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 e5e3804b71c5ee0a6282be87865a3579c6768b08..4321082ad0243b0d29ab4a0c01e464d34c2656b2 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 {
@@ -73,12 +74,9 @@ void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) {
 }
 
 void ChannelProxy::RegisterSenderCongestionControlObjects(
-    RtpPacketSender* rtp_packet_sender,
-    TransportFeedbackObserver* transport_feedback_observer,
-    PacketRouter* packet_router) {
+    RtpTransportControllerSenderInterface* transport) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  channel()->RegisterSenderCongestionControlObjects(
-      rtp_packet_sender, transport_feedback_observer, packet_router);
+  channel()->RegisterSenderCongestionControlObjects(transport);
 }
 
 void ChannelProxy::RegisterReceiverCongestionControlObjects(