kjellander@webrtc.org 14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00

82 lines
2.3 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
#define WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
#include <list>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/interface/module.h"
namespace webrtc {
class CallStatsObserver;
class CriticalSectionWrapper;
class RtcpRttStats;
// CallStats keeps track of statistics for a call.
class CallStats : public Module {
public:
friend class RtcpObserver;
CallStats();
~CallStats();
// Implements Module, to use the process thread.
int64_t TimeUntilNextProcess() override;
int32_t Process() override;
// Returns a RtcpRttStats to register at a statistics provider. The object
// has the same lifetime as the CallStats instance.
RtcpRttStats* rtcp_rtt_stats() const;
// Registers/deregisters a new observer to receive statistics updates.
void RegisterStatsObserver(CallStatsObserver* observer);
void DeregisterStatsObserver(CallStatsObserver* observer);
// Helper struct keeping track of the time a rtt value is reported.
struct RttTime {
RttTime(int64_t new_rtt, int64_t rtt_time)
: rtt(new_rtt), time(rtt_time) {}
const int64_t rtt;
const int64_t time;
};
protected:
void OnRttUpdate(int64_t rtt);
int64_t avg_rtt_ms() const;
private:
// Protecting all members.
rtc::scoped_ptr<CriticalSectionWrapper> crit_;
// Observer receiving statistics updates.
rtc::scoped_ptr<RtcpRttStats> rtcp_rtt_stats_;
// The last time 'Process' resulted in statistic update.
int64_t last_process_time_;
// The last RTT in the statistics update (zero if there is no valid estimate).
int64_t max_rtt_ms_;
int64_t avg_rtt_ms_;
// All Rtt reports within valid time interval, oldest first.
std::list<RttTime> reports_;
// Observers getting stats reports.
std::list<CallStatsObserver*> observers_;
DISALLOW_COPY_AND_ASSIGN(CallStats);
};
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_CALL_STATS_H_