Add send transports to individual webrtc::Call streams.

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 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_TEST_COMMON_CALL_TEST_H_
@@ skipping 35 matching lines @@
   static const int kAbsSendTimeExtensionId;
 
  protected:
   void RunBaseTest(BaseTest* test);
 
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
 
-  void CreateSendConfig(size_t num_streams);
-  void CreateMatchingReceiveConfigs();
+  void CreateSendConfig(size_t num_streams, newapi::Transport* send_transport);
+  void CreateMatchingReceiveConfigs(newapi::Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturer();
 
   void CreateStreams();
   void Start();
   void Stop();
   void DestroyStreams();
 
   Clock* const clock_;
 
@@ skipping 51 matching lines @@
   explicit EndToEndTest(unsigned int timeout_ms);
   EndToEndTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_COMMON_CALL_TEST_H_