One concept that is fairly easy to explain is smoothing filters. After a signal has been converted from digital to analog, it's a series of stairsteps. The smoothing filter smooths out the stairsteps to create a continuous waveform, but more importantly, re-create the original waveform.
 
Here's an analogy. A baseball player hits a fly ball to center field and you're in right field. Plot the height of the ball every 10 ms. This produces a series of points that correspond to the height. If you connect these points with a line, you will describe the ball's trajectory. Now, you might think "so if the points are closer together, the arc will be described more accurately." And that's true, BUT like a baseball audio doesn't move in a series of straight lines from point to point, and a smoothing filter recreates a smoothed curve based on the points. If you compare that smoothed curve to the ball's actual trajectory, you'll find the correlation is for all practical purposes identical.
 
The smoothing fllter is an important part of the "sound" of digital audio. Many people think the reason why DSD sounds more "analog" (whatever that means, LOL) is because the sampling rate is so high - a minimum of 2.8 MHz, with 5.6 MHz also being common - you can basically filter out the clock by twisting a couple wires together and hanging them across the output. Well maybe not quite, but you get the idea.
 
Those who think sample rates need to be much higher would make the argument in the baseball analogy of "Well, what happens if the ball hits a fly in between the two points where it's measured? Wouldn't that knock it off course somewhat?" The counter-argument would be that even at 44.1 kHz, the distance between the two points would be sufficiently close to take that into account.

mettelus
Already this thread is rehashing epic threads of old so I checked the OP again to be sure, which echoes the thread title of greater than 96kHz.

 
Yes! I think we've already covered 96 kHz pretty well. In addition to what you said, many plug-ins simply won't work at sample rates like 192 kHz.
 
If there was a demonstrable advantage to higher sample rates, I'm sure the industry would follow along and accommodate this change, especially as technology improves. I think what we need to be wary of are solutions in search of problems. As I demonstrated in a previous thread, there ARE problems that can occur when recording at 44.1 kHz, and a simple fix is to record at 96 kHz (and remember, I was talking only about recording, not playback). I'm not convinced a problem with recording at 96 kHz has been found yet, but as always, I have an open mind...which is why I ask these kinds of questions.

Hmmm- I guess I am still confused.  I can understand that the frequencies beyond a certain level may be insignificant or not matter at all.  To me, resolution is different than the frequency range of the recording.  Resolution to me, means how many spaces are there between samples.  And it seems that the less space, the higher the resolution.  But maybe after a certain resolution is achieved, the human brain cannot perceived any higher resolution.
 
For instance if a movies is 1000 frames per second or 2000 frames per second, can we tell the difference?  It is 1,000,000 frames per second versus 2,000,000 frames per second, can we tell.  It does seem to make sense that after a certain amount of frames (or samples) per second, the human brain cannot tell the difference.
 
 

deswind
Hmmm- I guess I am still confused.  I can understand that the frequencies beyond a certain level may be insignificant or not matter at all.  To me, resolution is different than the frequency range of the recording.  Resolution to me, means how many spaces are there between samples.  And it seems that the less space, the higher the resolution.  But maybe after a certain resolution is achieved, the human brain cannot perceived any higher resolution.

Again, there is nothing lost between the samples. Everything between the samples is already stored in the stream of samples. It has nothing to do with the brain - the sampling theorem mathematically proves that nothing is lost between the samples. There is no "guestimating" what's between the samples - the actual waveform at every point between the samples is stored in the samples. Since the entire waveform can be, and is, reconstructed from the samples, there is no space between the samples. You can't get any higher resolution because you already have everything - except for the high frequencies that are filtered out at the beginning of the process.
 
The problem is that it is simply not remotely intuitive how the part between the samples is stored, so people just assume it is lost. But it isn't - it's right there in the samples you have.

drewfx1
Craig (Anderton) had a long thread a while back on this (that got similarly sidetracked like this one!) after he found that some synths/effects sounded better when run at a higher rate.
 
Basically some, but not all, Digital Signal Processing (DSP) benefits from being done at higher sampling rates. Anything that produces overtones higher than one half the sampling rate will alias and sound somewhere between "ugly" and "less smooth" at a lower rate. 
 
So if a synth or effect that would benefit from this doesn't oversample internally (as many of us would argue it should), then running Sonar at a higher rate can improve things. In my case, everything I use that would benefit from oversampling happens to already do so internally, so there's no benefit to me. 
 
I would suspect that older stuff - written when CPU power was less abundant - would be more likely to not oversample internally. 

What sample rate(s) do you use, Drew? Just out of curiosity.
 
And (if you know) does everything that comes standard with X3 oversample internally? That's most of what I use, personally.

Anderton
If there was a demonstrable advantage to higher sample rates, I'm sure the industry would follow along and accommodate this change, especially as technology improves. I think what we need to be wary of are solutions in search of problems. As I demonstrated in a previous thread, there ARE problems that can occur when recording at 44.1 kHz, and a simple fix is to record at 96 kHz (and remember, I was talking only about recording, not playback). I'm not convinced a problem with recording at 96 kHz has been found yet, but as always, I have an open mind...which is why I ask these kinds of questions.

Craig, do you find that recording at 48 kHz suffers the same problems as 44.1? Is it significantly enough better, or do you feel it warrants the extra jump to 96 kHz?

drewfx1
Again, there is nothing lost between the samples. Everything between the samples is already stored in the stream of samples. It has nothing to do with the brain - the sampling theorem mathematically proves that nothing is lost between the samples. There is no "guestimating" what's between the samples - the actual waveform at every point between the samples is stored in the samples. Since the entire waveform can be, and is, reconstructed from the samples, there is no space between the samples. You can't get any higher resolution because you already have everything - except for the high frequencies that are filtered out at the beginning of the process.
 
The problem is that it is simply not remotely intuitive how the part between the samples is stored, so people just assume it is lost. But it isn't - it's right there in the samples you have.

Might some kind of diagram that points out what is what as far as sample rate and bit depth in relation to the peaks, troughs, wavelengths, etc. of a sound wave be helpful here?...
 
(Not saying that I have such a diagram, just spitballing the suggestion.)

Anderton
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."

 
In that situation you are comparing a 44.1K sampling rate to 192K and yes that certainly could sound different. (A bit unfair don't you think)  But how would a comparison go between a playback 96K sample rate and 192K then. I wonder if there is any audible difference.
 
I think the whole topic of this thread is a bit moot and uninteresting to say the least.  There are far better things one could talk about eg how to get better mixes! And that has nothing to do with sampling rate!
 
I certainly agree that the 44.1 to 96K thing is noticeable especially with some VST's as we have found out. Enough to make me re think my next setup and build an all 96K system from end to end.  I have worked at both 44.1K and 96K even for just straight audio recording  (in a variety of studios)  and it does sound better to me. It is almost a feeling rather than a sound too.
 
I have read some very interesting books by people such as Bob Katz and others and most agree that a higher sampling rate is better but we only really need to go to about 60K in order to really get the improvement.  So the closest thing to that we have today is either 88.2K or 96K.  Once you start going much higher you are getting into that very subtle hi fi territory where tons of expense will yield only a very small or even inaudible result.
 

Jeff Evans

Anderton
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."

In that situation you are comparing a 44.1K sampling rate to 192K and yes that certainly could sound different. (A bit unfair don't you think)  But how would a comparison go between a playback 96K sample rate and 192K then. I wonder if there is any audible difference.

 
I have mixed feelings about PCM vs. DSD because there are too many variables. At a listening test at AES among multiple formats, DSD sounded the "smoothest" or "most analog" to me. Although I'm not a cork-sniffer audio dude I could hear a difference. But that's the word - difference. I'm not sure that translates to "better" just because I subjectively found the sound more pleasing. Also, the technologies are so different - one bit with a multi-megahertz clock is very different from lots of bits with a kilohertz clock. 
 

I think the whole topic of this thread is a bit moot and uninteresting to say the least.  There are far better things one could talk about eg how to get better mixes! And that has nothing to do with sampling rate!

No, but the topic affects how our decisions will be made in the present in terms of recording. Pioneer is offering DJ tracks at 192 kHz. I've heard that in Japan, several companies offer music at 192 kHz...and interface manufacturers who do 192 kHz are telling stores that any interface that does less is destined to be a doorstop. There's a vocal minority, albeit a growing one, in the audio community that swears 192 kHz sounds better than 96 kHz. (I haven't seen a UFO, but if a career pilot sees one and it's picked up on radar, I'm going to ascribe at least some credibility to the pilot and the radar rather than flat out say UFOs can't exist.)
 
So, if we.re going to be expected to make masters available at 192 kHz, we need to know that now. A lot of people on these forums make a living from what they do with SONAR. Some people have already transitioned to 96/24 not because they think it sounds better, but because that's what Apple wants. When I was doing sample libraries back in the days when 44.1/16 was king, I recorded at 24-bit resolution and was very glad I did when sample libraries pretty much insisted that material be 24-bit.
 
In terms of art higher sample rates may not be that relevant but in terms of commerce, they are.

dimelives1
What sample rate(s) do you use, Drew? Just out of curiosity.
 

 
I use 44.1kHz the overwhelming majority of the time. But I have some older external digital processors that I connect digitally using Sonar's external inserts and their digital I/O only works at 44.1/48, so I'd need a reasonably compelling reason to switch things around. So far there isn't any.
 

 
And (if you know) does everything that comes standard with X3 oversample internally? That's most of what I use, personally.

Well, for one I only really pay attention to what I personally use and for two lots of things don't really benefit from it. If you look up Craig's old thread on this subject, you can find some examples that he and others found that didn't oversample internally and benefited from running Sonar at 96kHz instead of 44.1/48kHz.
 
I happen to know that lots of my stuff oversamples either because it has options for it you can set or because specified it's in the feature list. But generally if something really benefited from oversampling but didn't have it, I probably wouldn't use it in the first place - just because I had other stuff that sounded better when I auditioned it. This is of course one very good reason why the programmers should oversample whenever beneficial - otherwise those of us at 44.1/48kHz might not like the sound and thus won't buy it.