1 /* 2 * libjingle 3 * Copyright 2012 Google Inc. 4 * 5 * Redistribution and use in source and binary forms, with or without 6 * modification, are permitted provided that the following conditions are met: 7 * 8 * 1. Redistributions of source code must retain the above copyright notice, 9 * this list of conditions and the following disclaimer. 10 * 2. Redistributions in binary form must reproduce the above copyright notice, 11 * this list of conditions and the following disclaimer in the documentation 12 * and/or other materials provided with the distribution. 13 * 3. The name of the author may not be used to endorse or promote products 14 * derived from this software without specific prior written permission. 15 * 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 26 */ 27 28 // This file contains a class used for gathering statistics from an ongoing 29 // libjingle PeerConnection. 30 31 #ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_ 32 #define TALK_APP_WEBRTC_STATSCOLLECTOR_H_ 33 34 #include <map> 35 #include <string> 36 #include <vector> 37 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" 42 43 namespace webrtc { 44 45 class PeerConnection; 46 47 // Conversion function to convert candidate type string to the corresponding one 48 // from enum RTCStatsIceCandidateType. 49 const char* IceCandidateTypeToStatsType(const std::string& candidate_type); 50 51 // Conversion function to convert adapter type to report string which are more 52 // fitting to the general style of http://w3c.github.io/webrtc-stats. This is 53 // only used by stats collector. 54 const char* AdapterTypeToStatsType(rtc::AdapterType type); 55 56 // A mapping between track ids and their StatsReport. 57 typedef std::map<std::string, StatsReport*> TrackIdMap; 58 59 class StatsCollector { 60 public: 61 // The caller is responsible for ensuring that the pc outlives the 62 // StatsCollector instance. 63 explicit StatsCollector(PeerConnection* pc); 64 virtual ~StatsCollector(); 65 66 // Adds a MediaStream with tracks that can be used as a |selector| in a call 67 // to GetStats. 68 void AddStream(MediaStreamInterface* stream); 69 70 // Adds a local audio track that is used for getting some voice statistics. 71 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc); 72 73 // Removes a local audio tracks that is used for getting some voice 74 // statistics. 75 void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc); 76 77 // Gather statistics from the session and store them for future use. 78 void UpdateStats(PeerConnectionInterface::StatsOutputLevel level); 79 80 // Gets a StatsReports of the last collected stats. Note that UpdateStats must 81 // be called before this function to get the most recent stats. |selector| is 82 // a track label or empty string. The most recent reports are stored in 83 // |reports|. 84 // TODO(tommi): Change this contract to accept a callback object instead 85 // of filling in |reports|. As is, there's a requirement that the caller 86 // uses |reports| immediately without allowing any async activity on 87 // the thread (message handling etc) and then discard the results. 88 void GetStats(MediaStreamTrackInterface* track, 89 StatsReports* reports); 90 91 // Prepare a local or remote SSRC report for the given ssrc. Used internally 92 // in the ExtractStatsFromList template. 93 StatsReport* PrepareReport(bool local, 94 uint32_t ssrc, 95 const StatsReport::Id& transport_id, 96 StatsReport::Direction direction); 97 98 // Method used by the unittest to force a update of stats since UpdateStats() 99 // that occur less than kMinGatherStatsPeriod number of ms apart will be 100 // ignored. 101 void ClearUpdateStatsCacheForTest(); 102 103 private: 104 friend class StatsCollectorTest; 105 106 // Overridden in unit tests to fake timing. 107 virtual double GetTimeNow(); 108 109 bool CopySelectedReports(const std::string& selector, StatsReports* reports); 110 111 // Helper method for AddCertificateReports. 112 StatsReport* AddOneCertificateReport( 113 const rtc::SSLCertificate* cert, const StatsReport* issuer); 114 115 // Helper method for creating IceCandidate report. |is_local| indicates 116 // whether this candidate is local or remote. 117 StatsReport* AddCandidateReport(const cricket::Candidate& candidate, 118 bool local); 119 120 // Adds a report for this certificate and every certificate in its chain, and 121 // returns the leaf certificate's report. 122 StatsReport* AddCertificateReports(const rtc::SSLCertificate* cert); 123 124 StatsReport* AddConnectionInfoReport(const std::string& content_name, 125 int component, int connection_id, 126 const StatsReport::Id& channel_report_id, 127 const cricket::ConnectionInfo& info); 128 129 void ExtractDataInfo(); 130 void ExtractSessionInfo(); 131 void ExtractVoiceInfo(); 132 void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level); 133 void BuildSsrcToTransportId(); 134 webrtc::StatsReport* GetReport(const StatsReport::StatsType& type, 135 const std::string& id, 136 StatsReport::Direction direction); 137 138 // Helper method to get stats from the local audio tracks. 139 void UpdateStatsFromExistingLocalAudioTracks(); 140 void UpdateReportFromAudioTrack(AudioTrackInterface* track, 141 StatsReport* report); 142 143 // Helper method to get the id for the track identified by ssrc. 144 // |direction| tells if the track is for sending or receiving. 145 bool GetTrackIdBySsrc(uint32_t ssrc, 146 std::string* track_id, 147 StatsReport::Direction direction); 148 149 // Helper method to update the timestamp of track records. 150 void UpdateTrackReports(); 151 152 // A collection for all of our stats reports. 153 StatsCollection reports_; 154 TrackIdMap track_ids_; 155 // Raw pointer to the peer connection the statistics are gathered from. 156 PeerConnection* const pc_; 157 double stats_gathering_started_; 158 ProxyTransportMap proxy_to_transport_; 159 160 // TODO(tommi): We appear to be holding on to raw pointers to reference 161 // counted objects? We should be using scoped_refptr here. 162 typedef std::vector<std::pair<AudioTrackInterface*, uint32_t> > 163 LocalAudioTrackVector; 164 LocalAudioTrackVector local_audio_tracks_; 165 }; 166 167 } // namespace webrtc 168 169 #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_ 170
