when I record audio in cubase, it is offset by about 1700 samples prior to when I actually played the instrument. I’m on my Dell XPS Desktop 8960 recording through my UR-RT4. I have updated all the drivers and tried all the latency options I could find, but, if anything, things get worse. Here I recorded my Yamaha DX7 to a clicktrack and recorded the loopback as well:
I notice that you have the “monitor” button active on the input track, Do you monitor your DX7 through Cubase? Do you have direct Monitoring activated for your interface?
And just so I understand correctly: the loopback track in your screenshot is the recording of an audio click track in Cubase via a loopback cable?
If yes, I would say that overall the audio system seems to work correctly.
Yes, I monitor through cubase while recording. What do you mean by “direct Monitoring activated for your interface”?
I recorded the DX7 on input 1 on the interface, and recorded the clicktrack and the monitored dx7 (so basically what I was listening to while recording) via a loopback cable from the headphone output to input 2.
The recording of the loopback is timed exactly how I wanted it and how I played it, but somehow the actual recording on the input track is offset
Do you mean the “Record Shift” option? I have that set to 0, but probably could shift the recording by the average amount of samples the recording is usually offset. It varies though, so I still would have to do slight adjustments.
@fese you’re a godsend! It actually was the “Direct Monitoring” option in the device setup. I don’t know why exactly, but now it is being recorded just as I hear it. Thank you so much and have a great rest of your day!
When you monitor through Cubase without any direct monitoring, you will have the latency of AD/DA and the audio buffer added, plus any latency added by plugins in the signal path, which essentially means that you will hear the signal several milliseconds later than it actually is.
For that reason, it is usually recommended to use direct monitoring, which means that the input signal is directly routed to the outputs in the audio interface, without going through the computer. That way, the only latency that is added is the AD/DA, which is negligible.
I don’t know whether that has something to do with your issue, but I’d suggest trying direct monitoring and see whether that helps.
