Engineering Guide  |  Audio Latency

BROADCAST BASICS

Buffer Size vs. Audio Latency Explained

Understand the difference between buffer size and latency in professional broadcast audio, including ms calculations for 128 samples at 48kHz.

Essential for diagnosing lip-sync drift or configuring live monitors. For instant results, use our audio latency calculator. Explore more broadcast engineering tools for IP video and media networking.

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 the path uses Dante or AES67, a software mixer, or a local audio interface, separate the theoretical buffer duration from the measured end-to-end result.

How to Calculate Audio Latency

Buffer size and sample rate define the duration of one buffer. They do not by themselves define the full latency of OBS, vMix, a DAW, an interface, or a networked audio path.

The Formula:
Latency (ms) = (Buffer Size / Sample Rate) * 1000
Example: (128 / 48000) * 1000 = 2.67 milliseconds.

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:

Driver Architecture: ASIO vs. WDM

On Windows-based production systems, the available driver path can materially affect latency and stability, but it is one part of the complete path:

Workflow note: direct monitoring
Many interfaces can route an input to headphones without passing through the application's main buffer path. That can reduce monitoring latency substantially, but conversion, internal routing, and any digital processing still take finite time. Verify the routing and measured result for the actual interface.

Total Round-Trip Latency (RTL)

Buffer latency is only one stage. Audio round-trip latency can include:

  1. A/D Conversion: device-dependent input conversion
  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: device-dependent output conversion

The driver-reported round-trip value and a loopback measurement may differ because safety buffers, conversion, USB or network transport, and plugins are not always represented the same way. Measure the real path when the number matters.

Common Mistakes

Related Tools & Next Step

References