Use separate rtp module lists for send and receive in PacketRouter.

webrtc/voice_engine/channel.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 311 matching lines @@
     int SetLocalSSRC(unsigned int ssrc);
     int GetLocalSSRC(unsigned int& ssrc);
     int GetRemoteSSRC(unsigned int& ssrc);
     int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
     int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
     int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
     int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
     void EnableSendTransportSequenceNumber(int id);
     void EnableReceiveTransportSequenceNumber(int id);
 
-    void SetCongestionControlObjects(
+    void RegisterSenderCongestionControlObjects(
         RtpPacketSender* rtp_packet_sender,
         TransportFeedbackObserver* transport_feedback_observer,
         PacketRouter* packet_router);
+    void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
+    void ResetCongestionControlObjects();
 
     void SetRTCPStatus(bool enable);
     int GetRTCPStatus(bool& enabled);
     int SetRTCP_CNAME(const char cName[256]);
     int GetRemoteRTCP_CNAME(char cName[256]);
     int GetRemoteRTCPData(unsigned int& NTPHigh, unsigned int& NTPLow,
                           unsigned int& timestamp,
                           unsigned int& playoutTimestamp, unsigned int* jitter,
                           unsigned short* fractionLost);
     int SendApplicationDefinedRTCPPacket(unsigned char subType,
@@ skipping 257 matching lines @@
     PacketRouter* packet_router_ = nullptr;
     rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
     rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
     rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_