| /* |
| * 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 VIDEO_CALL_STATS_H_ |
| #define VIDEO_CALL_STATS_H_ |
| |
| #include <list> |
| #include <memory> |
| |
| #include "modules/include/module.h" |
| #include "rtc_base/constructormagic.h" |
| #include "rtc_base/criticalsection.h" |
| #include "system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| |
| class CallStatsObserver; |
| class RtcpRttStats; |
| |
| // CallStats keeps track of statistics for a call. |
| class CallStats : public Module { |
| public: |
| friend class RtcpObserver; |
| |
| explicit CallStats(Clock* clock); |
| ~CallStats(); |
| |
| // Implements Module, to use the process thread. |
| int64_t TimeUntilNextProcess() override; |
| void 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: |
| void UpdateHistograms(); |
| |
| Clock* const clock_; |
| // Protecting all members. |
| rtc::CriticalSection crit_; |
| // Observer receiving statistics updates. |
| std::unique_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_; |
| int64_t sum_avg_rtt_ms_ RTC_GUARDED_BY(crit_); |
| int64_t num_avg_rtt_ RTC_GUARDED_BY(crit_); |
| int64_t time_of_first_rtt_ms_ RTC_GUARDED_BY(crit_); |
| |
| // All Rtt reports within valid time interval, oldest first. |
| std::list<RttTime> reports_; |
| |
| // Observers getting stats reports. |
| std::list<CallStatsObserver*> observers_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(CallStats); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // VIDEO_CALL_STATS_H_ |