Track SSRCs configured for each media type in event log visualization tool.

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 {
 
@@ skipping 18 matching lines @@
 
   void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
 
   void CreateBweGraph(Plot* plot);
 
  private:
   class StreamId {
    public:
     StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
         : ssrc_(ssrc), direction_(direction) {}
-    bool operator<(const StreamId& other) const;
-    bool operator==(const StreamId& other) const;
+    bool operator<(const StreamId& other) const {
+      return std::tie(ssrc_, direction_) <
+             std::tie(other.ssrc_, other.direction_);
+    }
+    bool operator==(const StreamId& other) const {
+      return std::tie(ssrc_, direction_) ==
+             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)
@@ skipping 15 matching lines @@
     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.
   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.
+  // Tracks what each stream is configured for. Note that a single SSRC can be
+  // in several sets. For example, the SSRC used for sending video over RTX
+  // will appear in both video_ssrcs_ and rtx_ssrcs_. In the unlikely case that
+  // an SSRC is reconfigured to a different media type mid-call, it will also
+  // appear in multiple sets.
+  std::set<StreamId> rtx_ssrcs_;
+  std::set<StreamId> video_ssrcs_;
+  std::set<StreamId> audio_ssrcs_;
+
+  // Maps a stream identifier consisting of ssrc and direction 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_;
 
   std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_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_;
 
   // Duration (in seconds) of log file.
   float call_duration_s_;
 };
 
 }  // namespace plotting
 }  // namespace webrtc
 
 #endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_