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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 <cstdint>
#include <memory>
#include "absl/base/attributes.h"
#include "absl/base/nullability.h"
#include "api/environment/environment.h"
#include "api/media_types.h"
#include "api/sequence_checker.h"
#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/include/module_common_types.h"
#include "modules/remote_bitrate_estimator/congestion_control_feedback_generator.h"
#include "modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.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(
const Environment& env,
TransportSequenceNumberFeedbackGenenerator::RtcpSender feedback_sender,
RembThrottler::RembSender remb_sender);
~ReceiveSideCongestionController() override = default;
void EnableSendCongestionControlFeedbackAccordingToRfc8888();
void OnReceivedPacket(const RtpPacketReceived& packet, MediaType media_type);
// 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);
// 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 PickEstimator(bool has_absolute_send_time)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
const Environment env_;
RembThrottler remb_throttler_;
// TODO: bugs.webrtc.org/42224904 - Use sequence checker for all usage of
// ReceiveSideCongestionController. At the time of
// writing OnReceivedPacket and MaybeProcess can unfortunately be called on an
// arbitrary thread by external projects.
SequenceChecker sequence_checker_;
bool send_rfc8888_congestion_feedback_ = false;
TransportSequenceNumberFeedbackGenenerator
transport_sequence_number_feedback_generator_;
CongestionControlFeedbackGenerator congestion_control_feedback_generator_
RTC_GUARDED_BY(sequence_checker_);
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_
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