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Process RTP before RTCP in RTC event log analyzer.

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This handles an unlikely corner case where you receive a RTCP feedback for a packet the same millisecond that you send it.

Bug: None
Change-Id: I77f460bef4073d4d9c5633c88f4d2dd8470f8577
Reviewed-on: https://webrtc-review.googlesource.com/c/113305
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25911}
This commit is contained in:
Bjorn Terelius
2018-12-05 21:15:30 +01:00
committed by Commit Bot
parent 75726f2852
commit c60a77731d
2 changed files with 32 additions and 32 deletions
Download Patch File Download Diff File Expand all files Collapse all files

36
rtc_tools/event_log_visualizer/analyzer.cc
View File

@ -1124,6 +1124,24 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
int64_t last_update_us = 0;
while (time_us != std::numeric_limits<int64_t>::max()) {
clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
if (clock.TimeInMicroseconds() >= NextRtpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
const RtpPacketType& rtp_packet = *rtp_iterator->second;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
RTC_DCHECK(rtp_packet.rtp.header.extension.hasTransportSequenceNumber);
transport_feedback.AddPacket(
rtp_packet.rtp.header.ssrc,
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.total_length, PacedPacketInfo());
rtc::SentPacket sent_packet(
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.log_time_us() / 1000);
auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
if (sent_msg)
observer.Update(goog_cc->OnSentPacket(*sent_msg));
}
++rtp_iterator;
}
if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
@ -1154,24 +1172,6 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
++rtcp_iterator;
}
if (clock.TimeInMicroseconds() >= NextRtpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
const RtpPacketType& rtp_packet = *rtp_iterator->second;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
RTC_DCHECK(rtp_packet.rtp.header.extension.hasTransportSequenceNumber);
transport_feedback.AddPacket(
rtp_packet.rtp.header.ssrc,
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.total_length, PacedPacketInfo());
rtc::SentPacket sent_packet(
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.log_time_us() / 1000);
auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
if (sent_msg)
observer.Update(goog_cc->OnSentPacket(*sent_msg));
}
++rtp_iterator;
}
if (clock.TimeInMicroseconds() >= NextProcessTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
ProcessInterval msg;