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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
104 lines
3.6 KiB
C++
104 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 "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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namespace webrtc {
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class RemoteBitrateEstimator;
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class RemoteBitrateObserver;
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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 Module {
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public:
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ReceiveSideCongestionController(
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const Clock* clock,
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PacketRouter* packet_router);
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virtual ~ReceiveSideCongestionController() {}
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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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// TODO(nisse): Delete these methods, design a more specific interface.
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virtual RemoteBitrateEstimator* GetRemoteBitrateEstimator(bool send_side_bwe);
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virtual const RemoteBitrateEstimator* GetRemoteBitrateEstimator(
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bool send_side_bwe) const;
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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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// Implements Module.
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int64_t TimeUntilNextProcess() override;
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void Process() override;
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private:
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class WrappingBitrateEstimator : public RemoteBitrateEstimator {
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public:
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WrappingBitrateEstimator(RemoteBitrateObserver* observer,
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const Clock* clock);
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virtual ~WrappingBitrateEstimator() {}
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void IncomingPacket(int64_t arrival_time_ms,
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size_t payload_size,
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const RTPHeader& header) override;
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void Process() override;
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int64_t TimeUntilNextProcess() override;
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void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
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void RemoveStream(unsigned int ssrc) override;
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bool LatestEstimate(std::vector<unsigned int>* ssrcs,
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unsigned int* bitrate_bps) const override;
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void SetMinBitrate(int min_bitrate_bps) override;
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private:
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void PickEstimatorFromHeader(const RTPHeader& header)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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void PickEstimator() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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RemoteBitrateObserver* observer_;
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const Clock* const clock_;
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rtc::CriticalSection crit_sect_;
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std::unique_ptr<RemoteBitrateEstimator> rbe_;
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bool using_absolute_send_time_;
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uint32_t packets_since_absolute_send_time_;
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int min_bitrate_bps_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
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};
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WrappingBitrateEstimator remote_bitrate_estimator_;
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RemoteEstimatorProxy remote_estimator_proxy_;
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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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