This is a step on the way to have variable bitrate for audio and is intended to be as much of a no-op as possible. BUG=webrtc:5079 Review URL: https://codereview.webrtc.org/1441673002 Cr-Commit-Position: refs/heads/master@{#10630}
90 lines
2.8 KiB
C++
90 lines
2.8 KiB
C++
/*
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* Copyright (c) 2012 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 WEBRTC_CALL_CONGESTION_CONTROLLER_H_
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#define WEBRTC_CALL_CONGESTION_CONTROLLER_H_
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#include <vector>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/socket.h"
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#include "webrtc/stream.h"
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namespace webrtc {
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class BitrateController;
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class BitrateObserver;
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class CallStats;
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class Config;
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class PacedSender;
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class PacketRouter;
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class ProcessThread;
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class RemoteBitrateEstimator;
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class RemoteEstimatorProxy;
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class RtpRtcp;
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class SendStatisticsProxy;
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class TransportFeedbackAdapter;
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class TransportFeedbackObserver;
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class ViEEncoder;
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class VieRemb;
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class CongestionController {
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public:
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CongestionController(ProcessThread* process_thread, CallStats* call_stats,
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BitrateObserver* bitrate_observer);
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~CongestionController();
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void AddEncoder(ViEEncoder* encoder);
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void RemoveEncoder(ViEEncoder* encoder);
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void SetBweBitrates(int min_bitrate_bps,
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int start_bitrate_bps,
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int max_bitrate_bps);
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void SetChannelRembStatus(bool sender, bool receiver, RtpRtcp* rtp_module);
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void SignalNetworkState(NetworkState state);
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BitrateController* GetBitrateController() const;
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RemoteBitrateEstimator* GetRemoteBitrateEstimator(bool send_side_bwe) const;
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int64_t GetPacerQueuingDelayMs() const;
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PacedSender* pacer() const { return pacer_.get(); }
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PacketRouter* packet_router() const { return packet_router_.get(); }
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TransportFeedbackObserver* GetTransportFeedbackObserver();
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void UpdatePacerBitrate(int bitrate_kbps, int max_bitrate_kbps,
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int min_bitrate_kbps);
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void OnSentPacket(const rtc::SentPacket& sent_packet);
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private:
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rtc::scoped_ptr<VieRemb> remb_;
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rtc::scoped_ptr<PacketRouter> packet_router_;
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rtc::scoped_ptr<PacedSender> pacer_;
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rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
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rtc::scoped_ptr<RemoteEstimatorProxy> remote_estimator_proxy_;
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mutable rtc::CriticalSection encoder_crit_;
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std::vector<ViEEncoder*> encoders_ GUARDED_BY(encoder_crit_);
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// Registered at construct time and assumed to outlive this class.
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ProcessThread* const process_thread_;
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CallStats* const call_stats_;
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rtc::scoped_ptr<ProcessThread> pacer_thread_;
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rtc::scoped_ptr<BitrateController> bitrate_controller_;
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rtc::scoped_ptr<TransportFeedbackAdapter> transport_feedback_adapter_;
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int min_bitrate_bps_;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_CONGESTION_CONTROLLER_H_
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