dad91a69bf
Today, behaviour is decided based on if transport sequence number v2 is in the SDP answer. But it might be better to decide based on received packets since it is valid to negotiate both extensions. Another bonus With this solution is that Call does not need to know about receive header exensions. This is an alternative to https://webrtc-review.googlesource.com/c/src/+/291337 Bug: webrtc:7135 Change-Id: Ib75474127d6e2e2029557b8bb2528eaac66979f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291525 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39226}
94 lines
3.6 KiB
C++
94 lines
3.6 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
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#define MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
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#include <memory>
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#include <vector>
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#include "api/transport/network_control.h"
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#include "api/units/data_rate.h"
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#include "api/units/time_delta.h"
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#include "modules/congestion_controller/remb_throttler.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class RemoteBitrateEstimator;
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// This class represents the congestion control state for receive
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// streams. For send side bandwidth estimation, this is simply
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// relaying for each received RTP packet back to the sender. While for
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// receive side bandwidth estimation, we do the estimation locally and
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// send our results back to the sender.
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class ReceiveSideCongestionController : public CallStatsObserver {
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public:
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ReceiveSideCongestionController(
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Clock* clock,
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RemoteEstimatorProxy::TransportFeedbackSender feedback_sender,
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RembThrottler::RembSender remb_sender,
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NetworkStateEstimator* network_state_estimator);
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~ReceiveSideCongestionController() override {}
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void OnReceivedPacket(const RtpPacketReceived& packet, MediaType media_type);
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// TODO(perkj, bugs.webrtc.org/14859): Remove all usage. This method is
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// currently not used by PeerConnections.
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virtual void OnReceivedPacket(int64_t arrival_time_ms,
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size_t payload_size,
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const RTPHeader& header);
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// Implements CallStatsObserver.
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void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
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// This is send bitrate, used to control the rate of feedback messages.
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void OnBitrateChanged(int bitrate_bps);
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// Ensures the remote party is notified of the receive bitrate no larger than
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// `bitrate` using RTCP REMB.
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void SetMaxDesiredReceiveBitrate(DataRate bitrate);
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void SetTransportOverhead(DataSize overhead_per_packet);
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// Returns latest receive side bandwidth estimation.
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// Returns zero if receive side bandwidth estimation is unavailable.
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DataRate LatestReceiveSideEstimate() const;
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// Removes stream from receive side bandwidth estimation.
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// Noop if receive side bwe is not used or stream doesn't participate in it.
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void RemoveStream(uint32_t ssrc);
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// Runs periodic tasks if it is time to run them, returns time until next
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// call to `MaybeProcess` should be non idle.
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TimeDelta MaybeProcess();
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private:
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void PickEstimatorFromHeader(const RTPHeader& header)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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void PickEstimator() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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Clock& clock_;
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RembThrottler remb_throttler_;
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RemoteEstimatorProxy remote_estimator_proxy_;
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mutable Mutex mutex_;
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std::unique_ptr<RemoteBitrateEstimator> rbe_ RTC_GUARDED_BY(mutex_);
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bool using_absolute_send_time_ RTC_GUARDED_BY(mutex_);
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uint32_t packets_since_absolute_send_time_ RTC_GUARDED_BY(mutex_);
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};
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} // namespace webrtc
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#endif // MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
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