| /* |
| * 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); |
| |
| // 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_); |
| |
| 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_ |