Mic Latency Test

Microphone latency is the time delay between sound physical pressure waves hitting the microphone capsule and the digital audio samples becoming available to an application or operating system.

Click 'START MIC TEST', grant microphone permission, calibrate the background noise floor, and click 'START TEST'. The browser plays a short acoustic pulse through your speakers, detects it through your microphone, and computes the observed loopback delay.

Microphone delay is caused by analog-to-digital conversion, USB or Bluetooth controller packet buffering, operating system audio driver pipelines, browser audio process buffering, and device DSP algorithms (like noise cancellation).

Round-trip latency is the total elapsed time required for a digital audio signal to travel from software generation to speaker output, across the air to a microphone, through the capture driver, and back into software.

Acoustic loopback is a testing technique where an audio output signal played through physical speakers travels through the air and is picked up by a nearby microphone.

A browser can measure end-to-end observed acoustic loopback delay using high-resolution Web Audio API clocks. It cannot isolate the physical microphone hardware delay from the rest of the OS and browser pipeline.

No. The measurement is an observed acoustic loopback delay that includes speaker output buffering, air propagation, microphone capture latency, OS driver buffers, and browser processing.

Yes, as an observed loopback measurement. Bluetooth microphones typically exhibit 120 ms to 300 ms of delay due to wireless packetization, RF retransmission buffers, and low-complexity audio codecs.

Bluetooth audio standards prioritize link reliability over low latency. Digital audio packets must be encoded, transmitted over 2.4 GHz radio, acknowledged, and decoded through operating system Bluetooth stacks.

USB microphones use internal USB audio class controllers that buffer audio packets (typically 5 ms to 20 ms) before handing them to Windows WASAPI, macOS CoreAudio, or ALSA drivers.

If you listen to your own voice through headphones while speaking, the sound passes through the operating system and recording software buffers before returning to your ears, causing a distracting echo.

Operating systems dynamically schedule audio buffer threads, browser processes face CPU load variations, and ambient room acoustics can cause slight jitter in pulse arrival detection.

AudioContext latency includes baseLatency (the minimum hardware audio buffer delay) and outputLatency (the estimated delay between buffer submission and physical sound production).

Input latency is the time from physical acoustic sound capture at the microphone capsule until the audio samples are processed by the software application.

Output latency is the delay between a software audio command and the physical acoustic emission from your speakers or headphones.

Input latency measures only the capture side. Round-trip latency includes both the output playback chain, the acoustic path, and the input capture chain.

Yes. Algorithmic noise suppression analyzes spectral frames in chunks (often 10 ms to 40 ms) to calculate noise profiles, introducing additional processing delay.

Yes. Acoustic Echo Cancellation (AEC) buffers incoming audio and compares it against captured microphone audio to cancel speaker bleed, adding buffering overhead.

Automatic Gain Control (AGC) introduces slight envelope lookahead buffering to adjust volume smoothly without sudden clipping.

The most common reasons are: output volume is too low for the mic to hear, headphones are being used instead of speakers, background noise is louder than the reference signal, or the wrong microphone is selected.

If using headphones, sound is sealed inside the earcups and cannot reach the microphone. Also verify that your speaker volume is set to a moderate listening level and output is not muted.

Acoustic loopback requires sound to travel through the open air from speaker to microphone. Sealed headphones keep sound away from the microphone capsule.

Browser Web Audio API clocks use microsecond-resolution performance.now() and AudioContext.currentTime, providing timing precision within 1 to 5 milliseconds for observed loopback delay.

No. This tool uses automatic Web Audio API signal detection (RMS energy and peak thresholding) to eliminate human biological reaction delay (~200 ms) from the measurement.

No. Measuring physical sound pressure level (decibels SPL) requires a laboratory-calibrated physical measurement microphone. This tool measures digital full-scale levels (dBFS).

No. Microphone audio is analyzed in real-time transient memory using AnalyserNode.getFloatTimeDomainData(). No audio recordings are saved or uploaded.

Yes. All processing runs 100% locally in your browser. No microphone data or telemetry leaves your computer.

For real-time voice chat and streaming, observed round-trip loopback delay below 60 ms is considered excellent. Professional studio ASIO recording setups target under 15 ms.

Use a wired USB or XLR microphone, switch to speakers placed near the mic during testing, disable browser echo cancellation and noise suppression, and set your OS audio buffer to 128 or 256 samples.