Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/call/call.h

Index: webrtc/call/call.h
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index f67b6907e5f7a1c3465b984dc23de768f6130631..5719ac08e623f44da7752fb516a0d70a21f41cad 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -30,6 +30,8 @@ namespace webrtc {
 
 class AudioProcessing;
 class RtcEventLog;
+class RtpTransportControllerSendInterface;
+class SendSideCongestionController;
 
 enum class MediaType {
   ANY,
@@ -86,6 +88,12 @@ class Call {
     // RtcEventLog to use for this call. Required.
     // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
     RtcEventLog* event_log = nullptr;
+
+    RtpTransportControllerSendInterface* audio_rtp_transport_send = nullptr;
+    RtpTransportControllerSendInterface* video_rtp_transport_send = nullptr;
+    // Even if we have separate transports (unbundled case), we can have only
+    // one congestion controller.
+    SendSideCongestionController* send_side_cc = nullptr;
   };
 
   struct Stats {