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Fix the driver behind crashes, sound loss and screen glitchesFind Drivers →Repair Windows errors before they cause bigger problemsFix Now →Scan for outdated or missing drivers - takes under a minuteDriver Scan →There is no universal FFmpeg flag that fixes every audio/video sync problem. First determine whether the stream has a steady offset, progressive drift, or sudden jumps; then inspect timestamps, choose the right video frame-sync mode, and test the complete setup before going live. A constant offset and a growing timing error have different causes—and need different remedies.
Diagnose the kind of sync problem first
Write down your FFmpeg version and build, input types, full command, and whether audio and video come from one file or device or from independent sources. Then describe what you observe:
- Steady offset: audio consistently leads or trails video by about the same amount.
- Progressive drift: the offset grows during the stream. This can point to different source clocks or timing errors that accumulate over time.
- Sudden jumps: sync changes after a reconnect or interruption; inspect timestamp discontinuities and reconnect behavior.
- Intermittent stutter: playback may be dropping or buffering data rather than suffering a simple fixed offset.
These symptoms are clues, not diagnoses. Check both the FFmpeg input/output timing and playback before changing options.
Inspect timestamps before adding timing options
Review FFmpeg’s input stream information and logs for missing, irregular, discontinuous, or non-monotonic timestamps. Compare what the input reports with what the output reports. If the sources have separate clocks, note that too: a timestamp offset can correct measured steady lead or lag, but it cannot correct an error that keeps growing.
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FFmpeg’s documentation describes timestamp and muxer options, but which adjustment is appropriate depends on the source and the timing assumptions in your command. Do not add -copyts, wall-clock timestamping, timestamp offsets, or asynchronous resampling as generic fixes. Each changes how timing is interpreted or handled; validate any change against the actual input.
Use -re only when pacing a media file
For a file being sent as though it were live, -re reads the input at its native frame rate. It is principally useful for pacing file input in a live-output workflow; it does not synchronize independent audio and video clocks.
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Do not apply it reflexively to a real-time capture device or live network input. FFmpeg warns that a low read rate on an actual live source can cause packet loss. Check the FFmpeg manual for the behavior of -re and -readrate in your build.
Choose video frame-sync behavior deliberately
The output -fps_mode setting governs video frame timing; it is not, by itself, an audio-clock correction. FFmpeg documents these modes:
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| Mode | Effect on video frames and timestamps | When to consider it |
|---|---|---|
passthrough |
Passes demuxer timestamps through. | When you want to retain the input timing rather than force a target frame rate. |
cfr |
Duplicates or drops frames to reach the requested constant frame rate. | When the output needs a constant frame rate and the source can tolerate frame duplication or removal. |
vfr |
Passes timestamps through or drops frames to avoid duplicate timestamps. | When variable timing is acceptable but duplicate output timestamps should be avoided. |
The muxer may subsequently modify timestamps. Consult the FFmpeg documentation and verify output behavior rather than assuming a frame-sync mode will solve a growing audio/video mismatch.
Measure playback and test the full path
FFplay can report audio/video synchronization drift and identify the master clock it is using. Use its documentation to interpret those statistics; playback behavior is useful evidence, but it is not a substitute for checking input and output timing.
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Before relying on the setup live, run a test stream for a representative duration using material with both speech and movement. YouTube recommends testing with similar audio and motion, then monitoring stream health and event messages during the event. A short test may not reveal a timing error that accumulates over time.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Keep YouTube encoder compliance separate from sync diagnosis
YouTube’s live encoder guidance describes delivery settings, not a cure for source-clock drift. It recommends CBR bitrate encoding, AAC or MP3 audio, and a 2-second keyframe frequency (not exceeding 4 seconds). Its stated sample-rate guidance is 44.1 kHz for stereo and 48 kHz for 5.1 surround. Follow the platform guidance for a compatible stream, but do not treat those settings as proof that audio and video clocks are synchronized.
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What to do when the sources use separate clocks
If audio and video come from independent clocks and the mismatch progressively grows, investigate whether both can be captured against a common clock. If not, the remedy may require source-specific resampling or rate correction. The right configuration depends on the devices, capture path, timestamps, and FFmpeg command; the available documentation does not establish one universal command for this case.
Common troubleshooting mistakes
- Applying a fixed offset to progressive drift: measure whether the gap grows. A fixed offset only addresses a stable lead or lag.
- Using
-reon a live device or network input: it is intended chiefly for pacing file input; low read rates can cause packet loss on live sources. - Forcing CFR to repair audio timing: CFR changes video frame delivery by duplicating or dropping frames. It does not synchronize an independent audio clock.
- Changing several timestamp options at once: make one evidence-based change, then compare logs and playback in a representative test.
- Assuming YouTube’s encoder settings cure drift: bitrate mode, codec, sample rate, and keyframe interval address delivery compatibility, not necessarily source timing.
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