Milton
What is puzzling to me is that at lower sample rates (44/48) I get "lower" buffer settings but "higher" latency times. Conversely, at higher sample rates (96/192) I get "higher" buffer settings and "lower" latency. Haven't got my brain around this yet. Can anyone clarify this confusion for me? p.s., This is why 96KHz was the "sweet spot" for my system as I got useable enough low latency and some "wiggle" room to increase my buffer if needed.

Higher sample rates simply fill the buffer faster. At a given buffer size, it takes twice as long to fill it at 44.1KHz than it does at 88.2KHz. Since how long it takes to fill the buffer is the primary factor in determining latency, higher sample rates reduce latency given the same buffer size.
 
However, because your CPU is working twice as hard, you might not be able to realize the potential latency reduction if your computer can't keep up and you're forced to use larger buffers.

Another way of wording this is by showing how latency is calculated.
 
At 44100 samples per second:
A 1024 (1K) sample buffer corresponds to 23.21 msec of latency
 
At 88200 samples per second:
The very same buffer size corresponds to 11.61 msec of latency
 
So you can see that doubling the sample rate halved the latency for the same buffer size.

A second is a second.

Noel Borthwick [Cakewalk]
Another way of wording this is by showing how latency is calculated.
 
At 44100 samples per second:
A 1024 (1K) sample buffer corresponds to 23.21 msec of latency
 
At 88200 samples per second:
The very same buffer size corresponds to 11.61 msec of latency
 
So you can see that doubling the sample rate halved the latency for the same buffer size.

It also means that to produce the same output the CPU has to do twice the work,  or the same amount of work in half the time, which brings up the issues of number of plugs, soft synths, etc. but for what gain?  To work at a frequency response that no human can hear.

AT

 
Ear training can indeed allow one to hear small details if they are audible.
 
But the "only with super gear in special room" is nonsense. Different artifacts caused by different things (including listening systems and the listening environment) just do not conveniently line up with each other that way to get masked. 
 
Can't hear an artifact in a given situation (assuming it's real and audible by humans)? Try turning up the volume a little. Or playing a "worst case" signal. Or moving closer to the speaker to reduce the role of the room or environmental noise. Suddenly it's audible with any gear.
 
The stuff that can only ever be heard with special equipment under any conditions always turns out to be imaginary. 
"

Drew, I'm not sure what you're arguing here.  If you are sticking to the sample rate part of the thread, I agree with you.  As stated, I use 44.1 since I can't hear any difference worth the bother.  And I agree if you can only hear a difference in an anechoic chamber wearing a tin-foil hat it probably doesn't have any real-world use, esp. since it likely doesn't exist.  But you are too categoric in your dismissal of gear, room and training as far as the art of music, and as I argued, the psychology, too.  Some days in the studio I hear different things as related to mixing before I touch a knob.  Maybe I need that tinfoil hat? ;-)
 
@
 

I do agree that training can most definitely make a difference.
 
But what I am arguing is that the notion that certain things can ONLY ever be heard with super special gear under perfect conditions is false. Gear and room can make a difference in specific conditions but not every condition. 
 
If you turn up the volume, you can hear details you couldn't hear before, regardless of the gear. If you move closer to the speaker, the room has less of an effect. If you understand how a given artifact occurs, then it's generally not hard to pick a signal that makes it much more audible. Any masking done by a piece of gear doesn't automatically scale and move and jump around to continue to mask an artifact when you start making changes like these.
 
 
Mostly I'm arguing against a notion in audiophool circles that the reason no one else can hear the imaginary nonsense that they hear is because you need to spend $2000 on "more revealing cables" or whatever. The way you worded things just stepped a little too close to that and I don't want people to think that the reason they can't hear something is because they haven't spent enough money on gear - they might not be hearing it because it isn't there. And if it is there, then there are usually ways of testing for it or isolating it to make it more easily audible.

Whoa! I'm lost now. A bit too technical for me. What I concluded from recording at several sample rates, was that at lower sample rates I got lower buffers (good for adding intensive processing) but high latency (not good for low latency soft synth input recording and large sample libraries). And conversely at higher sample rates, I got higher buffers but lower latency. This puzzled me as I assumed a lower sample rate would also give me a lower latency. So this is what I'm really confused about. Like I've already mentioned, I found that I got a good balance (sweet spot) at 96KHz on my system in terms of a reasonable and workable enough low latency and also a low buffer. Note: my Lynx Aurora Thunderbolt converter's highest buffer setting is 1024. Here's what I achieved; (96KHz = 256 buffer and 3.9 milliseconds latency) (192KHz = 512 buffer and 6.2 milliseconds latency) (44.1 = 128 buffer and 11.2 milliseconds latency). So is this normal or is my computer system not correctly optimized or some other issue that needs to be attended to?  So at 96 I still had 2 buffers left higher to work with, 192 1 buffer, and 44.1 3 buffers to go up to if needed. My system sings at 96KHz. Best advice is from the quote from John T. "Basically, I decided to stop worrying about it and get on with making records."

Thanks bitflipper "However, because your CPU is working twice as hard, you might not be able to realize the potential latency reduction if your computer can't keep up and you're forced to use larger buffers". This is exactly why I decided NOT to record at 192KHz as I had only one more higher buffer to use and felt it was too risky not knowing ahead of time how much intensive processing (or adding more soft synths and large sample instruments) I was going to use. Your explanation of "Higher sample rates simply fill the buffer faster. At a given buffer size, it takes twice as long to fill it at 44.1KHz than it does at 88.2KHz. Since how long it takes to fill the buffer is the primary factor in determining latency, higher sample rates reduce latency given the same buffer size", helped me understand what I was perplexed about. Thanks.
 

Anderton

cclarry
So, no, it's not necessary...this debate has gone on for years...
If you go by the science...NO ...not required....

I don't think there's much debate that theoretically, higher sample rates aren't necessary. But there are so many wild cards. For example, one reason why some people might hear the difference between 192 kHz audio and 44.1 CDs in tests may have nothing to do with the sample rate, but instead be due to the 192 kHz signal being played from a hard drive, which has less jitter than something played back from an optical drive.
 
I couldn't hear a significant difference between 44.1 and 96 kHz until I started doing lots of ITB work with amp sims, virtual instruments, and dynamics processors. But it had nothing to do with human hearing, it was all about technological limitations that caused foldover distortion in the audible range at lower sampling rates.
 
Filtering has always been a consideration too, although filtering technology has improved dramatically since the CD was introduced. So the reason I'm curious is because some people swear they hear a difference with 192 compared to 96. In the case of the Be6 speakers, the response is only up to 40 kHz so in theory, 96 kHz and 192 kHz should have the high frequency components reproduced equally well.
 
The whole debate reminds me of cables. I was in a studio in Chicago and there was a vehement argument going on about whether cables made a difference. It was the old "it's just wire, you moron" vs. "but I can hear a difference." I finally stepped in and asked what the outputs and inputs feeding the cable were...and yes, with a tube amp and a long cable, capacitance can affect pickup tone...but with a high-output synth going into a mixer, "it's just wire."
 
It would be nice to determine once and for all whether people can hear a difference with double-blind testing that goes beyond Meyer-Moran, but it would be even nicer to find out why people hear a difference if there is a technological reason. I'm not ruling out sample rates per se, but I tend to think it might be something that's a byproduct of sample rates.
 
And I STILL think DSD sounds better than CDs...but in the immortal words of Herman Cain, "I don't have facts to back me up."

THIS is definitely true...if you are using Amp Sims, higher sample rates will do wonders...
just as Craig said...as it is the way the signal is process by the software that makes
the difference...and coming directly off the hard drive before Mastering or processing
will matter also....not that the 44.1 isn't sufficient...it is a limitation of the "reprocessing" 
Software that has the impact on the signal being processed...

This is not due to the "recording" sample rate...it's due to the processing of the recorded 
signal BY THE SOFTWARE, and you WILL hear the difference...

This all is due to the way the signal is processed by the MEDIUM (the Software)..and NOT due to the sample
itself...which will be completely fine at 44.1 Khz...

So, in this aspect, higher sample rates WILL matter...and I agree 100%

As an aside, am I right in assuming that if your intention is to slow audio down (time stretch), the end result will sound better for audio recorded at 96kHz than audio recorded at 48kHz? In much the same way as film recorded at a higher frame rate looks smoother when slowed down.

sharke
As an aside, am I right in assuming that if your intention is to slow audio down (time stretch), the end result will sound better for audio recorded at 96kHz than audio recorded at 48kHz? In much the same way as film recorded at a higher frame rate looks smoother when slowed down.

That's a really interesting question for which I have no answer.