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 <string> 35#include <map> 36 37#include "talk/app/webrtc/mediastreaminterface.h" 38#include "talk/app/webrtc/statstypes.h" 39#include "talk/app/webrtc/webrtcsession.h" 40 41#include "talk/base/timing.h" 42 43namespace webrtc { 44 45class StatsCollector { 46 public: 47 StatsCollector(); 48 49 // Register the session Stats should operate on. 50 // Set to NULL if the session has ended. 51 void set_session(WebRtcSession* session) { 52 session_ = session; 53 } 54 55 // Adds a MediaStream with tracks that can be used as a |selector| in a call 56 // to GetStats. 57 void AddStream(MediaStreamInterface* stream); 58 59 // Gather statistics from the session and store them for future use. 60 void UpdateStats(); 61 62 // Gets a StatsReports of the last collected stats. Note that UpdateStats must 63 // be called before this function to get the most recent stats. |selector| is 64 // a track label or empty string. The most recent reports are stored in 65 // |reports|. 66 bool GetStats(MediaStreamTrackInterface* track, StatsReports* reports); 67 68 // Prepare an SSRC report for the given ssrc. Used internally 69 // in the ExtractStatsFromList template. 70 StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport); 71 // Prepare an SSRC report for the given remote ssrc. Used internally. 72 StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport); 73 // Extracts the ID of a Transport belonging to an SSRC. Used internally. 74 bool GetTransportIdFromProxy(const std::string& proxy, 75 std::string* transport_id); 76 77 private: 78 bool CopySelectedReports(const std::string& selector, StatsReports* reports); 79 80 // Helper method for AddCertificateReports. 81 std::string AddOneCertificateReport( 82 const talk_base::SSLCertificate* cert, const std::string& issuer_id); 83 84 // Adds a report for this certificate and every certificate in its chain, and 85 // returns the leaf certificate's report's ID. 86 std::string AddCertificateReports(const talk_base::SSLCertificate* cert); 87 88 void ExtractSessionInfo(); 89 void ExtractVoiceInfo(); 90 void ExtractVideoInfo(); 91 double GetTimeNow(); 92 void BuildSsrcToTransportId(); 93 WebRtcSession* session() { return session_; } 94 webrtc::StatsReport* GetOrCreateReport(const std::string& type, 95 const std::string& id); 96 97 // A map from the report id to the report. 98 std::map<std::string, StatsReport> reports_; 99 // Raw pointer to the session the statistics are gathered from. 100 WebRtcSession* session_; 101 double stats_gathering_started_; 102 talk_base::Timing timing_; 103 cricket::ProxyTransportMap proxy_to_transport_; 104}; 105 106} // namespace webrtc 107 108#endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_ 109