Audio Latency Calculator

Frequently Asked Questions

How is audio buffer latency calculated?

One-way buffer latency (ms) = buffer size in samples ÷ sample rate × 1000. A 256-sample buffer at 48,000 Hz is 256 ÷ 48000 × 1000 ≈ 5.33 ms per pass.

What is round-trip latency?

Round-trip is input plus output buffering (plus converter and driver overhead), often roughly 2× the buffer latency. A 5.33 ms buffer typically yields ~11–15 ms round trip including A/D and D/A delays.

What latency is acceptable for live monitoring?

Under ~10 ms round trip feels immediate; 10–20 ms is usable for most overdubs; above ~25–30 ms players hear a distracting echo. Lower buffers reduce latency but raise CPU load and dropout risk.

How do I reduce latency without dropouts?

Lower the buffer size, raise the sample rate, use ASIO/Core Audio drivers, freeze or disable heavy plugins while tracking, and use direct/hardware monitoring for zero-latency input feedback.

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Estimates for informational purposes only.

Important Disclaimer: Estimates for informational purposes only.

This calculator provides estimates for informational purposes only. Results are based on assumptions and may not reflect actual outcomes. Consult qualified professionals in relevant fields before making important decisions based on these results.

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How This Calculator Works

Latency is the delay between making a sound (or playing a virtual instrument) and hearing it back through your audio interface. It is set almost entirely by the driver's buffer size and the sample rate. This calculator converts a buffer size in samples and a sample rate into the latency of a single buffer, then estimates the round-trip figure - input buffer plus output buffer, the delay a performer actually feels while monitoring through the computer. It also expresses the total in samples and gives a plain-language verdict against the rough thresholds at which musicians start to notice and then struggle with the delay. The reason this matters in practice is that the same project can feel effortless to track at one buffer setting and unplayable at another, even though nothing about the music changed. Knowing the millisecond figure before you hit record lets you choose a buffer that balances a comfortable feel against the CPU headroom your plugins need, instead of discovering mid-take that the delay is throwing off your timing.

Formula / Method

A buffer holds a fixed number of samples, and the converter empties it at the sample rate. The time one buffer represents is (buffer-samples ÷ sample-rate) × 1000 milliseconds. Round-trip monitoring latency is at least the input buffer plus the output buffer, so it is modelled here as one-buffer-ms × passes, where passes defaults to 2 (in and out); real interfaces add a small fixed converter/driver overhead on top that this estimate omits. Total samples of delay is simply buffer × passes.

Worked Example

At 48,000 Hz with a 128-sample buffer, one buffer is (128 ÷ 48000) × 1000 ≈ 2.67 ms. With two passes the round-trip estimate is about 5.33 ms (256 samples) - comfortably playable for most monitoring. Drop the buffer to 64 samples and one buffer halves to about 1.33 ms (round-trip ≈ 2.67 ms), tighter but more demanding on the CPU. Conversely a 512-sample buffer at 44,100 Hz is (512 ÷ 44100) × 1000 ≈ 11.6 ms per buffer, roughly 23 ms round-trip - fine for mixing but distracting for tracking a fast part.

Common Mistakes

Quoting only one-buffer latency when the felt delay is round-trip (input plus output).

Forgetting that doubling the sample rate halves latency at the same buffer size - but doubles CPU load.

Ignoring the interface's fixed converter/driver overhead, which adds a real (often 1–3 ms) extra.

Pushing the buffer so low the CPU glitches; clicks and dropouts are worse than a little latency.

Adding software-monitoring plugin latency (look-ahead limiters, linear-phase EQ) which can dwarf the buffer figure.

Tips & FAQ

What latency is acceptable? Under about 5 ms feels immediate; up to ~10 ms is fine for most players; beyond ~20 ms percussive and fast parts feel sluggish. Why not always use the smallest buffer? Small buffers force the CPU to service audio more often, raising the risk of dropouts - use a small buffer while tracking and a large one while mixing. What about direct/hardware monitoring? Many interfaces route the input to the headphones in analogue with near-zero latency, sidestepping the buffer entirely for the performer. Does sample rate help? Yes for latency, no for free - 96 kHz halves buffer latency versus 48 kHz but roughly doubles processing load; treat these numbers as a planning estimate, not a guaranteed spec for your specific interface.

Frequently Asked Questions

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