How is buffer latency calculated?

Buffer latency in milliseconds equals buffer size in samples divided by sample rate in Hz, times 1000. A 128-sample buffer at 48000 Hz is 128/48000 times 1000, which is 2.67 ms for one buffer pass.

What is round-trip latency?

Round-trip latency is the total delay from a sound entering the interface to the processed sound coming back out. It includes the input buffer, the output buffer, and analog-to-digital plus digital-to-analog converter delay. A common approximation is twice the buffer latency plus a small fixed converter overhead.

Does a higher sample rate reduce latency?

Yes, for the same buffer size in samples. At 96000 Hz a 128-sample buffer is 1.33 ms versus 2.67 ms at 48000 Hz, because each sample represents less time. The trade-off is roughly double the CPU load and disk throughput.

What buffer size should I use for recording versus mixing?

Use a small buffer (64 to 128 samples) when recording or monitoring through plugins, so latency stays under about 6 ms and feels immediate. Switch to a large buffer (512 to 1024 samples) when mixing, where latency does not matter but you need the CPU headroom for many plugins.

At what point does latency become noticeable?

Most players notice round-trip latency above roughly 10 ms, and it becomes disruptive above 20 to 30 ms, similar to standing a few metres from a speaker. Under about 6 ms feels effectively instant. Singers and percussionists are the most sensitive.

What is the Audio Buffer Size & Latency Calculator?

Compute input, output, and round-trip audio latency in milliseconds from buffer size (64 to 4096 samples) and sample rate. Helps producers and live performers choose the right ASIO or Core Audio buffer size for low-latency monitoring without audio dropouts. It runs free in your browser on Gera Tools, with nothing uploaded.

Audio Buffer Size & Latency Calculator

Name: Audio Buffer Size & Latency Calculator
Creator: Gera Tools
License: https://creativecommons.org/licenses/by/4.0/

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This calculator turns your audio interface buffer settings into the latency you will actually feel when monitoring through your DAW. Enter the buffer size and sample rate and read the one-way and round-trip delay in milliseconds.

How it works

A digital audio system processes sound in blocks called buffers. The time one buffer represents is:

buffer latency (ms) = (buffer size in samples / sample rate in Hz) x 1000

So a 256-sample buffer at 44100 Hz holds 256 / 44100 x 1000 = 5.8 ms of audio.

Round-trip latency

When you monitor through software you pay this cost twice — once on the way in and once on the way out — plus a fixed delay from the AD and DA converters:

round trip (ms) = 2 x buffer latency + converter overhead

The converter overhead is typically about 1 to 2 ms on modern interfaces. This tool uses a representative 1.5 ms to give a realistic estimate; your interface’s reported figure may differ slightly.

Choosing a buffer size

Tracking or live monitoring: use 64 to 128 samples so round-trip latency stays under ~6 ms and playing feels immediate.
Mixing and mastering: use 512 to 1024 samples; latency is irrelevant once you stop recording, and the larger buffer frees CPU for plugins.

Worked example

At 48000 Hz with a 128-sample buffer, one buffer is 2.67 ms. Round-trip is roughly 2 x 2.67 + 1.5 = 6.83 ms — low enough for comfortable real-time monitoring. Drop to a 64-sample buffer and it falls to about 4.2 ms, but the CPU works twice as hard per second.

All calculations run locally in your browser.

Buffer size quick-reference table

Buffer (samples)	44,100 Hz	48,000 Hz	96,000 Hz
32	0.73 ms	0.67 ms	0.33 ms
64	1.45 ms	1.33 ms	0.67 ms
128	2.90 ms	2.67 ms	1.33 ms
256	5.80 ms	5.33 ms	2.67 ms
512	11.6 ms	10.7 ms	5.33 ms
1024	23.2 ms	21.3 ms	10.7 ms

These are one-way buffer latencies. Add converter overhead and double for a round-trip estimate.

Why round-trip latency matters more than buffer latency alone

Buffer latency is what the math gives you. Round-trip latency is what you actually feel as a performer monitoring through software. The round trip includes input buffering, processing, output buffering, and the physical analog-to-digital plus digital-to-analog conversion that the converter hardware performs independently of the buffer setting.

Most modern audio interfaces add roughly 1 to 3 ms of fixed converter overhead that does not change regardless of how small you make the buffer. This puts a practical floor on round-trip latency even if you set your buffer to the minimum supported value. Running a 32-sample buffer at 96 kHz gives a buffer latency of just 0.33 ms, but the converter overhead means the performer still feels 2 to 4 ms — which is perfectly acceptable but not quite as dramatic as the raw buffer number implies.

CPU considerations

Smaller buffers do not just reduce latency — they increase the rate at which your processor must service audio interrupts. A 64-sample buffer at 48 kHz means the CPU must deliver fresh audio approximately every 1.33 ms, or roughly 750 times per second. At 1024 samples the same rate drops to about 47 times per second, which is why mixing sessions with many plugins run far more stably at larger buffers. A good workflow is to record at the smallest buffer that keeps your system stable, then increase the buffer when you move into mixing mode.