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source: webrtc/talk/app/webrtc/statscollector.h @ 0:4bda6873e34c

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Last change on this file since 0:4bda6873e34c was 0:4bda6873e34c, checked in by Michael Prittie <mprittie@…>, 6 years ago

Scrubbed password for publication.

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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 <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_
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