Repo created

TMessagesProj/jni/voip/webrtc/video/call_stats2.cc

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+/*
+ *  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 "rtc_base/checks.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(SafeTask(task_safety_.flag(), 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