Audio latency explained: where the milliseconds come from and how to get them back

Latency is the delay between a sound existing and you hearing it after it passed through your system. It is measured in milliseconds, and it accumulates across every stage of the chain rather than sitting in one place.

Where the delay comes from

• Analogue to digital conversion: typically 0.5 to 1.5 ms, fixed by the converter design.
• Input buffer: samples are handed to software in blocks. A 128 sample buffer at 48 kHz is 2.7 ms.
• Processing: plugins that look ahead, such as linear phase EQs and limiters, add their own delay.
• Output buffer plus digital to analogue conversion: roughly the same again.
• Wireless: Bluetooth adds anywhere from 30 to 200 ms and cannot be used for monitoring.

The buffer trade off

Smaller buffers mean lower latency and more frequent interrupts, so the CPU has less slack and you risk clicks and dropouts. Larger buffers are stable but sluggish. The usual approach is a small buffer while recording and a large one while mixing.

What counts as acceptable

Below roughly 10 ms round trip, most performers stop noticing. Between 10 and 20 ms it feels slightly soft. Above 30 ms, timing suffers. For reference, sound travels about 34 cm per millisecond in air, so 10 ms of latency is like standing three and a half metres further away.

Plugin delay compensation

Your DAW compensates for plugin latency by delaying every other track to match the slowest one. This keeps the mix in phase, but it also means a single look ahead limiter can push the whole session into sluggish monitoring. Bypass those plugins while tracking.

Latency is not sync error

A constant delay through a monitoring path is not the same as audio and picture drifting apart. Drift comes from mismatched clocks or pull up and pull down frame rates, and it grows over the length of a take. If your sync error increases as the take goes on, look at sample rate and frame rate, not at buffers.