Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/ortc/rtptransportcontrolleradapter.cc

Index: webrtc/ortc/rtptransportcontrolleradapter.cc
diff --git a/webrtc/ortc/rtptransportcontrolleradapter.cc b/webrtc/ortc/rtptransportcontrolleradapter.cc
index b482e475fdd508f58da4753888cf6140bb9e580b..573d3198602f9fe897f13ed80130b37d1ffcad18 100644
--- a/webrtc/ortc/rtptransportcontrolleradapter.cc
+++ b/webrtc/ortc/rtptransportcontrolleradapter.cc
@@ -18,6 +18,8 @@
 
 #include "webrtc/api/proxy.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/ptr_util.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/ortc/ortcrtpreceiveradapter.h"
 #include "webrtc/ortc/ortcrtpsenderadapter.h"
@@ -620,13 +622,17 @@ void RtpTransportControllerAdapter::Init_w() {
   const int kStartBandwidthBps = 300000;
   const int kMaxBandwidthBps = 2000000;
 
+  rtp_transport_controller_send_ = rtc::MakeUnique<RtpTransportControllerSend>(
+      Clock::GetRealTimeClock(), event_log_);
   webrtc::Call::Config call_config(event_log_);
   call_config.audio_state = channel_manager_->media_engine()->GetAudioState();
   call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
-
-  call_.reset(webrtc::Call::Create(call_config));
+  call_config.audio_rtp_transport_send = rtp_transport_controller_send_.get();
+  call_config.video_rtp_transport_send = rtp_transport_controller_send_.get();
+  call_config.send_side_cc = rtp_transport_controller_send_->send_side_cc();
+  call_ = rtc::WrapUnique(webrtc::Call::Create(call_config));
 }
 
 void RtpTransportControllerAdapter::Close_w() {