Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/pc/peerconnection.h

Index: webrtc/pc/peerconnection.h
diff --git a/webrtc/pc/peerconnection.h b/webrtc/pc/peerconnection.h
index 1e2ca3bb95487baa5d519967c3237314c0726c6c..fbb132e9a1e0114dff18339ded5d1a8f12c19357 100644
--- a/webrtc/pc/peerconnection.h
+++ b/webrtc/pc/peerconnection.h
@@ -31,6 +31,7 @@ namespace webrtc {
 class MediaStreamObserver;
 class VideoRtpReceiver;
 class RtcEventLog;
+class RtpTransportControllerSendInterface;
 
 // Populates |session_options| from |rtc_options|, and returns true if options
 // are valid.
@@ -438,6 +439,12 @@ class PeerConnection : public PeerConnectionInterface,
 
   bool remote_peer_supports_msid_ = false;
 
+  // TODO(nisse): Should use separate audio and video transports in
+  // the unbundled case. And potentially more than two in the
+  // completely unbundled case.
+  std::unique_ptr<RtpTransportControllerSendInterface>
+      rtp_transport_controller_send_;
+
   std::unique_ptr<Call> call_;
   std::unique_ptr<WebRtcSession> session_;
   std::unique_ptr<StatsCollector> stats_;  // A pointer is passed to senders_