The problem with gmock is worked around by commenting out any other override declarations in classes using gmock. NOPRESUBMIT=True BUG=webrtc:3970 Review-Url: https://codereview.webrtc.org/1921653002 Cr-Commit-Position: refs/heads/master@{#12563}
251 lines
8.8 KiB
C++
251 lines
8.8 KiB
C++
/*
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* Copyright (c) 2012 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 "webrtc/modules/congestion_controller/include/congestion_controller.h"
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#include <algorithm>
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#include <memory>
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#include <vector>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/socket.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
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#include "webrtc/modules/pacing/paced_sender.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
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#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
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#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
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#include "webrtc/modules/utility/include/process_thread.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/video/payload_router.h"
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namespace webrtc {
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namespace {
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static const uint32_t kTimeOffsetSwitchThreshold = 30;
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class WrappingBitrateEstimator : public RemoteBitrateEstimator {
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public:
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WrappingBitrateEstimator(RemoteBitrateObserver* observer, Clock* clock)
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: observer_(observer),
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clock_(clock),
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crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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rbe_(new RemoteBitrateEstimatorSingleStream(observer_, clock_)),
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using_absolute_send_time_(false),
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packets_since_absolute_send_time_(0),
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min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {}
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virtual ~WrappingBitrateEstimator() {}
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void IncomingPacket(int64_t arrival_time_ms,
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size_t payload_size,
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const RTPHeader& header,
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bool was_paced) override {
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CriticalSectionScoped cs(crit_sect_.get());
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PickEstimatorFromHeader(header);
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rbe_->IncomingPacket(arrival_time_ms, payload_size, header, was_paced);
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}
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void Process() override {
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CriticalSectionScoped cs(crit_sect_.get());
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rbe_->Process();
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}
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int64_t TimeUntilNextProcess() override {
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CriticalSectionScoped cs(crit_sect_.get());
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return rbe_->TimeUntilNextProcess();
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}
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void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override {
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CriticalSectionScoped cs(crit_sect_.get());
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rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
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}
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void RemoveStream(unsigned int ssrc) override {
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CriticalSectionScoped cs(crit_sect_.get());
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rbe_->RemoveStream(ssrc);
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}
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bool LatestEstimate(std::vector<unsigned int>* ssrcs,
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unsigned int* bitrate_bps) const override {
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CriticalSectionScoped cs(crit_sect_.get());
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return rbe_->LatestEstimate(ssrcs, bitrate_bps);
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}
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void SetMinBitrate(int min_bitrate_bps) override {
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CriticalSectionScoped cs(crit_sect_.get());
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rbe_->SetMinBitrate(min_bitrate_bps);
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min_bitrate_bps_ = min_bitrate_bps;
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}
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private:
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void PickEstimatorFromHeader(const RTPHeader& header)
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EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
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if (header.extension.hasAbsoluteSendTime) {
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// If we see AST in header, switch RBE strategy immediately.
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if (!using_absolute_send_time_) {
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LOG(LS_INFO) <<
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"WrappingBitrateEstimator: Switching to absolute send time RBE.";
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using_absolute_send_time_ = true;
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PickEstimator();
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}
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packets_since_absolute_send_time_ = 0;
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} else {
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// When we don't see AST, wait for a few packets before going back to TOF.
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if (using_absolute_send_time_) {
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++packets_since_absolute_send_time_;
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if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
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LOG(LS_INFO) << "WrappingBitrateEstimator: Switching to transmission "
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<< "time offset RBE.";
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using_absolute_send_time_ = false;
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PickEstimator();
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}
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}
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}
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}
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// Instantiate RBE for Time Offset or Absolute Send Time extensions.
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void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
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if (using_absolute_send_time_) {
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rbe_.reset(new RemoteBitrateEstimatorAbsSendTime(observer_));
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} else {
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rbe_.reset(new RemoteBitrateEstimatorSingleStream(observer_, clock_));
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}
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rbe_->SetMinBitrate(min_bitrate_bps_);
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}
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RemoteBitrateObserver* observer_;
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Clock* const clock_;
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std::unique_ptr<CriticalSectionWrapper> crit_sect_;
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std::unique_ptr<RemoteBitrateEstimator> rbe_;
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bool using_absolute_send_time_;
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uint32_t packets_since_absolute_send_time_;
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int min_bitrate_bps_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
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};
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} // namespace
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CongestionController::CongestionController(
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Clock* clock,
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BitrateObserver* bitrate_observer,
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RemoteBitrateObserver* remote_bitrate_observer)
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: clock_(clock),
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pacer_(new PacedSender(clock_,
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&packet_router_,
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BitrateController::kDefaultStartBitrateKbps,
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PacedSender::kDefaultPaceMultiplier *
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BitrateController::kDefaultStartBitrateKbps,
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0)),
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remote_bitrate_estimator_(
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new WrappingBitrateEstimator(remote_bitrate_observer, clock_)),
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// Constructed last as this object calls the provided callback on
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// construction.
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bitrate_controller_(
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BitrateController::CreateBitrateController(clock_, bitrate_observer)),
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remote_estimator_proxy_(clock_, &packet_router_),
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transport_feedback_adapter_(bitrate_controller_.get(), clock_),
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min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {
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transport_feedback_adapter_.SetBitrateEstimator(
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new RemoteBitrateEstimatorAbsSendTime(&transport_feedback_adapter_));
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transport_feedback_adapter_.GetBitrateEstimator()->SetMinBitrate(
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min_bitrate_bps_);
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}
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CongestionController::~CongestionController() {
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}
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void CongestionController::SetBweBitrates(int min_bitrate_bps,
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int start_bitrate_bps,
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int max_bitrate_bps) {
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// TODO(holmer): We should make sure the default bitrates are set to 10 kbps,
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// and that we don't try to set the min bitrate to 0 from any applications.
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// The congestion controller should allow a min bitrate of 0.
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const int kMinBitrateBps = 10000;
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if (min_bitrate_bps < kMinBitrateBps)
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min_bitrate_bps = kMinBitrateBps;
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if (max_bitrate_bps > 0)
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max_bitrate_bps = std::max(min_bitrate_bps, max_bitrate_bps);
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if (start_bitrate_bps > 0)
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start_bitrate_bps = std::max(min_bitrate_bps, start_bitrate_bps);
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bitrate_controller_->SetBitrates(start_bitrate_bps,
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min_bitrate_bps,
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max_bitrate_bps);
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if (remote_bitrate_estimator_)
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remote_bitrate_estimator_->SetMinBitrate(min_bitrate_bps);
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min_bitrate_bps_ = min_bitrate_bps;
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transport_feedback_adapter_.GetBitrateEstimator()->SetMinBitrate(
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min_bitrate_bps_);
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}
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BitrateController* CongestionController::GetBitrateController() const {
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return bitrate_controller_.get();
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}
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RemoteBitrateEstimator* CongestionController::GetRemoteBitrateEstimator(
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bool send_side_bwe) {
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if (send_side_bwe) {
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return &remote_estimator_proxy_;
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} else {
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return remote_bitrate_estimator_.get();
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}
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}
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TransportFeedbackObserver*
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CongestionController::GetTransportFeedbackObserver() {
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return &transport_feedback_adapter_;
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}
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void CongestionController::UpdatePacerBitrate(int bitrate_kbps,
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int max_bitrate_kbps,
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int min_bitrate_kbps) {
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pacer_->UpdateBitrate(bitrate_kbps, max_bitrate_kbps, min_bitrate_kbps);
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}
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int64_t CongestionController::GetPacerQueuingDelayMs() const {
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return pacer_->QueueInMs();
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}
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void CongestionController::SignalNetworkState(NetworkState state) {
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if (state == kNetworkUp) {
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pacer_->Resume();
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} else {
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pacer_->Pause();
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}
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}
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void CongestionController::OnSentPacket(const rtc::SentPacket& sent_packet) {
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transport_feedback_adapter_.OnSentPacket(sent_packet.packet_id,
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sent_packet.send_time_ms);
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}
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void CongestionController::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
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remote_bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
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transport_feedback_adapter_.OnRttUpdate(avg_rtt_ms, max_rtt_ms);
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}
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int64_t CongestionController::TimeUntilNextProcess() {
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return std::min(bitrate_controller_->TimeUntilNextProcess(),
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remote_bitrate_estimator_->TimeUntilNextProcess());
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}
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void CongestionController::Process() {
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bitrate_controller_->Process();
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remote_bitrate_estimator_->Process();
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}
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} // namespace webrtc
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