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Side by Side Diff: talk/app/webrtc/statscollector.h

Issue 1403633005: Revert of Moving MediaStreamSignaling logic into PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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36 #include <vector> 36 #include <vector>
37 37
38 #include "talk/app/webrtc/mediastreaminterface.h" 38 #include "talk/app/webrtc/mediastreaminterface.h"
39 #include "talk/app/webrtc/mediastreamsignaling.h" 39 #include "talk/app/webrtc/mediastreamsignaling.h"
40 #include "talk/app/webrtc/peerconnectioninterface.h" 40 #include "talk/app/webrtc/peerconnectioninterface.h"
41 #include "talk/app/webrtc/statstypes.h" 41 #include "talk/app/webrtc/statstypes.h"
42 #include "talk/app/webrtc/webrtcsession.h" 42 #include "talk/app/webrtc/webrtcsession.h"
43 43
44 namespace webrtc { 44 namespace webrtc {
45 45
46 class PeerConnection;
47
48 // Conversion function to convert candidate type string to the corresponding one 46 // Conversion function to convert candidate type string to the corresponding one
49 // from enum RTCStatsIceCandidateType. 47 // from enum RTCStatsIceCandidateType.
50 const char* IceCandidateTypeToStatsType(const std::string& candidate_type); 48 const char* IceCandidateTypeToStatsType(const std::string& candidate_type);
51 49
52 // Conversion function to convert adapter type to report string which are more 50 // Conversion function to convert adapter type to report string which are more
53 // fitting to the general style of http://w3c.github.io/webrtc-stats. This is 51 // fitting to the general style of http://w3c.github.io/webrtc-stats. This is
54 // only used by stats collector. 52 // only used by stats collector.
55 const char* AdapterTypeToStatsType(rtc::AdapterType type); 53 const char* AdapterTypeToStatsType(rtc::AdapterType type);
56 54
57 // A mapping between track ids and their StatsReport. 55 // A mapping between track ids and their StatsReport.
58 typedef std::map<std::string, StatsReport*> TrackIdMap; 56 typedef std::map<std::string, StatsReport*> TrackIdMap;
59 57
60 class StatsCollector { 58 class StatsCollector {
61 public: 59 public:
62 // The caller is responsible for ensuring that the pc outlives the 60 // The caller is responsible for ensuring that the session outlives the
63 // StatsCollector instance. 61 // StatsCollector instance.
64 explicit StatsCollector(PeerConnection* pc); 62 explicit StatsCollector(WebRtcSession* session);
65 virtual ~StatsCollector(); 63 virtual ~StatsCollector();
66 64
67 // Adds a MediaStream with tracks that can be used as a |selector| in a call 65 // Adds a MediaStream with tracks that can be used as a |selector| in a call
68 // to GetStats. 66 // to GetStats.
69 void AddStream(MediaStreamInterface* stream); 67 void AddStream(MediaStreamInterface* stream);
70 68
71 // Adds a local audio track that is used for getting some voice statistics. 69 // Adds a local audio track that is used for getting some voice statistics.
72 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc); 70 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc);
73 71
74 // Removes a local audio tracks that is used for getting some voice 72 // Removes a local audio tracks that is used for getting some voice
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146 bool GetTrackIdBySsrc(uint32_t ssrc, 144 bool GetTrackIdBySsrc(uint32_t ssrc,
147 std::string* track_id, 145 std::string* track_id,
148 StatsReport::Direction direction); 146 StatsReport::Direction direction);
149 147
150 // Helper method to update the timestamp of track records. 148 // Helper method to update the timestamp of track records.
151 void UpdateTrackReports(); 149 void UpdateTrackReports();
152 150
153 // A collection for all of our stats reports. 151 // A collection for all of our stats reports.
154 StatsCollection reports_; 152 StatsCollection reports_;
155 TrackIdMap track_ids_; 153 TrackIdMap track_ids_;
156 // Raw pointer to the peer connection the statistics are gathered from. 154 // Raw pointer to the session the statistics are gathered from.
157 PeerConnection* const pc_; 155 WebRtcSession* const session_;
158 double stats_gathering_started_; 156 double stats_gathering_started_;
159 cricket::ProxyTransportMap proxy_to_transport_; 157 cricket::ProxyTransportMap proxy_to_transport_;
160 158
161 // TODO(tommi): We appear to be holding on to raw pointers to reference 159 // TODO(tommi): We appear to be holding on to raw pointers to reference
162 // counted objects? We should be using scoped_refptr here. 160 // counted objects? We should be using scoped_refptr here.
163 typedef std::vector<std::pair<AudioTrackInterface*, uint32_t> > 161 typedef std::vector<std::pair<AudioTrackInterface*, uint32_t> >
164 LocalAudioTrackVector; 162 LocalAudioTrackVector;
165 LocalAudioTrackVector local_audio_tracks_; 163 LocalAudioTrackVector local_audio_tracks_;
166 }; 164 };
167 165
168 } // namespace webrtc 166 } // namespace webrtc
169 167
170 #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_ 168 #endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_
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