Use bayesian estimate of acked bitrate.

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 972 matching lines @@
 
   auto NextProcessTime = [&]() {
     if (rtcp_iterator != incoming_rtcp.end() ||
         rtp_iterator != outgoing_rtp.end()) {
       return clock.TimeInMicroseconds() +
              std::max<int64_t>(cc.TimeUntilNextProcess() * 1000, 0);
     }
     return std::numeric_limits<int64_t>::max();
   };
 
-  RateStatistics acked_bitrate(1000, 8000);
+  RateStatistics acked_bitrate(250, 8000);
 
   int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
+  int64_t last_update_us = 0;
   while (time_us != std::numeric_limits<int64_t>::max()) {
     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
       const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
       if (rtcp.type == kRtcpTransportFeedback) {
         TransportFeedbackObserver* observer = cc.GetTransportFeedbackObserver();
         observer->OnTransportFeedback(*static_cast<rtcp::TransportFeedback*>(
             rtcp.packet.get()));
         std::vector<PacketInfo> feedback =
@@ skipping 24 matching lines @@
         rtc::SentPacket sent_packet(
             rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
         cc.OnSentPacket(sent_packet);
       }
       ++rtp_iterator;
     }
     if (clock.TimeInMicroseconds() >= NextProcessTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
       cc.Process();
     }
-    if (observer.GetAndResetBitrateUpdated()) {
+    if (observer.GetAndResetBitrateUpdated() ||
+        time_us - last_update_us >= 1e6) {
       uint32_t y = observer.last_bitrate_bps() / 1000;
       float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
                 1000000;
       time_series.points.emplace_back(x, y);
+      last_update_us = time_us;
     }
     time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
   }
   // Add the data set to the plot.
   plot->series_list_.push_back(std::move(time_series));
   plot->series_list_.push_back(std::move(acked_time_series));
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
   plot->SetTitle("Simulated BWE behavior");
@@ skipping 113 matching lines @@
     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.");
 }
 }  // namespace plotting
 }  // namespace webrtc