When your computer plays a sound, it doesn’t come out instantly. The audio passes through software buffers, the operating-system mixer, and the output hardware before it reaches your ears — a delay called output latency, sometimes referred to as sound latency or monitoring latency when the context is live recording or headphone monitoring. The size of those buffers (often expressed in samples or milliseconds in your audio driver settings) is a primary factor: larger buffer sizes reduce CPU dropouts but add more latency. This tool measures end-to-end output latency without any microphone, using a method borrowed from rhythm-game calibration screens: it plays a steady metronome beat, and you tap along to what you hear. Because the sound arrives at your ears late, your taps land late by roughly the same amount. The average lateness of your taps is your latency calibration value — the audio offset you dial in to keep audio in sync.
The tool times each tap against the beat it was scheduled to play, using the browser’s high-resolution clock mapped to the audio clock (via getOutputTimestamp when available). This tap latency test ignores the first couple of taps as warm-up and any wildly off-beat taps, then reports the average offset and how consistent your taps were. If you want to verify that a pure tone reaches your ears cleanly before running this test, you can generate one with the online tone generator to confirm your output is working.
What the number means
- Average offset — how late (positive) your taps land relative to the scheduled beat. This combines the real output latency with your personal tapping tendency, which is exactly the offset you’d enter into a game or DAW so things feel in sync.
- Consistency (±) — the spread of your taps. A small ± means a trustworthy measurement; a large ± means keep tapping or slow the tempo down.
- Base / output latency — the browser’s own reported figures, shown for reference. These are hardware/buffering estimates; the tapped value is the perceived end-to-end offset.
Tips for a good measurement
- Use the same output device (and settings) you actually use — Bluetooth headphones in particular add large, variable latency.
- Tap when you hear the beat, not when you expect it. Relax into the rhythm.
- Collect plenty of taps (16+). More taps tighten the average.
- If your ± is large, try the 80 BPM tempo — slower beats are easier to tap accurately.
- In your DAW, this offset is typically entered as a negative value in the hardware monitoring delay or input-monitoring offset field — check your DAW’s manual for the exact setting name (e.g. “audio device latency” in Ableton Live, “hardware buffer compensation” in Pro Tools).
- On Windows, drivers using ASIO (like ASIO4ALL or manufacturer drivers) generally produce lower and more consistent latency than the standard WDM/shared mode used by browsers — so your browser measurement may be higher than what you see in a DAW using ASIO.
How is this different from the Mic Latency Tester?
The
Mic Latency Tester measures the full acoustic
round trip — it plays a click and your microphone hears it — so it needs speakers and a mic. This tool needs
no microphone: it measures
output / perceived latency by having you tap along to a beat. Use this one to calibrate an audio offset for games or recording monitoring; use the mic tester to measure true hardware round-trip latency.
Does this need a microphone?
No. There’s no microphone access at all — it only plays sound and reads your key/tap timing. Nothing is recorded.
Is the number pure hardware latency?
Not exactly — and that’s fine. The measured offset blends the real audio output latency with your personal tapping bias (most people tap slightly early or late by a consistent amount). For calibration purposes that combined figure is precisely what you want, because it’s the offset that makes things feel in sync for you. If you need pure hardware round-trip latency, use the Mic Latency Tester or a DAW loopback.
Why ignore the first couple of taps?
It takes a beat or two to lock into a rhythm, so the earliest taps are usually less accurate. Discarding them gives a cleaner average. Keep tapping steadily and the average settles quickly.
Why is my consistency (±) so large?
Either you’re anticipating instead of reacting, the tempo is too fast for you, or your output has variable latency (common with Bluetooth). Try 80 BPM, focus on tapping right when you hear the click, and avoid wireless audio for the measurement.
Should I use headphones or speakers?
Use whatever you’ll actually be using, since the latency you want to calibrate is for that setup. Wired headphones and speakers have low, stable latency. Bluetooth devices add 100–300 ms and can vary, so calibrate them specifically if that’s your setup.
What are good and bad numbers?
For wired output, a calibration offset under ~50 ms is typical and feels responsive; 50–150 ms is noticeable but workable; over ~150 ms (often Bluetooth) is sluggish for rhythm work. There’s no single "correct" value — it depends on your hardware — but a consistent measurement (small ±) is what matters for calibration.
Is any data collected?
No. Everything runs in your browser — sound output and tap timing only. Nothing is recorded, stored, or sent anywhere.
How do I enter this number into my DAW as an audio offset?
Most DAWs have a hardware latency or audio device delay compensation field — the exact name varies. In Ableton Live, check Preferences → Audio → Overall Latency. In Logic Pro, it’s in Audio → Devices → I/O Buffer Size and latency compensation is automatic. In Reaper, look under Preferences → Audio → Playback. Enter the millisecond value you measured here (sometimes as a negative number, sometimes positive depending on the field’s convention — consult your DAW’s manual). The goal is that recorded audio lines up with the click track without you having to nudge clips manually.
Can I use this for rhythm game calibration (Beat Saber, Guitar Hero, etc.)?
Yes — that’s exactly what this method is modelled on. Rhythm games like Beat Saber, Osu!, and Clone Hero include an audio offset setting (usually in milliseconds) that shifts when notes appear relative to the audio. Enter the average offset value from this tool as your audio offset (positive if the game asks “how late is the audio?”). Make sure you measure with the same headphones or speakers you play with, since Bluetooth audio can add 100–300 ms of latency not present with wired output.
What is audio buffer size and how does it affect latency?
The audio buffer is a small block of samples your system fills before sending them to the hardware. Larger buffers (e.g. 512 or 1024 samples) give the CPU more time and reduce glitches, but they add more latency — at 44,100 Hz, 512 samples equals about 11.6 ms just for the buffer. Browsers typically run with relatively large shared-mode buffers (often 20–60 ms), which is why browser-measured latency tends to be higher than what a DAW with ASIO drivers reports. This tool measures the real perceived delay in the browser, which is the number relevant for calibrating browser-based games and any output you monitor through your browser.