HARMOVANT
GUIDE

Live Sound Latency: How Much Is Acceptable and How to Reduce It

Latency is the time between sound hitting the mic and coming out of the speaker or in-ear monitor. On stage, a few milliseconds matter, especially in IEMs. Here's where it comes from and how to keep it under control.

Updated October 30, 2026 · 4 min read

Every digital system has latency. A typical digital console already adds a few milliseconds between input and output, and nobody notices. Trouble starts when latency grows and you don't know where it's coming from. Understanding each part helps you make better decisions about buffer size, plugins and monitors.

Where latency comes from

  1. Converters. The interface converts analog to digital at the input and back again at the output. Each conversion takes a fraction of a millisecond.
  2. Input and output buffers. The computer receives and delivers audio in blocks. Each block is a buffer.
  3. Driver and safety margin. The driver and the system may hold one extra block to avoid dropouts when the CPU spikes.
  4. Plugins. Most EQs and compressors have zero latency. Linear-phase, look-ahead and analysis plugins add time, sometimes a lot.

How to calculate buffer size

The time of one buffer is the number of samples divided by the sample rate.

Buffer 48 kHz 96 kHz
32 samples 0.67 ms 0.33 ms
64 samples 1.33 ms 0.67 ms
128 samples 2.67 ms 1.33 ms
256 samples 5.33 ms 2.67 ms

Remember that audio passes through at least one buffer on the way in and another on the way out. Add converters, driver and plugins, and you get the total latency, which is always higher than the buffer number alone.

How much is acceptable

A useful reference: sound travels about 34 centimeters (roughly 13 inches) per millisecond in air. A musician standing 3 meters (about 10 feet) from their amp already hears it about 9 ms late, and thinks that's normal.

  • FOH / PA: the audience won't notice a few extra milliseconds. What matters is that the system is time-aligned.
  • Floor wedges: the distance between the musician and the wedge already adds acoustic latency. A few extra milliseconds rarely bother anyone.
  • In-ear monitors (IEMs): this is where latency shows up most. Singers hear their own voice inside their head and through the IEMs at the same time. With delay, the voice sounds hollow or comb-filtered. Keep this path as short as possible and avoid plugins that add latency.

How to reduce latency without losing stability

  • Lower the buffer size gradually. Run the whole show at 128 samples first, then try 64 or 32.
  • Use 96 kHz only if your Mac can handle it. Latency is cut in half, but processing doubles.
  • Split your signal paths. Heavy plugins can stay on FOH, while IEM mixes get a path with zero-latency plugins.
  • Avoid linear-phase and look-ahead on vocals and instruments that feed the IEMs.
  • Keep the Mac free. Quit other apps, turn off syncing, and don't record in the background to the same drive without testing first.

Latency compensation: why it matters

When two signals are summed, both need to arrive at the same time. If one channel goes through a plugin with 3 ms of latency and the other doesn't, the sum gets phase cancellation. Latency compensation delays the faster path so everything arrives aligned.

Ideally, compensation happens only where signals meet. If the whole console waits for the slowest plugin, a single heavy plugin delays every channel, including monitor mixes that didn't need it.

Where Live Rack fits in

In Live Rack, at each summing point only the signals that are summed together wait for the slowest one, so a heavy plugin on one channel doesn't delay the whole console. The status bar shows the real total latency, from input to the LR output, including buffer, driver, plugins and the safety buffer. With low-latency plugins, that path stays between 5 and 14 ms with buffers from 32 to 128 samples at 48 kHz.

To choose your hardware and software, see also the guide do you need a DSP server?.

Frequently asked questions

What buffer size should I use live?

Start at 128 samples at 48 kHz. If the show runs without clicks and with CPU headroom, try 64. Use 32 only if your Mac and interface can handle the entire show.

Why does my software show more latency than the buffer size?

Because the total includes input, output, converters, driver, safety margin and plugins. Be wary of numbers that show only the buffer.

Does latency change with the number of channels?

Not directly. More channels need more processing. If the CPU can't keep up, you need a larger buffer, and then latency goes up.

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