Need help getting latency down and crackling to go away!

Hi Everyone,
 
This is my first post on the forum.  I finally reached my breaking point with getting my rig setup that I thought id give this a try.  I hope that somebody can help figure out what is wrong.  I am not a computer guy and am not confidant that I have the basics down. 
 
My audio interface is a Presonus Audiobox 1818VSL (via USB 2.0).
 
My computer is a Dell XPS 8700 (about 2 yr old) that is running Windows 8.1.  The processor is a 4th generation Intel I-4 790 processor @ 3.6 Ghz.  Ram is 16 GB.  I have a dedicated Samaung 500 gb solid state hard drive just for music files. The rest of the computer operates on the HD that came with the computer. 
Also running a Creative Sound Blaster Recon3D PCIe OEM Sound Card. 
 
All drivers and updates are current for the computer, Sonar, and the Presonus Audiobox. everything. 
 
I should also point out that my computer and Audiobox both are powered by a UPS power supply.  All instrument cables are quality and in good condition.
 
I went through just about all of these steps recommended here to optimize my machine: 
 
These are my two problems:
 
1.) The best latency I can get without crackling is about a 18 ms round trip.  (from what ive read, that is pretty bad, even with computers 10 years ago!)  That is at a sampling rate of 44.1 kHz and a block size of 256.  I can set the block size down to 128 and it will throw a single crackle every 10 seconds or so, just enough to not want to use it. 
 
2.) I drove myself bananas today trying to figure out why I was experiencing very fine crackling, almost a crispy fuzz, when recording rhythm guitar.  It just gets distorted if strumming the guitar in a punchy fashion.  Whenever my level goes above about -24db, I get the super fine distortion.  If I dial back the gain on the Audiobox so the level doesn't exceed -24db ish, the distortion goes away.  The lousy thing about that is then my recording quality stinks.   I am sure that its not the amp or a blown speaker. I tried two different microphones and even plugging my guitar straight into the Audiobox, and they all had the same result.
 
I don't know where to start.  Something is very wrong here. 
 
Please help!
 
Best,
 
Ryan
 
 

Hi Chuck,
Latest driver is installed for the Audiobox.  Below are screenshots of what I see.  I think that I have the ASIO and selection of the AudioBox setup correctly.  I only use 3 inputs max, so I don't have all the channel drivers on (thinking it would help with speed)
Do you see anything here that is out of line?
 
 


 

Tom B
In your Driver Settings, "Audio Driver Bit Depth",  try unchecking "64-bit Double Precision Engine".  Let us know if that helps.

Thanks Tom!
 
I gave that a shot and it didn't seem to help.
 
Here is a new detail to share:
 
The tracks crackle up in sonar during playback but don't after rendering down.  (I rendered a track down to post and share but it didn't happen, so no sense in posting).  Does that give any clues?

Many, maybe most, effects can add some latency and some will add quite a lot because they need time to operate in. Processors that use "look ahead" such as some compressors generally suffer from the added latency issue, as can convolution reverbs. Waves tell you how much latency their plugins cause and it's sometimes a lot, unfortunately most manufacturers don't provide that information.

It doesn't matter whether the effect(s) that add latency are on the selected track on not, if they are on an active track in a project they will add latency no matter which track they're on. The usual work-round is to avoid latency adding effects until the recording side of things has finished - e.g. use a low-latency effect as a "stand in" while tracking then replace is with the more resource demanding one when mixing and latency can be raised without it causing problems.

Another factor that can cause dropouts and crackling is what's called dpc or PCI bus latency. This has nothing to do with audio latency directly, but is the time Windows needs to meet the requirements of other applications, services and device drivers the system is running. And any multi-tasking system like Windoes has a lot of stuff going on in the "background". For "normal" computer use, such as word processing, web browsing or even games it doesn't matter if Windows stops paying attention to the application you're using for 50ms, you'll almost certainly never notice it. But if your audio buffer is shorter than that 50ms you'll get a dropout.

If you've not tried it already, download the free latency.mon application, run it and see what it reports. It might be that something as simple as disabling wifi drivers in device manager while using Sonar will let you get much lower ASIO latency.

FInally, another factor that affects latency is if there are any "hidden buffers" in the interface itself. Sonar can only tell you what the audio driver says the latency is. Theinterface itself may have a built-in latency on top of that reported by the driver, so the figure you are seeong is wrong. Not a lot you can do about that unfortunately other than get an interface that does report this "hidden latency".

By the sounds of it though, you've already found a big part of the problem - some plugion(s) you are using are adding latency. They will also be using some cpu resouirces, and if the system is marginal for the set buffer size at idle the additonal load when running might be enough to just trip things over the edge.

Good Day, 
 
If your talking about for mixing already recorded material, and adding FX etc.., crank up latency to maximum (longest time), you dont need low latency for mixing. If your trying to record/add new tracks, into existing recorded material loaded with FX and other Virtual Instruments, and if for any reasons you cant monitor your input directly from your audio interface's mixer or your instrument/amplifier (e.g., adding new Sonar VI tracks, or needing to monitor through Sonar software FX while recording), then your going to have to freeze (or commit to bounce) tracks with VI's and/or lots of FX in order to be able to lower latency to acceptable levels without delay in signal or midi key response time, and/or not get the snap, crackle and popping. (also dont eat Rice Krispies while recording, ).
e.g.,
If Im recording vocals into a project already loaded with VI's and FX, I monitor my vocals directly from my audio interface, (not through Sonar, i.e., input echo off), so I can leave the latency high, (not have to freeze anything). If I need to monitor with reverb or delay FX, I use those built into audio interface, (but dont record FX).
Otherwise, its probably not a bad idea if your happy with VI and FX laden tracks, to freeze them anyway.
 
Hope this helps a bit
 
Cheers      

If it's using asio drivers then it take it buffers setting from Presonus. I use Rme and that's how it works.