Correctly handle retransmissions/padding in early loss detection.
This CL makes sure we don't cull packets from the history based on incorrect ack mapping, just like it's predecessor: https://webrtc-review.googlesource.com/c/src/+/218000 It also changes the logic to make sure retransmits counts towards history pruning - and properly ignores padding/fec. Bug: webrtc:12713 Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34293}
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WebRTC LUCI CQ
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@ -1267,7 +1267,7 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
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const RtpPacketType& rtp_packet = *rtp_iterator->second;
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if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
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RtpPacketSendInfo packet_info;
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packet_info.ssrc = rtp_packet.rtp.header.ssrc;
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packet_info.media_ssrc = rtp_packet.rtp.header.ssrc;
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packet_info.transport_sequence_number =
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rtp_packet.rtp.header.extension.transportSequenceNumber;
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packet_info.rtp_sequence_number = rtp_packet.rtp.header.sequenceNumber;
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@ -84,7 +84,7 @@ void LogBasedNetworkControllerSimulation::OnPacketSent(
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}
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RtpPacketSendInfo packet_info;
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packet_info.ssrc = packet.ssrc;
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packet_info.media_ssrc = packet.ssrc;
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packet_info.transport_sequence_number = packet.transport_seq_no;
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packet_info.rtp_sequence_number = packet.stream_seq_no;
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packet_info.length = packet.size;
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