Set RtcpSender transport at construction.

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index e0195f1758de681829944ea34c9550e9f1bb3859..0a0d94a45e3f37bde68150a31862ff4a2a0772c9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -76,11 +76,10 @@ public:
  RTCPSender(bool audio,
             Clock* clock,
             ReceiveStatistics* receive_statistics,
-            RtcpPacketTypeCounterObserver* packet_type_counter_observer);
+            RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+            Transport* outgoing_transport);
  virtual ~RTCPSender();
 
- int32_t RegisterSendTransport(Transport* outgoingTransport);
-
  RTCPMethod Status() const;
  void SetRTCPStatus(RTCPMethod method);
 
@@ -230,8 +229,7 @@ private:
  Clock* const clock_;
  RTCPMethod method_ GUARDED_BY(critical_section_rtcp_sender_);
 
- rtc::scoped_ptr<CriticalSectionWrapper> critical_section_transport_;
- Transport* cbTransport_ GUARDED_BY(critical_section_transport_);
+ Transport* const transport_;
 
  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
  bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);