commit c60a77731d68c8648135372dbe54c92dfba731bd
author Bjorn Terelius <terelius@webrtc.org> Wed Dec 05 21:15:30 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Dec 05 21:04:55 2018 +0000
tree 20f18c671e9af5fbdab651d2425a06cd69f2754f
parent 75726f2852b3161ebed9dcce9454dd19338db2d0

Process RTP before RTCP in RTC event log analyzer.

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}

rtc_tools/event_log_visualizer/analyzer.cc

diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index 9c0a1d6..8d346f0 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -1124,6 +1124,24 @@
   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 @@
 #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;