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Yeah the threshold is pretty brutal, but it is enough. Experimentally, I'd say you need under 2-3ms but even at 1ms you can start to hear some phase differences.

Most of the time, I think my synchronization algorithm is actually sub-1ms, but it can be worse depending on unstable network conditions.



How are you measuring this? I'm surprised the Web Audio API scheduling system has that much insight into the hardware latency.


I was wondering that too. It’s an impressive demo when used on devices with low latency audio drivers but I’m not convinced there’s any ability to detect drift beyond this. Might be interesting to have an option to use microphones to detect and calibrate this… …but then you have the same issue of an unknown delay on the microphone input too.




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