Move talk/app/webrtc to webrtc/api

talk/app/webrtc/statscollector.h

-/*
- * 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_