Add send transports to individual webrtc::Call streams.

webrtc/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_H_
@@ skipping 51 matching lines @@
  protected:
   virtual ~LoadObserver() {}
 };
 
 // A Call instance can contain several send and/or receive streams. All streams
 // are assumed to have the same remote endpoint and will share bitrate estimates
 // etc.
 class Call {
  public:
   struct Config {
-    Config() = delete;
-    explicit Config(newapi::Transport* send_transport)
-        : send_transport(send_transport) {}
-
     static const int kDefaultStartBitrateBps;
 
-    // TODO(solenberg): Need to add media type to the interface for outgoing
-    // packets too.
-    newapi::Transport* send_transport = nullptr;
-
     // VoiceEngine used for audio/video synchronization for this Call.
     VoiceEngine* voice_engine = nullptr;
 
     // Callback for overuse and normal usage based on the jitter of incoming
     // captured frames. 'nullptr' disables the callback.
     LoadObserver* overuse_callback = nullptr;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
@@ skipping 54 matching lines @@
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
   virtual void SignalNetworkState(NetworkState state) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_