How to Prevent Audio Latency When Recording 2026: Buffer Size, Drivers & Direct Monitoring

What Causes Audio Latency?

Audio latency is the delay between when you play or sing and when you hear it in your headphones. It is caused by the entire digital audio chain: analog-to-digital conversion → USB/Thunderbolt transfer → driver processing → DAW buffer → plugin processing → digital-to-analog conversion. Each step adds microseconds to milliseconds of delay.

1. Use Direct Monitoring (Zero Latency)

Every modern audio interface includes direct monitoring — a feature that routes your microphone signal directly to your headphones before it reaches the computer. This is true zero-latency monitoring. On the Focusrite Scarlett Solo, the Direct Monitor switch blends between input (zero-latency) and DAW playback. When recording, set it mostly to Input.

2. Lower Your Buffer Size

Buffer size is the most controllable latency factor. At 48 kHz: 32 samples = ~1.5ms round-trip (very low, high CPU), 64 samples = ~3ms (good for recording), 128 samples = ~6ms (acceptable for most instruments), 256+ samples = 10ms+ (fine for mixing, noticeable for performers). If you hear crackles, increase the buffer.

3. Install ASIO Drivers (Windows Only)

Windows' default audio drivers (MME/DirectSound) have 100ms+ latency. ASIO drivers bypass the Windows audio stack for direct communication with your interface. Every interface includes ASIO drivers — always use them for recording on Windows. Mac users do not need to worry about this — Core Audio is inherently low-latency.

4. Freeze or Bounce Tracks with Heavy Plugins

Some plugins add significant latency: linear-phase EQs (20–50ms), lookahead limiters (5–20ms), noise reduction (50–100ms). When recording new parts, temporarily disable or freeze these tracks. Most DAWs have a 'Low Latency Mode' that automatically bypasses high-latency plugins during recording.

For interface recommendations, see our audio interface buying guide and browse interfaces with live prices.