/* * 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. */ #include "webrtc/video_engine/call_stats.h" #include #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/tick_util.h" namespace webrtc { namespace { // Time interval for updating the observers. const int64_t kUpdateIntervalMs = 1000; // Weight factor to apply to the average rtt. const float kWeightFactor = 0.3f; void RemoveOldReports(int64_t now, std::list* reports) { // A rtt report is considered valid for this long. const int64_t kRttTimeoutMs = 1500; while (!reports->empty() && (now - reports->front().time) > kRttTimeoutMs) { reports->pop_front(); } } int64_t GetMaxRttMs(std::list* reports) { int64_t max_rtt_ms = 0; for (std::list::const_iterator it = reports->begin(); it != reports->end(); ++it) { max_rtt_ms = std::max(it->rtt, max_rtt_ms); } return max_rtt_ms; } int64_t GetAvgRttMs(std::list* reports) { if (reports->empty()) { return 0; } int64_t sum = 0; for (std::list::const_iterator it = reports->begin(); it != reports->end(); ++it) { sum += it->rtt; } return sum / reports->size(); } void UpdateAvgRttMs(std::list* reports, int64_t* avg_rtt) { uint32_t cur_rtt_ms = GetAvgRttMs(reports); if (cur_rtt_ms == 0) { // Reset. *avg_rtt = 0; return; } if (*avg_rtt == 0) { // Initialize. *avg_rtt = cur_rtt_ms; return; } *avg_rtt = *avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor; } } // namespace class RtcpObserver : public RtcpRttStats { public: explicit RtcpObserver(CallStats* owner) : owner_(owner) {} virtual ~RtcpObserver() {} virtual void OnRttUpdate(int64_t rtt) { owner_->OnRttUpdate(rtt); } // Returns the average RTT. virtual int64_t LastProcessedRtt() const { return owner_->avg_rtt_ms(); } private: CallStats* owner_; RTC_DISALLOW_COPY_AND_ASSIGN(RtcpObserver); }; CallStats::CallStats() : crit_(CriticalSectionWrapper::CreateCriticalSection()), rtcp_rtt_stats_(new RtcpObserver(this)), last_process_time_(TickTime::MillisecondTimestamp()), max_rtt_ms_(0), avg_rtt_ms_(0) { } CallStats::~CallStats() { assert(observers_.empty()); } int64_t CallStats::TimeUntilNextProcess() { return last_process_time_ + kUpdateIntervalMs - TickTime::MillisecondTimestamp(); } int32_t CallStats::Process() { CriticalSectionScoped cs(crit_.get()); int64_t now = TickTime::MillisecondTimestamp(); if (now < last_process_time_ + kUpdateIntervalMs) return 0; last_process_time_ = now; RemoveOldReports(now, &reports_); max_rtt_ms_ = GetMaxRttMs(&reports_); UpdateAvgRttMs(&reports_, &avg_rtt_ms_); // If there is a valid rtt, update all observers with the max rtt. // TODO(asapersson): Consider changing this to report the average rtt. if (max_rtt_ms_ > 0) { for (std::list::iterator it = observers_.begin(); it != observers_.end(); ++it) { (*it)->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_); } } return 0; } int64_t CallStats::avg_rtt_ms() const { CriticalSectionScoped cs(crit_.get()); return avg_rtt_ms_; } RtcpRttStats* CallStats::rtcp_rtt_stats() const { return rtcp_rtt_stats_.get(); } void CallStats::RegisterStatsObserver(CallStatsObserver* observer) { CriticalSectionScoped cs(crit_.get()); for (std::list::iterator it = observers_.begin(); it != observers_.end(); ++it) { if (*it == observer) return; } observers_.push_back(observer); } void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) { CriticalSectionScoped cs(crit_.get()); for (std::list::iterator it = observers_.begin(); it != observers_.end(); ++it) { if (*it == observer) { observers_.erase(it); return; } } } void CallStats::OnRttUpdate(int64_t rtt) { CriticalSectionScoped cs(crit_.get()); reports_.push_back(RttTime(rtt, TickTime::MillisecondTimestamp())); } } // namespace webrtc