Add loss-based BWE estimate to the outgoing bitrate plot.

webrtc/tools/event_log_visualizer/analyzer.h

 /*
  *  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 49 matching lines @@
     webrtc::MediaType media_type_;
   };
 
   struct LoggedRtpPacket {
     LoggedRtpPacket(uint64_t timestamp, RTPHeader header)
         : timestamp(timestamp), header(header) {}
     uint64_t timestamp;
     RTPHeader header;
   };
 
+  struct BwePacketLossEvent {
+    BwePacketLossEvent(uint64_t timestamp,
+                       int32_t new_bitrate,
+                       uint8_t fraction_loss,
+                       int expected_packets)
+        : timestamp(timestamp),
+          new_bitrate(new_bitrate),
+          fraction_loss(fraction_loss),
+          expected_packets(expected_packets) {}
+    uint64_t timestamp;
+    int32_t new_bitrate;
+    uint8_t fraction_loss;
+    int32_t expected_packets;
+  };
+
   const ParsedRtcEventLog& parsed_log_;
 
   // A list of SSRCs we are interested in analysing.
   // If left empty, all SSRCs will be considered relevant.
   std::vector<uint32_t> desired_ssrc_;
 
   // Maps a stream identifier consisting of ssrc, direction and MediaType
   // to the parsed RTP headers in that stream. Header extensions are parsed
   // if the stream has been configured.
   std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_;
 
+  // A list of all updates from the send-side loss-based bandwidth estimator.
+  std::vector<BwePacketLossEvent> bwe_loss_updates_;
+
   // Window and step size used for calculating moving averages, e.g. bitrate.
   // The generated data points will be |step_| microseconds apart.
   // Only events occuring at most |window_duration_| microseconds before the
   // current data point will be part of the average.
   uint64_t window_duration_;
   uint64_t step_;
 
   // First and last events of the log.
   uint64_t begin_time_;
   uint64_t end_time_;
 };
 
 }  // namespace plotting
 }  // namespace webrtc
 
 #endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_