Update libjingle 62364298->62472237

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/statscollector.h b/talk/app/webrtc/statscollector.h
index 6256d77..fdb2961 100644
--- a/talk/app/webrtc/statscollector.h
+++ b/talk/app/webrtc/statscollector.h
@@ -31,8 +31,9 @@
 #ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_
 #define TALK_APP_WEBRTC_STATSCOLLECTOR_H_
 
-#include <string>
 #include <map>
+#include <string>
+#include <vector>
 
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
@@ -57,6 +58,13 @@
   // 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
+  // statistics.
+  void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
+
   // Gather statistics from the session and store them for future use.
   void UpdateStats(PeerConnectionInterface::StatsOutputLevel level);
 
@@ -95,6 +103,13 @@
   WebRtcSession* session() { return session_; }
   webrtc::StatsReport* GetOrCreateReport(const std::string& type,
                                          const std::string& id);
+  webrtc::StatsReport* GetReport(const std::string& type,
+                                 const std::string& id);
+
+  // Helper method to get stats from the local audio tracks.
+  void UpdateStatsFromExistingLocalAudioTracks();
+  void UpdateReportFromAudioTrack(AudioTrackInterface* track,
+                                  StatsReport* report);
 
   // A map from the report id to the report.
   std::map<std::string, StatsReport> reports_;
@@ -103,6 +118,10 @@
   double stats_gathering_started_;
   talk_base::Timing timing_;
   cricket::ProxyTransportMap proxy_to_transport_;
+
+  typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
+      LocalAudioTrackVector;
+  LocalAudioTrackVector local_audio_tracks_;
 };
 
 }  // namespace webrtc