Convert buffer samples to milliseconds, compare input and output delay, and estimate round-trip latency for a DAW and audio-interface signal path.
Audio settings
Latency result
Estimated round-trip latency5.333 ms
256 samples total at 48 kHz, including 0 ms additional overhead
Input buffer delay2.667 ms
Output buffer delay2.667 ms
Buffer-only round trip5.333 ms
Estimated total samples256 samples
Samples per millisecond48 samples/ms
Lowest buffer service rate375 times/s
Important: 5.333 ms is the two-buffer mathematical result, not a measured analog loopback value. Real round-trip latency may be higher.
How to use this audio latency calculator
Match the interface sample rate. Enter the actual rate used by both the audio driver and DAW. If they differ, one device may resample or fail to connect as expected.
Enter the I/O buffer. Most setups use the same buffer in both directions. Enable the separate output field only when the driver or host reports different input and output sizes.
Add overhead only when known. A driver panel or DAW may report input and output latency beyond the raw buffers. A loopback test is better than guessing.
Use the buffer-only result as a floor. It is useful for comparing settings. The estimated total becomes more realistic only when the additional latency field reflects the actual signal chain.
Test stability. Reduce buffer size for live monitoring or software instruments, but raise it if the session clicks, pops, or drops audio.
Hardware direct monitoring bypasses the DAW round trip. For that route, this two-buffer model does not apply; converter and internal mixer/DSP latency are the relevant figures.
Audio buffer latency formula
Sample rate is the number of audio sample frames processed per second. A buffer containing B samples at sample rate fs spans B ÷ fs seconds:
Example: 128 input samples and 128 output samples at 48,000 Hz produce 2.6667 ms per direction and 5.3333 ms buffer-only round trip. If a measured or reported 3.2 ms of additional hardware and driver delay applies to the complete path, the estimate becomes 8.5333 ms.
The model assumes one entered buffer per direction. Actual drivers may use safety buffers or expose different reported latencies. Converter filters, USB or network transport, operating-system mixing, plug-in look-ahead, convolution, DSP, and wireless links can all add delay. Some DAWs compensate recorded material on the timeline, but that does not remove the delay a performer hears while monitoring through the software path.
Common buffer sizes at 44.1, 48, and 96 kHz
These are one-buffer durations. Double a value for a simple input-plus-output buffer round trip using equal buffers.
Buffer
44.1 kHz
48 kHz
96 kHz
Practical trade-off
32 samples
0.726 ms
0.667 ms
0.333 ms
Very short service interval; high CPU scheduling demand.
64 samples
1.451 ms
1.333 ms
0.667 ms
Common low-latency target for tracking on a capable system.
128 samples
2.902 ms
2.667 ms
1.333 ms
Often a workable balance between monitoring delay and stability.
256 samples
5.805 ms
5.333 ms
2.667 ms
More processing time, with increasingly noticeable software monitoring delay.
512 samples
11.610 ms
10.667 ms
5.333 ms
Useful for heavier mixing sessions where live input response is less important.
1,024 samples
23.220 ms
21.333 ms
10.667 ms
Large stability margin, but unsuitable for many real-time performance tasks.
There is no universal “best” buffer. Project load, driver quality, CPU scheduling, plug-ins, interface connection, and monitoring method determine the lowest stable value.
Audio latency FAQ
How do I calculate audio latency from buffer size?
Divide buffer samples by the sample rate in samples per second, then multiply by 1,000. At 48 kHz, a 128-sample buffer is 128 ÷ 48,000 × 1,000 = 2.667 ms.
How do I estimate round-trip audio latency?
Add input buffer time and output buffer time, then add known hardware, driver, transport, safety-buffer, and processing delay. A physical loopback measurement is the better way to find the complete real system value.
Why does my DAW report more latency than the buffer formula?
The DAW or driver may include A/D and D/A conversion, interface and stream latency, extra safety buffers, operating-system processing, transport, and plug-in delay. Reported values can also differ in accuracy between devices.
Does a higher sample rate reduce latency?
At the same buffer size in samples, yes. Doubling sample rate halves the time represented by that buffer. The trade-off is more samples to process and transfer each second, which can increase CPU and interface load.
Should I always choose the smallest buffer?
No. A smaller buffer gives the computer less time to finish each block. Choose the lowest setting that stays stable for the current project; raise it if you hear crackles, pops, or dropouts.
Is hardware direct-monitoring latency included?
No. Direct monitoring normally avoids the trip through the DAW buffers. Consult the interface specifications or measure the direct-monitor route separately.
Are my settings uploaded or saved?
No. The calculation, clipboard report, and CSV file are created locally by JavaScript in your browser. The page does not upload or store the numbers you enter.
Method and references
Formula and assumptions checked 2 August 2026. The equation follows Ableton’s explanation of expected latency as buffer samples divided by sample rate. Apple defines sample rate as frames per second and notes that stream and device latency may need to be summed. Microsoft’s low-latency audio documentation shows that buffers are only part of the capture and render paths.
