Hey all!
 
I've been using Sonar X3 for about a month now, and so far I've had a few problems but I've been able to smooth them out with a little bit of tweaking and tinkering, but one problem remains constant, and it's really getting on my nerves. I like to use the Leslie simulator from Amplitube, because it's an incredible piece of software, that I use on everything from grand pianos, to vocals, to guitars. The problem is, whenever I use that Leslie sim the CPU usage skyrockets, and it's sound is constantly lagging, clipping, and generally sounding pretty awful. Same thing with Arturia MiniMoog V, especially with polyphony. I have a Focusrite 2i2 that I use as a preamp to track guitars and vocals, and I've read that it's not only a preamp, but it can take on the job of a soundcard as well. I have no idea how to set that as the main soundcard, and I'm guessing once I do that it'll get rid of those lag/latency issues I've been experiencing with the Leslie sim and the rest of Amplitube, as well as the MiniMoog and a few other VSTs. Any help is greatly appreciated, thank you!
 
Specs:
Windows 8
Intel Core i7 3517U @ 1.9 Ghz (3 Ghz Turbo Boost)
nVidia GeForce GT 620M 1GB 
10 GB RAM
500 GB HDD
Realtek HD Audio and Intel Display Audio
 

I HAVE the Arturia V Collection (just bought the whole thing on sale for $199 - 50% off it's regular cost), which includes the Mini V. On my system, I have no latency issues with that synth, or any others that I use, many of which are from Native Instruments.
 
If your audio interface drivers are installed in Windows, you will need to tell Sonar to use it as the audio device.
 
Before you do that, I am going to suggest you set your audio interface settings to: Sample Rate = either 44.1 k or 48 k.  ASIO Buffer Size of 128.  If you make changes to your audio interface settings, it is generally recommended that you power cycle the interface, to get a nice clean fresh set of buffers and such with the new settings.
 
In Sonar, go to Edit>Preferences>File>Audio Data, and set the Record Bit-Depth to 24 bits, then click Apply.  This will globally set the audio depth for recording new clips to 24 bits.
 
Now, go to Edit>Preferences>Audio>Playback and Recording, and make sure you set the Driver Mode to ASIO.  Click Apply, and Sonar should automatically set a couple of other things for you in another area.  
 
Go to Preferences>Audio>Driver Settings, and Sonar should have populated your Playback and Record Timing Master values to your audio interface.  While here, set the Sample Rate to the exact same as you have set for your audio interface, and again click Apply.  Take note of the reported latency values, and when you post back, include them.
 
OK, so now, check Preferences>Audio>Devices, and make sure all of the ones you want to use from your audio interface are checked, for inputs and outputs.
 
To recap, I generally suggest folks start with the following settings, which seem to me to be a reasonable place to begin (These are also what I run with):
 
Audio interface: Sample Rate 44.1 k or 48 k, ASIO Buffer Size of 128.  (I use 48 k)
 
Sonar: Sample Rate matching what is set for audio interface, Record Bit-Depth of 24-bits, and check your reported Total Roundtrip Latency - hopefully it will be at or a little below 10 milliseconds.
 
Give the above a try and please post back.
 
Bob Bone
 
 

Thanks for the reply robert_e_bone! I did what you said last night, and here are the results I've gotten thus far. The Focusrite doesnt really have a control panel, only a small window that lets you set your buffer size in milliseconds. So I just set it to 10 ms, which is about 128 in buffer size (I'm guessing). I did everything else you said to the letter, and the results I got were interesting. In a mix with only one instrument on the track, the Leslie sim on Amplitube and Arturia, which gave me the most problems to begin with, were seemingly taken care of. CPU usage increased a bit when the Leslie sim was turned on, and went up a bit more when the speed on the cabinet was set to fast. So I then I tried running it on a more busy mix, with a drum track (Addictive Drums), rhythm guitar track, and a bass track (NI Scarbee Pre-Bass). Before I mic'd my cabinet to play the lead track, I played back the recorded stuff to adjust tempos and whatnot. The main problem was, the MIDI track would only play (about) 8 out every 10 MIDI notes at times. Sometimes it would play all of them, sometimes it would skip a note or two and play the next one. So I set the sample rate from 48k to 44.1k, and that sort of fixed the problem, because it would still skip a note or two, but not as much as before. So then I decided I wanted a Leslie sound on the lead guitar part. I mic'd up the cabinet, fired up Amplitube, and turned on the Leslie. The sound was still lagging, popping, and clipping, but to a smaller degree. The problems really began when I increased the speed on the Leslie, the clipping and popping became unbearable. So to a certain degree it was fixed, but the track I wanted to play on wasn't such a busy track, so I don't really understand why it was clipping so much. Thanks a ton for your help!

Alright - try setting the sample rate back to 48 k, and instead try increasing the Recording and Playback I/O Buffer sizes to 512, instead of their default 250 or 256, or whatever it defaults to.
 
So, if your Total Roundtrip Latency is at or a little less than 10 milliseconds, and you are still having midi notes drop out, increasing those buffers may help, 
 
Bob Bone
 

So where would I go about doing that? In Edit>Preferences>File>Audio Data? 

Edit>Preferences>Audio>Sync and Caching
 
Bob Bone
 

 

The problem is, whenever I use that Leslie sim the CPU usage skyrockets, and it's sound is constantly lagging, clipping, and generally sounding pretty awful. Same thing with Arturia MiniMoog V, especially with polyphony.

If you keep getting drop outs with your round trip latency set at "less than 10ms", then its either one of these 4 thnigs:
1. Your sound card buffers are too low for that Sonar project
2. Your PC cannot handle the current load for that current Sonar Project
3. Some effects are not meant for REAL Time processing!!
3. All sound card drivers are different. Some will give you great performance and some will not. You have to learn what and what not your card can do.
 
As far as your clipping, that is pretty easy to fix,. just lower your signal by checking the gain stage.
 
CJ
 

I concur with your assessment, CJ
 
Bob Bone
 

I concur with you concurring :)