Audio latency is the delay between when you play a note, speak into a mic, or trigger a sound, and when you hear it back through your speakers or headphones. Even a delay of 20–30ms becomes noticeable to musicians and vocalists. Understanding where it comes from helps you know which levers to pull.
Latency exists at multiple points: inside your audio interface, in your DAW or recording software, in your operating system’s audio drivers, and in any effects or plugins you’re running. Some of these you can control directly; others require tweaks to your system.
Lower Your Buffer Size (The Quickest Fix)
The most direct way to reduce latency is to shrink your buffer size — the amount of audio data your interface processes at once. Smaller buffers mean your computer sends audio to your speakers more frequently, with less delay between input and output.
In your DAW or audio interface control panel, you’ll typically find buffer size settings measured in samples. Common sizes are 64, 128, 256, or 512 samples. At a 48 kHz sample rate, a 64-sample buffer produces roughly 1.3ms of latency; a 256-sample buffer produces about 5.3ms.
The catch: smaller buffers demand more from your CPU. If you go too low, you’ll hear clicks, pops, and audio dropouts. Start by lowering your buffer size by one step (say, from 256 to 128) and see if your system can handle it without glitches. If it does, try going lower.
Use ASIO Instead of WASAPI on Windows
Windows offers two main audio driver protocols: WASAPI and ASIO. ASIO (Audio Stream Input/Output) was designed for low-latency recording and playback and consistently outperforms WASAPI in terms of round-trip delay.
If your audio interface came with an ASIO driver, use it. If you’re using your computer’s onboard audio and need low latency, install a universal ASIO driver like ASIO4ALL as a workaround. You can learn more about ASIO4ALL setup and configuration to maximize compatibility with your DAW.
For comparison, check how WASAPI and ASIO perform side-by-side to understand why the choice matters for your workflow.
Check and Fix DPC Latency Issues
DPC (Deferred Procedure Call) latency is a Windows-specific problem. When your system handles background tasks—driver updates, USB polling, antivirus scans—it can interrupt audio processing with delays that add up to audible glitches, even if your buffer size is set low.
Run a DPC latency check using the browser-based tool or the third-party app LatencyMon to measure how often and how long these interruptions occur. If you see spikes above 2–3ms, you’ve found a bottleneck.
Common culprits include USB drivers, chipset drivers, power management settings, and background processes. Update your chipset and USB drivers first. Then disable unnecessary background services, check Windows Update settings, and consider disabling C-state power saving in your BIOS if you’re serious about audio work.
Update Audio Drivers and Firmware
Outdated drivers are a silent killer of low latency. Audio interface manufacturers regularly release driver updates that improve timing and reduce CPU load. Check the maker’s website for your specific interface model and install the latest version.
Also update your motherboard chipset drivers from the manufacturer (not Windows Update), as these control how your computer communicates with external USB and Thunderbolt devices.
Optimize Your System for Audio Performance
Close unnecessary applications, disable Wi-Fi and Bluetooth if you’re not using them (they can cause DPC spikes), and disable visual effects in Windows. Set your DAW’s process priority to “high” or “realtime” if your OS allows it—this tells Windows to give audio processing more CPU attention.
If you’re using plugins, reduce the number of active effects on each track or freeze tracks to audio once you’re happy with the sound. This lightens the CPU load and lets you use smaller buffer sizes without dropouts.
When to Upgrade Your Audio Interface
If you’ve done all the above and latency is still unacceptable, your interface itself may be the limit. Budget interfaces and onboard audio often have higher latency specs than professional-grade gear. A dedicated audio interface with a modern ASIO driver can easily cut total latency in half.
Test your current setup’s latency with a roundtrip measurement so you have a baseline, then compare it to specs of interfaces you’re considering. Look for interfaces that promise sub-5ms roundtrip latency if you’re tracking live vocals, playing virtual instruments, or performing live.
Frequently Asked Questions
What’s the lowest latency I can realistically achieve?
Modern audio interfaces can achieve 2–5ms roundtrip latency with optimal driver and buffer settings. Some high-end interfaces go lower. Anything below 10ms is generally unnoticeable for most musicians.
Does latency differ between Mac and Windows?
Yes. macOS’s Core Audio driver is inherently low-latency by design. Windows requires more tweaking, especially around DPC latency and driver choice. Linux can achieve very low latency with the right kernel and ALSA/PulseAudio tuning, but setup is more technical.
Can plugins and effects add latency?
Yes. Many plugins, especially those with linear-phase EQ or look-ahead compression, introduce their own processing delay. Freeze tracks to audio once you’ve finalized effects to remove plugin latency from your session.
Should I use wireless headphones for monitoring?
Not for tracking. Bluetooth headphones add 40–500ms of latency depending on the codec. Use wired headphones, in-ear monitors, or the interface’s analog outputs for live monitoring while recording.
Dalton is an audio testing and latency optimization writer at SoundLatencyTest. He focuses on audio latency analysis, sound delay testing, recording performance, and audio troubleshooting tools for producers, gamers, streamers, musicians, and audio engineers.