Online Audio Latency Tester Measure Input Delay in Ms Free

The tool uses the Web Audio API to play a precise click tone through your output device, then records through the microphone to detect when that click arrives. The elapsed time between the click being scheduled for output and the moment the microphone signal exceeds the detection threshold is your round-trip latency. All audio processing is done entirely within your browser — no sound data is transmitted, uploaded, or stored anywhere. No registration is needed and it's completely free to use.

Under 20 ms is excellent for professional recording and competitive gaming. 20–50 ms is good for gaming, streaming, Discord and podcast recording. 50–100 ms is acceptable for video calls and casual use. Above 100 ms is noticeably delayed and will cause problems in most applications. For music production with live monitoring, even 20 ms feels slightly delayed — professionals often target 10 ms or below using low-buffer ASIO setups.

Bluetooth audio requires encoding the raw audio into a compressed codec (SBC, AAC, aptX, LDAC), transmitting it wirelessly, and decoding it on the receiving end. Each step adds processing delay. SBC — the default Bluetooth codec — adds 150–300 ms. AAC adds 60–120 ms. aptX adds 40–80 ms. aptX Low Latency can achieve 30–40 ms. aptX Adaptive varies from 30–80 ms. USB audio, by contrast, transfers raw PCM audio data digitally with minimal encoding overhead, achieving 5–35 ms total latency depending on buffer settings.

Jitter is the variation in timing between successive audio samples or measurements. High jitter means your latency is inconsistent — sometimes 20 ms, sometimes 60 ms — which causes audio to sound choppy or uneven. In music production, high jitter introduces timing artifacts and makes recordings difficult to edit. In VoIP calls, it causes clipping and gaps. In gaming, it makes positional audio cues unreliable. Our tool calculates jitter as the statistical standard deviation across your last 10 measurements. For professional use, target jitter under 3 ms.

Buffer latency = (Buffer Size ÷ Sample Rate) × 1000 ms. At 48kHz, a 512-sample buffer adds 10.7 ms per buffer pass. Round-trip requires at least two passes (input + output), so minimum buffer contribution is ~21 ms. Reducing to 128 samples brings this to ~5.3 ms round-trip buffer latency. However, smaller buffers require the CPU to process audio faster — if it can't keep up, you get dropouts and crackling. The Buffer Analysis mode in this tool shows your exact buffer configuration and theoretical minimum latency automatically.

Browser-based measurement is accurate for identifying latency categories and comparing devices, but has some limitations for absolute precision. The Web Audio API scheduling has 1–5 ms overhead and browsers add their own audio buffers. Chrome gives the best results due to AudioWorklet optimization. Measurements within 5 ms of each other should be treated as equivalent. For sub-1 ms precision in professional settings, a dedicated DAW with ASIO drivers is needed. This tool excels at relative comparison — testing USB vs Bluetooth, or before/after driver changes — and identifying whether you're in the excellent, good, or high-latency range.

Yes — USB audio interfaces like Focusrite Scarlett, PreSonus AudioBox, Universal Audio Volt, SSL 2, Audient iD, Behringer UMC and others all appear as standard audio devices in your browser. Select your interface as both the input and output device, enable loopback if supported (or use a cable from output to input), and run the round-trip test. Interfaces typically show 4–20 ms total latency in Chrome. Our breakdown panel shows how much of that is buffer vs system latency.

Each browser implements the Web Audio API differently. Chrome achieves the lowest latency — typically 10–30 ms round-trip — using aggressive AudioWorklet scheduling with low-latency hints. Firefox adds a compatibility buffer layer, typically resulting in 20–50 ms. Safari on macOS connects directly to Core Audio but applies security processing overhead, typically 20–45 ms. Edge (Chromium-based) performs similarly to Chrome. Brave and Opera also follow Chrome's implementation. For the lowest latency from a browser environment, use Chrome with a wired USB microphone for best results.

Input latency (one-way) is the delay from a physical sound entering the microphone to the digital audio appearing in software — typically measured at 5–50 ms. Round-trip latency is the total delay from output speaker, through the air, captured by the microphone, and back to software — typically double input latency. What you perceive when monitoring yourself through headphones while recording is round-trip latency. Our tool primarily measures round-trip because this reflects real-world experience, though you can estimate one-way input latency as approximately half the round-trip figure.

This tool is completely safe and private. All audio analysis runs entirely inside your browser using the Web Audio API. Your microphone audio stream never leaves your device — no audio data, recordings, or personal information is transmitted to any server at any time. The tool only stores millisecond timing numbers locally in memory for the current session to calculate statistics. No login, no account, no cookies, and no registration of any kind is required. When you close the tab, all data is cleared and microphone access is fully revoked.