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Instead of once a second, buffering level measurements are performed after each output packet for which an input packet has recently arrived. AudioDriftCorrection::RequestFrames() targets desired buffering at 13/10 * MeasuredSourceLatency() if actual buffering stays near desired. 20% less than this is 104/100 * MeasuredSourceLatency(). When input packets are much larger than output packets, then when this much is buffered after an input packet arrives, then the buffer was close to underruning before the packet arrived. Differential Revision: https://phabricator.services.mozilla.com/D215489