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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