18i20 Seemingly High Latency

xb2003

xb2003

New member
My bass rig consists of using Guitar Rig 5 into a power amp into a cabinet. I had been using my PreSonus AudioBox 22VSL as the interface, but recently I stuck my 18i20 in there for 2 reasons: it is rack mountable, and I can rout one of the extra outputs to be a line out to send to a mixer.

With my AudioBox I was getting about 20ms of latency. Like 6.5ish input and output and 5.8 for processing (taken from Guitar Rig). With my Scarlett I'm getting around 30ms. 13 for i/o and 6.2 ish for processing.

I'm running 44.1k (it's not like my cabinet can reproduce 22kHz) and with the Audiobox I have the buffer at 256 Samples. On the FocusRite, its at 6ms which is ~260 Samples. This makes sense as to why the processing Latency is greater. But why is the i/o latency so much worse?

To be honest, it isn't a huge deal... I can play fine with 30ms. It just seems weird to me considering how much more expensive the interface is.

Jesse
 
xb2003 said:
My bass rig consists of using Guitar Rig 5 into a power amp into a cabinet.

Jesse
Click to expand...

Well I am a bit confused here. Are you running outs from your daw (via amp sim) to an amp and recording it with a mic? Then recording?

And you are a monster of groove if you can play with 30ms of latency. That is just a ridiculous IMO.

Please explain further. I hope to help man.
 
Can you not lower the buffer to say, 64 Samples, and lower the latency. 30ms is shocking!

I don't use the same equipment but I have mine set to 64 Samples for tracking. I bump it up to 1024 for mixing.
 
Yes, I have the most up to date drivers.

No, there is no mic involved. I have my 18i20 interface and am running into input 1 from my bass. That goes into guitar rig, processing (blah blah blah) and back to the interface where I have line out 3 going to my power amp which goes to my cabinet.

I guess I am a little confused as to how the buffer works, because it obviously does not work how I had thought. I apologize for my ignorance. I understand that digital audio requires a certain amount of latency so it can process, but what exactly is the difference between as you said, 64 and 1024 samples, besides the obvious higher latency. I didn't realize you could switch. I thought that if I recorded at x samples than that was all the higher quality I would have.

My interface has buffer size in Milliseconds rather than samples. I set it down to 1.0ms, (48 samples) in MixControl with a sample rate of 48k. My total latency is 6.6ms, of which I have 2.8 input, 1.0 Processing, and 2.8 output (according to guitar rig.) The information in my first post was with the buffer at 6.0ms (288 samples, 48k) which yielded 31.6ms total (12.8ms i/o, 6.0ms processing). I have not tried playing through this yet.

Interestingly enough, I plug in my audiobox 22VSL and set it to 256 Samples of latency at 48k and get a total of 18.6ms of latency (9.3 in, 5.3 processing, 4.0 out). This seems weird to me, as I spent a lot more money on my 18i20 (obviously it has a few more mic preamps..) I drop down the buffer on this interface to 64 samples and i get a total latency of 4.6ms.

When I said I was playing with 30ms of latency fine, well that was a very dumb statement. I was monitoring through the interface, not through Guitar Rig. After I got to thinking about it, I realized that I had forgotten to mute that input in mix control. Yes, playing with 30ms of latency is extremely difficult.

So I guess it comes down to, can someone help me understand just how the buffer works and what it does?

Jesse
 
The buffer in the interface stores some of the data generated from the analog-to-digital converters in order to always have some ready to feed into the data stream going to the CPU/DAW. It's very similar to buffering of online video. The bigger the buffer the less likely you'll get dropouts but the longer the latency. If you absolutely must use the computer as a live bass amp (which seems nuts to me, but whatever) you're going to deal with latency.

If I wanted to record without bothering housemates I'd get a hardware amp sim (like a Pod) and record two tracks, before and after the Pod. If the Pod tone wasn't right you could reamp the dry track.
 
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