QuadCore

John T
There is a fairly cogent argument for using higher than 44.1 for recording, but it's got nothing to do with capturing higher frequencies. It's that it's easier to build the cutoff filter when there's a bit more bandwidth to play with. Not I just say easier, which you can read as "cheaper". There are many filters that work perfectly well in 44.1.
 

 
Yes.  Cheaper converters may sound better at higher sample rates because the requirements of the anti-aliasing filters are not as critical at higher sample rates for the reason you point out.  A high quality converter may sound just as good at 44.1 because the anti-aliasing filter is doing it's job well, even when the cut off frequency is very close to the highest frequencies we want to capture.  So i opted for a good converters (Lavry blue series).

 
Don't confuse purely analog antialiasing filters with oversampled converters. Oversampled converters use a gentle analog filter combined with a very steep digital one to combine to produce the required results. Unlike analog filters, you can easily make for extremely steep digital filters within certain mathematical limits and trade offs involving processing requirements, latency, ringing, etc., along with the filter shape. In the analog world you have all that same stuff along with physics to contend with and that makes it real hard. Which is a big reason for oversampling - you can get excellent, consistent and reliable performance very cheaply. "Excellent" is of course a dangerous word in audio circles where many people are either too young, or too old and forgetful to remember the days when pretty poor performance was commonplace, and thus they have completely lost all perspective.
 
As I noted earlier, in addition to ultrasonics question, the only question with commonly used oversampling converters is whether you can hear the filter artifacts at all. If you can't hear them, all this talk of gentle filters being better is irrelevant.
 

 
I am intrigued though by the idea that higher sample rate oversampling in plugins makes for better sound across the spectrum.  Do X3 Pro Channel effects oversample?

My impression is that Sonar's bakers are more than smart enough to oversample in the cases where they should.
 
For the most part, I would think that anyone writing DSP today with any degree of competence will know where using a higher rate (or greater bit depth) is beneficial and take appropriate action. Several years ago, it's more likely that people might have left it out to save CPU, but there isn't much need for that now.

mike_mccue

brconflict
Something else to consider with higher sampling rates, I didn't see in this thread (or overlooked): Accurate clocking. The higher the sample rates, the more accurate your clocking needs to be, since jitter and component issues may become more of a problem. This is why you see these oven-baked atomic clocks offered up from Antelope, for example.
 
If you're sampling at 192Khz with a low-quality clock, you may stand to benefit from halving the sampling rate and improving transient materials, for example. It could be a noticeable improvement.

I thought the conventional wisdom was the opposite of this. Back in the day overclocking was a strategy to mask clocking inaccuracy. As clocks became more stable and accurate, overclocking was deprecated in a lot of hi end audio playback gear, to the point where people were uninstalling the overclocking pieces/parts from their CD players and installing new improved replacement clocks as upgrades became popular. 
 
Is oversampling similar? Will oversampling make the results of clock that drifts by some small percentage seem like smaller absolute timing errors or is the effect of a clock spread evenly across the duration of the recording?

 
Conventional wisdom today is that the clocking system in today's devices are better than using a master clock. That said, if you have a studio full of digital gear, you'll need a master clock to keep it all in sync anyway. It's just that the jitter response from today's stuff is pretty good, and an external clock will induce bit of jitter which is the nature of PLL circuits. In the past, the jitter from syncing to a master clock would be lesser than a device's internal clock.
 

IfItMovesFunkIt
 
But seriously I decided that CD quality is good enough for me.... I'm a 56 year old bass player that records the ocassional song in a bedroom and so the technically inferior CD specification is more than adequate for my purposes

 
Ditto - I'm a 68 year old musician (bass/guitar/wind controller) and I record in an untreated room.  I tried higher kHz but I didn't notice any major difference, possibly do to my age even though I have very good hearing. Thus CD quality is fine with me.
 
Also no one has ever said this would sound better if it was recorded at a higher kHz.
 
So my question would be why record at a higher kHz if you can't tell the difference?

There's an expression in french that say:"there's no smoke without a fire".  So, 3 years ago I changed all my recording system.  Before, I was at 16/44.1   With my new gear and after alot of reading and questionning, I decided to go at 24/88.2  I don't know if the sample rate by itself made a difference.  I just feel that the hole thing is better now and I tried to put any small chance in my side.
I took many projet that I had recorded with my old gear and then I could hear the difference with my new stuff.  But, I still listening what I had record with my old gear in my new converters, monitors and another version of SONAR.
Very interesting conversation here and it happened before but, I can't see the end of that...

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.

I may have to revisit this but my question is although the higher sample rates may improve what is done in the middle can you hear the difference in a CD?  CD quality is what I'm recording for so if there is an audible difference to some maybe I should revisit this as my prior test was listening to CD quality.
 
I am not trying to be a jerk I would just like to know if a higher sample rate would improve my CDs.
 

Read an article last year about a dude recording tree sounds.
 
He had determined trees creak at about 70k.
 
Good reason to use way high rates.

I'll give my take for what it's worth....
 
I've tried testing this out at all different sampling rates over the years. I've done the blind tests, I've tried great interfaces, cheap interfaces, line recording, mic recording....I hear differences but I can't tell you what they are. The differences that I can sort of explain:
 
Cheap interfaces: They sound better at higher sampling rates but don't get too excited because they aren't sounding better at all when compared to something good. At higher rates they sound as good as a really good interface at 44.1 in MY experience.
 
Good interfaces sound the same to me at all sample rates. However, I HAVE (like Craig) noticed differences with guitar sims at higher rates too. BUT, to my ears it's not enough of a difference for me to go that high. As a matter of fact, I prefer the warmer, grittier sound of a Realtek stock card for guitar sims. Why you may ask? They aren't as clean yet they aren't dirty in a bad way. The filters make dirty rock sounds and lightly dirty sounds more pleasing to my ears. I know...that sounds ridiculous. But it's funny, all the guitar tunes I've done on my little internet machines running Realteks via ASIO I always know without thinking. I can't even explain what the heck it is...I just prefer some of the sounds I've created using guitar sims as well as some of my guitar pre-amps through those cheesy stock cards.
 
When I have noticed fairly noticeable differences in higher rates, it has been in a mic'd situation. Example...
 
I recorded an orchestra one time at 24/48. They tried a song they weren't quite ready to do and screwed up after about a minute. They wanted to try it one more time. Just for my head, I saved the other project, opened up a new one and set it for 24/96. When I got the projects home and worked on mixing them, I noticed the higher sample rate just sounded different. I can't even tell you it was better....it was just "different." The only way I can explain it would be "more brilliant sounding". Maybe a cleaner high end? I don't even know.
 
So in my opinion, I think it matters what you record as well. Meaning, a hard rock, metal band or techno artist is not going to benefit from 24/96 (unless a synth used might) but I sincerely believe an orchestra just might. The reason for my theory? The more sonic something is, the more I feel it will not benefit from higher sample rates. I have no idea if I'm right in my belief and to be honest, I haven't given it much thought. The reason for this?
 
Like John T, my main concern is getting all my sounds to be right before I print anything. Digital is going to give you exactly what you put into it.....even at 16/44. Good sound in, good sound comes out regardless of bit/sample rates. I feel a good interface is important...but sample rates....I'll never go higher than 48 on my end. No particular reason other than the majority of my clients use 24/48....the people that taught me used it and it's just become a part of the way I work. :)
 
-Danny

BlixYZ
Everyone should read this.
http://www.trustmeimascientist.com/2013/02/04/the-science-of-sample-rates-when-higher-is-better-and-when-it-isnt/

great read, i'd mark this helpful but someone beat me to it :)

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?

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.

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

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.