Work out audio buffer latency in milliseconds from buffer size and sample rate — for recording, live monitoring, and interface setup.
The Audio Latency Calculator converts an audio interface’s buffer size and sample rate into milliseconds — the delay you hear when monitoring through your computer. It reports one-way buffer time, the total round-trip (input + output + converters) and what that delay means for playing in time.
A 64-sample buffer at 48 kHz is only 1.3 ms per buffer, whereas 1024 samples is 21 ms: the table of results shows why tracking is done at small buffers and mixing at large ones.
Latency of one buffer = buffer size ÷ sample rate. Round trip ≈ input buffer + output buffers + converter/driver overhead, since audio must be captured, processed and played back. Rule of thumb: under about 6 ms feels instantaneous; 6–12 ms is playable; over 25 ms is clearly noticeable. Sound travels about 34 cm per millisecond, which the tool also reports as distance.
Pick the smallest stable buffer.
Keep keyboards responsive.
Decide when to use direct monitoring.
Understand audio lag.
| Buffers | 32–2048 samples |
|---|---|
| Sample rates | 44.1–192 kHz |
| Output | One buffer, round trip, distance equivalent |
64–128 for recording, 256–512 for general work, 1024+ for mixing.
It lowers latency per buffer but uses more CPU.
Latency = buffer size ÷ sample rate. A smaller buffer processes audio in smaller, more frequent chunks, so there's less delay between input and output — at the cost of more CPU overhead.
It's the theoretical latency for one direction based on buffer size alone — actual round-trip latency (input to output) is typically about double this, plus additional overhead from drivers, converters, and any plugins in the signal path.
This tool runs entirely in your browser using the Web Audio API. Nothing is uploaded, stored or shared — there is no server to send your file to, so it works offline once the page has loaded.
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