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28 // This file contains a class used for gathering statistics from an ongoing
29 // libjingle PeerConnection.
31 #ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_
32 #define TALK_APP_WEBRTC_STATSCOLLECTOR_H_
38 #include "talk/app/webrtc/mediastreaminterface.h"
39 #include "talk/app/webrtc/peerconnectioninterface.h"
40 #include "talk/app/webrtc/statstypes.h"
41 #include "talk/app/webrtc/webrtcsession.h"
43 #include "talk/base/timing.h"
47 class StatsCollector {
56 // Register the session Stats should operate on.
57 // Set to NULL if the session has ended.
58 void set_session(WebRtcSession* session) {
62 // Adds a MediaStream with tracks that can be used as a |selector| in a call
64 void AddStream(MediaStreamInterface* stream);
66 // Adds a local audio track that is used for getting some voice statistics.
67 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
69 // Removes a local audio tracks that is used for getting some voice
71 void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
73 // Gather statistics from the session and store them for future use.
74 void UpdateStats(PeerConnectionInterface::StatsOutputLevel level);
76 // Gets a StatsReports of the last collected stats. Note that UpdateStats must
77 // be called before this function to get the most recent stats. |selector| is
78 // a track label or empty string. The most recent reports are stored in
80 bool GetStats(MediaStreamTrackInterface* track,
81 StatsReports* reports);
83 // Prepare an SSRC report for the given ssrc. Used internally
84 // in the ExtractStatsFromList template.
85 StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport,
86 TrackDirection direction);
87 // Prepare an SSRC report for the given remote ssrc. Used internally.
88 StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport,
89 TrackDirection direction);
90 // Extracts the ID of a Transport belonging to an SSRC. Used internally.
91 bool GetTransportIdFromProxy(const std::string& proxy,
92 std::string* transport_id);
95 bool CopySelectedReports(const std::string& selector, StatsReports* reports);
97 // Helper method for AddCertificateReports.
98 std::string AddOneCertificateReport(
99 const talk_base::SSLCertificate* cert, const std::string& issuer_id);
101 // Adds a report for this certificate and every certificate in its chain, and
102 // returns the leaf certificate's report's ID.
103 std::string AddCertificateReports(const talk_base::SSLCertificate* cert);
105 void ExtractSessionInfo();
106 void ExtractVoiceInfo();
107 void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level);
109 void BuildSsrcToTransportId();
110 WebRtcSession* session() { return session_; }
111 webrtc::StatsReport* GetOrCreateReport(const std::string& type,
112 const std::string& id,
113 TrackDirection direction);
114 webrtc::StatsReport* GetReport(const std::string& type,
115 const std::string& id,
116 TrackDirection direction);
118 // Helper method to get stats from the local audio tracks.
119 void UpdateStatsFromExistingLocalAudioTracks();
120 void UpdateReportFromAudioTrack(AudioTrackInterface* track,
121 StatsReport* report);
123 // Helper method to get the id for the track identified by ssrc.
124 // |direction| tells if the track is for sending or receiving.
125 bool GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
126 TrackDirection direction);
128 // A map from the report id to the report.
129 std::map<std::string, StatsReport> reports_;
130 // Raw pointer to the session the statistics are gathered from.
131 WebRtcSession* session_;
132 double stats_gathering_started_;
133 talk_base::Timing timing_;
134 cricket::ProxyTransportMap proxy_to_transport_;
136 typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
137 LocalAudioTrackVector;
138 LocalAudioTrackVector local_audio_tracks_;
141 } // namespace webrtc
143 #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_