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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 // TODO(tommi): Change this contract to accept a callback object instead
76 // of filling in |reports|. As is, there's a requirement that the caller
77 // uses |reports| immediately without allowing any async activity on
78 // the thread (message handling etc) and then discard the results.
79 void GetStats(MediaStreamTrackInterface* track,
80 StatsReports* reports);
82 // Prepare an SSRC report for the given ssrc. Used internally
83 // in the ExtractStatsFromList template.
84 StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport,
85 TrackDirection direction);
86 // Prepare an SSRC report for the given remote ssrc. Used internally.
87 StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport,
88 TrackDirection direction);
90 // Method used by the unittest to force a update of stats since UpdateStats()
91 // that occur less than kMinGatherStatsPeriod number of ms apart will be
93 void ClearUpdateStatsCache();
96 bool CopySelectedReports(const std::string& selector, StatsReports* reports);
98 // Helper method for AddCertificateReports.
99 std::string AddOneCertificateReport(
100 const rtc::SSLCertificate* cert, const std::string& issuer_id);
102 // Adds a report for this certificate and every certificate in its chain, and
103 // returns the leaf certificate's report's ID.
104 std::string AddCertificateReports(const rtc::SSLCertificate* cert);
106 void ExtractSessionInfo();
107 void ExtractVoiceInfo();
108 void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level);
109 void BuildSsrcToTransportId();
110 webrtc::StatsReport* GetOrCreateReport(const std::string& type,
111 const std::string& id,
112 TrackDirection direction);
113 webrtc::StatsReport* GetReport(const std::string& type,
114 const std::string& id,
115 TrackDirection direction);
117 // Helper method to get stats from the local audio tracks.
118 void UpdateStatsFromExistingLocalAudioTracks();
119 void UpdateReportFromAudioTrack(AudioTrackInterface* track,
120 StatsReport* report);
122 // Helper method to get the id for the track identified by ssrc.
123 // |direction| tells if the track is for sending or receiving.
124 bool GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
125 TrackDirection direction);
127 // A map from the report id to the report.
129 // Raw pointer to the session the statistics are gathered from.
130 WebRtcSession* const session_;
131 double stats_gathering_started_;
132 cricket::ProxyTransportMap proxy_to_transport_;
134 typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
135 LocalAudioTrackVector;
136 LocalAudioTrackVector local_audio_tracks_;
139 } // namespace webrtc
141 #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_