Recording Latency Fix: 8 Easy Steps for Perfect Timing 2026

Recording latency is the delay between playing a note and hearing it back through your computer. You fix it by lowering the buffer size to 64–128 samples while recording, using your interface’s dedicated driver (ASIO on Windows), and using direct monitoring. Wired headphones help too. Under about 10 ms of round-trip latency feels instant to most players.

Latency is the number-one complaint from new home recordists, and it’s one of the first things to get right in any home recording studio setup. A delay of even 20 milliseconds makes playing in time feel like walking through mud.

This guide explains where the delay comes from and gives you a step-by-step recording latency fix that works on Mac and Windows. If you’re recording guitar through plugins, our guide on how to record electric guitar at home pairs well with it.

What Causes Recording Latency?

When you play through your computer, the sound goes on a round trip:

1. The interface converts your analog signal to digital (AD conversion).
2. The driver passes the audio to your DAW in chunks called buffers.
3. The DAW and plugins (amp sims, effects) process each buffer.
5. The driver sends it back, and the interface converts it to analog (DA conversion) for your headphones.

Latency matters most for performers. A drummer playing to a click, a singer hearing their own voice, or a guitarist playing through an amp sim all rely on instant feedback. When the sound arrives late, players tend to rush or drag without realising it, and the takes feel stiff. A proper recording latency fix often improves timing more than any amount of practice. If you’re just getting started, our guide on how to play guitar through computer amp sims covers the full setup.

Each step adds a little delay. The biggest variable, and the heart of any recording latency fix, is the buffer size.

Buffer Size: The Main Recording Latency Fix

The buffer size is how many audio samples the computer collects before processing them. Smaller buffers mean less delay but more CPU strain. Larger buffers are safer for the CPU but add delay. Here’s the one-way latency each setting adds, before converter and driver overhead:

Buffer size (samples)	At 44.1 kHz	At 48 kHz	At 96 kHz	Use it for
32	0.73 ms	0.67 ms	0.33 ms	Ultra-low latency on powerful systems
64	1.45 ms	1.33 ms	0.67 ms	Tracking guitar through amp sims
128	2.90 ms	2.67 ms	1.33 ms	Tracking with a few plugins (a good default)
256	5.80 ms	5.33 ms	2.67 ms	Light tracking, or tracking with direct monitoring
512	11.61 ms	10.67 ms	5.33 ms	Mixing
1024	23.22 ms	21.33 ms	10.67 ms	Heavy mixing and mastering

The formula is simple: latency (ms) = buffer size ÷ sample rate × 1,000. Round-trip latency is at least double that, plus a few milliseconds of converter and driver “safety” buffers. That’s why a 128-sample buffer at 48 kHz typically shows a round trip of around 5–8 ms in a DAW.

For perspective, sound travels about 1.1 feet per millisecond in air. Standing 10 feet from a guitar amp adds about 9 ms of acoustic delay, and nobody complains about that. Most players can’t feel under 10 ms of round trip, and many struggle above 15–20 ms.

Step-by-Step Recording Latency Fix

1. Use a real audio interface. Built-in laptop sound cards and USB guitar cables have high latency. See best guitar audio interfaces.
2. Install the manufacturer’s driver. On Windows, use the interface’s own ASIO driver. Generic drivers like ASIO4ALL or Windows audio are a last resort. On Mac, Core Audio is low-latency by default.
4. Select the driver in your DAW. In your DAW’s audio preferences, choose your interface’s ASIO driver (Windows) or the interface itself (Mac).
5. Lower the buffer to 64 or 128 samples while recording.
7. Disable heavy plugins on other tracks. Look-ahead limiters, linear-phase EQs and some mastering plugins add their own delay. Bypass them or use your DAW’s low-latency mode while tracking.
8. Use direct monitoring when you don’t need to hear plugins. The interface sends your input straight to the headphones with near-zero delay.
10. Use wired headphones. Bluetooth adds roughly 100–300 ms, which is unusable for tracking.
11. Raise the buffer again for mixing (512–1024) so you can run more plugins without clicks.

Home Recording – Understanding Latency and the Buffer

Audio Interface Latency: Why Some Interfaces Are Faster

Audio interface latency varies more than most spec sheets admit. Two interfaces at the same buffer size can show different round-trip times because of:

• Driver quality. This is the biggest factor. Well-optimised drivers add fewer safety buffers.
• Connection type. USB, Thunderbolt and PCIe all achieve low latency with good drivers. A solid USB driver beats a poor Thunderbolt one.
• Converter design. AD/DA filters add a fraction of a millisecond to a millisecond or more.
• Onboard DSP. Some interfaces (for example, Universal Audio Apollo models) run effects on the interface itself, so you can monitor through them at near-zero latency.

Your DAW shows reported latency in its audio settings. Treat it as a guide, because some drivers under-report. You can measure the true round trip by recording a click back into an input through a loopback cable and comparing the waveform offset.

Direct vs Software Monitoring as a Recording Latency Fix

	Direct (hardware) monitoring	Software (DAW) monitoring
Latency	Near zero	Depends on buffer (usually 3–12 ms)
Hear plugins (amp sims, effects)?	No (dry signal only)	Yes
Best for	Vocals, acoustic guitar, mic’d amps	Guitar or bass through amp sims, virtual instruments
Risk	Hearing yourself twice if DAW monitoring is also on	Clicks and pops if the buffer is too low

If you use direct monitoring, turn off input monitoring on the DAW track. Otherwise you’ll hear a dry signal and a delayed copy, which sounds like a short slapback echo.

Recording Latency Fix Settings by Task

Use this table as a starting point for your recording latency fix. Adjust one step up or down depending on how powerful your computer is.

Task	Sample rate	Buffer size	Monitoring
Electric guitar through an amp sim or NAM	48 kHz	64–128	Software (DAW)
Bass through an amp sim	48 kHz	64–128	Software (DAW)
Vocals with no effects	48 kHz	128–256	Direct (interface)
Vocals with reverb in headphones	48 kHz	128	Software, or interface DSP if available
Acoustic guitar with a mic	48 kHz	128–256	Direct (interface)
Playing virtual instruments (MIDI keyboard)	48 kHz	64–128	Software (DAW)
Mixing a full song	48 kHz	512–1024	Not applicable

Many engineers use 48 kHz as a sensible default for music recording. It’s the video standard, it gives a little more latency headroom than 44.1 kHz at the same buffer, and it doesn’t double the CPU load the way 96 kHz does.

Plugin Delay Compensation: The Other Kind of Latency