Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/call/fake_rtp_transport_controller_send.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
 #define WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
 
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 
 namespace webrtc {
 
+// TODO(nisse): This is almost the same as RtpTransportControllerSend,
+// only difference is that it doesn't own the PacketRouter and
+// SendSideCongestionController. However, RtpTransportControllerSend
+// should likely move in that direction too, and then this class
+// becomes redundant.
 class FakeRtpTransportControllerSend
     : public RtpTransportControllerSendInterface {
  public:
   explicit FakeRtpTransportControllerSend(
       PacketRouter* packet_router,
       SendSideCongestionController* send_side_cc)
       : packet_router_(packet_router), send_side_cc_(send_side_cc) {
     RTC_DCHECK(send_side_cc);
   }
 
@@ skipping 10 matching lines @@
   RtpPacketSender* packet_sender() override { return send_side_cc_->pacer(); }
 
  private:
   PacketRouter* packet_router_;
   SendSideCongestionController* send_side_cc_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_