Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/call/call.h

 /*
  *  Copyright (c) 2013 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_CALL_H_
@@ skipping 12 matching lines @@
 #include "webrtc/call/flexfec_receive_stream.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
 class RtcEventLog;
+class RtpTransportControllerSendInterface;
+class SendSideCongestionController;
 
 enum class MediaType {
   ANY,
   AUDIO,
   VIDEO,
   DATA
 };
 
 class PacketReceiver {
  public:
@@ skipping 36 matching lines @@
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     rtc::scoped_refptr<AudioState> audio_state;
 
     // Audio Processing Module to be used in this call.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     AudioProcessing* audio_processing = nullptr;
 
     // 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 {
     std::string ToString(int64_t time_ms) const;
 
     int send_bandwidth_bps = 0;       // Estimated available send bandwidth.
     int max_padding_bitrate_bps = 0;  // Cumulative configured max padding.
     int recv_bandwidth_bps = 0;       // Estimated available receive bandwidth.
     int64_t pacer_delay_ms = 0;
     int64_t rtt_ms = -1;
@@ skipping 65 matching lines @@
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_CALL_H_