Moving overhead counting to bitrate estimators.

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 1087 matching lines @@
       }
       ++rtcp_iterator;
     }
     if (clock.TimeInMicroseconds() >= NextRtpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
       const LoggedRtpPacket& rtp = *rtp_iterator->second;
       if (rtp.header.extension.hasTransportSequenceNumber) {
         RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
         cc.AddPacket(rtp.header.ssrc,
                      rtp.header.extension.transportSequenceNumber,
-                     rtp.total_length, PacedPacketInfo());
+                     rtp.total_length - rtp.header.headerLength,
+                     rtp.header.headerLength, PacedPacketInfo());
         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();
     }
@@ skipping 79 matching lines @@
       }
       ++rtcp_iterator;
     }
     if (clock.TimeInMicroseconds() >= NextRtpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
       const LoggedRtpPacket& rtp = *rtp_iterator->second;
       if (rtp.header.extension.hasTransportSequenceNumber) {
         RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
         feedback_adapter.AddPacket(rtp.header.ssrc,
                                    rtp.header.extension.transportSequenceNumber,
-                                   rtp.total_length, PacedPacketInfo());
+                                   rtp.total_length - rtp.header.headerLength,
+                                   rtp.header.headerLength, PacedPacketInfo());
         feedback_adapter.OnSentPacket(
             rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
       }
       ++rtp_iterator;
     }
     time_us = std::min(NextRtpTime(), NextRtcpTime());
   }
   // We assume that the base network delay (w/o queues) is the min delay
   // observed during the call.
   for (TimeSeriesPoint& point : time_series.points)
@@ skipping 169 matching lines @@
       },
       audio_network_adaptation_events_, begin_time_, &time_series);
   plot->AppendTimeSeries(std::move(time_series));
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
                           kBottomMargin, kTopMargin);
   plot->SetTitle("Reported audio encoder number of channels");
 }
 }  // namespace plotting
 }  // namespace webrtc