Add method needed to extract frame capture and arrival timestamps from rtc event logs.

webrtc/tools/event_log_visualizer/analyzer.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1149 matching lines @@
   // observed during the call.
   for (TimeSeriesPoint& point : time_series.points)
     point.y -= estimated_base_delay_ms;
   // Add the data set to the plot.
   plot->series_list_.push_back(std::move(time_series));
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
   plot->SetTitle("Network Delay Change.");
 }
+
+std::vector<int64_t> EventLogAnalyzer::GetFrameTimestamps() const {
+  std::vector<int64_t> timestamps;
+  size_t largest_stream_size = 0;
+  const std::vector<LoggedRtpPacket>* largest_video_stream = nullptr;
+  // Find the incoming video stream with the most number of packets that is
+  // not rtx.
+  for (const auto& kv : rtp_packets_) {
+    if (kv.first.GetDirection() == kIncomingPacket &&
+        video_ssrcs_.find(kv.first) != video_ssrcs_.end() &&
+        rtx_ssrcs_.find(kv.first) == rtx_ssrcs_.end() &&
+        kv.second.size() > largest_stream_size) {
+      largest_stream_size = kv.second.size();
+      largest_video_stream = &kv.second;
+    }
+  }
+  if (largest_video_stream == nullptr)
+    return std::vector<int64_t>();

[terelius, 2016/12/20 15:25:08]

Maybe:
if (largest_video_stream != nullptr) {
  for (auto& packet : *largest_video_stream) {...}
}

WDYT?

[stefan-webrtc, 2016/12/20 15:28:39]

Done.

+  for (auto& packet : *largest_video_stream) {
+    if (packet.header.markerBit) {
+      int64_t capture_ms = packet.header.timestamp / 90.0;
+      timestamps.push_back(capture_ms);

[terelius, 2016/12/20 15:25:08]

I think it would be nicer if timestamps was a vector of pairs.

[stefan-webrtc, 2016/12/20 15:28:39]

Done.

+      int64_t arrival_ms = packet.timestamp / 1000.0;
+      timestamps.push_back(arrival_ms);
+    }
+  }
+  return timestamps;
+}
 }  // namespace plotting
 }  // namespace webrtc