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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"
45 class StatsCollector {
52 // The caller is responsible for ensuring that the session outlives the
53 // StatsCollector instance.
54 explicit StatsCollector(WebRtcSession* session);
55 virtual ~StatsCollector();
57 // Adds a MediaStream with tracks that can be used as a |selector| in a call
59 void AddStream(MediaStreamInterface* stream);
61 // Adds a local audio track that is used for getting some voice statistics.
62 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
64 // Removes a local audio tracks that is used for getting some voice
66 void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
68 // Gather statistics from the session and store them for future use.
69 void UpdateStats(PeerConnectionInterface::StatsOutputLevel level);
71 // Gets a StatsReports of the last collected stats. Note that UpdateStats must
72 // be called before this function to get the most recent stats. |selector| is
73 // a track label or empty string. The most recent reports are stored in
75 bool GetStats(MediaStreamTrackInterface* track,
76 StatsReports* reports);
78 // Prepare an SSRC report for the given ssrc. Used internally
79 // in the ExtractStatsFromList template.
80 StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport,
81 TrackDirection direction);
82 // Prepare an SSRC report for the given remote ssrc. Used internally.
83 StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport,
84 TrackDirection direction);
86 // Method used by the unittest to force a update of stats since UpdateStats()
87 // that occur less than kMinGatherStatsPeriod number of ms apart will be
89 void ClearUpdateStatsCache();
92 bool CopySelectedReports(const std::string& selector, StatsReports* reports);
94 // Helper method for AddCertificateReports.
95 std::string AddOneCertificateReport(
96 const rtc::SSLCertificate* cert, const std::string& issuer_id);
98 // Adds a report for this certificate and every certificate in its chain, and
99 // returns the leaf certificate's report's ID.
100 std::string AddCertificateReports(const rtc::SSLCertificate* cert);
102 void ExtractSessionInfo();
103 void ExtractVoiceInfo();
104 void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level);
105 void BuildSsrcToTransportId();
106 webrtc::StatsReport* GetOrCreateReport(const std::string& type,
107 const std::string& id,
108 TrackDirection direction);
109 webrtc::StatsReport* GetReport(const std::string& type,
110 const std::string& id,
111 TrackDirection direction);
113 // Helper method to get stats from the local audio tracks.
114 void UpdateStatsFromExistingLocalAudioTracks();
115 void UpdateReportFromAudioTrack(AudioTrackInterface* track,
116 StatsReport* report);
118 // Helper method to get the id for the track identified by ssrc.
119 // |direction| tells if the track is for sending or receiving.
120 bool GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
121 TrackDirection direction);
123 // A map from the report id to the report.
124 std::map<std::string, StatsReport> reports_;
125 // Raw pointer to the session the statistics are gathered from.
126 WebRtcSession* const session_;
127 double stats_gathering_started_;
128 cricket::ProxyTransportMap proxy_to_transport_;
130 typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
131 LocalAudioTrackVector;
132 LocalAudioTrackVector local_audio_tracks_;
135 } // namespace webrtc
137 #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_