ea86bb74fc
This reverts commit 65792c5a5c542201f7b9feefded505842692e6ed. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Reland "Moved congestion controller to task queue.""" > > This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Reland "Moved congestion controller to task queue."" > > > > This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793. > > > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > > > Original change's description: > > > Reland "Moved congestion controller to task queue." > > > > > > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. > > > > > > Original change's description: > > > > Moved congestion controller to task queue. > > > > > > > > The goal of this work is to make it easier to experiment with the > > > > bandwidth estimation implementation. For this reason network control > > > > functionality is moved from SendSideCongestionController(SSCC), > > > > PacedSender and BitrateController to the newly created > > > > GoogCcNetworkController which implements the newly created > > > > NetworkControllerInterface. This allows the implementation to be > > > > replaced at runtime in the future. > > > > > > > > This is the first part of a split of a larger CL, see: > > > > https://webrtc-review.googlesource.com/c/src/+/39788/8 > > > > For further explanations. > > > > > > > > Bug: webrtc:8415 > > > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > > > > Reviewed-on: https://webrtc-review.googlesource.com/43840 > > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#21868} > > > > > > Bug: webrtc:8415 > > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > > > Reviewed-on: https://webrtc-review.googlesource.com/48000 > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21899} > > > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:8415 > > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > > Reviewed-on: https://webrtc-review.googlesource.com/52980 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22017} > > TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8415 > Reviewed-on: https://webrtc-review.googlesource.com/53262 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22023} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53360 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22024}
87 lines
3.1 KiB
C++
87 lines
3.1 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/pacing/alr_detector.h"
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#include <algorithm>
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#include <string>
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#include <cstdio>
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#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/experiments/alr_experiment.h"
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#include "rtc_base/format_macros.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/timeutils.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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AlrDetector::AlrDetector() : AlrDetector(nullptr) {}
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AlrDetector::AlrDetector(RtcEventLog* event_log)
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: bandwidth_usage_percent_(kDefaultAlrBandwidthUsagePercent),
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alr_start_budget_level_percent_(kDefaultAlrStartBudgetLevelPercent),
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alr_stop_budget_level_percent_(kDefaultAlrStopBudgetLevelPercent),
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alr_budget_(0, true),
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event_log_(event_log) {
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RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
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rtc::Optional<AlrExperimentSettings> experiment_settings =
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AlrExperimentSettings::CreateFromFieldTrial(
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AlrExperimentSettings::kScreenshareProbingBweExperimentName);
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if (!experiment_settings) {
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experiment_settings = AlrExperimentSettings::CreateFromFieldTrial(
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AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
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}
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if (experiment_settings) {
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alr_stop_budget_level_percent_ =
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experiment_settings->alr_stop_budget_level_percent;
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alr_start_budget_level_percent_ =
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experiment_settings->alr_start_budget_level_percent;
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bandwidth_usage_percent_ = experiment_settings->alr_bandwidth_usage_percent;
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}
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}
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AlrDetector::~AlrDetector() {}
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void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t delta_time_ms) {
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alr_budget_.UseBudget(bytes_sent);
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alr_budget_.IncreaseBudget(delta_time_ms);
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bool state_changed = false;
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if (alr_budget_.budget_level_percent() > alr_start_budget_level_percent_ &&
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!alr_started_time_ms_) {
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alr_started_time_ms_.emplace(rtc::TimeMillis());
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state_changed = true;
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} else if (alr_budget_.budget_level_percent() <
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alr_stop_budget_level_percent_ &&
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alr_started_time_ms_) {
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state_changed = true;
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alr_started_time_ms_.reset();
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}
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if (event_log_ && state_changed) {
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event_log_->Log(
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rtc::MakeUnique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
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}
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}
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void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
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RTC_DCHECK(bitrate_bps);
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const auto target_rate_kbps = static_cast<int64_t>(bitrate_bps) *
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bandwidth_usage_percent_ / (1000 * 100);
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alr_budget_.set_target_rate_kbps(rtc::dchecked_cast<int>(target_rate_kbps));
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}
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rtc::Optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
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const {
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return alr_started_time_ms_;
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}
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} // namespace webrtc
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