| /* |
| * Copyright (c) 2020 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 "video/call_stats2.h" |
| |
| #include <algorithm> |
| #include <memory> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "modules/utility/include/process_thread.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/task_utils/to_queued_task.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| namespace webrtc { |
| namespace internal { |
| namespace { |
| |
| void RemoveOldReports(int64_t now, std::list<CallStats::RttTime>* reports) { |
| static constexpr const int64_t kRttTimeoutMs = 1500; |
| reports->remove_if( |
| [&now](CallStats::RttTime& r) { return now - r.time > kRttTimeoutMs; }); |
| } |
| |
| int64_t GetMaxRttMs(const std::list<CallStats::RttTime>& reports) { |
| int64_t max_rtt_ms = -1; |
| for (const CallStats::RttTime& rtt_time : reports) |
| max_rtt_ms = std::max(rtt_time.rtt, max_rtt_ms); |
| return max_rtt_ms; |
| } |
| |
| int64_t GetAvgRttMs(const std::list<CallStats::RttTime>& reports) { |
| RTC_DCHECK(!reports.empty()); |
| int64_t sum = 0; |
| for (std::list<CallStats::RttTime>::const_iterator it = reports.begin(); |
| it != reports.end(); ++it) { |
| sum += it->rtt; |
| } |
| return sum / reports.size(); |
| } |
| |
| int64_t GetNewAvgRttMs(const std::list<CallStats::RttTime>& reports, |
| int64_t prev_avg_rtt) { |
| if (reports.empty()) |
| return -1; // Reset (invalid average). |
| |
| int64_t cur_rtt_ms = GetAvgRttMs(reports); |
| if (prev_avg_rtt == -1) |
| return cur_rtt_ms; // New initial average value. |
| |
| // Weight factor to apply to the average rtt. |
| // We weigh the old average at 70% against the new average (30%). |
| constexpr const float kWeightFactor = 0.3f; |
| return prev_avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor; |
| } |
| |
| } // namespace |
| |
| constexpr TimeDelta CallStats::kUpdateInterval; |
| |
| CallStats::CallStats(Clock* clock, TaskQueueBase* task_queue) |
| : clock_(clock), |
| max_rtt_ms_(-1), |
| avg_rtt_ms_(-1), |
| sum_avg_rtt_ms_(0), |
| num_avg_rtt_(0), |
| time_of_first_rtt_ms_(-1), |
| task_queue_(task_queue) { |
| RTC_DCHECK(task_queue_); |
| RTC_DCHECK_RUN_ON(task_queue_); |
| } |
| |
| CallStats::~CallStats() { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| RTC_DCHECK(observers_.empty()); |
| |
| repeating_task_.Stop(); |
| |
| UpdateHistograms(); |
| } |
| |
| void CallStats::EnsureStarted() { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| repeating_task_ = |
| RepeatingTaskHandle::DelayedStart(task_queue_, kUpdateInterval, [this]() { |
| UpdateAndReport(); |
| return kUpdateInterval; |
| }); |
| } |
| |
| void CallStats::UpdateAndReport() { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| |
| RemoveOldReports(clock_->CurrentTime().ms(), &reports_); |
| max_rtt_ms_ = GetMaxRttMs(reports_); |
| avg_rtt_ms_ = GetNewAvgRttMs(reports_, avg_rtt_ms_); |
| |
| // If there is a valid rtt, update all observers with the max rtt. |
| if (max_rtt_ms_ >= 0) { |
| RTC_DCHECK_GE(avg_rtt_ms_, 0); |
| for (CallStatsObserver* observer : observers_) |
| observer->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_); |
| // Sum for Histogram of average RTT reported over the entire call. |
| sum_avg_rtt_ms_ += avg_rtt_ms_; |
| ++num_avg_rtt_; |
| } |
| } |
| |
| void CallStats::RegisterStatsObserver(CallStatsObserver* observer) { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| if (!absl::c_linear_search(observers_, observer)) |
| observers_.push_back(observer); |
| } |
| |
| void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| observers_.remove(observer); |
| } |
| |
| int64_t CallStats::LastProcessedRtt() const { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| // No need for locking since we're on the construction thread. |
| return avg_rtt_ms_; |
| } |
| |
| void CallStats::OnRttUpdate(int64_t rtt) { |
| // This callback may for some RtpRtcp module instances (video send stream) be |
| // invoked from a separate task queue, in other cases, we should already be |
| // on the correct TQ. |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| auto update = [this, rtt, now_ms]() { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| reports_.push_back(RttTime(rtt, now_ms)); |
| if (time_of_first_rtt_ms_ == -1) |
| time_of_first_rtt_ms_ = now_ms; |
| UpdateAndReport(); |
| }; |
| |
| if (task_queue_->IsCurrent()) { |
| update(); |
| } else { |
| task_queue_->PostTask(ToQueuedTask(task_safety_, std::move(update))); |
| } |
| } |
| |
| void CallStats::UpdateHistograms() { |
| RTC_DCHECK_RUN_ON(task_queue_); |
| |
| if (time_of_first_rtt_ms_ == -1 || num_avg_rtt_ < 1) |
| return; |
| |
| int64_t elapsed_sec = |
| (clock_->TimeInMilliseconds() - time_of_first_rtt_ms_) / 1000; |
| if (elapsed_sec >= metrics::kMinRunTimeInSeconds) { |
| int64_t avg_rtt_ms = (sum_avg_rtt_ms_ + num_avg_rtt_ / 2) / num_avg_rtt_; |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms); |
| } |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |