Move Ownership of RtpModules to VideoSendStream.

webrtc/video/vie_receiver.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 42 matching lines @@
   // received.
   void SetUseRtxPayloadMappingOnRestore(bool val);
   void SetRtxSsrc(uint32_t ssrc);
   bool GetRtxSsrc(uint32_t* ssrc) const;
 
   bool IsFecEnabled() const;
 
   uint32_t GetRemoteSsrc() const;
   int GetCsrcs(uint32_t* csrcs) const;
 
-  void Init(const std::vector<RtpRtcp*>& modules);
+  void Init(RtpRtcp* rtp_rtcp);
 
   RtpReceiver* GetRtpReceiver() const;
 
   void EnableReceiveRtpHeaderExtension(const std::string& extension, int id);
 
   void StartReceive();
   void StopReceive();
 
   bool DeliverRtp(const uint8_t* rtp_packet,
                   size_t rtp_packet_length,
@@ skipping 21 matching lines @@
   void NotifyReceiverOfFecPacket(const RTPHeader& header);
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
   void UpdateHistograms();
 
   Clock* const clock_;
   VideoCodingModule* const vcm_;
   RemoteBitrateEstimator* const remote_bitrate_estimator_;
 
   // TODO(pbos): Make const and set on construction.
-  std::vector<RtpRtcp*> rtp_rtcp_;
+  RtpRtcp* rtp_rtcp_;  // Owned by ViEChannel
 
   RemoteNtpTimeEstimator ntp_estimator_;
   RTPPayloadRegistry rtp_payload_registry_;
 
   const std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   const std::unique_ptr<RtpReceiver> rtp_receiver_;
   const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<FecReceiver> fec_receiver_;
 
   rtc::CriticalSection receive_cs_;
   bool receiving_ GUARDED_BY(receive_cs_);
   uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
   bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
   int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIE_RECEIVER_H_