Duet 2 settings to record acoustic guitar

[satellite maps]

Hi again folks,

I have been trying to figure out how to use the Maestro 2 software that comes with Duet 2 and although some stuff is obvious i'm at a lost to get the most optimal settings to get a nice pristine acoustic guitar sound.

Setting the inputs make sense to me but can someone shed some light on:

1. What to set the outputs at?
2. The Mixer, software return & mixer master (I have no idea what to set these at)

Guitar is a 1959 Harmony Cremona archtop acoustic
Mics are: SM81(about 6 inches away from the 12th fret) & Rode NT1 (off the body angled toward the f-holes)

Buffer is set at 128. I'm applying no plugins on the guitar channel strips in L9Express, preferring to get the best natural sound and add EQ and compression later.

I would love some advice from the infinitely smarter and seasoned group of veterans out there 😉

thanks in advance
Gary

 

[georgelegeriii]

Well, it's all about mic technique then. Best thing it to set up one mic, set the duets input level to where you get a few DB from clipping when you play your hardest chord, and move the mic around and find the guitars sweet spot.

As for the second mic, do the same, and once you add the second mic, be sure to check out mono compatibility by adding a gain plug-in and turning the mono setting on and off.

You can mic a guitar a number of ways, stereo, with one on the body and one down at the end of the neck, panned left and right... that is, if your player is willing to be pretty still.

Or you can go into mono, and use the second mic just for ambience if you have a good room sound (which could also be panned left and right for a kinda cool sound).

Be careful you don't get too close to the body around where the arm or elbow is: there tends to be a pretty strong harmonic there that can overwhelm your overall tone.

The duet has no eq, so all you can do is mic placement. And try using both mics in as many places as you can, one close, then switch it with the other, record a bit and listen back to see what you have. Put down a click and play the same part with each mic to get a good idea of the differences.

That's my 2 cents.

 

[Peter Ostry]

1. What to set the outputs at?
2. The Mixer, software return & mixer master (I have no idea what to set these at)

I neither know the Duet nor the Maestro software but "software return" sounds like the playback signal coming from Logic and "mixer master" could be the fader for mixing the playback with the recording signal. If this is correct, you should use both to find a good balance of playback and guitar.

Here is the standard setup:

​

For this kind of monitoring make sure that Logic does not send the recording signal out. You would hear it twice, once from the guitar and a little later from Logic. What you want is a perfect relation between direct signal and playback and you should be able to set it with the Maestro mixer.

If the playback is too loud you tend to play harder and vice versa. You may prefer a monitoring signal that sounds like the final piece, or you want to hear the guitar as good as possible and use the playback just as a pilot track. You can also pan the guitar signal to one side and the playback to the other if you like that. It needs a little experimentation to find the best way for you.

 

[satellite maps]

Thanks for this info George. Thankfully i have more understanding of mic placement then i do with the Maestro softward ;-)

What i really need to understand is what signal the output is affecting and where do master mixer and software return come in to play. Is the output signal the direct signal before D/A conversion etc...

 

[satellite maps]

1. What to set the outputs at?
2. The Mixer, software return & mixer master (I have no idea what to set these at)

I neither know the Duet nor the Maestro software but "software return" sounds like the playback signal coming from Logic and "mixer master" could be the fader for mixing the playback with the recording signal. If this is correct, you should use both to find a good balance of playback and guitar.

Here is the standard setup:

​

For this kind of monitoring make sure that Logic does not send the recording signal out. You would hear it twice, once from the guitar and a little later from Logic. What you want is a perfect relation between direct signal and playback and you should be able to set it with the Maestro mixer.

If the playback is too loud you tend to play harder and vice versa. You may prefer a monitoring signal that sounds like the final piece, or you want to hear the guitar as good as possible and use the playback just as a pilot track. You can also pan the guitar signal to one side and the playback to the other if you like that. It needs a little experimentation to find the best way for you.

Peter thank you so much for this very in depth response. Just a couple of questions below.

For this kind of monitoring make sure that Logic does not send the recording signal out How exactly do i do this. Is this where the Maestro mixer comes into play?

Also can you explain where in the chain i'm hearing the unprocessed direct signal before A/D conversion. I think my problem might be that i'm getting all recorded signal and none of the natural direct signal. Sorry Peter, I'm a bit confused by all of this.

 

[Peter Ostry]

For this kind of monitoring make sure that Logic does not send the recording signal out

How exactly do i do this.

If you don't need software monitoring (= monitoring the recording signal through Logic) you can generally switch it off. Go to Logic Pro -> Preferences -> Audio and deactivate the "Software Monitoring" checkbox. Afterwards no incoming signal leaves Logic, only the playback.

Also can you explain where in the chain i'm hearing the unprocessed direct signal before A/D conversion.

In your interface (and I think in most) the signal gets A/D converted, runs through the mixer and gets D/A converted for the outputs. I don't know the latency of the Duet but it should be close to mine (RME Fireface, about 2 ms latency from analog in to analog out). Not likely that you hear that, although some people do.

I think my problem might be that i'm getting all recorded signal and none of the natural direct signal.

Then, after you switched software monitoring off in Logic, you would get nothing of the recording signal. If this happens, your Maestro Mixer is disabled. Consult the Duet manual. In my downloaded PDF the chapter "low latency mixing" starts on page 19 and what they actually mean is low latency monitoring.

 

[satellite maps]

Thanks Peter! starting to make a bit more sense now. I will try all this next time and test it out. You've been very very helpful and I just have one last thing to ask. Wondering if you (or any other Logic 9 pro/express experts) can take a look at this screenshot of my audio settings and let me know if there's any that sticks out as wrong.

Buffer size for recording: (click on thumb)




Buffer size for mix: (click on thumb)


thanks again
Gary

 

[Peter Ostry]

The settings are ok.

Your "Recording Delay" is at 0 samples. Have you tested it and 0 is the correct value or didn't you test the delay with your interface yet?

 

[satellite maps]

The settings are ok.

Your "Recording Delay" is at 0 samples. Have you tested it and 0 is the correct value or didn't you test the delay with your interface yet?

I'm sorry Peter but I'm not really sure what this is for and what it should be set at. For that matter, I'm not sure what the following are or should be set to:

Processing threads
Process buffer range
Scrub speed
scrub response

Do any of these have any major factor on the overall sound quality.

 

[Peter Ostry]

Your "Recording Delay" is at 0 samples. Have you tested it and 0 is the correct value or didn't you test the delay with your interface yet?

I'm not really sure what this is for and what it should be set at.

This is the latency of the hardware. An audio interface tells Logic about it's own latency but this information is not always correct. If the value is wrong, Logic writes the recorded signal too late or too early. For your recording setting you should test the latency of the interface and adjust it if necessary.

1. Open a fresh Logic Project with two mono audio tracks and no plugins at all.
2. Record a very short percussive sound or draw a spike in the Sample Editor.
3. Cut the region so that the sample or spike is not too far away from the beginning of the track and make it short. Move the region exactly to bar 1.
4. Set the Sample Editor's view to show samples.
5. Connect one output of your interface to one input by a cable. Don't use internal routing, you have to use a cable.
6. Let Logic play and record the spikes to another audio track.
7. On the original track, with the pointer tool, measure the distance between the spike and the start of the region.
8. Measure the distance of the spike in the recorded region.
9. The difference in samples is the recording delay.
10. Adjust the recording delay slider in the audio preferences according to your sample difference. Positive or negative.
    
11. Repeat steps 6-10 until there is no time difference between original and recorded signal.

If you don't see a difference at the first attempt, leave the recording delay at zero.

There is an old but still useful detailled description with a slightly different measurement method on John Pitcairn's website:
http://www.opuslocus.com/logic/record_offset.php

I'm not sure what the following are or should be set to:

Processing threads
Process buffer range
Scrub speed
scrub response

Please consult the Logic Pro Manual for these parameters. They are explained from page 1274 on in the English PDF: "Core Audio Device Preferences". I found this chapter quickly by searching for the word "buffer".

 

[satellite maps]

Your "Recording Delay" is at 0 samples. Have you tested it and 0 is the correct value or didn't you test the delay with your interface yet?

I'm not really sure what this is for and what it should be set at.

This is the latency of the hardware. An audio interface tells Logic about it's own latency but this information is not always correct. If the value is wrong, Logic writes the recorded signal too late or too early. For your recording setting you should test the latency of the interface and adjust it if necessary.

1. Open a fresh Logic Project with two mono audio tracks and no plugins at all.
2. Record a very short percussive sound or draw a spike in the Sample Editor.
3. Cut the region so that the sample or spike is not too far away from the beginning of the track and make it short. Move the region exactly to bar 1.
4. Set the Sample Editor's view to show samples.
5. Connect one output of your interface to one input by a cable. Don't use internal routing, you have to use a cable.
6. Let Logic play and record the spikes to another audio track.
7. On the original track, with the pointer tool, measure the distance between the spike and the start of the region.
8. Measure the distance of the spike in the recorded region.
9. The difference in samples is the recording delay.
10. Adjust the recording delay slider in the audio preferences according to your sample difference. Positive or negative.
    
11. Repeat steps 6-10 until there is no time difference between original and recorded signal.

If you don't see a difference at the first attempt, leave the recording delay at zero.

There is an old but still useful detailled description with a slightly different measurement method on John Pitcairn's website:
http://www.opuslocus.com/logic/record_offset.php

I'm not sure what the following are or should be set to:

Processing threads
Process buffer range
Scrub speed
scrub response

Please consult the Logic Pro Manual for these parameters. They are explained from page 1274 on in the English PDF: "Core Audio Device Preferences". I found this chapter quickly by searching for the word "buffer".

Thank you Peter, I will try all this shortly. Really really appreciate all this help.