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Latency

Mic Latency Test

Measure the real round-trip delay between making a sound and your computer hearing it.

The short answer

A mic latency test measures round-trip delay: output buffer, air gap, input buffer. Under 60 ms feels instant. Over 120 ms makes live monitoring unusable and singing to a backing track impossible.

Mic Latency Test — how to do it

  1. Use speakers, not headphones

    The test plays a click and listens for it coming back through the microphone. Headphones break that loop and the measurement fails.

  2. Start the microphone test

    Latency measurement needs a live input stream, so press Start test first and confirm the meter is moving.

  3. Set a moderate volume

    Loud enough for the microphone to hear clearly, quiet enough not to overload it. Around half volume works for most systems.

  4. Quieten the room

    Any background noise can be mistaken for the returning click. Close windows and pause fans if you can.

  5. Run the measurement

    Five clicks are played and timed. The result is the average of the tightest passes, with the single best pass shown separately.

  6. Compare against buffer size

    The panel also reports the browser's own output latency and buffer figures so you can see how much of the total is software.

How to reduce audio latency

  • Use ASIO on WindowsAudio interfaces with ASIO drivers bypass the Windows audio stack and typically achieve under 10 ms output latency.
  • Lower the buffer sizeIn an interface control panel, a smaller buffer means less delay and more risk of dropouts. 128 or 256 samples suits most machines.
  • Avoid USB hubs and long cablesEvery extra hop adds latency and jitter to a USB audio device.
  • Turn off audio effectsVendor enhancement software processes audio in additional buffers, which shows up directly as delay.
  • Use a wired connectionBluetooth adds 100 to 250 ms in most cases. Any wired headset or microphone removes that in one step.

What round trip actually measures

Round-trip latency is the sum of several delays: the application preparing audio, the output driver buffering it, the digital-to-analogue conversion, the travel time through air, the analogue-to-digital conversion, the input buffer, and finally the application receiving it.

Software is responsible for nearly all of it. The air gap at typical desk distance contributes about 3 ms, while buffering on a consumer system routinely contributes 40 to 100 ms. That is why reducing buffer size and avoiding Bluetooth make far more difference than moving the microphone closer.

Privacy: every measurement on this page happens inside your browser. No audio is uploaded, and recordings exist only in the tab until you download them or close it.

Common questions

Is Bluetooth latency fixable?

Not really. Codec choice helps a little, but the protocol overhead sets a floor. Wired is the only reliable answer for live monitoring.

What is a good latency figure?

Under 60 ms round trip feels instant for monitoring. Under 20 ms is professional territory. Above 120 ms is distracting for anything live.

Why is my browser latency higher than my DAW?

Browsers use conservative buffers for stability across unknown hardware. A DAW with a dedicated driver can be several times faster.

Why do results vary between runs?

Buffer scheduling and CPU load fluctuate. That is why the test averages several passes and reports the best one separately.

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