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/*
* 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
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