Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

webrtc/video/video_receive_stream.h

 /*
  *  Copyright (c) 2013 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 @@
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video/rtp_stream_receiver.h"
 #include "webrtc/video/rtp_streams_synchronizer.h"
 #include "webrtc/video/transport_adapter.h"
 #include "webrtc/video/video_stream_decoder.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
 class CallStats;
-class CongestionController;
 class IvfFileWriter;
 class ProcessThread;
 class RTPFragmentationHeader;
 class VieRemb;
 class VCMTiming;
 class VCMJitterEstimator;
 
 namespace internal {
 
 class VideoReceiveStream : public webrtc::VideoReceiveStream,
                            public rtc::VideoSinkInterface<VideoFrame>,
                            public EncodedImageCallback,
                            public NackSender,
                            public KeyFrameRequestSender,
                            public video_coding::OnCompleteFrameCallback,
                            public Syncable {
  public:
   VideoReceiveStream(int num_cpu_cores,
                      bool protected_by_flexfec,
-                     CongestionController* congestion_controller,
                      PacketRouter* packet_router,
                      VideoReceiveStream::Config config,
                      ProcessThread* process_thread,
                      CallStats* call_stats,
                      VieRemb* remb);
   ~VideoReceiveStream() override;
 
   const Config& config() const { return config_; }
 
   void SignalNetworkState(NetworkState state);
@@ skipping 54 matching lines @@
 
   TransportAdapter transport_adapter_;
   const VideoReceiveStream::Config config_;
   const int num_cpu_cores_;
   const bool protected_by_flexfec_;
   ProcessThread* const process_thread_;
   Clock* const clock_;
 
   rtc::PlatformThread decode_thread_;
 
-  CongestionController* const congestion_controller_;
   CallStats* const call_stats_;
 
   std::unique_ptr<VCMTiming> timing_;  // Jitter buffer experiment.
   vcm::VideoReceiver video_receiver_;
   std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
   ReceiveStatisticsProxy stats_proxy_;
   RtpStreamReceiver rtp_stream_receiver_;
   std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
   RtpStreamsSynchronizer rtp_stream_sync_;
 
   rtc::CriticalSection ivf_writer_lock_;
   std::unique_ptr<IvfFileWriter> ivf_writer_ GUARDED_BY(ivf_writer_lock_);
 
   // Members for the new jitter buffer experiment.
   std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
   std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_