28e2075280
trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
96 lines
3.6 KiB
C++
96 lines
3.6 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 <string>
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#include <map>
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#include "talk/app/webrtc/mediastreaminterface.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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#include "talk/base/timing.h"
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namespace webrtc {
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class StatsCollector {
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public:
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StatsCollector();
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// Register the session Stats should operate on.
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// Set to NULL if the session has ended.
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void set_session(WebRtcSession* session) {
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session_ = session;
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}
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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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// Gather statistics from the session and store them for future use.
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void UpdateStats();
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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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bool GetStats(MediaStreamTrackInterface* track, StatsReports* reports);
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WebRtcSession* session() { return session_; }
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// Prepare an SSRC report for the given ssrc. Used internally.
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StatsReport* PrepareReport(uint32 ssrc, const std::string& transport);
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// Extracts the ID of a Transport belonging to an SSRC. Used internally.
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bool GetTransportIdFromProxy(const std::string& proxy,
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std::string* transport_id);
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private:
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bool CopySelectedReports(const std::string& selector, StatsReports* reports);
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void ExtractSessionInfo();
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void ExtractVoiceInfo();
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void ExtractVideoInfo();
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double GetTimeNow();
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void BuildSsrcToTransportId();
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// A map from the report id to the report.
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std::map<std::string, webrtc::StatsReport> reports_;
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// Raw pointer to the session the statistics are gathered from.
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WebRtcSession* session_;
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double stats_gathering_started_;
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talk_base::Timing timing_;
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cricket::ProxyTransportMap proxy_to_transport_;
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};
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} // namespace webrtc
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#endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_
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