/* * 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 #include "absl/base/nullability.h" #include "api/environment/environment.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/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, absl::Nullable network_state_estimator); ~ReceiveSideCongestionController() override = default; void EnablSendCongestionControlFeedbackAccordingToRfc8888(); 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); 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 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 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_