I currently record 24 bit 44.1khz higher sampling rate?

I've been recording for a while now at 24bit 44.1khz. Can anyone actually hear a difference with higher sampling rates? It seems to me that  at higher sampling rates, your average listener in the modern world (high end topping out at 15-18khz) wouldn't be able to even hear it. Nyquist seems to suggest that 44.1khz gets you up in the near 20khz range ( although you're not getting more than a couple of sample points @ 20khz). What say you? I know I may be opening up a can of worms and this has been talked to death, but I'm seriously considering it for some projects and could use some input.

Vastman
Most of us are at 44.1 or 48/24 and that's what cakewalk recommends. I've tried others...higher reduces latency with more cpu load.

48 will lower your latency relative to 44.1 for a given buffer size, however most audio interfaces I've seen double their buffer size for 88.2 or 96 negating any gains.

It is thoroughly explained in Ethan Winer book "The Audio Expert". Together with other audio myths.   http://ethanwiner.com/book.htm 

I use 44.1/24 all of the time

[link=mailto:44.1@24]44.1@24bit[/link] all the time.  I do note many professional studios are using 96 and I may start using 96 for my next CD just because.  One of the reasons to use higher rates is for soft synth rendition, but many synths (and FX), upsample anyway these days.  Be aware many of theoretical benefits get lost in home studios (noise in the electrical lines, less than ideal rooms, etc).  But if you have plenty of CPU power and storage it probably won't hurt.

With well designed A/D conversion, 44.1kHz recording is perfectly adequate.  The problem is that many audio interfaces have mediocre A/D converters circuits that introduce ripple, phase shift and distortion at frequencies well below 20kHz.  In fact, any audible differences between 44.1kHz and 96kHz recordings can almost always be attributed to the quality of the converter design.
 
For example, a cheap converter design might have a 20kHz anti-aliasing filter that actually starts adding ripple and phase shift starting around 5kHz.  By running that same converter at 96kHz, the anti-aliasing filter can be moved up to around 40kHz, which moves up the start frequency of ripple and phase shift to about 10kHz.  With less of the musical spectrum affected, the 96kHz sample rate will sound better.  Not because 96kHz is inherently better, but because the higher sample rate lessens the negative effects of bad A/D conversion.

Thanks for the input. Thanks for not getting too technical about it as  well. I was wondering who was using higher rates and if they were actually hearing results that compelled them to continue. My interface (komplete audio6) uses cirrus logic converters, Rick I would be interested in your take on those. I most likely will try a higher rate for a smallish project (less than 20 tracks) and see if 1) my computer doesn't chug along too much and 2) if it improves the sound of synths and plugins. I'll let Ethan Winer ponder on the other stuff. I caught myself listening to some Bowie stuff yesterday (R I P) and had to remind myself of some of the limitations of the technology of the mid to late 70s. My take away was that good  performances,mixes and arrangements trump all.

Rick is correct about bad convertors, however most modern delta-sigma convertors internally upsample so the problems he describes are not as prevalent today.
If you have old converters that introduce phase issues at 5Khz, get another interface!
Your Cirrus convertors should be fine.

As far as I know,
There are pros and cons.

I'm not a specialist or scientist but have been internated in technical things about sampling rate so I searched and read articles, forums, books.
I think Aliasing, EQ, IMD and SRC, are the important things.

For aliasing,
compressor, limiter, distortion, saturation and these kinds of process usually produce upper frequency information, and this could go above nyquist frequency, which will result aliasing noise. So, technically, with higher sampling rate, there will be less aliasing noise when you are mixing or mastering because nyquist frequency is set higher.

Also, when you tweak EQ near the nyquist frequency, the EQ curve doesn't be like you'd expect. It acts differently when it gets close to the nyquist frequency. (in a bad way i think)
So if you think about these, higher sampling rate like 96khz can be better.

Some plugins internally process at doubled of multipled sampling rate (oversampling) to avoid aliasing or that. The higher sampling rate is better in some case.

However, whenever you use non-linear processes like I mentioned above (compressor, distortion, etc.), they also produce IMD.
If the original material or recorded track has
a lot of information that we can't hear, lile above 20khz, these unnecessary information produces more IMD. So this is a downside. To avoid this, you can use a filter to cut these frequency before you put non-linear process.
So with higher sampling rate with unnecessary information, you might get even worse result because of IMD. I read this thing on gearslutz forum.

For SRC, I think you eventually release your songs at 44khz, so if you produce your music at other sampling rate, you need to convert the sampling rate with SRC. You can use SRC on your DAW but the quality of SRC is different on each program. And whenever you use SRC, the sound get worse and peak changes. (only just slightly and no one might notice though) anyway, usually SRC in your DAW is not ideal, so if you don't have decent SRC, the track converted to 44khz may sound worse than one with 96khz. IFAIK SRC on ableton live is great among DAW. SRC by Saracon or izotope are considered to be best I think. You can check the quality of SRC on SRC comparison website.

Reverb or other process may act differently on different sampling rate but I just don't know it.
Intersampling peak at higher sampling rate is beyond this topic I think, too.

So the end result will be really different not because you record it on higher sampling rate but you processd it on higher sampling rate.

I currently master at 96khz with ultrasonic filter to lessen IMD but for other thing like mixing or producing. I don't know which is better because higher sampling rate seems to require more CPU.


So, for producing, mixing mastering, there will be some difference in sound. But for recording, there might be subtle differences technically but these might not be significant.
If you mix at higher sampling rate you can record them at higher sampling rate to avoid unnecessary process.

It´s important to separate music making and playback samperates. 44.1kHz/16bit is perfect for playback only. 96kHz/24bit is perfect for making music.
All non-linear processing will benefit a lot of 96k samplerate. Also audiorate modulation is beneficial. If you have the CPU power then go for it.

@Craig Anderton: You did not mention the [x2] button (Upsampling) that has been introduced in Sonar.
Do I understand correctly, that if in doubt, using this feature will solve the VSTi Synth/Sampler artefact problem, allowing users to continue recording at 44k1 (or 48k)?

I can tell you where it is: It is (on my Lexington) in the cluster of buttons to the right of the light blue time display.
Hover over the button marked [2x], bottom row, third from the left.
The tooltip states "Enable/Bypass Plug-in Upsampling"
 
But yeah, it would be good to get some guidelines on it's use from those who know.
Should I just turn it on the moment I use any VSTi?
Are there any drawbacks to taking this naive approach?

jpetersen
But yeah, it would be good to get some guidelines on it's use from those who know.
Should I just turn it on the moment I use any VSTi?
Are there any drawbacks to taking this naive approach?

The main drawback is that upsampling adds time to the rendering process.
 
There's no need to turn it on the moment you use any VSTi, compressor, limiter, etc. If the VSTi already does oversampling (most modern ones do, or give the option to do so), or doesn't generate energy above the Nyquist frequency, upsampling is not needed.
 
After upsampling was introduced, shortly thereafter SONAR introduced upsampling on playback. This makes it easy to determine whether something sounds better or not when upsampled. For example, I've found the SI-Strings sound better when upsampled. Also, some amp sims sound better when upsampled, even if they do internal oversampling...haven't figured out why, though, or whether the sound is perhaps "different" instead of "better."
 
Bottom line: Use your ears

the way i tend to view things and ask is how well the anti aliasing filtering has been implemented, in the software and hardware domain, the daw its self, the plugins, sound interface, so many variables