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