Latency Comparison: MAC vs PC!

The actual latency of MIDI messages and soft synth response should be pretty consistent. The big difference when playing soft synths will be outbound audio latency. What's you buffer size in SONAR, and what does it report for outbound latency under Preferences > Audio > Driver Setting? I don't know Logic, but I assume it has a similar configuration menu somewhere that will give you this info for comparison.

That's a huge amount of round-trip latency (and thus the lag you feel).
Reduce the ASIO buffer size to 64-samples... and your round-trip latency will drop significantly.
 
It's possible to achieve sub 5ms total round-trip latency under Windows (64-sample ASIO buffer size/44.1k).
The audio interface driver's hidden safety-buffer is the X-factor.
If your audio interface uses a large hidden safety-buffer, it'll have high round-trip latency.
If your audio interface uses a small hidden safety-buffer, it'll have low round-trip latency.
Your particular unit offers decent round-trip latency (not particularly high... not super low).
 

You should be able to get a 10ms round trip latency or maybe better than that. To an extent it depends on how many tracks you are input echoing in the DAW and the cpu load plugins are causing (and some plugins create quite substantial latency fhemselves).

It will also mean making sure the PC is optimally configured for real time audio. Things like the various C sleep states (controlled in the BIOS), cpu core parking and things like drivers hogging too much Windows/hardware time (a different kind of latency on the PCI bus itself caused by delayed procedure calls) can all cause dropouts or crackles if latency is too low. Wifi adaptors are a major source of trouble unless disabled in Window's device manager while you're using the DAW software.

I can get a 32 sample buffer at 44.1KHz on the PC in my sig, so long as I don't load up the project too heavily. 64 samples is more stable though. That's a 5 or 6ms round trip latency. I can also get the same on my MacBook Pro which is quite a bit less powerful and needed no tweaking of anything to get latency that low. Apple's core audio handles things better than Windows, but a well configured PC can equal a Mac. It just requires some fine tuning.

charlyg
BTW from what I know, that effective latency at the top only regardsWDM. Your ASIO setting is too high. If you contact Focusrite, they may send you the new driver(it was in beta when I got it and may still be). It gives you 3 choices Recording  Balanced Mixing with the ability to change the sample rate. If it was me, and it was.......I'd get it. The old focusrite driver, while stable, was a kludge to configure correctly. Fortunately we usually have leeway and more than one setting will work well.
 

I have the latest driver (1.8) from Focusrite.
What settings are you currently using with your Focusrite soundcard?
Also, my Playback Timing Master is set to the Focusrite's 'SP/DIF L' but I don't use any other gear besides my audio interface, so it's currently set to 'Internal' as the sync source. Is this fine?

thechurchboi
Buffer Size: 5.0 msec

Input: 11.4 msec, 501 samples
Output: 16.4 msec, 722 samples
Total Roundtrip: 27.7 msec, 1223 samples

 
That's quite a bit of bus and A/D/A conversion latency. Even if you had no buffering at all, your round-trip would be 27.7ms - 2 x 5ms = 17.7ms, which would be a little bothersome with input-monitored instruments, and just tolerable with soft synths that are only subject outbound latency.
 
If the latency is a lot better on your Mac with comparable buffer sizes, it would seem that the USB system and interface drivers are working better there.  SONAR is at the mercy of you Windows hardware performance.
 
 
 
 
 
 
 
 
 

How is this a bug?

Why the heck aren't hosts moving these posts? Have they stopped monitoring these forums? Are they even alive?
There are loads of posts around like this as well, it's not like this is an isolated thing... But this is a forum for recipes to reproduce issues and get them validated.
 
Nevermind I gave in ages ago..

Well it wore off on me.Looks like this finally got moved. I think fresh blood is called for myself. Anyways moving on...

While I CAN run at an ASIO Buffer Size of 64, I generally run at 128, and 48K sample rate and 24 bit record depth, and I am right around 9.7 ms to 11.3, depending on which of my audio interfaces I am using.  These numbers are for tracking/recording, by the way.  When mixing/mastering, I jack up the ASIO Buffer Size to either 1024 or 2048.  I then switch back and forth between low for tracking and high for mixing, and I never have performance/latency issues.
 
Hope that helps, 
 
Bob Bone