/* * libjingle * Copyright 2012 Google Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: * * 1. Redistributions of source code must retain the above copyright notice, * this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright notice, * this list of conditions and the following disclaimer in the documentation * and/or other materials provided with the distribution. * 3. The name of the author may not be used to endorse or promote products * derived from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ // This file contains a class used for gathering statistics from an ongoing // libjingle PeerConnection. #ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_ #define TALK_APP_WEBRTC_STATSCOLLECTOR_H_ #include #include #include #include "talk/app/webrtc/mediastreaminterface.h" #include "talk/app/webrtc/mediastreamsignaling.h" #include "talk/app/webrtc/peerconnectioninterface.h" #include "talk/app/webrtc/statstypes.h" #include "talk/app/webrtc/webrtcsession.h" namespace webrtc { // Conversion function to convert candidate type string to the corresponding one // from enum RTCStatsIceCandidateType. const char* IceCandidateTypeToStatsType(const std::string& candidate_type); // Conversion function to convert adapter type to report string which are more // fitting to the general style of http://w3c.github.io/webrtc-stats. This is // only used by stats collector. const char* AdapterTypeToStatsType(rtc::AdapterType type); class StatsCollector { public: enum TrackDirection { kSending = 0, kReceiving, }; // The caller is responsible for ensuring that the session outlives the // StatsCollector instance. StatsCollector(WebRtcSession* session); virtual ~StatsCollector(); // Adds a MediaStream with tracks that can be used as a |selector| in a call // 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); // Gets a StatsReports of the last collected stats. Note that UpdateStats must // be called before this function to get the most recent stats. |selector| is // a track label or empty string. The most recent reports are stored in // |reports|. // TODO(tommi): Change this contract to accept a callback object instead // of filling in |reports|. As is, there's a requirement that the caller // uses |reports| immediately without allowing any async activity on // the thread (message handling etc) and then discard the results. void GetStats(MediaStreamTrackInterface* track, StatsReports* reports); // Prepare an SSRC report for the given ssrc. Used internally // in the ExtractStatsFromList template. StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport, TrackDirection direction); // Prepare an SSRC report for the given remote ssrc. Used internally. StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport, TrackDirection direction); // Method used by the unittest to force a update of stats since UpdateStats() // that occur less than kMinGatherStatsPeriod number of ms apart will be // ignored. void ClearUpdateStatsCacheForTest(); private: friend class StatsCollectorTest; bool CopySelectedReports(const std::string& selector, StatsReports* reports); // Helper method for AddCertificateReports. std::string AddOneCertificateReport( const rtc::SSLCertificate* cert, const std::string& issuer_id); // Helper method for creating IceCandidate report. |is_local| indicates // whether this candidate is local or remote. std::string AddCandidateReport(const cricket::Candidate& candidate, const std::string& report_type); // Adds a report for this certificate and every certificate in its chain, and // returns the leaf certificate's report's ID. std::string AddCertificateReports(const rtc::SSLCertificate* cert); void ExtractDataInfo(); void ExtractSessionInfo(); void ExtractVoiceInfo(); void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level); void BuildSsrcToTransportId(); webrtc::StatsReport* GetOrCreateReport(const std::string& type, const std::string& id, TrackDirection direction); webrtc::StatsReport* GetReport(const std::string& type, const std::string& id, TrackDirection direction); // Helper method to get stats from the local audio tracks. void UpdateStatsFromExistingLocalAudioTracks(); void UpdateReportFromAudioTrack(AudioTrackInterface* track, StatsReport* report); // Helper method to get the id for the track identified by ssrc. // |direction| tells if the track is for sending or receiving. bool GetTrackIdBySsrc(uint32 ssrc, std::string* track_id, TrackDirection direction); // A map from the report id to the report. StatsSet reports_; // Raw pointer to the session the statistics are gathered from. WebRtcSession* const session_; double stats_gathering_started_; cricket::ProxyTransportMap proxy_to_transport_; typedef std::vector > LocalAudioTrackVector; LocalAudioTrackVector local_audio_tracks_; }; } // namespace webrtc #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_