NOT to oversample, that way I could compare) recorded at 44.1 and recorded at 96 then converted back down to 44.1. It was pretty obvious the highs were cleaner and more accurate when the instrument was recorded at 96.

 
This gets curiouser and curiouser.
 
Does this mean Zeta (easier to say and write than Zee three Tee A plus) code is different for 96K?
 
Or does it mean Sonar code is different when recording at 96, and/or for converting?
 
Or the sound drivers are involved with how Sonar records/renders/converts?
 
I have read your speculations and findings and the great articles by/about Lavry, and historically by David Moulton,  and the other posts here, and it is clear We "Know Nothing!, Nothing!" (Hogan's Heros)
 
Is it only Zeta that is better at 96? Could this be because Zeta is more recent code than DP, and  Rapture?   Can the CW dudes help with us understanding this? Do they mostly run their computers at 96?
 
Lavry sorta suggests 88 would be the best choice. Especially for down sampling to 44. And that 96 may introduce more distortion. Could it be the sound of Zeta at 96 is introducing some pleasing harmonic distoriton?
 
Have you tested at 88?
===============
 
And, since you seem to be indicating that you are recording new projects at 96, are you converting pre-96 projects?  If so, tips about doing this?
 
 
All hail THE CRAIG!
 

YouDontHasToCallMeJohnson

Anderton
 
I don't recall anyone not noticing a difference in the audio example I posted way back when of then z3ta+ (set specifically NOT to oversample, that way I could compare) recorded at 44.1 and recorded at 96 then converted back down to 44.1. It was pretty obvious the highs were cleaner and more accurate when the instrument was recorded at 96.

 
This gets curiouser and curiouser.
 
Does this mean Zeta (easier to say and write than Zee three Tee A plus) code is different for 96K?
 
Or does it mean Sonar code is different when recording at 96, and/or for converting?

 
It means that under at least some conditions Z3ta+ can generate frequencies > 24kHz (i.e. 48kHz/2), which causes aliasing (imaging) distortion because it is > 1/2 the sample rate. If you run the same code at 96kHz, only frequencies > 48kHz (= 1/2 * 96kHz) would cause this type of distortion.
 

Lavry sorta suggests 88 would be the best choice. Especially for down sampling to 44. And that 96 may introduce more distortion.

My recollection is that Lavry was talking about non-oversampling converters being less accurate at higher clock rates (due to analog components accuracy in measuring declining at higher rates), and balancing this against the difficulty of creating steep purely analog filters. This is a converter design issue and the game changes when we are talking about oversampling converters. And once the signal is digital, the analog limitations are irrelevant.
 
 
Unfortunately it is more complicated than might be ideal, but it may help to divide things between the converters (ADC/DAC) and DSP processing.
 
For converters, higher sampling rate = higher frequencies can be present in the signal, and that's pretty much it. So there is no benefit to increasing the sampling rate > twice the highest frequency needed (plus an appropriate margin for error).
 
But any type of DSP that creates frequencies > 1/2 the sampling rate that the processing is done at will create (aliasing/imaging) distortion, so increasing the sampling rate can improve the quality. To complicate it more, note that this doesn't apply to every type of DSP, but only certain types. Ideally, the programmers would take care of this kind of thing behind the scenes (and often they do), but there are cases where they haven't.

I don't and never have.  Since 90% or so of my recording involves sample libraries recorded in 24-bit, 44.1 and synthesizers, I've never felt the need to increase file size a lot for little, if any, sonic gain. 
 
Remember, the overall impact of a recording is based on many factors:
1.  Quality of the composition
2.  Quality of the performance and/or sequencing depth
3.  Quality of the musical instruments
4.  Quality and placement of the microphone(s) if recording vocals or acoustic instruments
5.  Quality of the AD converters (I think this is more important than bumping up the sample rate to 96)
6.  Quality of the mix
7.  Quality of the post-processing/mastering
8.  Quality and attentiveness of the listener (no control over this, but it is a factor)
 
Some people say they can hear the difference between 44.1 and 96.  Perhaps, in some cases there is more "air" as the listener might be sensing more smoothness in the higher harmonics.  A Grammy-winning colleague I know says he doesn't think 96khz is worth using for recording, but good or archiving purposes. 
 
Jerry
www.jerrygerber.com/symphony9.htm