7623ce4aeb
Reason for revert: Bot breakage caused by TickTime::UseFakeClock has been removed. Original issue's description: > Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) > > Reason for revert: > Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots. > > Original issue's description: > > Merge webrtc/video_engine/ into webrtc/video/ > > > > BUG=webrtc:1695 > > R=mflodman@webrtc.org > > > > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646 > > Cr-Commit-Position: refs/heads/master@{#10926} > > TBR=mflodman@webrtc.org,pbos@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:1695 > > Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518 > Cr-Commit-Position: refs/heads/master@{#10937} BUG=webrtc:1695 TBR=mflodman@webrtc.org,kjellander@webrtc.org NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1510183002 . Cr-Commit-Position: refs/heads/master@{#10948}
84 lines
2.4 KiB
C++
84 lines
2.4 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_CALL_STATS_H_
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#define WEBRTC_VIDEO_CALL_STATS_H_
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#include <list>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/include/module.h"
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#include "webrtc/system_wrappers/include/clock.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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explicit CallStats(Clock* clock);
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~CallStats();
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// Implements Module, to use the process thread.
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int64_t TimeUntilNextProcess() override;
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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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Clock* const clock_;
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// Protecting all members.
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rtc::scoped_ptr<CriticalSectionWrapper> crit_;
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// Observer receiving statistics updates.
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rtc::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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RTC_DISALLOW_COPY_AND_ASSIGN(CallStats);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_CALL_STATS_H_
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