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One free scan finds every outdated or missing driver and matches the right update for your exact hardware.Free scan · exact hardware matchIf audio drifts farther out of sync with video as a 24/7 lecture stream runs, measure the problem at playback and compare it with the encoded output before changing settings. A fixed offset, steadily increasing drift and sudden jump are different symptoms, but none identifies the cause by itself. Trace the stream through capture, audio processing, encoding, timestamps, transport and playback to find where the offset begins.
First identify what kind of sync problem you have
Write down when the stream starts, when the mismatch first becomes noticeable and the approximate offset at each check. If safe, repeat the observation after a restart. Do not treat a restart as a fix until you know whether it changes the drift over a sufficiently long run.
- Constant offset: audio and video are out of sync by roughly the same amount over time.
- Gradual drift: the mismatch increases as the stream runs.
- Sudden jump: sync changes abruptly, possibly around a reconnect or other event.
These patterns help narrow where to investigate, but they do not prove whether capture, encoding, transport or playback is responsible.
Measure drift at playback
Use a non-disruptive test receiver
When possible, monitor a test copy or a separate receiver rather than interrupting the lecture that viewers are watching. FFmpeg’s ffplay documentation describes a -stats display that includes audio/video synchronization drift, along with synchronization-clock and frame-drop controls. For a local test file, run:
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ffplay -stats lecture.mp4
For a live test, use the actual input URL or stream source supported in your setup. Record the displayed drift at consistent intervals. The reading tells you what the player observes; by itself, it does not prove the encoder caused the problem. FFmpeg explains that “The master clock is used to control audio-video synchronization.” That describes ffplay’s clock behavior, not a universal explanation of every streaming player.
Find the stage where sync starts changing
Compare capture, encoded output and received playback
Compare the source or capture, the encoded stream before distribution where available, and playback after the destination’s ingest and delivery path. If sync is already changing in a local recording or encoded output, focus first on capture and encoding. If that output remains stable but the distributed stream drifts, investigate ingest, transport, any transcoding stage and the player. This comparison localizes the fault domain; it cannot name a cause without the actual stream and logs.
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Inspect timestamps and packet order
When your tools expose them, inspect presentation timestamps (PTS), decode timestamps (DTS) and the timebase for both tracks throughout the run. The OBS encoder API reference documents these packet fields. OBS’s backend design documentation describes output as audio and video packets interleaved in monotonic timestamp order. Look for timestamps that stop advancing coherently, become irregular, or diverge between tracks; use the surrounding logs and output to establish what happened rather than diagnosing from one value alone.
Check audio format and processing
Confirm the audio source’s sample rate and channel layout against OBS’s configured audio backend and the destination’s supported output. OBS can remix or resample when a source’s sample rate or channel count differs from the configured backend, so do not assume audio passes through unchanged. OBS describes its audio processing and format handling in its backend design documentation and audio/video formats guide.
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Also check CPU or encoder load and any dropped or duplicated video frames as context. A dropped frame alone does not establish the cause of gradual audio drift: OBS documents its video queue behavior separately from its timestamp-interleaved packet output.
Check the transport settings for your actual output path
Do not transfer settings from one protocol to another. Confirm which ingest path the lecture stream uses, then follow the destination’s codec, frame-rate and latency requirements.
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If the stream uses SRT
Measure round-trip time between the encoder and ingest server, then compare it with the configured SRT latency. The OBS SRT guide lists a 120 ms default and recommends latency of at least 2.5 times the encoder-to-ingest round-trip time. These figures are guidance for SRT, not universal streaming settings. After any change, measure sync again end to end.
If it uses WHIP/WebRTC
Check the ingest provider’s supported OBS version, audio and video encoders, and frame rate. The OBS WHIP guide, dated January 14, 2025, describes OBS’s WHIP/WebRTC workflow. Cloudflare’s WebRTC guide gives one provider-specific example using OBS 31.0 or later, Opus audio and 30 fps video. Those are examples for that path, not settings to apply to every WebRTC service.
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If it uses RTMP
Check the destination’s current ingest requirements and inspect the encoded result. FFmpeg documents RTMP as multimedia streaming over TCP/IP and provides publishing and playback examples in its protocol documentation. That protocol description does not, by itself, identify the cause of sync drift.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Change settings methodically and verify the result
- Save the current settings and note the time, observed offset, stream path and relevant logs.
- Choose one setting that matches evidence from the stage you are investigating, such as a confirmed audio-format mismatch or an SRT latency value below the documented guidance.
- Change only that setting. Keep the same observation points and measurement method.
- Measure long enough to tell whether the drift rate changed. A single session that looks better does not establish a durable correction for a 24/7 run.
- If the fault remains unclear, preserve timestamps, logs and a sample of the output from before and after the suspected point in the pipeline.
For any two available ingest or delivery options, compare measured sync stability over the intended run length, timestamp coherence from encoder to receiver, behavior under the actual network conditions, codec compatibility and whether monitoring can locate a failure without taking the lecture offline. No single service or setting can be selected as a fix without evidence about your stream path.
Common troubleshooting mistakes
- Changing latency without measuring the transport: a larger buffer is not a universal sync repair. For SRT, use the encoder-to-ingest RTT and OBS’s protocol-specific guidance, then test end to end.
- Blaming dropped frames for all drift: record frame drops as context, but check the actual audio and video output and their timestamps.
- Assuming a format mismatch is harmless: verify sample rate and channel layout across source, OBS and output because OBS may resample or remix.
- Applying instructions for the wrong protocol: confirm whether the path is SRT, WHIP/WebRTC or RTMP before changing settings.
- Calling a short-term improvement a permanent fix: verify the stream over a run long enough to represent its normal continuous operation.
Or let it run in the cloud
If the lecture recording is already in sync and the goal is to keep a prerecorded YouTube stream running without leaving a computer on, StreamNeo can loop uploaded videos or playlists from the cloud. It will not repair audio that is already out of sync in the recording, and it streams to YouTube only.
- Upload the recording or build a playlist.
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