Moving MediaStreamSignaling logic into PeerConnection.

talk/app/webrtc/statscollector.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 25 matching lines @@
 #include <vector>
 
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/mediastreamsignaling.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/statstypes.h"
 #include "talk/app/webrtc/webrtcsession.h"
 
 namespace webrtc {
 
+class PeerConnection;
+
 // Conversion function to convert candidate type string to the corresponding one
 // from  enum RTCStatsIceCandidateType.
 const char* IceCandidateTypeToStatsType(const std::string& candidate_type);
 
 // Conversion function to convert adapter type to report string which are more
 // fitting to the general style of http://w3c.github.io/webrtc-stats. This is
 // only used by stats collector.
 const char* AdapterTypeToStatsType(rtc::AdapterType type);
 
 // A mapping between track ids and their StatsReport.
 typedef std::map<std::string, StatsReport*> TrackIdMap;
 
 class StatsCollector {
  public:
-  // The caller is responsible for ensuring that the session outlives the
-  // StatsCollector instance.
-  explicit StatsCollector(WebRtcSession* session);
+  // The caller is responsible for ensuring that the pc and session
+  // outlive the StatsCollector instance.
+  explicit StatsCollector(PeerConnection* pc, WebRtcSession* session);

[Taylor Brandstetter, 2015/10/07 00:26:19]

I don't really like passing the StatsCollector a PeerConnection pointer, but we
already pass it a WebRtcSession pointer... And later WebRtcSession will become
PeerConnection. Maybe in the future we should define a new interface that
StatsCollector depends on, distinct from PeerConnection and
PeerConnectionInterface?

[pthatcher1, 2015/10/07 02:50:51]

It makes sense to pass a PeerConnection in.  It's passing both a pc and a
session that's lame.  

Could we just make PeerConnection implement all of the methods that
StatsCollector needs from WebRtcSession by calling down to WebRtcSession inside
of it?  Then this test won't need to change again when WebRtcSession and
PeerConnection are merged.

[pthatcher1, 2015/10/08 22:05:36]

Or maybe pc_->session()?

[Taylor Brandstetter, 2015/10/09 19:54:09]

Done.

   virtual ~StatsCollector();
 
   // Adds a MediaStream with tracks that can be used as a |selector| in a call
   // to GetStats.
   void AddStream(MediaStreamInterface* stream);
 
   // Adds a local audio track that is used for getting some voice statistics.
   void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
 
   // Removes a local audio tracks that is used for getting some voice
@@ skipping 68 matching lines @@
   // |direction| tells if the track is for sending or receiving.
   bool GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
                         StatsReport::Direction direction);
 
   // Helper method to update the timestamp of track records.
   void UpdateTrackReports();
 
   // A collection for all of our stats reports.
   StatsCollection reports_;
   TrackIdMap track_ids_;
-  // Raw pointer to the session the statistics are gathered from.
+  // Raw pointers to the peer connection and session the statistics are
+  // gathered from.
+  PeerConnection* const pc_;
   WebRtcSession* const session_;
   double stats_gathering_started_;
   cricket::ProxyTransportMap proxy_to_transport_;
 
   // TODO(tommi): We appear to be holding on to raw pointers to reference
   // counted objects?  We should be using scoped_refptr here.
   typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
       LocalAudioTrackVector;
   LocalAudioTrackVector local_audio_tracks_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_STATSCOLLECTOR_H_