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
+  // The caller is responsible for ensuring that the pc outlives the
   // StatsCollector instance.
-  explicit StatsCollector(WebRtcSession* session);
+  explicit StatsCollector(PeerConnection* pc);
   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_t ssrc);
 
   // Removes a local audio tracks that is used for getting some voice
@@ skipping 71 matching lines @@
   bool GetTrackIdBySsrc(uint32_t 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.
-  WebRtcSession* const session_;
+  // Raw pointer to the peer connection the statistics are gathered from.
+  PeerConnection* const pc_;
   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_t> >
       LocalAudioTrackVector;
   LocalAudioTrackVector local_audio_tracks_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_STATSCOLLECTOR_H_