Electric Guitar with Focusrite 8i6

[DonaldM]

So last night I was experimenting with getting some electric guitar tracks down. Haven't really tried this before. I plugged the output of the guitar to one of the front inputs on the 8i6 and made the proper adjustments on the Mix Setup to get a good signal to the PT audio track. It all works fine. However, I wanted to use SansAmp to add some nice distortion and such. The problem is the input delay. If I mute the track so I hear the guitar signal through the focusrite mix setup output, then I have no delay, but if I want to hear the part as I play it WITH the SansAmp, I have to unmute the track, but then I can't avoid the delay.

Is there a way to do this so I can hear what I'm doing WITH SansAmp (or any other one for that matter) and NOT have the delay from the signal latency?

It appears to be a catch 22.

[Darryl Ramm]

We really need more information.

Start at the " Help us help you..." Link on this page and read the stuff there and describe your setup.

What version of Pro Tools are you running, on what computer hardware and OS? (Post a Sandra report if on Windows).

The latency is directly caused by the buffering to the plugin.

What is the playback engine buffer size you are using?

What are all the other playback engine dialog panel settings?

What is the smallest buffer size value you can use and Pro Tools still reliably run?

How small you can set that is determined by what you are doing in that session, how much your computer meets/exceeds the system requirements, wether your computer is properly set up (especially with a dedicated audio/session disk) and has been properly optimized (have you done all the standard Pro Tools optimizations?).

What is the session sample rate? How many track total? Exactly what plugins and how many instances? Especially what VIs are in use?

An outboard DSP based guitar processor like an Eleven Rack or Axe FX-II let's you avoid the latency of internal plugin processing.

In what you are doing you have to be sure to always totally mute the interface's hardware monitoring even with a small buffer size as there will always be some delay between that and the Pro Tools sansamp playback. You should hopefully be able to get the buffer size small enough to be usable when playing, but don't try to also listen to the dry signals, your brain will detect the difference.

You can always split dry signal going into the interface the use a physical amp/cab to provide a wet signal for monitoring. Once you have the dry signal captured you can reamp that to your hearts content. There are much better amp sim plugins than sansamp, including Avid's Eleven plugin, which often appear for sale on DUC at bargain prices. A "better" sim plugin won't fundamentally change the latency issue, you have to get the buffer size down.

Darryl

[nst7]

Donald, I think you had an Mbox 2 Mini before this, so you're used to dealing with "direct monitoring", which is also what tbe Mix Control software does.

Direct monitoring bypasses the conversion to digital (at least for monitoring purposes) and allows you to hear the direct audio signal just as if you were using a mixer. Therefore it can not go through any plugins.

So what you want to do is avoid direct monitoring and actually monitor through the normal conversion process of the computer, but at a low buffer setting like 64, or even 32. This should still sound like little to no latency. The tradeoff is that this will use more computer power, and if you try to do this with many tracks and plugins, you may have trouble. In PT11, you will be able to this more than in previous versions.

The workaround if you are wanting to record a track this way and already have many tracks/plugs and straining the CPU is to either deactivate most of the other plugs, or bounce the other tracks down to 2 tracks and deactivate the individual tracks, while recording, and then reactivate them all when you're done recording and want to mix.

Depending on your computer, you may be able to work at the low buffer for quite a while before needing to do workarounds.

[DonaldM]

Originally Posted by nst7

Donald, I think you had an Mbox 2 Mini before this, so you're used to dealing with "direct monitoring", which is also what tbe Mix Control software does.

Direct monitoring bypasses the conversion to digital (at least for monitoring purposes) and allows you to hear the direct audio signal just as if you were using a mixer. Therefore it can not go through any plugins.

So what you want to do is avoid direct monitoring and actually monitor through the normal conversion process of the computer, but at a low buffer setting like 64, or even 32. This should still sound like little to no latency. The tradeoff is that this will use more computer power, and if you try to do this with many tracks and plugins, you may have trouble. In PT11, you will be able to this more than in previous versions.

The workaround if you are wanting to record a track this way and already have many tracks/plugs and straining the CPU is to either deactivate most of the other plugs, or bounce the other tracks down to 2 tracks and deactivate the individual tracks, while recording, and then reactivate them all when you're done recording and want to mix.

Depending on your computer, you may be able to work at the low buffer for quite a while before needing to do workarounds.

Okay, that's just what I needed to know. I did understand the the bypass to direct monitoring thing. I didn't think about the buffer size...duh! I'm on 10.3.6 at the moment, but have 11 ready to go. As of TODAY, I'll be at 16gig RAM, so should be able to handle most of what I need to do.

Thanks a bunch! You, too, Darryl. Should have thought of buffer size right off! Most of the time that's only something I need to adjust when I'm recording VI's on an instrument/midi track.