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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 "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