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Unified Diff: talk/app/webrtc/statscollector.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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Index: talk/app/webrtc/statscollector.h
diff --git a/talk/app/webrtc/statscollector.h b/talk/app/webrtc/statscollector.h
deleted file mode 100644
index 56db79de209d8a83b85bb61aa3943a7a15115e7f..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/statscollector.h
+++ /dev/null
@@ -1,169 +0,0 @@
-/*
- * 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_
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