Why, in 2016, is latency still an issue in recording applications?
Once you are finished recording, why can't the software simply calculate the latency and then shift the whole recording automatically a few milliseconds earlier so that the latency in playback no longer exists? Would this solution not work in practice?
I don't know much about audio software, so I am likely missing something here.
If you are talking about recording audio, latency is not an issue in 2016. Today, audio interfaces that are designed for multitracking have zero-latency monitoring. The audio input is split and you monitor an analog loop back, while the other side of the split goes into the recorder. You don’t monitor what is coming out of the recorder.
Latency is not computer-specific. If you monitor through a tape recorder you also get latency. You have to monitor the input signal itself. It will always take too long to go through the recorder because anything other than instant is too long. So we have always had to monitor the input signal in a music context, where timing is critical.
Early audio interfaces omitted the zero-latency monitoring to save money. That is why latency used to be an issue. Today, if an audio interface lacks zero-latency monitoring, it was simply not designed for multitrack recording. It is designed for speech, or recording a stereo music signal with no multitracking.
In short, a multitrack recorder will always have latency, and it has always been solved (even in 1975) by monitoring the audio input.
So if you are having trouble with latency, maybe you are not using your audio interface’s zero-latency monitoring? Or maybe your audio interface is not designed for multitracking.
