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wait 10s before growing jitter
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@ -420,7 +420,7 @@ void InboundAudioStream::packetReceivedUpdateTimingStats() {
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// update our timegap stats and desired jitter buffer frames if necessary
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// discard the first few packets we receive since they usually have gaps that aren't represensative of normal jitter
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const quint32 NUM_INITIAL_PACKETS_DISCARD = 3;
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const quint32 NUM_INITIAL_PACKETS_DISCARD = 1000; // 10s
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quint64 now = usecTimestampNow();
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if (_incomingSequenceNumberStats.getReceived() > NUM_INITIAL_PACKETS_DISCARD) {
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quint64 gap = now - _lastPacketReceivedTime;
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