Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

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 1052 matching lines @@
                   1000000;
         acked_time_series.points.emplace_back(x, y);
       }
       ++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.extension.transportSequenceNumber,
+        cc.AddPacket(rtp.header.ssrc,
+                     rtp.header.extension.transportSequenceNumber,
                      rtp.total_length, 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 78 matching lines @@
           time_series.points.emplace_back(x, y);
         }
       }
       ++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.extension.transportSequenceNumber,
+        feedback_adapter.AddPacket(rtp.header.ssrc,
+                                   rtp.header.extension.transportSequenceNumber,
                                    rtp.total_length, 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.
@@ skipping 168 matching lines @@
         return rtc::Optional<float>();
       },
       audio_network_adaptation_events_, begin_time_, 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