Latency Test
Lower = more sensitive. Increase if getting false triggers.
Setup Instructions
- Position your microphone near your speakers (or use headphones with a mic)
- Ensure quiet environment to avoid false triggers
- Click "Run Test" — a click sound will play
- The mic will detect when it hears the click
- Latency is calculated from the time difference
Tip: For accurate results, the microphone should be able to hear the speakers. If using headphones, hold the mic near an earcup, or use external speakers.
Understanding Latency
| Latency (ms) | Quality | Use Case |
|---|---|---|
| < 10 ms | Excellent | Professional recording, live performance |
| 10-20 ms | Good | Music production, real-time monitoring |
| 20-40 ms | Acceptable | General music production, gaming |
| 40-100 ms | Noticeable | Casual recording, video calls |
| > 100 ms | High | Problematic for real-time audio |
What This Measures
This tool measures round-trip latency: the time from when audio is sent to your speakers until it's received back by your microphone. This includes:
- Output buffer (audio interface → speakers)
- Acoustic travel time (speakers → microphone)
- Input buffer (microphone → audio interface)
- Processing latency in your browser/OS
Tips to Reduce Latency
- Use a dedicated audio interface instead of built-in sound card
- Lower buffer size in your audio settings (e.g., 128 or 256 samples)
- Use ASIO drivers on Windows, Core Audio on Mac
- Close other audio applications
- Disable audio enhancements/effects in system settings
About This Tool
Audio Latency Tester is a free browser-based tool that measures the round-trip delay in your audio setup. It plays a short click through your speakers, listens for that click on your microphone, and reports the elapsed time in milliseconds.
This measurement includes every stage of the audio path: the output buffer (your system sending audio to the speakers), acoustic travel time (sound moving through air), the input buffer (your microphone capturing the sound), and browser/OS processing overhead.
Worked Example
A musician connects a Focusrite Scarlett 2i2 audio interface to their laptop, sets the buffer to 128 samples at 44.1 kHz, and runs 5 tests. Results: 14 ms, 15 ms, 13 ms, 15 ms, 14 ms. Average: 14.2 ms. This falls in the "Good" range — comfortable for tracking vocals or guitar with real-time monitoring.
The same musician then switches to the laptop's built-in sound card with default drivers. Results jump to 45-60 ms — clearly in the "Noticeable" range and uncomfortable for live monitoring. The difference is the audio interface's low-latency ASIO/Core Audio driver versus the generic system driver.
Buffer Size vs. Latency Reference
| Buffer (samples) | At 44.1 kHz | At 48 kHz | At 96 kHz |
|---|---|---|---|
| 64 | 1.5 ms | 1.3 ms | 0.7 ms |
| 128 | 2.9 ms | 2.7 ms | 1.3 ms |
| 256 | 5.8 ms | 5.3 ms | 2.7 ms |
| 512 | 11.6 ms | 10.7 ms | 5.3 ms |
| 1024 | 23.2 ms | 21.3 ms | 10.7 ms |
| 2048 | 46.4 ms | 42.7 ms | 21.3 ms |
These are one-way figures. Round-trip latency is roughly double (input + output buffers), plus acoustic travel time and driver overhead.
Privacy and Limitations
All processing runs in your browser using the Web Audio API. No audio is recorded, stored, or sent to any server. Microphone access is used only to detect the test click — no audio data leaves your device. Results are cleared when you refresh the page.
This tool measures round-trip latency through speakers and microphone. It does not measure internal DAW latency or driver-reported buffer latency. Results include acoustic travel time (roughly 0.3 ms per 10 cm of distance between speaker and mic). For the most consistent readings, keep the mic close to the speaker and run at least 5 tests.
Audio Latency Tester FAQ
What is audio latency?
Audio latency is the delay between when a sound is produced and when you hear it through your system. In a digital audio setup, latency comes from output buffers, DA/AD conversion, acoustic travel time, input buffers, and software processing. It is measured in milliseconds (ms). Under 10 ms is excellent for live performance; 10–20 ms is good for music production; above 40 ms is noticeable and problematic for real-time monitoring.
How does this audio latency tester work?
The tool plays a short click sound through your speakers using the Web Audio API, then listens on your microphone for that click. It measures the time between sending the click and detecting it on the mic input. This gives you the round-trip latency — output buffer plus acoustic travel plus input buffer plus browser processing. Run multiple tests and take the average for a reliable number.
What is a good audio latency for music production?
For music production with real-time monitoring, aim for under 10 ms round-trip latency. At 10–20 ms, most musicians can still play comfortably. Above 20 ms, you may notice a delay when monitoring through headphones. Above 40 ms, the delay becomes distracting for most performers. If your DAW reports 5 ms input and 5 ms output, your round-trip latency is roughly 10 ms.
Why is my audio latency so high?
Common causes of high audio latency: large buffer sizes (512+ samples), built-in sound cards with generic drivers, background audio processing or effects, other applications using the audio device, and Bluetooth audio devices which add 40–200 ms. To reduce latency, use a dedicated audio interface with ASIO (Windows) or Core Audio (Mac) drivers, lower your buffer size to 128 or 256 samples, and close other audio applications.
Does this tool work with Bluetooth headphones?
Bluetooth headphones will show much higher latency than wired audio — typically 40–200 ms or more depending on the codec (SBC, AAC, aptX, aptX LL). This is expected and reflects the actual delay you experience. For accurate latency testing of your audio interface, use wired speakers or headphones.
What is the difference between buffer size and latency?
Buffer size is the number of audio samples processed in one batch. Latency depends on buffer size and sample rate. At 44.1 kHz, a 128-sample buffer adds about 2.9 ms of latency; 256 samples adds 5.8 ms; 512 samples adds 11.6 ms. The formula is: latency (ms) = buffer size / sample rate × 1000. Round-trip latency is roughly double because you have both input and output buffers.
How do I reduce audio latency on Windows?
On Windows, install ASIO drivers for your audio interface (or ASIO4ALL for built-in sound cards). In your DAW or audio application, select the ASIO driver and lower the buffer size to 128 or 256 samples. Disable audio enhancements in Windows Sound settings. Close other applications that use the sound device. A dedicated USB or Thunderbolt audio interface typically achieves 5–10 ms round-trip latency.
Does this tool store any data or recordings?
No. All audio processing runs locally in your browser using the Web Audio API. No audio is recorded, stored, or sent to any server. Microphone access is only used to detect the click sound. Refreshing the page clears all test results.
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