I'm using win2K with the delta1010 and n-track. When i'm dubbing over audio tracks I set my ASIO buffer size down pretty low to the point where I'm probably getting around 10 ms of latency. My question is - how does one eliminate latency entirely from the final recording. It seems to me that if the dubbed over track is consistantly 10ms behind the original, it could just be shifted over by 10ms - then the 2 tracks would be perfectly in time. Seems as if this is an option audio software should have. Does anyone out there know anything about this?