Define RtpTransportControllerSendInterface.

webrtc/voice_engine/channel_proxy.cc

Index: webrtc/voice_engine/channel_proxy.cc
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index a58b2e19557790c9248ae0133a714fa4d50729ae..204e219034141fc82c2009e043e565f1c391e4a7 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -14,6 +14,7 @@
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/call/rtp_transport_controller.h"
 #include "webrtc/voice_engine/channel.h"
 
 namespace webrtc {
@@ -73,14 +74,11 @@ void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) {
 }
 
 void ChannelProxy::RegisterSenderCongestionControlObjects(
-    RtpPacketSender* rtp_packet_sender,
-    TransportFeedbackObserver* transport_feedback_observer,
-    PacketRouter* packet_router,
+    RtpTransportControllerSendInterface* transport,
     RtcpBandwidthObserver* bandwidth_observer) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  channel()->RegisterSenderCongestionControlObjects(
-      rtp_packet_sender, transport_feedback_observer, packet_router,
-      bandwidth_observer);
+  channel()->RegisterSenderCongestionControlObjects(transport,
+                                                    bandwidth_observer);
 }
 
 void ChannelProxy::RegisterReceiverCongestionControlObjects(