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 a9cf2b26dc8edea51bb8d56b57d13a52275e58b2..c5b4c881a40626cd289aabd4db23f01c4b126708 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"
@@ -2440,10 +2441,14 @@ void Channel::EnableReceiveTransportSequenceNumber(int id) {
 }
 
 void Channel::RegisterSenderCongestionControlObjects(
-    RtpPacketSender* rtp_packet_sender,
-    TransportFeedbackObserver* transport_feedback_observer,
-    PacketRouter* packet_router,
+    RtpTransportControllerSendInterface* transport,
     RtcpBandwidthObserver* bandwidth_observer) {
+  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_);