ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
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/base/scoped_ptr.h"
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#include "webrtc/modules/include/module.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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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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// 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_ENGINE_CALL_STATS_H_
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