Refactor NACK bitrate allocation

webrtc/modules/congestion_controller/include/congestion_controller.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 14 matching lines @@
 namespace rtc {
 struct SentPacket;
 }
 
 namespace webrtc {
 
 class BitrateController;
 class BitrateObserver;
 class Clock;
 class ProcessThread;
+class RateLimiter;
 class RemoteBitrateEstimator;
 class RemoteBitrateObserver;
 class TransportFeedbackObserver;
 
 class CongestionController : public CallStatsObserver, public Module {
  public:
   // Observer class for bitrate changes announced due to change in bandwidth
   // estimate or due to that the send pacer is full. Fraction loss and rtt is
   // also part of this callback to allow the observer to optimize its settings
   // for different types of network environments. The bitrate does not include
@@ skipping 31 matching lines @@
                                    int min_bitrate_bps,
                                    int max_bitrate_bps);
   virtual void SignalNetworkState(NetworkState state);
   virtual BitrateController* GetBitrateController() const;
   virtual RemoteBitrateEstimator* GetRemoteBitrateEstimator(
       bool send_side_bwe);
   virtual int64_t GetPacerQueuingDelayMs() const;
   virtual PacedSender* pacer() { return pacer_.get(); }
   virtual PacketRouter* packet_router() { return packet_router_.get(); }
   virtual TransportFeedbackObserver* GetTransportFeedbackObserver();
+  RateLimiter* GetRetransmissionRateLimiter();
 
   // SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
   // settings.
   // |min_send_bitrate_bps| is the total minimum send bitrate required by all
   // sending streams.  This is the minimum bitrate the PacedSender will use.
   // Note that CongestionController::OnNetworkChanged can still be called with
   // a lower bitrate estimate.
   // |max_padding_bitrate_bps| is the max bitrate the send streams request for
   // padding. This can be higher than the current network estimate and tells
   // the PacedSender how much it should max pad unless there is real packets to
@@ skipping 18 matching lines @@
   bool IsNetworkDown() const;
   bool HasNetworkParametersToReportChanged(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt);
   Clock* const clock_;
   Observer* const observer_;
   const std::unique_ptr<PacketRouter> packet_router_;
   const std::unique_ptr<PacedSender> pacer_;
   const std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
   const std::unique_ptr<BitrateController> bitrate_controller_;
+  const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
   RemoteEstimatorProxy remote_estimator_proxy_;
   TransportFeedbackAdapter transport_feedback_adapter_;
   int min_bitrate_bps_;
   rtc::CriticalSection critsect_;
   uint32_t last_reported_bitrate_bps_ GUARDED_BY(critsect_);
   uint8_t last_reported_fraction_loss_ GUARDED_BY(critsect_);
   int64_t last_reported_rtt_ GUARDED_BY(critsect_);
   NetworkState network_state_ GUARDED_BY(critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CongestionController);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_CONGESTION_CONTROLLER_H_