[SOLVED] Which of these equates to lower latency?

All things being equal the higher the sample rate, you will always get lower latency.
The trade off is the higher you go the more room the files take, the more load is on your CPU and your hard drive because you have to move more data in the same time.

If your system is powerful enough then it's no problem.

The question I would ask is Why? Why do you want to record at 48khz instead of 44.1khz. Do you know of some improvement of one over the other or are you preparing audio for DVD media?

Thanks for the replies, guys.
 
 

mudgel
All things being equal the higher the sample rate, you will always get lower latency.
The trade off is the higher you go the more room the files take, the more load is on your CPU and your hard drive because you have to move more data in the same time.

 
Thanks, Mike. I do know all of that already, though. My main question was really about in what way the Record Latency Manual Offset factors in to my final latency result.
 
Obviously the 2 msec @ 48k is smaller than the 3 msec @ 44.1k, but then the 48k's recordings require a higher manual offset (46 samples) as compared to that of 44.1k (2 samples) for exact round-trip synchronization.
 
My assumption would be that, since the looped recordings were coming in 46 samples late in the 48k tests, the total latency would be 2 msec + 46 samples (or about 2.96 msec). And the same for the 44.1k tests, which came in 2 samples late, I would assume to be a final figure of 3 msec + 2 samples (about 3.05 msec).
 
But I don't want to just assume... For all I know, the actual latency might simply be the total offset amounts required to get things in sync. Or, I might be on the right track up there^ but, as I've read in prior threads, SONAR's reported effective latency figures may not be entirely trustworthy.
 
This is why I wanted to pick the brains of the experienced forum users first.
 

mudgel
The question I would ask is Why? Why do you want to record at 48khz instead of 44.1khz. Do you know of some improvement of one over the other or are you preparing audio for DVD media?

 
Well the main driving force was the matter of latency... if there was enough of a gain from 44.1 to 48 to make it worth using. I realize that the difference in that regard (especially if it's only around 1 msec or less) is miniscule, and almost certainly imperceptible. But on a more academic level, just "to know", I am curious how the manual offset factors in to the final figures.
 
There is also the matter of aliasing with certain plugins when using one sampling rate over another, but that's a whole other enchilada that's been covered in other threads.
 
 

thomasabarnes
Do you know what the record latency offset is? Have you actually tested your record latency to see if you have the correct record latency offset set under "sync and Caching" in the Unified Preferences view?

 
Yes, of course. Did you read all of my original post? I gave the offset parameters needed for proper synchronization under both sampling rates.
 

dimelives1:
 
I did read all of your post, but it seemed like you didn't really understand what the Record Latency Offset is.
 
But to answer your question of which the title of this thread is: I think the 2ms is the lower latency of the two. That seems obvious to me, but maybe I'm missing something and you're looking for one of Jim Roseberry's magnificent analyses. 

No with WDM the latency is listed next to the buffer size slider under Audio Driver Settings.

thomasabarnes
But to answer your question of which the title of this thread is: I think the 2ms is the lower latency of the two. That seems obvious to me, but maybe I'm missing something and you're looking for one of Jim Roseberry's magnificent analyses. 

 
Yes, yes, obviously 2 msec is lower than 3, I'm not that hopeless What was making the grand total unclear to me was that the 2 msec tests required an offset of 46 samples to get things properly synced, while the 3 msec tests required an offset of only 2 samples. So the sampling rate that offered a lower base latency (48k) required a larger offset than the sampling rate that offered a slightly higher base frequency (44.1k). I just wasn't clear on which way the manual offset swung the results, i.e. if the fact that the 2 msec setup required a higher offset actually made it have a higher total latency than the other setup, which needed a smaller offset.
 
Hopefully that makes more sense and I don't sound too lost in the woods here.
 
 

John
No with WDM the latency is listed next to the buffer size slider under Audio Driver Settings.

 
Thanks, John. But it was my understanding that that given figure from SONAR may not be entirely trustworthy, and/or that it may not account for the other factors that the ASIO reported latency appears to (A/D conversion latency, etc.).
 
And if it is accurate, is that telling you the round-trip latency, or just one way?
 
 

mudgel
The only way to know true latency is to measure round trip latency or loop back as you've mentioned. But I don't understand your figures.

 
What seems unclear about my results, Mike? I simply gave the buffer sizes and manual offsets required to get stable, in-sync loopback recordings.
 
 

mudgel
Take note of the sample offset that Sonar introduces,

 
Like I mentioned to scook, if I'm using a driver mode other than ASIO, that figure is not present.
 
 

mudgel
Do the same recording but this time route the audio out of your device via a cable and back in via an input and record that signal.

The difference between the two tracks in samples is your round trip latency. Don't forget to factor in any sample offsets introduced by Sonar. If that's what you've done I'm sorry but I don't get that from you're figures.

 
That is precisely what I've done, my friend Only I did it without SONAR's automatic offset (since I wasn't using ASIO for these tests). I apologize for not making that more clear.
 

Perhaps in my original post I should have withheld the exposition and simply asked:
 
Does a positive manual offset figure ADD TO or SUBTRACT FROM your total latency?
 
 
That might have spared a bit of confusion here... (Unless that too is a muddled question.)

The sample rate discussion is one that I suspect will never end.  After much reading back/forth on the subject, I found a couple of articles by folks that know this stuff.  I'm sure there are a bazillion yeah-buts out there, but I describe my decision in this thread:
 
http://forum.cakewalk.com/FindPost/3121040

mudgel

dimelives1
Thanks a lot, guys. The last few posts answered what I was wondering about.
 
 
@ mudgel, BobF, & brundlefly:  Just out of curiosity, why do you prefer working at 48k instead of 44.1?
 

When I first started using 48khz it was because my efforts were heading to DVD. It kind of stuck, at one time thinking that the higher sampling rate would result in a better final sound. These day It's more of a habit. Despite all the reading I've done there are so many differing opinions in what I thought should be a scientific black and white subject.
Apparently that's not the case.
I'm not absolutely sure what my own listening experience tells me. Too many variables and too subjective. I don't think that 48 "sounds" better than 44.1 but I do "feel" that when working with acoustic/mic'd instruments including vocals that 96khz recordings end up as more "spacious" finished recordings even if they're mixed down to 44.1.
 
c'est la vie

yeah, the discussions on sample rate usually span too many pages to follow through - 189 replies when Craig raised the question last time ...
 
anyway, for about a year now we have changed our sample rate up to 96 kHz because we have found audible evidence (for us!) that it sounds better (to us!) ... totally subjective I know ... it's only one particular guitarist with high-end gear who spends aeons on finding the right sound and records directly, so 96 kHz really does him justice ... other projects I've done in the same period it would not have made a difference but I also recorded at 96kHz because the system can easily handle it and changing samples rates between projects/sessions is too much of a hassle ;-) of course, it's a bit of a disadvantage when doing long distance collaborations (due to really large files), but that we handle by using mp3 in pre-production and accept longer downloads for production files ...
 
try it, see if your system can handle it for large projects, and use what makes you feel comfortable
 
BTW, some synths/samplers may sound quite a bit different when changing between 44.1 kHz and 96 kHz (depends on synth internal and samples) ... and when going after industrial style synth sounds you better stick with 44.1, 96 khz may make sound more open and transparent and reduce some of the industrial flair ...
 
 
 

brundleflyA much quicker way of finding the correct Manual Offset is to measure the actual round-trip latency using the free CEntrance ASIO Latency Test Utility (https://centrance.com/downloads/ltu/).

 
It says it can' measure it. I took left output if 2i2  and connected to left input.. turned down volume, ran program, and it fails.
 
In strep 3 of the help it says this.....
3) Select the device driver you would like to test from the drop down menu and select the input and output
channels you have physically connected together in Step 2. and yes I chose Focusrite Scarlett  ASIO
 
There is nowhere to select input and output channels?
 
 

Thanks, all.