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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:13 +00003 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00004 *
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;
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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
Henrik Kjellander15583c12016-02-10 10:53:12 +010031#ifndef WEBRTC_API_STATSCOLLECTOR_H_
32#define WEBRTC_API_STATSCOLLECTOR_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000033
henrike@webrtc.org28e20752013-07-10 00:45:36 +000034#include <map>
henrike@webrtc.org40b3b682014-03-03 18:30:11 +000035#include <string>
36#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037
Henrik Kjellander15583c12016-02-10 10:53:12 +010038#include "webrtc/api/mediastreaminterface.h"
39#include "webrtc/api/peerconnectioninterface.h"
40#include "webrtc/api/statstypes.h"
41#include "webrtc/api/webrtcsession.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043namespace webrtc {
44
deadbeefab9b2d12015-10-14 11:33:11 -070045class PeerConnection;
46
guoweis@webrtc.org950c5182014-12-16 23:01:31 +000047// Conversion function to convert candidate type string to the corresponding one
48// from enum RTCStatsIceCandidateType.
49const 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.
54const char* AdapterTypeToStatsType(rtc::AdapterType type);
55
jbauchbe24c942015-06-22 15:06:43 -070056// A mapping between track ids and their StatsReport.
57typedef std::map<std::string, StatsReport*> TrackIdMap;
58
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059class StatsCollector {
60 public:
deadbeefab9b2d12015-10-14 11:33:11 -070061 // The caller is responsible for ensuring that the pc outlives the
tommi@webrtc.org03505bc2014-07-14 20:15:26 +000062 // StatsCollector instance.
deadbeefab9b2d12015-10-14 11:33:11 -070063 explicit StatsCollector(PeerConnection* pc);
tommi@webrtc.org03505bc2014-07-14 20:15:26 +000064 virtual ~StatsCollector();
henrike@webrtc.org28e20752013-07-10 00:45:36 +000065
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
henrike@webrtc.org40b3b682014-03-03 18:30:11 +000070 // Adds a local audio track that is used for getting some voice statistics.
Peter Boström0c4e06b2015-10-07 12:23:21 +020071 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc);
henrike@webrtc.org40b3b682014-03-03 18:30:11 +000072
73 // Removes a local audio tracks that is used for getting some voice
74 // statistics.
Peter Boström0c4e06b2015-10-07 12:23:21 +020075 void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc);
henrike@webrtc.org40b3b682014-03-03 18:30:11 +000076
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077 // Gather statistics from the session and store them for future use.
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000078 void UpdateStats(PeerConnectionInterface::StatsOutputLevel level);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000079
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|.
tommi@webrtc.org5b06b062014-08-15 08:38:30 +000084 // 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,
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000089 StatsReports* reports);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090
tommi@webrtc.org4fb7e252015-01-21 11:36:18 +000091 // Prepare a local or remote SSRC report for the given ssrc. Used internally
wu@webrtc.org97077a32013-10-25 21:18:33 +000092 // in the ExtractStatsFromList template.
Peter Boström0c4e06b2015-10-07 12:23:21 +020093 StatsReport* PrepareReport(bool local,
94 uint32_t ssrc,
95 const StatsReport::Id& transport_id,
96 StatsReport::Direction direction);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000097
xians@webrtc.org01bda202014-07-09 07:38:38 +000098 // 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.
tommi@webrtc.org69bc5a32014-12-15 09:44:48 +0000101 void ClearUpdateStatsCacheForTest();
xians@webrtc.org01bda202014-07-09 07:38:38 +0000102
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103 private:
guoweis@webrtc.org950c5182014-12-16 23:01:31 +0000104 friend class StatsCollectorTest;
105
decurtis@webrtc.org322a5642015-02-03 22:09:37 +0000106 // Overridden in unit tests to fake timing.
107 virtual double GetTimeNow();
108
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000109 bool CopySelectedReports(const std::string& selector, StatsReports* reports);
110
wu@webrtc.org4551b792013-10-09 15:37:36 +0000111 // Helper method for AddCertificateReports.
tommi@webrtc.orgd3900292015-03-12 16:35:55 +0000112 StatsReport* AddOneCertificateReport(
113 const rtc::SSLCertificate* cert, const StatsReport* issuer);
wu@webrtc.org4551b792013-10-09 15:37:36 +0000114
guoweis@webrtc.org950c5182014-12-16 23:01:31 +0000115 // Helper method for creating IceCandidate report. |is_local| indicates
116 // whether this candidate is local or remote.
tommi@webrtc.orgd3900292015-03-12 16:35:55 +0000117 StatsReport* AddCandidateReport(const cricket::Candidate& candidate,
118 bool local);
guoweis@webrtc.org950c5182014-12-16 23:01:31 +0000119
wu@webrtc.org4551b792013-10-09 15:37:36 +0000120 // Adds a report for this certificate and every certificate in its chain, and
tommi@webrtc.orgd3900292015-03-12 16:35:55 +0000121 // 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);
wu@webrtc.org4551b792013-10-09 15:37:36 +0000128
decurtis@webrtc.org487a4442015-01-15 22:55:07 +0000129 void ExtractDataInfo();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000130 void ExtractSessionInfo();
131 void ExtractVoiceInfo();
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000132 void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000133 void BuildSsrcToTransportId();
tommi@webrtc.org4fb7e252015-01-21 11:36:18 +0000134 webrtc::StatsReport* GetReport(const StatsReport::StatsType& type,
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000135 const std::string& id,
tommi@webrtc.org4fb7e252015-01-21 11:36:18 +0000136 StatsReport::Direction direction);
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000137
138 // Helper method to get stats from the local audio tracks.
139 void UpdateStatsFromExistingLocalAudioTracks();
140 void UpdateReportFromAudioTrack(AudioTrackInterface* track,
141 StatsReport* report);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000143 // Helper method to get the id for the track identified by ssrc.
144 // |direction| tells if the track is for sending or receiving.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200145 bool GetTrackIdBySsrc(uint32_t ssrc,
146 std::string* track_id,
tommi@webrtc.org4fb7e252015-01-21 11:36:18 +0000147 StatsReport::Direction direction);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000148
jbauchbe24c942015-06-22 15:06:43 -0700149 // Helper method to update the timestamp of track records.
150 void UpdateTrackReports();
151
tommi@webrtc.org4fb7e252015-01-21 11:36:18 +0000152 // A collection for all of our stats reports.
153 StatsCollection reports_;
jbauchbe24c942015-06-22 15:06:43 -0700154 TrackIdMap track_ids_;
deadbeefab9b2d12015-10-14 11:33:11 -0700155 // Raw pointer to the peer connection the statistics are gathered from.
156 PeerConnection* const pc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000157 double stats_gathering_started_;
deadbeefd59daf82015-10-14 15:02:44 -0700158 ProxyTransportMap proxy_to_transport_;
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000159
tommi@webrtc.orgd3900292015-03-12 16:35:55 +0000160 // TODO(tommi): We appear to be holding on to raw pointers to reference
161 // counted objects? We should be using scoped_refptr here.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200162 typedef std::vector<std::pair<AudioTrackInterface*, uint32_t> >
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000163 LocalAudioTrackVector;
164 LocalAudioTrackVector local_audio_tracks_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000165};
166
167} // namespace webrtc
168
Henrik Kjellander15583c12016-02-10 10:53:12 +0100169#endif // WEBRTC_API_STATSCOLLECTOR_H_