The term latency refers to the period of time the computer needs to process an audio signal. It is usually measured in milliseconds. If the latency is too high, you will notice an undesirable lag between any input (recording a microphone signal, playing a software instrument via a MIDI keyboard) and the respective output (monitoring your recording in realtime, listening to the output of your software instrument).
The latency is determined by the size of the audio buffer (in samples). The size of the audio buffer can be set in your ASIO driver settings (Win) or the preferences of your audio software (Mac). A small audio buffer (e.g. 32, 64, 128 samples) will give you the desired low latency, however it will also put more load on the computer. If the computer is not able to provide the amount of processing power needed to complete all audio calculations in time, you will get crackles and drop-outs in the audio playback. In this case, gradually increase the size of the audio buffer until the playback is clean.
Note: If you cannot find a latency that is both low enough and interference-free, also read the "Windows Tuning Tips" linked in the Related Articles area at the bottom of this page.