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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 StatsCollector {
public:
enum TrackDirection {
kSending = 0,
kReceiving,
};
// The caller is responsible for ensuring that the session outlives the
// StatsCollector instance.
explicit StatsCollector(WebRtcSession* session);
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 ssrc);
// Removes a local audio tracks that is used for getting some voice
// statistics.
void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32 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 an SSRC report for the given ssrc. Used internally
// in the ExtractStatsFromList template.
StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport,
TrackDirection direction);
// Prepare an SSRC report for the given remote ssrc. Used internally.
StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport,
TrackDirection 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 ClearUpdateStatsCache();
private:
bool CopySelectedReports(const std::string& selector, StatsReports* reports);
// Helper method for AddCertificateReports.
std::string AddOneCertificateReport(
const rtc::SSLCertificate* cert, const std::string& issuer_id);
// Adds a report for this certificate and every certificate in its chain, and
// returns the leaf certificate's report's ID.
std::string AddCertificateReports(const rtc::SSLCertificate* cert);
void ExtractSessionInfo();
void ExtractVoiceInfo();
void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level);
void BuildSsrcToTransportId();
webrtc::StatsReport* GetOrCreateReport(const std::string& type,
const std::string& id,
TrackDirection direction);
webrtc::StatsReport* GetReport(const std::string& type,
const std::string& id,
TrackDirection 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 ssrc, std::string* track_id,
TrackDirection direction);
// A map from the report id to the report.
StatsSet reports_;
// Raw pointer to the session the statistics are gathered from.
WebRtcSession* const session_;
double stats_gathering_started_;
cricket::ProxyTransportMap proxy_to_transport_;
typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
LocalAudioTrackVector;
LocalAudioTrackVector local_audio_tracks_;
};
} // namespace webrtc
#endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_
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