Buffer Size vs. Audio Latency Explained

Quick Answer

Audio latency is usually a buffer-and-processing problem, not a mystery number. The buffer formula gives you the starting point, but the real-world result depends on drivers, plugins, conversion stages, and whether you are measuring one-way delay or full round-trip latency.

What is Audio Latency?

Audio latency is the time delay between an audio signal entering a system and emerging from it. In broadcasting, live events, or studio recording, this delay accumulates as the audio travels through ADC converters, computer processing buffers, digital network protocols, and finally the DAC interface.

Whether you're doing live audio over Dante and AES67 networks, or trying to synchronize lip sync in a production camera path, understanding how digital audio latency works is paramount for a professional engineer.

How to Calculate Audio Latency

Inside a digital audio system (like OBS, vMix, Pro Tools, or Reaper), the latency is primarily defined by the buffer size and the sample rate.

To eliminate manual calculations in the heat of a production, use our audio buffer latency calculation tool. It gives you the calculated delay for common sample-rate and buffer combinations, which you can then verify against the actual signal path.

Examples: 128, 256, and 512 Samples

Here are the theoretical single-buffer durations for common buffer sizes at 48 kHz; driver, safety buffer, conversion, and application stages add more delay:

• 128 samples = 2.67 ms: Excellent for in-ear monitoring (IEM) and real-time processing. Requires a fast CPU.
• 256 samples = 5.33 ms: The sweet spot for live production processing. A safe balance between CPU overhead and acceptable vocal delay.
• 512 samples = 10.67 ms: Noticeable to a performer wearing headphones, but excellent for stream encoding buffers or mastering chains.
• 1024 samples = 21.33 ms: Often used in purely post-production environments where rendering stability is more important than latency.

Driver Architecture: ASIO vs. WDM

On Windows-based broadcast systems (vMix, OBS, Wirecast), the driver type you choose is more important than the buffer size itself:

• WDM / WASAPI: The standard Windows driver path. It often adds more latency and less predictable behavior than an ASIO workflow, though the exact result depends on the device and software stack.
• ASIO (Audio Stream Input/Output): A direct driver model commonly used for lower-latency professional work. It is often the better choice for sub-10 ms targets on Windows, but the result still depends on hardware, plugins, and routing.

Total Round-Trip Latency (RTL)

Keep in mind that buffer latency is only one layer of the cake. Total system latency (or glass-to-glass latency) involves:

1. A/D Conversion: ~1ms (microphone to digital)
2. Input Buffer: (e.g., 128 samples @ 48kHz = 2.67ms)
3. Processing: VST plugins, EQ, Compression (variable)
4. Output Buffer: (e.g., 128 samples @ 48kHz = 2.67ms)
5. D/A Conversion: ~1ms (digital to speaker)

Typical: a software mixer running at 128 samples may land in roughly the 8–10 ms round-trip range before heavier processing is added. Look-ahead limiters, denoisers, and other buffered plugins can push it higher quickly.

Common Mistakes

• Treating buffer math like the full end-to-end latency answer.
• Ignoring plugin delay and only checking the interface buffer size.
• Confusing one-way latency with round-trip latency.
• Assuming one Windows driver path behaves like another.

References

• OBS Knowledge Base
• Audinate Learning
• Steinberg ASIO Overview