Monitoring with Reaper and Focusrite 2i2

S

Scooterj

New member
I have just learned what the monitoring modes do in my Reaper DAW. What I do not understand is what the monitoring with the Focusrite 2i2 gives me.

Normally I have my instruments going into the 2i2 interface, amp sim, and DAW and listen to them and and other audio tracks on my PC via studio monitors hooked up to the back of the Focusrite.

I am not used to using headphones, but will want to when I start recording my acoustic guitar with a LDC mic.

As I understand it, if I leave the Focusrite monitoring OFF, but have my headphones plugged into the Focusrite, I will hear playback from my DAW and my instrument input but AFTER the signal has gone into the PC and my DAW and then back out again possibly causing latency.

If I have Focusrite monitoring ON then I will hear my input instrument directly without any latency, but can still listen to the already recorded tracks in my DAW at the same time

Do I have this correct? Is it recommended to use direct monitoring on the Focusrite? Does it differ for guitar vs vocals?

Also I just realized that when I plug headphones into the 2i2, the studio monitors continue to sound and are not turned off and turning down the PC volume turns done both the monitor speaker volume and the headphone volume. Hmmmm, clearly I don't understand something here.

Thanks for helping me understand best practices. :-)
 
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Using the internal monitoring simply lets you hear your mic input without delay, but still allows you to hear the previously recorded tracks. Depending on the speed of your computer and any plugins you are using, there can be a delay in the processing. Some people can be quite sensitive to this timing.

Of course the downside is that you aren't getting any of the advantages of a plugin, such as reverb for vocals, or an amp simulator.

Latency will depend on several things, starting with your sample rate and buffer. The minimum latency one way can be calculated by dividing the buffer size by the sample rate x1000 to get the milliseconds of latency. 256 buffer/44100 sample rate gives 5.8ms latency in and you'll get the same coming back to your headphones for a total of 11.6ms. Then you have to add in any processing latency of your computer DAW and plugins. One of the advantages of high sample rates is that you can cut latency if your computer can process the audio quickly enough. 128/96000 is 1.3ms.

With hardware monitoring you have essentially zero delay.

As for the volume issue, I can't help with that. I use a Tascam which has both a headphone and monitor volume controls. You might be able to turn off the monitor feed via the control software.
 
Using the internal monitoring simply lets you hear your mic input without delay, but still allows you to hear the previously recorded tracks. Depending on the speed of your computer and any plugins you are using, there can be a delay in the processing. Some people can be quite sensitive to this timing.

Of course the downside is that you aren't getting any of the advantages of a plugin, such as reverb for vocals, or an amp simulator.

Latency will depend on several things, starting with your sample rate and buffer. The minimum latency one way can be calculated by dividing the buffer size by the sample rate x1000 to get the milliseconds of latency. 256 buffer/44100 sample rate gives 5.8ms latency in and you'll get the same coming back to your headphones for a total of 11.6ms. Then you have to add in any processing latency of your computer DAW and plugins. One of the advantages of high sample rates is that you can cut latency if your computer can process the audio quickly enough. 128/96000 is 1.3ms.

With hardware monitoring you have essentially zero delay.

As for the volume issue, I can't help with that. I use a Tascam which has both a headphone and monitor volume controls. You might be able to turn off the monitor feed via the control software.
Thanks so much for all those thoughts. Very helpful.

As for the volume in the monitor speaker issue, well duh!, the big fat monitor knob controls the sound coming out of the speakers and that turned all the way down does not affect the sound coming out of the headphone jack. So obvious that I could not even see it! I had a good laugh at myself.

Thanks again for your answers! Happy New Year to you.
 
My experience is 18i8.
You got it, separate knobs.
I never had a problem with latency. Happy Recording!
 
You will get delay if going through your computer, the question is "how much?" This is where you will get into ASIO drivers, as stated, plugins, etc. Direct monitoring is great if you use outboard equipment and your signal is already "wet" or as required other than final mixing. Me, I have very little outboard gear, 90+% of all my effects come from my plugins. To record with the plugin, as it does influence the performance, you will go through the Interface, computer, DAW, Plugin, Interface, headphones. That is a lot of processing.

However, if you start with just monitoring through the DAW, get your ASIO and drivers set, you can figure out how much delay you will have. Set your buffers as low as they will go until the computer starts to be taxed. I use 44.1 (CD quality) and 128 block. You can adjust as you figure it out. Then determine how much delay you can tolerate. After that, when you add in plugins, it will help you troubleshoot delay issues since you have a good baseline.

Once you start recording, have your drums. clicks, etc. you will the delay is very tolerable and has little impact on your performance. But it may take some getting use to. As most of us here will attest, we don't think much about it any longer. Kind of strange how it just becomes second nature.

Experiment, see what works. While digital gives us a lot of power and flexibility, there is a learning curve with it as well. But IMO the benefits more than offset the downsides. But I think you will adjust pretty quick.
 
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