Add send transports to individual webrtc::Call streams.

webrtc/audio_send_stream.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_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_SEND_STREAM_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/stream.h"
+#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioSendStream : public SendStream {
  public:
   struct Stats {};
 
   struct Config {
+    Config() = delete;
+    explicit Config(newapi::Transport* send_transport)
+        : send_transport(send_transport) {}
+
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
       // Sender SSRC.
       uint32_t ssrc = 0;
 
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
+    // Transport for outgoing packets.
+    newapi::Transport* send_transport = nullptr;
+
     rtc::scoped_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
     int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
   };
 
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_