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1618095100
This CL simplifies a lot of code that can be cleaned up after the merge of RtpTransportControllerSend and SendSideCongestionController. In particular, the role of CongestionControlHandler is reduced to only handle the pacer pushback and stream pausing mechanism. Bug: webrtc:9586 Change-Id: Idbc1e968efd35e6df6129bc307f6bc1db18d20f2 Reviewed-on: https://webrtc-review.googlesource.com/c/113947 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25994}
106 lines
3.9 KiB
C++
106 lines
3.9 KiB
C++
/*
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* Copyright (c) 2018 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/congestion_controller/rtp/control_handler.h"
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#include <algorithm>
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#include <vector>
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#include "api/units/data_rate.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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namespace {
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// When PacerPushbackExperiment is enabled, build-up in the pacer due to
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// the congestion window and/or data spikes reduces encoder allocations.
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bool IsPacerPushbackExperimentEnabled() {
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return field_trial::IsEnabled("WebRTC-PacerPushbackExperiment");
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}
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// By default, pacer emergency stops encoder when buffer reaches a high level.
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bool IsPacerEmergencyStopDisabled() {
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return field_trial::IsEnabled("WebRTC-DisablePacerEmergencyStop");
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}
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} // namespace
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CongestionControlHandler::CongestionControlHandler()
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: pacer_pushback_experiment_(IsPacerPushbackExperimentEnabled()),
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disable_pacer_emergency_stop_(IsPacerEmergencyStopDisabled()) {
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sequenced_checker_.Detach();
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}
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CongestionControlHandler::~CongestionControlHandler() {}
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void CongestionControlHandler::SetTargetRate(
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TargetTransferRate new_target_rate) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
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last_incoming_ = new_target_rate;
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}
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void CongestionControlHandler::SetNetworkAvailability(bool network_available) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
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network_available_ = network_available;
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}
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void CongestionControlHandler::SetPacerQueue(TimeDelta expected_queue_time) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
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pacer_expected_queue_ms_ = expected_queue_time.ms();
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}
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absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
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if (!last_incoming_.has_value())
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return absl::nullopt;
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TargetTransferRate new_outgoing = *last_incoming_;
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DataRate log_target_rate = new_outgoing.target_rate;
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bool pause_encoding = false;
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if (!network_available_) {
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pause_encoding = true;
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} else if (pacer_pushback_experiment_) {
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const int64_t queue_length_ms = pacer_expected_queue_ms_;
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if (queue_length_ms == 0) {
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encoding_rate_ratio_ = 1.0;
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} else if (queue_length_ms > 50) {
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double encoding_ratio = 1.0 - queue_length_ms / 1000.0;
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encoding_rate_ratio_ = std::min(encoding_rate_ratio_, encoding_ratio);
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encoding_rate_ratio_ = std::max(encoding_rate_ratio_, 0.0);
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}
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new_outgoing.target_rate = new_outgoing.target_rate * encoding_rate_ratio_;
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log_target_rate = new_outgoing.target_rate;
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if (new_outgoing.target_rate < DataRate::kbps(50))
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pause_encoding = true;
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} else if (!disable_pacer_emergency_stop_ &&
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pacer_expected_queue_ms_ > PacedSender::kMaxQueueLengthMs) {
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pause_encoding = true;
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}
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if (pause_encoding)
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new_outgoing.target_rate = DataRate::Zero();
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if (!last_reported_ ||
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last_reported_->target_rate != new_outgoing.target_rate ||
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(!new_outgoing.target_rate.IsZero() &&
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(last_reported_->network_estimate.loss_rate_ratio !=
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new_outgoing.network_estimate.loss_rate_ratio ||
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last_reported_->network_estimate.round_trip_time !=
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new_outgoing.network_estimate.round_trip_time))) {
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if (encoder_paused_in_last_report_ != pause_encoding)
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RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
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<< ToString(log_target_rate) << ".";
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encoder_paused_in_last_report_ = pause_encoding;
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last_reported_ = new_outgoing;
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return new_outgoing;
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}
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return absl::nullopt;
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}
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} // namespace webrtc
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