322a564f49
Makes the id now be "datachannel_#####" where '####' is the id number for the datachannel. Adds a timestamp to the data channel reports. Implements unit tests to verify that the timestamp is set correctly. BUG=1805 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33119004 Cr-Commit-Position: refs/heads/master@{#8236} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8236 4adac7df-926f-26a2-2b94-8c16560cd09d
155 lines
6.5 KiB
C++
155 lines
6.5 KiB
C++
/*
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* libjingle
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* Copyright 2012 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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// This file contains a class used for gathering statistics from an ongoing
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// libjingle PeerConnection.
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#ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_
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#define TALK_APP_WEBRTC_STATSCOLLECTOR_H_
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#include <map>
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#include <string>
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#include <vector>
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#include "talk/app/webrtc/mediastreaminterface.h"
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#include "talk/app/webrtc/mediastreamsignaling.h"
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#include "talk/app/webrtc/peerconnectioninterface.h"
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#include "talk/app/webrtc/statstypes.h"
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#include "talk/app/webrtc/webrtcsession.h"
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namespace webrtc {
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// Conversion function to convert candidate type string to the corresponding one
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// from enum RTCStatsIceCandidateType.
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const char* IceCandidateTypeToStatsType(const std::string& candidate_type);
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// Conversion function to convert adapter type to report string which are more
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// fitting to the general style of http://w3c.github.io/webrtc-stats. This is
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// only used by stats collector.
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const char* AdapterTypeToStatsType(rtc::AdapterType type);
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class StatsCollector {
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public:
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// The caller is responsible for ensuring that the session outlives the
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// StatsCollector instance.
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explicit StatsCollector(WebRtcSession* session);
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virtual ~StatsCollector();
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// Adds a MediaStream with tracks that can be used as a |selector| in a call
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// to GetStats.
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void AddStream(MediaStreamInterface* stream);
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// Adds a local audio track that is used for getting some voice statistics.
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void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
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// Removes a local audio tracks that is used for getting some voice
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// statistics.
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void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
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// Gather statistics from the session and store them for future use.
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void UpdateStats(PeerConnectionInterface::StatsOutputLevel level);
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// Gets a StatsReports of the last collected stats. Note that UpdateStats must
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// be called before this function to get the most recent stats. |selector| is
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// a track label or empty string. The most recent reports are stored in
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// |reports|.
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// TODO(tommi): Change this contract to accept a callback object instead
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// of filling in |reports|. As is, there's a requirement that the caller
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// uses |reports| immediately without allowing any async activity on
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// the thread (message handling etc) and then discard the results.
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void GetStats(MediaStreamTrackInterface* track,
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StatsReports* reports);
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// Prepare a local or remote SSRC report for the given ssrc. Used internally
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// in the ExtractStatsFromList template.
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StatsReport* PrepareReport(bool local, uint32 ssrc,
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const std::string& transport_id, StatsReport::Direction direction);
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// Method used by the unittest to force a update of stats since UpdateStats()
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// that occur less than kMinGatherStatsPeriod number of ms apart will be
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// ignored.
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void ClearUpdateStatsCacheForTest();
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private:
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friend class StatsCollectorTest;
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// Overridden in unit tests to fake timing.
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virtual double GetTimeNow();
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bool CopySelectedReports(const std::string& selector, StatsReports* reports);
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// Helper method for AddCertificateReports.
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std::string AddOneCertificateReport(
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const rtc::SSLCertificate* cert, const std::string& issuer_id);
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// Helper method for creating IceCandidate report. |is_local| indicates
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// whether this candidate is local or remote.
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std::string AddCandidateReport(const cricket::Candidate& candidate,
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bool local);
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// Adds a report for this certificate and every certificate in its chain, and
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// returns the leaf certificate's report's ID.
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std::string AddCertificateReports(const rtc::SSLCertificate* cert);
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void ExtractDataInfo();
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void ExtractSessionInfo();
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void ExtractVoiceInfo();
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void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level);
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void BuildSsrcToTransportId();
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webrtc::StatsReport* GetOrCreateReport(const StatsReport::StatsType& type,
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const std::string& id,
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StatsReport::Direction direction);
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webrtc::StatsReport* GetReport(const StatsReport::StatsType& type,
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const std::string& id,
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StatsReport::Direction direction);
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// Helper method to get stats from the local audio tracks.
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void UpdateStatsFromExistingLocalAudioTracks();
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void UpdateReportFromAudioTrack(AudioTrackInterface* track,
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StatsReport* report);
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// Helper method to get the id for the track identified by ssrc.
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// |direction| tells if the track is for sending or receiving.
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bool GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
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StatsReport::Direction direction);
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// A collection for all of our stats reports.
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StatsCollection reports_;
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// Raw pointer to the session the statistics are gathered from.
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WebRtcSession* const session_;
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double stats_gathering_started_;
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cricket::ProxyTransportMap proxy_to_transport_;
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typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
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LocalAudioTrackVector;
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LocalAudioTrackVector local_audio_tracks_;
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};
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} // namespace webrtc
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#endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_
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