Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/pc/peerconnection.cc

Index: webrtc/pc/peerconnection.cc
diff --git a/webrtc/pc/peerconnection.cc b/webrtc/pc/peerconnection.cc
index 852dd39de5bc21eed1461340f796d4ef0132e55f..ab69ce1cde87a09d9152718a26dfcc108bfe642c 100644
--- a/webrtc/pc/peerconnection.cc
+++ b/webrtc/pc/peerconnection.cc
@@ -22,10 +22,12 @@
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/call.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/sctp/sctptransport.h"
 #include "webrtc/pc/audiotrack.h"
@@ -460,6 +462,9 @@ bool PeerConnection::Initialize(
     return false;
   }
 
+  rtp_transport_controller_send_ = rtc::MakeUnique<RtpTransportControllerSend>(
+      Clock::GetRealTimeClock(), event_log_.get());
+
   // Call must be constructed on the worker thread.
   factory_->worker_thread()->Invoke<void>(
       RTC_FROM_HERE, rtc::Bind(&PeerConnection::CreateCall_w,
@@ -2341,7 +2346,11 @@ void PeerConnection::CreateCall_w() {
   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.send_side_cc = rtp_transport_controller_send_->send_side_cc();
+  call_config.audio_rtp_transport_send = rtp_transport_controller_send_.get();
+  call_config.video_rtp_transport_send = rtp_transport_controller_send_.get();
+
+  call_ = rtc::WrapUnique(webrtc::Call::Create(call_config));
 }
 
 }  // namespace webrtc