Define RtpTransportControllerSendInterface.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 2791f7f8b4ed555ed9da3e8d3f45ab6b9fa18171..303ad57e7ed62e676f78ed19bc03f75791d6320a 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -22,6 +22,7 @@
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
+#include "webrtc/call/rtp_transport_controller.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
@@ -2422,9 +2423,13 @@ void Channel::EnableReceiveTransportSequenceNumber(int id) {
 }
 
 void Channel::RegisterSenderCongestionControlObjects(
-    RtpPacketSender* rtp_packet_sender,
-    TransportFeedbackObserver* transport_feedback_observer,
-    PacketRouter* packet_router) {
+    RtpTransportControllerSenderInterface* transport) {
+  RtpPacketSender* rtp_packet_sender =
+      transport->congestion_controller()->pacer();
+  TransportFeedbackObserver* transport_feedback_observer =
+      transport->congestion_controller()->GetTransportFeedbackObserver();
+  PacketRouter* packet_router = transport->packet_router();
+
   RTC_DCHECK(rtp_packet_sender);
   RTC_DCHECK(transport_feedback_observer);
   RTC_DCHECK(packet_router && !packet_router_);