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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc')
-rw-r--r-- | third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc | 130 |
1 files changed, 130 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc new file mode 100644 index 0000000000..e042678897 --- /dev/null +++ b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc @@ -0,0 +1,130 @@ +/* + * 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::PickEstimator( + bool has_absolute_send_time) { + if (has_absolute_send_time) { + // 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; + rbe_ = std::make_unique<RemoteBitrateEstimatorAbsSendTime>( + &remb_throttler_, &clock_); + } + 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; + 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_); + PickEstimator(packet.HasExtension<AbsoluteSendTime>()); + rbe_->IncomingPacket(packet); + } +} + +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 |