Latency is the delay between a sound going into your computer and coming back out. A little is unavoidable: the computer processes audio in small chunks called buffers rather than one sample at a time.
How buffer size affects latency
The time one buffer takes is its size divided by the sample rate. At 48 kHz, a 64-sample buffer is about 1.3 ms and a 1,024-sample buffer about 21 ms. The delay you actually hear is larger — input and output buffers, converters and drivers all add a little — but the buffer is the part you control.
The trade-off
- Small buffers — low latency, but the computer has less time per chunk, so heavy sessions may crackle or drop out.
- Large buffers — stable playback with lots of plug-ins, but noticeable delay when you play or sing.
What to use when
- Recording — 64 to 128 samples, and disable heavy plug-ins (especially look-ahead limiters) on the tracks you're monitoring.
- Mixing — 512 or 1,024 samples; nobody is playing live, so stability matters more.
- Direct monitoring on your audio interface lets you hear your input with near-zero latency whatever the buffer.
Using a dedicated audio interface with its own driver (ASIO on Windows) usually gives far lower latency than a computer's built-in sound card.