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If audio gradually drifts away from the picture during a 24/7 lecture stream, measure the offset at playback and compare it with the stream before and after distribution. That helps locate whether the drift is already present in capture or encoding, or appears later in ingest, transport, transcoding, or playback. A constant offset, a steady drift, and a sudden jump are different symptoms—not proof of any one cause.
First identify how the sync error changes
Note when the stream starts, when you first notice a mismatch, and the approximate offset at each check. Repeat the observation after a restart only if doing so is safe for the lecture. The pattern narrows the investigation; it does not diagnose the cause by itself.
| What you observe | What it suggests | Next check |
|---|---|---|
| The same offset is present from the start and stays about the same | A fixed offset is different from accumulating drift. It may be introduced before playback or by the receiver. | Compare the source or local recording with the received stream at the same point in the lecture. |
| The mismatch grows over time | One or more clocks, timestamps, or processing stages may not be staying aligned. | Measure playback drift, then compare the encoded output with the received copy. |
| Sync is correct, then suddenly changes | A discrete event may have changed timing or playback state. | Record when the jump occurs and check logs, reconnects, dropped or duplicated frames, and receiver behavior around that moment. |
These are diagnostic patterns, not universal cause-and-effect rules. Without the actual stream and logs, the symptom alone cannot establish which stage is responsible.
Measure sync at playback
Use a test receiver where possible
When you can do so without disrupting the live lecture, inspect a test copy or a separate receiver. FFmpeg’s ffplay documentation describes the -stats display, which includes audio/video synchronization drift, and explains that the master clock is used to control audio-video synchronization. For a local input, run ffplay -stats -i input and observe whether the displayed drift changes over time. The readout describes that playback session; by itself, it does not prove the encoder is at fault.
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Keep the observation comparable
Use the same receiver, playback route, and measurement method for before-and-after checks. Write down the start time, the observed offset, and whether it is stable, growing, or jumping. A single check can reveal a mismatch, but it cannot establish a drift rate.
Compare the source, encoded output, and received stream
Check the same passage at multiple points, if your workflow allows it: the source or capture, the encoded output before distribution, and the received playback. This comparison is the practical way to separate a problem already present in production from one that appears farther downstream.
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- If the local recording or encoded output already drifts, investigate capture timing, audio processing, and encoding.
- If the encoded output stays in sync but the distributed version drifts, investigate ingest, transport, transcoding, and the player.
- If different receivers behave differently, compare their playback path and logs before changing the encoder.
This localization is a troubleshooting inference, not a guarantee that a particular stage is responsible in every setup. OBS documents how its audio and video processing and packet output work in its backend design; the actual chain and receiver determine what evidence is available.
Check OBS timestamps and audio configuration
Look for coherent timestamps
When your tools expose them, inspect presentation timestamps (PTS), decode timestamps (DTS), and the timebase for both audio and video over the run. OBS’s encoder API reference documents these packet fields. OBS describes its output as interleaving encoded audio and video packets in monotonic timestamp order in its backend design documentation. Look for timestamps that remain coherent and ordered throughout the stream; do not infer a fault from one isolated packet or from a player statistic alone.
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Verify audio format through the whole chain
Confirm the audio source’s sample rate and channel layout against OBS’s configured audio backend and the output expected by the destination. OBS documents that it can remix or resample when sample rate or channel count differs from the configured backend. Its audio/video formats guide covers format compatibility. Check the source, OBS, encoder, and destination settings rather than assuming the audio passes through unchanged.
Use load and frame data as context
Check CPU or encoder load and any dropped or duplicated video-frame counters alongside the sync measurements. OBS’s backend documentation describes a video queue that can duplicate a frame when full, while encoded packet output is interleaved separately by timestamp. A dropped or duplicated frame is useful context, but it does not by itself explain gradually accumulating audio drift; inspect the actual output.
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Match the fix to the stream transport
Do not transfer a latency or codec setting from one protocol to another. Use the destination’s ingest requirements and the guide for the transport actually in use.
SRT
For SRT, OBS’s SRT streaming guide lists a 120 ms default latency and recommends configuring latency to at least 2.5 times the round-trip time between encoder and ingest server. Measure that RTT and treat the recommendation as SRT-specific guidance, not a general audio-sync repair. After changing latency, remeasure end-to-end drift.
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WHIP/WebRTC
For WHIP/WebRTC, check the ingest provider’s supported OBS version, audio and video encoders, and frame-rate requirements. OBS’s WHIP streaming guide, dated January 14, 2025, describes its WHIP workflow. Cloudflare’s WebRTC guide gives one provider-specific example using OBS 31.0 or later, Opus audio, and 30 fps video. Those example settings are not universal requirements for other services.
RTMP
For RTMP, use the destination’s own ingest requirements and inspect the encoded result. FFmpeg documents RTMP separately as a multimedia streaming protocol over TCP/IP, with publishing and playback examples in its protocols documentation. That protocol description does not identify the cause of sync drift in a particular stream.
Change settings methodically
- Record the current configuration and the measured sync at each observation point.
- Choose one setting relevant to the evidence—for example, an audio format mismatch or SRT latency that is below the documented guidance for measured RTT.
- Change only that setting, then repeat the same measurements with the same receiver and route.
- Compare whether the drift rate or offset changed. If it did not, restore the original value before testing another setting.
- Keep monitoring long enough for the relevant operating period. One improved session does not establish a durable correction for a continuous broadcast.
Avoid restarting as the first response if doing so would interrupt a lecture and you have not established that a restart addresses the fault. If a restart is necessary, capture the timing, logs, and measurements beforehand where possible; otherwise, useful evidence may disappear.
Common troubleshooting outcomes
- ffplay shows changing drift, but the encoded output is stable: compare another receiver and inspect ingest, distribution, transcoding, and player behavior before changing encoder settings.
- The local recording and outgoing stream both drift: focus on source timing, audio format, processing, and timestamps in the production path.
- Audio format settings differ across components: align the source, OBS backend, encoder, and destination expectations, then measure again. OBS may remix or resample mismatched input.
- SRT latency is below the measured-RTT guidance: test an appropriate SRT latency change and verify the resulting playback; do not apply the SRT value to RTMP or WHIP.
- Only one player or receiver is affected: compare its playback and clock behavior with a second receiver before disrupting a stable production stream.
- The symptom remains unexplained: collect the source hardware, sample rate and channel layout, OBS and encoder settings, transport, ingest service, player, logs, and measurements at each pipeline point. Those details are needed for a case-specific diagnosis.
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For any rebroadcast, confirm that you have the rights to the lecture and that the content complies with YouTube’s applicable rules.
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