Thank you … I really like your channel, too 🙂 I have an answer for you regarding the recording delay: This is for more complicated setups. If you have a simple setup with just one audio interface, the driver of the Audiointerface tells logic how much time it needs for input and a roundtrip. Things get a little more complicated if you have separated converters. In our studio we have for example on every Mac a RME Digiface USB as an audio interface without built in converter. Connected to this interface are several converters (Prisms, JCF, RME ADI 2 pro fs, Millennia and so on) THey all need little different time to process the signal. However, the audio interface is not aware of this and tells logic still just the processing time of the interface without the converters. For the different converters we have values between -220 and -575 samples. With the correct setting your recording is in this setup still perfect in place :)
This means that you would not be using multiple converters for the same recording scenario right? Or at least, this would be the less ideal situation. Any idea on how to measure the difference in time between converters? Maybe recording the same signal through both units and sample measure it in the DAW?
@@nuestudi524 Yes, you are absolutely right. Especially for stereo recordings it‘s an absolute no go to work with different converters hooked up on an audio interface. We measured this several times. Word clock ensures, that the converters are in „sync“ and don‘t drift. But wordclock has no idea about to which sample to lock. Especially for coincident microphones this is pretty bad, if on microphone is on the first converter the other one is on the other one. This leads to comb filtering effects, even if both converters are from the same company or the same model. You can measure the needed recording delay quite easily. Just bounce a track (1 measure is already enough) with test oscillator plugin set to needle impulse to 1 Hz. Adjust the tempo of this logic project to 60 or 120. When you import this file just adjust the region that the needle impulse is right on a second on a beat (this makes navigation just a little easier but is not really needed). Open a second ruler. Then record the track with the needle impulse audio file on a different track going through your AD and DA converters. After the recording you will notice that the recorded track will have some delay when you zoom into sample level. In the marquee tool when you select the space between where the needle should be and where it is, there is a window that shows how many frames and samples you are off. Apply this to the recording delay (with a minus) and make another test recording. You should notice that the needle impulses are aligned now :) -> save these compensation recording delay times per converter, so that you have it right away… Oterwise you get crazy :)
3:46 how do I see that help tag box about the plugin? if I check settings>view>general show help tags is enabled but I don't see that box when I hover over a plugin. thanks!
Your videos are always helpful. I think this is geared towards beginners, not too much detail? but, I find it VERY interesting that you have never checked and set your recording delay and that you leave it at 0? Curious to know if you have tested using the ping function inside of the I/O plugin, with a loop wired from output to an input? or tested by recording a click coming out of your speakers with a microphone right up to the tweeter? then inspect what that deviation is? I am running through an Orion32, and my recording delay is set to -16 samples.
This is very useful info. I mostly record vocals and I find the easiest way to eliminate latency all together is using the direct monitor capability of my interface so I don't listen to Logic on the track I'm recording at all.
While I always do this and it makes for an in-time performance, I find the recording is sometimes itself out of time and requires moving to line it up.
I just looked up latency of AD converters and it’s negligible. From the Focusrite website the minimum latency the human ear can pick up is 11 ms and one would have to put 10 to 20 passes through AD converters to get to that.
Hi Derek, I just found out by myself: the channel where the plugin is inserted has to be "active", clicked in other words (in the mixer page that's more obvious - they become highlighted - for the track channel you've to pretend to insert a new plugin and then they become active - at least on my system:) - ciao from Italy.
The only time you may need this, is for example you are using (monitoring via) the external Dolby ATMOS renderer. (or any other latency inducing monitoring solution) and for example, you are running RME Totalmix (or some other direct monitoring device)for a near zero monitoring for a singer or other.... it is really handy, for making sure resultant recordings are exactly where they need to be, without affecting artist feel/performance. (important to calculate carefully, the whole monitoring path).
been there! anyway I have a question regarding the software monitoring, recently the auto input monitoring wasn't working (you have it deselected in your settings I noticed) however in the recent update 11.1.2, they said that they fixed the issue but it still not working! any of you encountered the same issue?
I recently have been having this issue, when trying to record into Live Loops, it seems that the recording always misses or is late to getting the beginning of the riff. Could this be solved with these tips? Coming from Ableton, and never had an issue with that software, seems to be something in Logic Pro.
Thank you … I really like your channel, too 🙂
I have an answer for you regarding the recording delay: This is for more complicated setups. If you have a simple setup with just one audio interface, the driver of the Audiointerface tells logic how much time it needs for input and a roundtrip. Things get a little more complicated if you have separated converters. In our studio we have for example on every Mac a RME Digiface USB as an audio interface without built in converter. Connected to this interface are several converters (Prisms, JCF, RME ADI 2 pro fs, Millennia and so on) THey all need little different time to process the signal. However, the audio interface is not aware of this and tells logic still just the processing time of the interface without the converters. For the different converters we have values between -220 and -575 samples. With the correct setting your recording is in this setup still perfect in place :)
This means that you would not be using multiple converters for the same recording scenario right? Or at least, this would be the less ideal situation. Any idea on how to measure the difference in time between converters? Maybe recording the same signal through both units and sample measure it in the DAW?
@@nuestudi524 Yes, you are absolutely right. Especially for stereo recordings it‘s an absolute no go to work with different converters hooked up on an audio interface. We measured this several times. Word clock ensures, that the converters are in „sync“ and don‘t drift. But wordclock has no idea about to which sample to lock. Especially for coincident microphones this is pretty bad, if on microphone is on the first converter the other one is on the other one. This leads to comb filtering effects, even if both converters are from the same company or the same model.
You can measure the needed recording delay quite easily. Just bounce a track (1 measure is already enough) with test oscillator plugin set to needle impulse to 1 Hz. Adjust the tempo of this logic project to 60 or 120. When you import this file just adjust the region that the needle impulse is right on a second on a beat (this makes navigation just a little easier but is not really needed). Open a second ruler. Then record the track with the needle impulse audio file on a different track going through your AD and DA converters. After the recording you will notice that the recorded track will have some delay when you zoom into sample level. In the marquee tool when you select the space between where the needle should be and where it is, there is a window that shows how many frames and samples you are off. Apply this to the recording delay (with a minus) and make another test recording. You should notice that the needle impulses are aligned now :)
-> save these compensation recording delay times per converter, so that you have it right away… Oterwise you get crazy :)
5 years 9 still learning more about LP X thanks
Very clear, simple, essential and helpful. Thanks!
Excellent! Very much needed!
Very good. thanks!
Thanks CW dig the sample part, always wondered about that
Needed this-thanks!!!!!
I appreciate you Chris!
THANK YOU!
3:46 how do I see that help tag box about the plugin? if I check settings>view>general show help tags is enabled but I don't see that box when I hover over a plugin. thanks!
Your videos are always helpful. I think this is geared towards beginners, not too much detail? but, I find it VERY interesting that you have never checked and set your recording delay and that you leave it at 0? Curious to know if you have tested using the ping function inside of the I/O plugin, with a loop wired from output to an input? or tested by recording a click coming out of your speakers with a microphone right up to the tweeter? then inspect what that deviation is? I am running through an Orion32, and my recording delay is set to -16 samples.
Another question, do Drum tracks always seem late to you? how can i fix that?
This is very useful info. I mostly record vocals and I find the easiest way to eliminate latency all together is using the direct monitor capability of my interface so I don't listen to Logic on the track I'm recording at all.
While I always do this and it makes for an in-time performance, I find the recording is sometimes itself out of time and requires moving to line it up.
I just looked up latency of AD converters and it’s negligible. From the Focusrite website the minimum latency the human ear can pick up is 11 ms and one would have to put 10 to 20 passes through AD converters to get to that.
@@johnvender Interesting. Can you elaborate?
can you explain how to get the help tags to show the latency when hovering over plugin?
Hi Derek, I just found out by myself: the channel where the plugin is inserted has to be "active", clicked in other words (in the mixer page that's more obvious - they become highlighted - for the track channel you've to pretend to insert a new plugin and then they become active - at least on my system:) - ciao from Italy.
The only time you may need this, is for example you are using (monitoring via) the external Dolby ATMOS renderer. (or any other latency inducing monitoring solution) and for example, you are running RME Totalmix (or some other direct monitoring device)for a near zero monitoring for a singer or other.... it is really handy, for making sure resultant recordings are exactly where they need to be, without affecting artist feel/performance. (important to calculate carefully, the whole monitoring path).
been there! anyway I have a question regarding the software monitoring, recently the auto input monitoring wasn't working (you have it deselected in your settings I noticed) however in the recent update 11.1.2, they said that they fixed the issue but it still not working! any of you encountered the same issue?
I recently have been having this issue, when trying to record into Live Loops, it seems that the recording always misses or is late to getting the beginning of the riff. Could this be solved with these tips? Coming from Ableton, and never had an issue with that software, seems to be something in Logic Pro.
🤘🙏
🙏