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https://github.com/kunkundi/crossdesk.git
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[feat] receive and parse congestion control feedback supported
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@@ -1,5 +1,201 @@
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#include "congestion_control.h"
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#include <algorithm>
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#include <numeric>
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#include <vector>
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#include "log.h"
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constexpr int64_t kLossUpdateInterval = 1000;
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// Pacing-rate relative to our target send rate.
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// Multiplicative factor that is applied to the target bitrate to calculate
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// the number of bytes that can be transmitted per interval.
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// Increasing this factor will result in lower delays in cases of bitrate
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// overshoots from the encoder.
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constexpr float kDefaultPaceMultiplier = 2.5f;
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// If the probe result is far below the current throughput estimate
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// it's unlikely that the probe is accurate, so we don't want to drop too far.
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// However, if we actually are overusing, we want to drop to something slightly
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// below the current throughput estimate to drain the network queues.
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constexpr double kProbeDropThroughputFraction = 0.85;
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CongestionControl::CongestionControl() {}
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CongestionControl::~CongestionControl() {}
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CongestionControl::~CongestionControl() {}
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NetworkControlUpdate CongestionControl::OnTransportPacketsFeedback(
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TransportPacketsFeedback report) {
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if (report.packet_feedbacks.empty()) {
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// TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
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// against building very large network queues.
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return NetworkControlUpdate();
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}
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if (congestion_window_pushback_controller_) {
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congestion_window_pushback_controller_->UpdateOutstandingData(
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report.data_in_flight);
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}
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int64_t max_feedback_rtt = std::numeric_limits<int64_t>::min();
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int64_t min_propagation_rtt = std::numeric_limits<int64_t>::max();
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int64_t max_recv_time = std::numeric_limits<int64_t>::min();
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std::vector<PacketResult> feedbacks = report.ReceivedWithSendInfo();
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for (const auto& feedback : feedbacks)
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max_recv_time = std::max(max_recv_time, feedback.receive_time);
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for (const auto& feedback : feedbacks) {
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int64_t feedback_rtt =
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report.feedback_time - feedback.sent_packet.send_time;
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int64_t min_pending_time = max_recv_time - feedback.receive_time;
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int64_t propagation_rtt = feedback_rtt - min_pending_time;
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max_feedback_rtt = std::max(max_feedback_rtt, feedback_rtt);
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min_propagation_rtt = std::min(min_propagation_rtt, propagation_rtt);
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}
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if (max_feedback_rtt != std::numeric_limits<int64_t>::min() &&
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min_propagation_rtt != std::numeric_limits<int64_t>::max()) {
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feedback_max_rtts_.push_back(max_feedback_rtt);
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const size_t kMaxFeedbackRttWindow = 32;
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if (feedback_max_rtts_.size() > kMaxFeedbackRttWindow)
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feedback_max_rtts_.pop_front();
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// TODO(srte): Use time since last unacknowledged packet.
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// bandwidth_estimation_->UpdatePropagationRtt(report.feedback_time,
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// min_propagation_rtt);
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}
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if (packet_feedback_only_) {
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if (!feedback_max_rtts_.empty()) {
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int64_t sum_rtt_ms =
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std::accumulate(feedback_max_rtts_.begin(), feedback_max_rtts_.end(),
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static_cast<int64_t>(0));
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// int64_t mean_rtt_ms = sum_rtt_ms / feedback_max_rtts_.size();
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// if (delay_based_bwe_) delay_based_bwe_->OnRttUpdate(mean_rtt_ms);
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}
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int64_t feedback_min_rtt = std::numeric_limits<int64_t>::max();
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for (const auto& packet_feedback : feedbacks) {
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int64_t pending_time = max_recv_time - packet_feedback.receive_time;
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int64_t rtt = report.feedback_time -
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packet_feedback.sent_packet.send_time - pending_time;
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// Value used for predicting NACK round trip time in FEC controller.
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feedback_min_rtt = std::min(rtt, feedback_min_rtt);
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}
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if (feedback_min_rtt != std::numeric_limits<int64_t>::max() &&
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feedback_min_rtt != std::numeric_limits<int64_t>::min()) {
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// bandwidth_estimation_->UpdateRtt(feedback_min_rtt,
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// report.feedback_time);
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}
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expected_packets_since_last_loss_update_ +=
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report.PacketsWithFeedback().size();
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for (const auto& packet_feedback : report.PacketsWithFeedback()) {
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if (!packet_feedback.IsReceived())
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lost_packets_since_last_loss_update_ += 1;
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}
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if (report.feedback_time > next_loss_update_) {
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next_loss_update_ = report.feedback_time + kLossUpdateInterval;
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// bandwidth_estimation_->UpdatePacketsLost(
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// lost_packets_since_last_loss_update_,
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// expected_packets_since_last_loss_update_, report.feedback_time);
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expected_packets_since_last_loss_update_ = 0;
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lost_packets_since_last_loss_update_ = 0;
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}
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}
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// std::optional<int64_t> alr_start_time =
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// alr_detector_->GetApplicationLimitedRegionStartTime();
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// if (previously_in_alr_ && !alr_start_time.has_value()) {
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// int64_t now_ms = report.feedback_time;
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// acknowledged_bitrate_estimator_->SetAlrEndedTime(report.feedback_time);
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// probe_controller_->SetAlrEndedTimeMs(now_ms);
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// }
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// previously_in_alr_ = alr_start_time.has_value();
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// acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
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// report.SortedByReceiveTime());
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// auto acknowledged_bitrate = acknowledged_bitrate_estimator_->bitrate();
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// bandwidth_estimation_->SetAcknowledgedRate(acknowledged_bitrate,
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// report.feedback_time);
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for (const auto& feedback : report.SortedByReceiveTime()) {
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if (feedback.sent_packet.pacing_info.probe_cluster_id !=
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PacedPacketInfo::kNotAProbe) {
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// probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(feedback);
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}
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}
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// if (network_estimator_) {
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// network_estimator_->OnTransportPacketsFeedback(report);
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// // SetNetworkStateEstimate(network_estimator_->GetCurrentEstimate());
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// }
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// std::optional<int64_t> probe_bitrate =
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// probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate();
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// if (ignore_probes_lower_than_network_estimate_ && probe_bitrate &&
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// estimate_ && *probe_bitrate < delay_based_bwe_->last_estimate() &&
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// *probe_bitrate < estimate_->link_capacity_lower) {
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// probe_bitrate.reset();
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// }
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// if (limit_probes_lower_than_throughput_estimate_ && probe_bitrate &&
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// acknowledged_bitrate) {
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// Limit the backoff to something slightly below the acknowledged
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// bitrate. ("Slightly below" because we want to drain the queues
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// if we are actually overusing.)
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// The acknowledged bitrate shouldn't normally be higher than the delay
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// based estimate, but it could happen e.g. due to packet bursts or
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// encoder overshoot. We use std::min to ensure that a probe result
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// below the current BWE never causes an increase.
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// int64_t limit =
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// std::min(delay_based_bwe_->last_estimate(),
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// *acknowledged_bitrate * kProbeDropThroughputFraction);
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// probe_bitrate = std::max(*probe_bitrate, limit);
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// }
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NetworkControlUpdate update;
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bool recovered_from_overuse = false;
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// DelayBasedBwe::Result result;
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// result = delay_based_bwe_->IncomingPacketFeedbackVector(
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// report, acknowledged_bitrate, probe_bitrate, estimate_,
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// alr_start_time.has_value());
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// if (result.updated) {
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// if (result.probe) {
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// bandwidth_estimation_->SetSendBitrate(result.target_bitrate,
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// report.feedback_time);
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// }
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// Since SetSendBitrate now resets the delay-based estimate, we have to
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// call UpdateDelayBasedEstimate after SetSendBitrate.
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// bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time,
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// result.target_bitrate);
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// }
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// bandwidth_estimation_->UpdateLossBasedEstimator(
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// report, result.delay_detector_state, probe_bitrate,
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// alr_start_time.has_value());
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// if (result.updated) {
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// // Update the estimate in the ProbeController, in case we want to probe.
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// MaybeTriggerOnNetworkChanged(&update, report.feedback_time);
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// }
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// recovered_from_overuse = result.recovered_from_overuse;
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// if (recovered_from_overuse) {
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// probe_controller_->SetAlrStartTimeMs(alr_start_time);
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// auto probes = probe_controller_->RequestProbe(report.feedback_time);
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// update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
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// probes.begin(), probes.end());
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// }
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// No valid RTT could be because send-side BWE isn't used, in which case
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// we don't try to limit the outstanding packets.
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// if (rate_control_settings_.UseCongestionWindow() &&
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// max_feedback_rtt.IsFinite()) {
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// UpdateCongestionWindowSize();
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// }
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if (congestion_window_pushback_controller_ && current_data_window_) {
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congestion_window_pushback_controller_->SetDataWindow(
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*current_data_window_);
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} else {
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update.congestion_window = current_data_window_;
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}
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return update;
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}
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