Thank goodness for that 
“LOL! But how do I know? How can I verify this? I don’t wanna guess or assume. I want to be sure. I mean, I don’t have to measure it for myself, but I’d like some sort of verification that (or reason why) “Input Latency” is compensated for when I record.”
“Play any Cubase wave file test signal from the output of your converter.
In the real world that is what you’re trying to sync to.
Plug that back into the converter’s input and record it.
Play the two files side by side and reverse one’s polarity.
If you hear dead silence, you’re good to go.”
“That’ll never completely null, with the artifacts from DA and AD conversion, the cable, the circuitry, possibly a pre-amp on the input etc.”
I just now did an export using a drum snippet, and it nulled perfectly, down to the sample.
so I have to disagree.
(I have MOTU PCI converters.)
Ok, surprising. Good for you though 
I know this is old but I am wondering if anyone has any more info on this.
For example I use Mytek converters, into an RME Madi card. If I do a round trip I get about 90 samples of latency, as does another colleague of mine with Eclipse converters and gets similar results when running through an RME AES card vs the stock eclipse USB.
I am wondering:
-
Are we sure that Cubase compensates for the Mytek in addition to the RME card alone for input latency?
-
How can I figure out just the input latency of my Myteks vs the input latency of the RME card, vs the DA conversion latency?
Maybe it is psychological but if I use the “record shift” option in Cubase to compensate by 90 samples, my tracks seem to feel more tight to what I actually played. I know some may not agree but a couple of milliseconds can make a large difference in “feel” and if there is some conversion latency that is not being compensated for I would like to correct for it.
I agree with Dudleys on this one. I did the following:
- Recorded a click track into cubase (just routed the click back into a channel and recorded it).
- Killed the click from cubase.
- Now the recorded click is my reference, so i played it and routed it back into a channel in Cubase, then recorded it.
- Zoomed WAY in to compare the tracks, and behold there WAS a difference; visually I could see the waveforms not exactly lined up. Close tho.
- Repeatedly adjusted record latency compensation until visually I could see it was sample accurate.
I found my record latency compensation to be 57 samples. At 48Khz it is probably fractions of a ms difference. I don’t know I don’t feel like doing the math. I mean, this could even be the difference in time for the signal to travel through the cables in my studio. I don’t know.
Incidentally throwing the tracks out of phase does not cancel completely, however the sample accurate track cut out MUCH more of the volume in comparison to the track with no compensation. Hmmm…
Does anyone see a hole in this method?
But seriously, the distance from your ears to the speakers probably creates a timing shift greater than 57 samples. The farther you are the worse it gets. Unless you are using headphones.
As stated earlier in this thread, Cubase takes it’s latency compensation from the reported buffer setting from the ASIO driver. Once you go outside that, Cubase will no longer know what latency is being introduced, so if you’re hanging converters onto a digital bus, then Cubase has no idea what latency those converters are adding, then manual adjustment may be required.