Audio Latency Test
Lip-sync problems on Bluetooth headphones or video calls almost always trace back to audio latency, and the Audio Latency Test gives you a reaction-based estimate of yours. Click Start Test and tap I Heard It! the moment each of the 8 tones plays, and you'll get an Estimated Latency Range — reported honestly as a range rather than a single false number — alongside any Hardware-Reported base and output latency your browser can access directly. The free hearing test is free to use with no sign-up, and works on both desktop and mobile browsers.
Ever wondered why your Bluetooth earbuds feel slightly out of sync during a clip, or why your interactive headset seems to make every gunshot land a split-second late? Running an audio latency test reveals the exact milliseconds of delay sitting between when a signal is triggered and when you actually hear sound — and that number determines whether your setup is fast enough for what you need it to do. Whether you're a gamer chasing gaming response, a producer monitoring live instruments, or someone troubleshooting lip-sync on a soundbar, understanding your latency data puts you in control.
How the Latency Test Measures Your Audio Delay in Milliseconds
What Is Audio Latency and Why Does It Accumulate in Milliseconds?
Audio latency is the time elapsed between a signal being triggered and the moment it reaches your ears. It is not a single fixed value — it accumulates across every link in your chain: the app generating the signal, the operating system processing stack, the converter changing digital to analogue, any radio transmission layer, and finally the driver physically moving air. A wired 3.5mm output might add only 5–20 ms of total delay, while older SBC wireless encoding can stack up to around 250 ms of accumulation before output reaches you. In studio engineering contexts, even a few extra milliseconds can undermine synchronization between live instruments and monitoring. The stereo left right test is free to use with no sign-up, and works on both desktop and mobile browsers.
What Is the Reported Output Latency Shown in the Web Browser?
Modern browsers expose two estimates through the browser's Web Audio API: baseLatency and outputLatency. The baseLatency value reflects the browser's internal processing buffer — the minimum delay the browser's engine introduces before handing output off to the OS processing stack. The outputLatency value is the browser's best estimate of the delay from internal buffer all the way through the operating system stack and on to the output equipment. Together these give you a useful browser-reported latency baseline, but they cannot see inside wireless over-ear devices — which is exactly why a perceptual test typically reads higher than the reported value on radio-connected devices. The audio output latency shown on this page combines both estimates to give you a system delay reference before you begin the interactive measurement. This reflects underlying device behaviour that the browser can only partially observe, and affects overall browser performance during the test.
How Does This Audio Delay Detection Method Work?
The perceptual slider method used here plays a repeating click and shows a visual flash on a one-second cycle. Because of audio latency, the click arrives after the flash. You drag the perceptual adjustment slider to delay the flash until the simultaneous flash and click feel perfectly aligned — watch the flash in your peripheral vision while listening for the tone for the most accurate result. The slider value at that point is your end-to-end output latency relative to your display baseline.
An alternative approach lets you tap button on beat as a beep plays to calculate latency via a trailing average of the last 10 beats: the tool measures the interval between your tap and the expected play-start time, and you hear sound at the moment you compare that offset. The measured offset represents the gap between when you tap button and when output actually begins. The measured offset calculation uses a modulo formula to handle cases where device performance means latency extends beyond an entire beat duration:
The formula for detecting latency when a missed beat may have occurred is:
Adjusted Latency = Current Latency bmod ms between beatsIn code, this is written as: [current latency] % [ms between beats]
This matters because if your output delay is long enough to span a whole beat, it becomes ambiguous whether you tapped on beat or simply heard the signal one full beat late. The modulo operation strips out complete beat durations, leaving only the fractional beat duration offset — ideally matching your current BPM and your detected BPM readout. The tool shows the interval value alongside your modulo between beats figure so you can verify the two align. A stable countdown timer gives you a starting beat reference so you can settle into rhythm before the measurement locks in.
The round-trip method used by some implementations goes further: a beep is emitted through device speakers and captured by the device microphone, measuring the gap between emission and capture directly to calculate latency end-to-end. To use this approach you need to be in a quiet place, set both input volume and output volume near maximum, and grant microphone access (browser permission) when prompted. Note that not all browsers on all devices will allow microphone access — if the access request is declined, the output-and-capture method will not function. You can also embed iframe the latency widget directly into your own web app using a standard iframe embed. This offline ad-free audio latency widget works without an internet connection in supported browsers, making it a reliable latency tester for any environment. It is also a useful tool for audio development workflows where consistent sdk-level measurement is needed.
Typical Audio Latency by Device — Your Web Browser Audio Latency Test Reference Table
Latency by Connection Type: Wired Headphones vs. Wireless Audio Delay
Your measured latency data should be interpreted against real-world benchmarks. The table below covers typical latency ranges by connection type alongside the practical use cases each latency range is fast enough for. Use wired output first to establish your display and reaction baseline, then switch to a radio-connected device — the difference between the two readings isolates your wireless delay. A bluetooth delay checker or dedicated bluetooth latency test can further drill into encoding-level differences. The frequency sweep test is free to use with no sign-up, and works on both desktop and mobile browsers.
| Connection Type | Typical Latency (ms) | Good Enough For |
|---|---|---|
| Wired 3.5mm / wired speakers | 5–20 ms | Everything: live instruments, professional audio recording, gaming, calls |
| USB Audio (DAC) | 5–15 ms | Studio production, monitoring, singing through monitors |
| HDMI / DisplayPort | 10–40 ms | Home theatre, interactive play, TV latency, broadcasting, clip playback |
| Bluetooth aptX Adaptive / LE Audio | 40–80 ms | Interactive play, clip playback, av sync, live instrument timing |
| Bluetooth aptX / aptX Low Latency | 60–120 ms | Clip playback, casual interactive play, music latency |
| Bluetooth AAC (typical wireless earbuds) | 100–200 ms | Casual listening, most clip playback |
| Bluetooth SBC (older encoding) | 150–300 ms | Listening only — not suitable for interactive play or clip synchronization |
| Wi-Fi / network streaming | 80–200 ms | Casual streaming, internet calls — potential lip-sync error at upper range |
Why Is Bluetooth Audio Latency So High? The Role of Audio Codec and Wireless Transmission
Bluetooth output cannot be sent as a raw waveform. Every packet must go through encoding on the transmitter side, travel over radio as packets (subject to interference and radio-link buffering), then undergo decoding inside your receiving device before the driver can move. Each of those steps adds delay, and the output buffer introduces additional latency to protect against dropped packets. The encoding format chosen determines most of the outcome:
- SBC codec (standard encoding): 150–250 ms — baseline Bluetooth output, widest device compatibility, highest latency
- AAC: ~100–180 ms — common in Apple devices and typical wireless in-ear units
- aptX: ~60–120 ms — Qualcomm standard for Android and many over-ear radio devices
- aptX Low Latency: ~40 ms — purpose-built for interactive play and clip synchronization
- aptX Adaptive and LE Audio: 40–80 ms — next-generation reduced-latency Bluetooth with variable bitrate
- LDAC: optimized for output quality over reduced latency — typically 100–180 ms range
Critically, both the transmitter and the receiving device must match their encoding format for you to benefit from a faster option. If your phone supports aptX Adaptive but your wireless headphones only decode the standard format, you'll get standard-format latency. A dedicated Bluetooth latency test can confirm which format is active on your connection.
How Much Audio Delay Is Acceptable? Thresholds for Gaming, Music Production, and VoIP
Acceptable latency is entirely task-dependent. The key thresholds used in studio and consumer contexts are:
- Under 10ms or lower — professional audio quality. Required for studio production with live instruments, recording vocals, and real-time applications where latency accumulation would disrupt performance. This is the target for USB interfaces and wired converter setups.
- Under 20 ms — required for singing through monitors or playing live instrument parts without perceivable echo or drift.
- Under 40 ms — excellent for interactive response and apps where input lag must feel immediate.
- Under 60 ms — acceptable for most interactive scenarios and clip playback with reasonable av sync.
- Under 80 ms to 100 ms — borderline for clip playback; lip-sync errors become noticeable around that range on a TV or projector.
- Under 150 ms — acceptable for internet calls and remote conferencing; beyond this, conversation feels noticeably delayed.
- Around 200 ms and beyond — fine for casual listening where nothing needs to synchronize, but unsuitable for interactive use or real-time communication. At around 250 ms, sync is clearly broken for any visual media.
Remember: lower is better for every use case. Even if 150 ms is technically acceptable for calls, reducing to around 40 ms will make every conversation feel more natural.
Worked Examples: Interpreting Your Latency Measurement Results
Example 1 — Gamer with Bluetooth headphones: A gamer running the browser audio latency test on a radio-connected headset measures 180 ms. Consulting the table, they can see that 180 ms falls in the standard-encoding range — well above the threshold for interactive response. Switching to wired output drops the measured delay to under 10 ms, confirming the issue is entirely the radio transmission and legacy encoding overhead, not the game or the browser. The measured delay difference — roughly 170 ms — is the real-world cost of wireless output for competitive play.
Example 2 — Music producer checking output latency: A producer opens the latency widget and reads a browser-reported outputLatency of 6 ms on a wired USB interface. Cross-referencing the professional audio threshold of 10ms or lower, the setup passes comfortably for live recording and real-time monitoring. They note the internal buffer value separately to understand how much of that 6 ms is browser buffering versus equipment overhead. This five-minute check becomes part of their standard session startup routine before any recording session.
Example 3 — Video call user on Wi-Fi: A remote worker uses the latency tester on a network connection and reads 120 ms. Referencing the acceptable thresholds, 120 ms falls within the usable range for internet calls — technically under the 150 ms ceiling. However, the same measurement flags a potential lip-sync risk: at around 100 ms, clip playback can already start to feel misaligned. The worker learns that switching to a wired Ethernet connection or enabling a reduced-latency encoding format on their wireless headset could push the measured delay down toward the 60 ms range, improving communication quality noticeably.
Using Your Round-Trip Audio Latency Results to Diagnose and Improve Audio Performance
Troubleshooting When Your Web Browser Returns 0 ms or Shows No Audio Support
If the latency test shows 0 ms or fails to run, the most likely cause is that your web browser does not support the browser's Web Audio API, or that microphone access was declined. Check the following:
- Ensure you are using a modern browser (Chrome, Edge, Firefox, or Safari with Web Audio API support enabled)
- Grant microphone access when the browser permission prompt appears — without it, the output-and-capture method cannot measure round-trip latency
- Confirm the device microphone is not muted at the OS level and that input volume is set near maximum
- Test in a quiet place to prevent ambient noise from confusing the capture detection
- If your browser reports 0 ms for the output latency value, this may indicate the browser has not yet implemented the full latency reporting spec — use the perceptual slider test as your primary audio latency measurement instead
Some older mobile browsers and restricted environments block microphone access entirely. In these cases, switching to a desktop browser or one optimized for output (such as Chrome with web performance settings enabled) is the most reliable fix. The indicator returned by these browsers may also reflect only part of the processing stack — the OS-level components below the browser may add additional unmeasured delay, affecting overall browser performance.
Related Hearing and Audio Tools for Complete Audio Diagnostics
Your setup involves more than latency alone. Once you have your latency data, these related tools help you build a complete picture of your perception and output performance:
- Hearing Test — find out how high you can hear across the full audible frequency range
- Hearing Age Test — assess how old your ears are based on high-frequency sensitivity
- Mosquito Tone Test — a high-frequency age test using tones most adults cannot perceive
- Subwoofer Test — test bass response down to 1 Hz to verify your subwoofer or output drivers
- Tinnitus Matcher — identify the pitch and character of your tinnitus
- Vocal Range Test — discover your vocal range and voice type
- AV Sync Test — calibrate lip-sync on a TV, soundbar, projector, or monitor for accurate synchronization
- Bluetooth Latency Test — isolate wireless delay and identify which encoding format your connection is using
Together these tools cover the full spectrum of diagnostics — from acoustics and perception to encoding-level latency measurement and output optimization — so you can trust that every part of your setup is performing as it should.
Frequently Asked Questions
- What is audio latency?
- Audio latency is the delay between when a sound is triggered and when you actually hear it through your output device. It is measured in milliseconds (ms). Lower latency is better, especially for live instruments, gaming, and professional audio production.
- How does this audio latency test work?
- Enter your BPM and connection type. The tool calculates the millisecond gap between each beat at that tempo, then compares your connection's typical latency against that window. If you have a measured latency figure, you can enter it to see how it stacks up against your connection type's expected range.
- How much latency is acceptable?
- For professional audio production and live instruments, 10 ms or lower is ideal. For gaming and video, under 80 ms is generally acceptable. For casual music listening, even 150–300 ms (common with SBC Bluetooth) is tolerable. The acceptable threshold depends heavily on the use case.
- Why is Bluetooth audio latency so high?
- Bluetooth audio introduces latency due to wireless encoding, decoding, and buffering processes. Older codecs like SBC can add 150–300 ms, while newer codecs like aptX Adaptive or LE Audio have reduced this to 40–80 ms. Wired connections bypass this entirely and typically measure under 20 ms.
- What does 'milliseconds per beat' mean in this context?
- At any given BPM, each beat spans a fixed number of milliseconds (ms per beat = 60,000 ÷ BPM). If your audio latency is a large fraction of that window, you will perceive a noticeable delay between a trigger and the sound — especially critical at higher tempos.
- What is the difference between output latency and round-trip latency?
- Output latency measures the delay from triggering a sound to hearing it. Round-trip latency includes both output and input (microphone capture), measuring the full loop. Round-trip latency is what tools like the Superpowered Web Latency Test measure using a speaker-to-microphone beep test.
- What BPM should I use for the latency test?
- Use the BPM of the music or metronome you are working with. A common reference is 120 BPM (500 ms per beat). Higher tempos make latency more perceptible because the ms window per beat is smaller, so even moderate delays become a noticeable fraction of the beat.
- Which connection type has the lowest audio latency?
- Wired connections (3.5mm jack or USB audio) consistently offer the lowest latency, typically 5–20 ms. Among Bluetooth codecs, aptX Adaptive and LE Audio are the best, ranging from 40–80 ms. Older SBC Bluetooth is the worst, often adding 150–300 ms of delay.