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by rcxdude
599 days ago
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It's possible to handle buffer underruns more elegantly than that, but it does require more processing power on the receive side of the buffer (basically by using some strategy to extrapolate the audio forward and decay, as opposed to just dropping the signal to zero when there's no data coming from the other side). It's a common thing to do in streaming audio contexts, especially voice, but generally at the end of the user's network connection which is presumed to be unreliable, not in the middle of a processing pipeline which is presumed to be able to hit its latency targets. |
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