| /* |
| * 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 <map> |
| #include <string> |
| #include <vector> |
| |
| #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 { |
| |
| class PeerConnection; |
| |
| // 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); |
| |
| // A mapping between track ids and their StatsReport. |
| typedef std::map<std::string, StatsReport*> TrackIdMap; |
| |
| class StatsCollector { |
| public: |
| // The caller is responsible for ensuring that the pc outlives the |
| // StatsCollector instance. |
| explicit StatsCollector(PeerConnection* pc); |
| 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_t ssrc); |
| |
| // Removes a local audio tracks that is used for getting some voice |
| // statistics. |
| void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t 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 a local or remote SSRC report for the given ssrc. Used internally |
| // in the ExtractStatsFromList template. |
| StatsReport* PrepareReport(bool local, |
| uint32_t ssrc, |
| const StatsReport::Id& transport_id, |
| StatsReport::Direction 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; |
| |
| // Overridden in unit tests to fake timing. |
| virtual double GetTimeNow(); |
| |
| bool CopySelectedReports(const std::string& selector, StatsReports* reports); |
| |
| // Helper method for AddCertificateReports. |
| StatsReport* AddOneCertificateReport( |
| const rtc::SSLCertificate* cert, const StatsReport* issuer); |
| |
| // Helper method for creating IceCandidate report. |is_local| indicates |
| // whether this candidate is local or remote. |
| StatsReport* AddCandidateReport(const cricket::Candidate& candidate, |
| bool local); |
| |
| // Adds a report for this certificate and every certificate in its chain, and |
| // returns the leaf certificate's report. |
| StatsReport* AddCertificateReports(const rtc::SSLCertificate* cert); |
| |
| StatsReport* AddConnectionInfoReport(const std::string& content_name, |
| int component, int connection_id, |
| const StatsReport::Id& channel_report_id, |
| const cricket::ConnectionInfo& info); |
| |
| void ExtractDataInfo(); |
| void ExtractSessionInfo(); |
| void ExtractVoiceInfo(); |
| void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level); |
| void BuildSsrcToTransportId(); |
| webrtc::StatsReport* GetReport(const StatsReport::StatsType& type, |
| const std::string& id, |
| StatsReport::Direction 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_t ssrc, |
| std::string* track_id, |
| StatsReport::Direction direction); |
| |
| // Helper method to update the timestamp of track records. |
| void UpdateTrackReports(); |
| |
| // A collection for all of our stats reports. |
| StatsCollection reports_; |
| TrackIdMap track_ids_; |
| // Raw pointer to the peer connection the statistics are gathered from. |
| PeerConnection* const pc_; |
| double stats_gathering_started_; |
| ProxyTransportMap proxy_to_transport_; |
| |
| // TODO(tommi): We appear to be holding on to raw pointers to reference |
| // counted objects? We should be using scoped_refptr here. |
| typedef std::vector<std::pair<AudioTrackInterface*, uint32_t> > |
| LocalAudioTrackVector; |
| LocalAudioTrackVector local_audio_tracks_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_ |