
Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
82 lines
2.3 KiB
C++
82 lines
2.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
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#define WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
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#include <list>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/interface/module.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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class CallStatsObserver;
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class CriticalSectionWrapper;
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class RtcpRttStats;
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// CallStats keeps track of statistics for a call.
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class CallStats : public Module {
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public:
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friend class RtcpObserver;
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CallStats();
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~CallStats();
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// Implements Module, to use the process thread.
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virtual int64_t TimeUntilNextProcess() OVERRIDE;
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virtual int32_t Process() OVERRIDE;
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// Returns a RtcpRttStats to register at a statistics provider. The object
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// has the same lifetime as the CallStats instance.
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RtcpRttStats* rtcp_rtt_stats() const;
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// Registers/deregisters a new observer to receive statistics updates.
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void RegisterStatsObserver(CallStatsObserver* observer);
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void DeregisterStatsObserver(CallStatsObserver* observer);
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// Helper struct keeping track of the time a rtt value is reported.
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struct RttTime {
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RttTime(int64_t new_rtt, int64_t rtt_time)
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: rtt(new_rtt), time(rtt_time) {}
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const int64_t rtt;
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const int64_t time;
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};
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protected:
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void OnRttUpdate(int64_t rtt);
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int64_t avg_rtt_ms() const;
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private:
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// Protecting all members.
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scoped_ptr<CriticalSectionWrapper> crit_;
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// Observer receiving statistics updates.
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scoped_ptr<RtcpRttStats> rtcp_rtt_stats_;
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// The last time 'Process' resulted in statistic update.
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int64_t last_process_time_;
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// The last RTT in the statistics update (zero if there is no valid estimate).
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int64_t max_rtt_ms_;
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int64_t avg_rtt_ms_;
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// All Rtt reports within valid time interval, oldest first.
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std::list<RttTime> reports_;
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// Observers getting stats reports.
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std::list<CallStatsObserver*> observers_;
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DISALLOW_COPY_AND_ASSIGN(CallStats);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
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