Visualize delay changes based on both abs-send-time and capture time.

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.
  */
 
 #ifndef WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
 #define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
 
 #include <vector>
 #include <map>
 #include <memory>
 #include <set>
 #include <utility>
 
 #include "webrtc/call/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/tools/event_log_visualizer/plot_base.h"
 
 namespace webrtc {
 namespace plotting {
 
+struct LoggedRtpPacket {
+  LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length)
+      : timestamp(timestamp), header(header), total_length(total_length) {}
+  uint64_t timestamp;
+  RTPHeader header;
+  size_t total_length;
+};
+
+struct LoggedRtcpPacket {
+  LoggedRtcpPacket(uint64_t timestamp,
+                   RTCPPacketType rtcp_type,
+                   std::unique_ptr<rtcp::RtcpPacket> rtcp_packet)
+      : timestamp(timestamp), type(rtcp_type), packet(std::move(rtcp_packet)) {}
+  uint64_t timestamp;
+  RTCPPacketType type;
+  std::unique_ptr<rtcp::RtcpPacket> packet;
+};
+
+struct BwePacketLossEvent {
+  uint64_t timestamp;
+  int32_t new_bitrate;
+  uint8_t fraction_loss;
+  int32_t expected_packets;
+};
+
 class EventLogAnalyzer {
  public:
   // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
   // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
   // modified while the EventLogAnalyzer is being used.
   explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
 
   void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
 
   void CreatePlayoutGraph(Plot* plot);
@@ skipping 28 matching lines @@
              std::tie(other.ssrc_, other.direction_);
     }
     uint32_t GetSsrc() const { return ssrc_; }
     webrtc::PacketDirection GetDirection() const { return direction_; }
 
    private:
     uint32_t ssrc_;
     webrtc::PacketDirection direction_;
   };
 
-  struct LoggedRtpPacket {
-    LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length)
-        : timestamp(timestamp), header(header), total_length(total_length) {}
-    uint64_t timestamp;
-    RTPHeader header;
-    size_t total_length;
-  };
-
-  struct LoggedRtcpPacket {
-    LoggedRtcpPacket(uint64_t timestamp,
-                     RTCPPacketType rtcp_type,
-                     std::unique_ptr<rtcp::RtcpPacket> rtcp_packet)
-        : timestamp(timestamp),
-          type(rtcp_type),
-          packet(std::move(rtcp_packet)) {}
-    uint64_t timestamp;
-    RTCPPacketType type;
-    std::unique_ptr<rtcp::RtcpPacket> packet;
-  };
-
-  struct BwePacketLossEvent {
-    uint64_t timestamp;
-    int32_t new_bitrate;
-    uint8_t fraction_loss;
-    int32_t expected_packets;
-  };
-
   bool IsRtxSsrc(StreamId stream_id);
 
   bool IsVideoSsrc(StreamId stream_id);
 
   bool IsAudioSsrc(StreamId stream_id);
 
   const ParsedRtcEventLog& parsed_log_;
 
   // A list of SSRCs we are interested in analysing.
   // If left empty, all SSRCs will be considered relevant.
@@ skipping 30 matching lines @@
   uint64_t end_time_;
 
   // Duration (in seconds) of log file.
   float call_duration_s_;
 };
 
 }  // namespace plotting
 }  // namespace webrtc
 
 #endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_