When I play a note, the sound arrives after a long delay... (What is audio latency?)

Understanding Audio Latency in Digital Music Production

What is latency?

Latency refers to the short delay between an input (like pressing a key on a MIDI controller) and the resulting sound being heard. In digital audio systems, this is a normal and expected behavior caused by the time it takes for your computer or device to process audio signals.

Why Does Latency Happen?

Digital audio processing involves several steps:

• Input conversion: Analog signals (e.g., microphone or instrument) are converted into digital data.
• Processing: The plugin or DAW applies effects, instruments, or other processing.
• Output conversion: The processed digital signal is converted back to analog so you can hear it.
• Each of these steps takes time, and the total delay is what we call latency.

Can Latency Be Reduced?

Yes, but it can’t be eliminated entirely. Here are some ways to minimize it:

• Lower buffer size in your audio settings (e.g., 128 or 64 samples). This reduces processing time but increases CPU load.
• Use an audio interface with optimized drivers (ASIO on Windows, Core Audio on macOS).
• Disable unnecessary plugins during recording to reduce processing overhead.
• Freeze or bounce tracks to free up system resources.

Common Misunderstandings

Some users expect instant sound with no delay, especially when using virtual instruments. While modern systems can achieve very low latency, a small delay is always present. It’s not a bug—it’s a technical limitation of digital audio.