| /* |
| * Copyright (c) 2018 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 "modules/congestion_controller/rtp/control_handler.h" |
| |
| #include <algorithm> |
| #include <vector> |
| |
| #include "api/units/data_rate.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/numerics/safe_minmax.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| // By default, pacer emergency stops encoder when buffer reaches a high level. |
| bool IsPacerEmergencyStopDisabled() { |
| return field_trial::IsEnabled("WebRTC-DisablePacerEmergencyStop"); |
| } |
| |
| } // namespace |
| CongestionControlHandler::CongestionControlHandler() |
| : disable_pacer_emergency_stop_(IsPacerEmergencyStopDisabled()) { |
| sequenced_checker_.Detach(); |
| } |
| |
| CongestionControlHandler::~CongestionControlHandler() {} |
| |
| void CongestionControlHandler::SetTargetRate( |
| TargetTransferRate new_target_rate) { |
| RTC_DCHECK_RUN_ON(&sequenced_checker_); |
| RTC_CHECK(new_target_rate.at_time.IsFinite()); |
| last_incoming_ = new_target_rate; |
| } |
| |
| void CongestionControlHandler::SetNetworkAvailability(bool network_available) { |
| RTC_DCHECK_RUN_ON(&sequenced_checker_); |
| network_available_ = network_available; |
| } |
| |
| void CongestionControlHandler::SetPacerQueue(TimeDelta expected_queue_time) { |
| RTC_DCHECK_RUN_ON(&sequenced_checker_); |
| pacer_expected_queue_ms_ = expected_queue_time.ms(); |
| } |
| |
| absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() { |
| RTC_DCHECK_RUN_ON(&sequenced_checker_); |
| if (!last_incoming_.has_value()) |
| return absl::nullopt; |
| TargetTransferRate new_outgoing = *last_incoming_; |
| DataRate log_target_rate = new_outgoing.target_rate; |
| bool pause_encoding = false; |
| if (!network_available_) { |
| pause_encoding = true; |
| } else if (!disable_pacer_emergency_stop_ && |
| pacer_expected_queue_ms_ > PacedSender::kMaxQueueLengthMs) { |
| pause_encoding = true; |
| } |
| if (pause_encoding) |
| new_outgoing.target_rate = DataRate::Zero(); |
| if (!last_reported_ || |
| last_reported_->target_rate != new_outgoing.target_rate || |
| (!new_outgoing.target_rate.IsZero() && |
| (last_reported_->network_estimate.loss_rate_ratio != |
| new_outgoing.network_estimate.loss_rate_ratio || |
| last_reported_->network_estimate.round_trip_time != |
| new_outgoing.network_estimate.round_trip_time))) { |
| if (encoder_paused_in_last_report_ != pause_encoding) |
| RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: " |
| << ToString(log_target_rate) << "."; |
| encoder_paused_in_last_report_ = pause_encoding; |
| last_reported_ = new_outgoing; |
| return new_outgoing; |
| } |
| return absl::nullopt; |
| } |
| |
| } // namespace webrtc |