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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.
*/
#ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
#define MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
#include <memory>
#include <vector>
#include "api/transport/network_control.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "modules/congestion_controller/remb_throttler.h"
#include "modules/pacing/packet_router.h"
#include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class RemoteBitrateEstimator;
// This class represents the congestion control state for receive
// streams. For send side bandwidth estimation, this is simply
// relaying for each received RTP packet back to the sender. While for
// receive side bandwidth estimation, we do the estimation locally and
// send our results back to the sender.
class ReceiveSideCongestionController : public CallStatsObserver {
public:
ReceiveSideCongestionController(
Clock* clock,
RemoteEstimatorProxy::TransportFeedbackSender feedback_sender,
RembThrottler::RembSender remb_sender,
NetworkStateEstimator* network_state_estimator);
~ReceiveSideCongestionController() override {}
void OnReceivedPacket(const RtpPacketReceived& packet, MediaType media_type);
// TODO(perkj, bugs.webrtc.org/14859): Remove all usage. This method is
// currently not used by PeerConnections.
virtual void OnReceivedPacket(int64_t arrival_time_ms,
size_t payload_size,
const RTPHeader& header);
// Implements CallStatsObserver.
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
// This is send bitrate, used to control the rate of feedback messages.
void OnBitrateChanged(int bitrate_bps);
// Ensures the remote party is notified of the receive bitrate no larger than
// `bitrate` using RTCP REMB.
void SetMaxDesiredReceiveBitrate(DataRate bitrate);
void SetTransportOverhead(DataSize overhead_per_packet);
// Returns latest receive side bandwidth estimation.
// Returns zero if receive side bandwidth estimation is unavailable.
DataRate LatestReceiveSideEstimate() const;
// Removes stream from receive side bandwidth estimation.
// Noop if receive side bwe is not used or stream doesn't participate in it.
void RemoveStream(uint32_t ssrc);
// Runs periodic tasks if it is time to run them, returns time until next
// call to `MaybeProcess` should be non idle.
TimeDelta MaybeProcess();
private:
void PickEstimatorFromHeader(const RTPHeader& header)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void PickEstimator() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
Clock& clock_;
RembThrottler remb_throttler_;
RemoteEstimatorProxy remote_estimator_proxy_;
mutable Mutex mutex_;
std::unique_ptr<RemoteBitrateEstimator> rbe_ RTC_GUARDED_BY(mutex_);
bool using_absolute_send_time_ RTC_GUARDED_BY(mutex_);
uint32_t packets_since_absolute_send_time_ RTC_GUARDED_BY(mutex_);
};
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_