| /* |
| * Copyright (c) 2017 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/include/receive_side_congestion_controller.h" |
| |
| #include "api/media_types.h" |
| #include "api/units/data_rate.h" |
| #include "modules/pacing/packet_router.h" |
| #include "modules/remote_bitrate_estimator/include/bwe_defines.h" |
| #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h" |
| #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" |
| #include "rtc_base/logging.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| static const uint32_t kTimeOffsetSwitchThreshold = 30; |
| } // namespace |
| |
| void ReceiveSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms, |
| int64_t max_rtt_ms) { |
| MutexLock lock(&mutex_); |
| rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms); |
| } |
| |
| void ReceiveSideCongestionController::RemoveStream(uint32_t ssrc) { |
| MutexLock lock(&mutex_); |
| rbe_->RemoveStream(ssrc); |
| } |
| |
| DataRate ReceiveSideCongestionController::LatestReceiveSideEstimate() const { |
| MutexLock lock(&mutex_); |
| return rbe_->LatestEstimate(); |
| } |
| |
| void ReceiveSideCongestionController::PickEstimatorFromHeader( |
| const RTPHeader& header) { |
| if (header.extension.hasAbsoluteSendTime) { |
| // If we see AST in header, switch RBE strategy immediately. |
| if (!using_absolute_send_time_) { |
| RTC_LOG(LS_INFO) |
| << "WrappingBitrateEstimator: Switching to absolute send time RBE."; |
| using_absolute_send_time_ = true; |
| PickEstimator(); |
| } |
| packets_since_absolute_send_time_ = 0; |
| } else { |
| // When we don't see AST, wait for a few packets before going back to TOF. |
| if (using_absolute_send_time_) { |
| ++packets_since_absolute_send_time_; |
| if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) { |
| RTC_LOG(LS_INFO) |
| << "WrappingBitrateEstimator: Switching to transmission " |
| "time offset RBE."; |
| using_absolute_send_time_ = false; |
| PickEstimator(); |
| } |
| } |
| } |
| } |
| |
| // Instantiate RBE for Time Offset or Absolute Send Time extensions. |
| void ReceiveSideCongestionController::PickEstimator() { |
| if (using_absolute_send_time_) { |
| rbe_ = std::make_unique<RemoteBitrateEstimatorAbsSendTime>(&remb_throttler_, |
| &clock_); |
| } else { |
| rbe_ = std::make_unique<RemoteBitrateEstimatorSingleStream>( |
| &remb_throttler_, &clock_); |
| } |
| } |
| |
| ReceiveSideCongestionController::ReceiveSideCongestionController( |
| Clock* clock, |
| RemoteEstimatorProxy::TransportFeedbackSender feedback_sender, |
| RembThrottler::RembSender remb_sender, |
| NetworkStateEstimator* network_state_estimator) |
| : clock_(*clock), |
| remb_throttler_(std::move(remb_sender), clock), |
| remote_estimator_proxy_(std::move(feedback_sender), |
| network_state_estimator), |
| rbe_(new RemoteBitrateEstimatorSingleStream(&remb_throttler_, clock)), |
| using_absolute_send_time_(false), |
| packets_since_absolute_send_time_(0) {} |
| |
| void ReceiveSideCongestionController::OnReceivedPacket( |
| const RtpPacketReceived& packet, |
| MediaType media_type) { |
| bool has_transport_sequence_number = |
| packet.HasExtension<TransportSequenceNumber>() || |
| packet.HasExtension<TransportSequenceNumberV2>(); |
| if (media_type == MediaType::AUDIO && !has_transport_sequence_number) { |
| // For audio, we only support send side BWE. |
| return; |
| } |
| |
| if (has_transport_sequence_number) { |
| // Send-side BWE. |
| remote_estimator_proxy_.IncomingPacket(packet); |
| } else { |
| // Receive-side BWE. |
| MutexLock lock(&mutex_); |
| RTPHeader header; |
| packet.GetHeader(&header); |
| PickEstimatorFromHeader(header); |
| rbe_->IncomingPacket(packet.arrival_time().ms(), |
| packet.payload_size() + packet.padding_size(), header); |
| } |
| } |
| |
| void ReceiveSideCongestionController::OnReceivedPacket( |
| int64_t arrival_time_ms, |
| size_t payload_size, |
| const RTPHeader& header) { |
| remote_estimator_proxy_.IncomingPacket(arrival_time_ms, payload_size, header); |
| if (!header.extension.hasTransportSequenceNumber) { |
| // Receive-side BWE. |
| MutexLock lock(&mutex_); |
| PickEstimatorFromHeader(header); |
| rbe_->IncomingPacket(arrival_time_ms, payload_size, header); |
| } |
| } |
| |
| void ReceiveSideCongestionController::OnBitrateChanged(int bitrate_bps) { |
| remote_estimator_proxy_.OnBitrateChanged(bitrate_bps); |
| } |
| |
| TimeDelta ReceiveSideCongestionController::MaybeProcess() { |
| Timestamp now = clock_.CurrentTime(); |
| mutex_.Lock(); |
| TimeDelta time_until_rbe = rbe_->Process(); |
| mutex_.Unlock(); |
| TimeDelta time_until_rep = remote_estimator_proxy_.Process(now); |
| TimeDelta time_until = std::min(time_until_rbe, time_until_rep); |
| return std::max(time_until, TimeDelta::Zero()); |
| } |
| |
| void ReceiveSideCongestionController::SetMaxDesiredReceiveBitrate( |
| DataRate bitrate) { |
| remb_throttler_.SetMaxDesiredReceiveBitrate(bitrate); |
| } |
| |
| void ReceiveSideCongestionController::SetTransportOverhead( |
| DataSize overhead_per_packet) { |
| remote_estimator_proxy_.SetTransportOverhead(overhead_per_packet); |
| } |
| |
| } // namespace webrtc |