Index: talk/app/webrtc/statscollector.h |
diff --git a/talk/app/webrtc/statscollector.h b/talk/app/webrtc/statscollector.h |
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-/* |
- * libjingle |
- * Copyright 2012 Google Inc. |
- * |
- * Redistribution and use in source and binary forms, with or without |
- * modification, are permitted provided that the following conditions are met: |
- * |
- * 1. Redistributions of source code must retain the above copyright notice, |
- * this list of conditions and the following disclaimer. |
- * 2. Redistributions in binary form must reproduce the above copyright notice, |
- * this list of conditions and the following disclaimer in the documentation |
- * and/or other materials provided with the distribution. |
- * 3. The name of the author may not be used to endorse or promote products |
- * derived from this software without specific prior written permission. |
- * |
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
- */ |
- |
-// This file contains a class used for gathering statistics from an ongoing |
-// libjingle PeerConnection. |
- |
-#ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_ |
-#define TALK_APP_WEBRTC_STATSCOLLECTOR_H_ |
- |
-#include <map> |
-#include <string> |
-#include <vector> |
- |
-#include "talk/app/webrtc/mediastreaminterface.h" |
-#include "talk/app/webrtc/peerconnectioninterface.h" |
-#include "talk/app/webrtc/statstypes.h" |
-#include "talk/app/webrtc/webrtcsession.h" |
- |
-namespace webrtc { |
- |
-class PeerConnection; |
- |
-// Conversion function to convert candidate type string to the corresponding one |
-// from enum RTCStatsIceCandidateType. |
-const char* IceCandidateTypeToStatsType(const std::string& candidate_type); |
- |
-// Conversion function to convert adapter type to report string which are more |
-// fitting to the general style of http://w3c.github.io/webrtc-stats. This is |
-// only used by stats collector. |
-const char* AdapterTypeToStatsType(rtc::AdapterType type); |
- |
-// A mapping between track ids and their StatsReport. |
-typedef std::map<std::string, StatsReport*> TrackIdMap; |
- |
-class StatsCollector { |
- public: |
- // The caller is responsible for ensuring that the pc outlives the |
- // StatsCollector instance. |
- explicit StatsCollector(PeerConnection* pc); |
- virtual ~StatsCollector(); |
- |
- // Adds a MediaStream with tracks that can be used as a |selector| in a call |
- // to GetStats. |
- void AddStream(MediaStreamInterface* stream); |
- |
- // Adds a local audio track that is used for getting some voice statistics. |
- void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc); |
- |
- // Removes a local audio tracks that is used for getting some voice |
- // statistics. |
- void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc); |
- |
- // Gather statistics from the session and store them for future use. |
- void UpdateStats(PeerConnectionInterface::StatsOutputLevel level); |
- |
- // Gets a StatsReports of the last collected stats. Note that UpdateStats must |
- // be called before this function to get the most recent stats. |selector| is |
- // a track label or empty string. The most recent reports are stored in |
- // |reports|. |
- // TODO(tommi): Change this contract to accept a callback object instead |
- // of filling in |reports|. As is, there's a requirement that the caller |
- // uses |reports| immediately without allowing any async activity on |
- // the thread (message handling etc) and then discard the results. |
- void GetStats(MediaStreamTrackInterface* track, |
- StatsReports* reports); |
- |
- // Prepare a local or remote SSRC report for the given ssrc. Used internally |
- // in the ExtractStatsFromList template. |
- StatsReport* PrepareReport(bool local, |
- uint32_t ssrc, |
- const StatsReport::Id& transport_id, |
- StatsReport::Direction direction); |
- |
- // Method used by the unittest to force a update of stats since UpdateStats() |
- // that occur less than kMinGatherStatsPeriod number of ms apart will be |
- // ignored. |
- void ClearUpdateStatsCacheForTest(); |
- |
- private: |
- friend class StatsCollectorTest; |
- |
- // Overridden in unit tests to fake timing. |
- virtual double GetTimeNow(); |
- |
- bool CopySelectedReports(const std::string& selector, StatsReports* reports); |
- |
- // Helper method for AddCertificateReports. |
- StatsReport* AddOneCertificateReport( |
- const rtc::SSLCertificate* cert, const StatsReport* issuer); |
- |
- // Helper method for creating IceCandidate report. |is_local| indicates |
- // whether this candidate is local or remote. |
- StatsReport* AddCandidateReport(const cricket::Candidate& candidate, |
- bool local); |
- |
- // Adds a report for this certificate and every certificate in its chain, and |
- // returns the leaf certificate's report. |
- StatsReport* AddCertificateReports(const rtc::SSLCertificate* cert); |
- |
- StatsReport* AddConnectionInfoReport(const std::string& content_name, |
- int component, int connection_id, |
- const StatsReport::Id& channel_report_id, |
- const cricket::ConnectionInfo& info); |
- |
- void ExtractDataInfo(); |
- void ExtractSessionInfo(); |
- void ExtractVoiceInfo(); |
- void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level); |
- void BuildSsrcToTransportId(); |
- webrtc::StatsReport* GetReport(const StatsReport::StatsType& type, |
- const std::string& id, |
- StatsReport::Direction direction); |
- |
- // Helper method to get stats from the local audio tracks. |
- void UpdateStatsFromExistingLocalAudioTracks(); |
- void UpdateReportFromAudioTrack(AudioTrackInterface* track, |
- StatsReport* report); |
- |
- // Helper method to get the id for the track identified by ssrc. |
- // |direction| tells if the track is for sending or receiving. |
- bool GetTrackIdBySsrc(uint32_t ssrc, |
- std::string* track_id, |
- StatsReport::Direction direction); |
- |
- // Helper method to update the timestamp of track records. |
- void UpdateTrackReports(); |
- |
- // A collection for all of our stats reports. |
- StatsCollection reports_; |
- TrackIdMap track_ids_; |
- // Raw pointer to the peer connection the statistics are gathered from. |
- PeerConnection* const pc_; |
- double stats_gathering_started_; |
- ProxyTransportMap proxy_to_transport_; |
- |
- // TODO(tommi): We appear to be holding on to raw pointers to reference |
- // counted objects? We should be using scoped_refptr here. |
- typedef std::vector<std::pair<AudioTrackInterface*, uint32_t> > |
- LocalAudioTrackVector; |
- LocalAudioTrackVector local_audio_tracks_; |
-}; |
- |
-} // namespace webrtc |
- |
-#endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_ |